From e01b4f624278d5efe5fb5da585ca371947b16680 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:26:42 +0100 Subject: [PATCH 001/529] ASoC: dapm: Fix potential DAI widget pointer deref when linking DAIs Sometime a component or topology may configure a DAI widget with no private data leading to a dev_dbg() dereferencne of this data. Fix this to check for non NULL private data and let users know if widget is missing DAI. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 36a39ba30226..8ede773b1db8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4073,6 +4073,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) continue; } + /* let users know there is no DAI to link */ + if (!dai_w->priv) { + dev_dbg(card->dev, "dai widget %s has no DAI\n", + dai_w->name); + continue; + } + dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ From c60b613a7097cff20fdd05e2891ce69542f0d5a3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:50:37 +0100 Subject: [PATCH 002/529] ASoC: topology: Give more data to clients via callbacks Give topology clients more access to the topology data by passing index, pcm, link_config and dai_driver to clients. This allows clients to fully instantiate and track topology objects. The SOF driver is the first user of these new APIs and needs them to build component topology driver and FW objects. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 23 +++++++++++-------- sound/soc/intel/skylake/skl-pcm.c | 7 +++--- sound/soc/intel/skylake/skl-topology.c | 5 +++-- sound/soc/intel/skylake/skl-topology.h | 5 +++-- sound/soc/soc-topology.c | 31 +++++++++++++++----------- 5 files changed, 42 insertions(+), 29 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f552c3f56368..e1f265e21ee1 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -30,6 +30,8 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; +struct snd_soc_dai_driver; +struct snd_soc_dai; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -109,35 +111,38 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, + int (*control_load)(struct snd_soc_component *, int index, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, + int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, + int (*widget_ready)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + int (*dai_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); + int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, - struct snd_soc_dai_link *link); + int (*link_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, + int (*vendor_load)(struct snd_soc_component *, int index, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -146,7 +151,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, + int (*manifest)(struct snd_soc_component *, int index, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..1f4dd08d36c5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1017,10 +1017,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index fcdc716754b6..647e52aecdc3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3024,7 +3024,7 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { @@ -3131,6 +3131,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { @@ -3619,7 +3620,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..af198ea0379e 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 53f121a50c97..9b33260fd537 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -259,7 +259,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -291,7 +291,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -302,27 +303,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -643,7 +647,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -1702,7 +1707,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1772,7 +1777,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2080,7 +2085,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2202,7 +2207,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2311,7 +2316,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); From 503e79b793fea5de626db73accf8e8994bc4289d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:53:59 +0100 Subject: [PATCH 003/529] ASoC: topology: Add callback for DAPM route load/unload Add a callback fro clients for notification about DAPM route loading and unloading. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 7 +++++++ sound/soc/soc-topology.c | 13 +++++++++++++ 2 files changed, 20 insertions(+) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index e1f265e21ee1..401ef2c45d6c 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -32,6 +32,7 @@ struct snd_kcontrol_new; struct snd_soc_dai_link; struct snd_soc_dai_driver; struct snd_soc_dai; +struct snd_soc_dapm_route; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -116,6 +117,12 @@ struct snd_soc_tplg_ops { int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); + /* DAPM graph route element loading and unloading */ + int (*dapm_route_load)(struct snd_soc_component *, int index, + struct snd_soc_dapm_route *route); + int (*dapm_route_unload)(struct snd_soc_component *, + struct snd_soc_dobj *); + /* external widget init - used for any driver specific init */ int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 9b33260fd537..05d177d689e2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1105,6 +1105,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic kcontrol to component driver. */ +static int soc_tplg_add_route(struct soc_tplg *tplg, + struct snd_soc_dapm_route *route) +{ + if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + return tplg->ops->dapm_route_load(tplg->comp, tplg->index, + route); + + return 0; +} + static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1153,6 +1164,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, else route.control = elem->control; + soc_tplg_add_route(tplg, &route); + /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, &route, 1); } From 134c875bff58ac988c64a1ea2a337acba1711d4b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:56:51 +0000 Subject: [PATCH 004/529] ASoC: fsi: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3bae06dd121f..aa7e902f0c02 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1,16 +1,12 @@ -/* - * Fifo-attached Serial Interface (FSI) support for SH7724 - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on ssi.c - * Copyright (c) 2007 Manuel Lauss - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Fifo-attached Serial Interface (FSI) support for SH7724 +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on ssi.c +// Copyright (c) 2007 Manuel Lauss #include #include From cb006e7b1712bb9507a218be7ed811ca8e65fc4d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:57:13 +0000 Subject: [PATCH 005/529] ASoC: hac: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 624aaf569fef..c2b496398e6b 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -1,13 +1,11 @@ -/* - * Hitachi Audio Controller (AC97) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable HAC output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Hitachi Audio Controller (AC97) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable HAC output pins! /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only * the FIRST can be used since ASoC does not pass any information to the @@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = { module_platform_driver(hac_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); MODULE_AUTHOR("Manuel Lauss "); From 217bc8c898b24fa2098a375a575f6170cfac45a9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:57:47 +0000 Subject: [PATCH 006/529] ASoC: ssi: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/ssi.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 89ed1b107ac5..8125fa3840b6 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -1,14 +1,11 @@ -/* - * Serial Sound Interface (I2S) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss - * - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable SSI output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Serial Sound Interface (I2S) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable SSI output pins! /* * LIMITATIONS: @@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = { module_platform_driver(sh4_ssi_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); MODULE_AUTHOR("Manuel Lauss "); From 4e6fdaf1bd334e7e3ad9d6ef7aef4a433c770952 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:07 +0000 Subject: [PATCH 007/529] ASoC: siu: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/siu.h | 26 ++++++-------------------- sound/soc/sh/siu_dai.c | 26 ++++++-------------------- sound/soc/sh/siu_pcm.c | 27 +++++++-------------------- 3 files changed, 19 insertions(+), 60 deletions(-) diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 6088d627c0e4..63a508fdfe78 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -1,23 +1,9 @@ -/* - * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz #ifndef SIU_H #define SIU_H diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index ee2211635e92..f2a386fcd92e 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -1,23 +1,9 @@ -/* - * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz #include #include diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 172909570ed5..e263757e4a69 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -1,23 +1,10 @@ -/* - * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz + #include #include #include From 1e0edd4deadbbacd3b35179c233efa26624ab2af Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:38 +0000 Subject: [PATCH 008/529] ASoC: rsnd: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/Makefile | 1 + sound/soc/sh/rcar/adg.c | 15 ++++++--------- sound/soc/sh/rcar/cmd.c | 17 +++++++---------- sound/soc/sh/rcar/core.c | 24 ++++++++++-------------- sound/soc/sh/rcar/ctu.c | 15 ++++++--------- sound/soc/sh/rcar/dma.c | 17 +++++++---------- sound/soc/sh/rcar/dvc.c | 16 ++++++---------- sound/soc/sh/rcar/gen.c | 16 ++++++---------- sound/soc/sh/rcar/mix.c | 14 +++++--------- sound/soc/sh/rcar/rsnd.h | 17 +++++++---------- sound/soc/sh/rcar/src.c | 16 ++++++---------- sound/soc/sh/rcar/ssi.c | 22 +++++++++------------- sound/soc/sh/rcar/ssiu.c | 15 ++++++--------- 13 files changed, 82 insertions(+), 123 deletions(-) diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 9c3d5aed99d1..5d1ff8ef26f9 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,3 @@ +# SPDX-License-Identifier: GPL-2.0 snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 4672688cac32..3a3064dda57f 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -1,12 +1,9 @@ -/* - * Helper routines for R-Car sound ADG. - * - * Copyright (C) 2013 Kuninori Morimoto - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Helper routines for R-Car sound ADG. +// +// Copyright (C) 2013 Kuninori Morimoto + #include #include "rsnd.h" diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 5900fb535a2b..d8043ad33540 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car CMD support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car CMD support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto + #include "rsnd.h" struct rsnd_cmd { diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f237002180c0..eac22fef4543 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on fsi.c - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRU/SCU/SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on fsi.c +// Kuninori Morimoto /* * Renesas R-Car sound device structure @@ -1606,7 +1602,7 @@ static struct platform_driver rsnd_driver = { }; module_platform_driver(rsnd_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Renesas R-Car audio driver"); MODULE_AUTHOR("Kuninori Morimoto "); MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 83be7d3ae0a8..6a55aa753003 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -1,12 +1,9 @@ -/* - * ctu.c - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ctu.c +// +// Copyright (c) 2015 Kuninori Morimoto + #include "rsnd.h" #define CTU_NAME_SIZE 16 diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ef82b94d038b..fe63ef8600d0 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car Audio DMAC support - * - * Copyright (C) 2015 Renesas Electronics Corp. - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Audio DMAC support +// +// Copyright (C) 2015 Renesas Electronics Corp. +// Copyright (c) 2015 Kuninori Morimoto + #include #include #include "rsnd.h" diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ca1780e0b830..2b16e0ce6bc5 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car DVC support - * - * Copyright (C) 2014 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car DVC support +// +// Copyright (C) 2014 Renesas Solutions Corp. +// Kuninori Morimoto /* * Playback Volume diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 25642e92dae0..0230301fe078 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car Gen1 SRU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Gen1 SRU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto /* * #define DEBUG diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 1881b2de9126..8e3b57eaa708 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -1,12 +1,8 @@ -/* - * mix.c - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// mix.c +// +// Copyright (c) 2015 Kuninori Morimoto /* * CTUn MIXn diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6d7280d2d9be..96d93330b1e1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -1,13 +1,10 @@ -/* - * Renesas R-Car - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto + #ifndef RSND_H #define RSND_H diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6c72d1a81cf5..beccfbac7581 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car SRC support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRC support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto /* * you can enable below define if you don't need diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6e1166ec24a0..3d9ea100a64f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on fsi.c - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on fsi.c +// Kuninori Morimoto /* * you can enable below define if you don't need diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 47bdba9fc582..016fbf5ac242 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -1,12 +1,9 @@ -/* - * Renesas R-Car SSIU support - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU support +// +// Copyright (c) 2015 Kuninori Morimoto + #include "rsnd.h" #define SSIU_NAME "ssiu" From 0026c551bacd9654eaa6ee8e4aa19d52730e74f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:54 +0000 Subject: [PATCH 009/529] ASoC: migor: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index ecb057ff9fbb..8739c9f60672 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -1,12 +1,8 @@ -/* - * ALSA SoC driver for Migo-R - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC driver for Migo-R +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski #include #include From ddfe227c0cbfb97b37afe123e014b9bc4df217b5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:59:11 +0000 Subject: [PATCH 010/529] ASoC: dma-sh7760: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 2dc3b762fdd9..922fb6aa3ed1 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -1,16 +1,14 @@ -/* - * SH7760 ("camelot") DMABRG audio DMA unit support - * - * Copyright (C) 2007 Manuel Lauss - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which - * trigger an interrupt when one half of the programmed transfer size - * has been xmitted. - * - * FIXME: little-endian only for now - */ +// SPDX-License-Identifier: GPL-2.0 +// +// SH7760 ("camelot") DMABRG audio DMA unit support +// +// Copyright (C) 2007 Manuel Lauss +// +// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which +// trigger an interrupt when one half of the programmed transfer size +// has been xmitted. +// +// FIXME: little-endian only for now #include #include @@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = { module_platform_driver(sh7760_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); MODULE_AUTHOR("Manuel Lauss "); From 04433977b164c8ed0b65cd023c87c74da0fff2bf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:59:27 +0000 Subject: [PATCH 011/529] ASoC: sh7760-ac97: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/sh7760-ac97.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 4a3568a9bf59..4bb4c13cf860 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -1,10 +1,8 @@ -/* - * Generic AC97 sound support for SH7760 - * - * (c) 2007 Manuel Lauss - * - * Licensed under the GPLv2. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Generic AC97 sound support for SH7760 +// +// (c) 2007 Manuel Lauss #include #include @@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void) module_init(sh7760_ac97_init); module_exit(sh7760_ac97_exit); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); MODULE_AUTHOR("Manuel Lauss "); From 7cc90a5cadb1733d95d3c2bc147cbcf7843aa585 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:52:00 +0000 Subject: [PATCH 012/529] ASoC: rsnd: has .symmetric_rates if SSIs are sharing WS pin If SSIs are sharing WS pin, it should has .symmetric_rates. This patch sets it. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 6 ++++++ sound/soc/sh/rcar/ssi.c | 5 +++-- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index af04d41a4274..6bbdddef426e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1085,6 +1085,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, of_node_put(capture); } + if (rsnd_ssi_is_pin_sharing(io_capture) || + rsnd_ssi_is_pin_sharing(io_playback)) { + /* should have symmetric_rates if pin sharing */ + drv->symmetric_rates = 1; + } + dev_dbg(dev, "%s (%s/%s)\n", rdai->name, rsnd_io_to_mod_ssi(io_playback) ? "play" : " -- ", rsnd_io_to_mod_ssi(io_capture) ? "capture" : " -- "); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9538f76f8e20..4e605648918b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1055,9 +1055,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + if (!mod) + return 0; - return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE)); + return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE)); } static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io, From 203cdf51f28820bee7893b4be392847418e6f4ec Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:52:17 +0000 Subject: [PATCH 013/529] ASoC: rsnd: SSI parent cares SWSP bit SSICR has SWSP bit (= Serial WS Polarity) which decides WS pin 1st channel polarity (low or hi). This bit shouldn't exchange after running. Current SSI "parent" doesn't care SSICR, just controls clock only. Because of this behavior, if platform uses SSI0 as playback, SSI1 as capture, and if user starts capture -> playback order, SSI0 SSICR::SWSP bit exchanged 0 -> 1 during captureing, and it makes capture noise. This patch cares SSICR on SSI parent, too. Special thanks to Yokoyama-san Reported-by: Hiroyuki Yokoyama Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 32 +++++++++++++++++++++----------- 1 file changed, 21 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4e605648918b..98dd120d830a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -37,6 +37,7 @@ #define CHNL_4 (1 << 22) /* Channels */ #define CHNL_6 (2 << 22) /* Channels */ #define CHNL_8 (3 << 22) /* Channels */ +#define DWL_MASK (7 << 19) /* Data Word Length mask */ #define DWL_8 (0 << 19) /* Data Word Length */ #define DWL_16 (1 << 19) /* Data Word Length */ #define DWL_18 (2 << 19) /* Data Word Length */ @@ -353,21 +354,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - u32 cr_own; - u32 cr_mode; - u32 wsr; + u32 cr_own = ssi->cr_own; + u32 cr_mode = ssi->cr_mode; + u32 wsr = ssi->wsr; int is_tdm; - if (rsnd_ssi_is_parent(mod, io)) - return; - is_tdm = rsnd_runtime_is_ssi_tdm(io); /* * always use 32bit system word. * see also rsnd_ssi_master_clk_enable() */ - cr_own = FORCE | SWL_32; + cr_own |= FORCE | SWL_32; if (rdai->bit_clk_inv) cr_own |= SCKP; @@ -377,9 +375,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= SDTA; if (rdai->sys_delay) cr_own |= DEL; + + /* + * We shouldn't exchange SWSP after running. + * This means, parent needs to care it. + */ + if (rsnd_ssi_is_parent(mod, io)) + goto init_end; + if (rsnd_io_is_play(io)) cr_own |= TRMD; + cr_own &= ~DWL_MASK; switch (snd_pcm_format_width(runtime->format)) { case 16: cr_own |= DWL_16; @@ -406,7 +413,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, wsr |= WS_MODE; cr_own |= CHNL_8; } - +init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; ssi->wsr = wsr; @@ -470,15 +477,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, return -EIO; } - if (!rsnd_ssi_is_parent(mod, io)) - ssi->cr_own = 0; - rsnd_ssi_master_clk_stop(mod, io); rsnd_mod_power_off(mod); ssi->usrcnt--; + if (!ssi->usrcnt) { + ssi->cr_own = 0; + ssi->cr_mode = 0; + ssi->wsr = 0; + } + return 0; } From 6e56e5d04191aa20e08430dcb203c081fa247e93 Mon Sep 17 00:00:00 2001 From: "Agrawal, Akshu" Date: Thu, 7 Jun 2018 14:48:43 +0800 Subject: [PATCH 014/529] ASoC: AMD: Add NULL pointer check Fix crash in those platforms whose machine driver does not expose platform_info. For those platforms we rely on default value and select I2SSP channel. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 77203841c535..1458b5048498 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -773,7 +773,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - rtd->i2s_instance = pinfo->i2s_instance; + if (pinfo) + rtd->i2s_instance = pinfo->i2s_instance; if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); From 187e01d0d56d1fd682dfaafb0b45d332abec6387 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Mon, 11 Jun 2018 17:13:59 +0200 Subject: [PATCH 015/529] ASoC: stm32: sai: add iec958 controls support Add support of iec958 controls for STM32 SAI. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 139 +++++++++++++++++++++++++++++++--- 2 files changed, 128 insertions(+), 12 deletions(-) diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 48f9ddd94016..9b2681397dba 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO + select SND_PCM_IEC958 help Say Y if you want to enable SAI for STM32 diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index cfeb219e1d78..c4f15ea14197 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -96,7 +96,8 @@ * @slot_mask: rx or tx active slots mask. set at init or at runtime * @data_size: PCM data width. corresponds to PCM substream width. * @spdif_frm_cnt: S/PDIF playback frame counter - * @spdif_status_bits: S/PDIF status bits + * @snd_aes_iec958: iec958 data + * @ctrl_lock: control lock */ struct stm32_sai_sub_data { struct platform_device *pdev; @@ -125,7 +126,8 @@ struct stm32_sai_sub_data { int slot_mask; int data_size; unsigned int spdif_frm_cnt; - unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES]; + struct snd_aes_iec958 iec958; + struct mutex ctrl_lock; /* protect resources accessed by controls */ }; enum stm32_sai_fifo_th { @@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) } } -static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = { - 0, 0, 0, IEC958_AES3_CON_FS_48000, -}; - static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, @@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .fast_io = true, }; +static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(uctl->value.iec958.status, sai->iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(sai->iec958.status, uctl->value.iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static const struct snd_kcontrol_new iec958_ctls = { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_pcm_iec958_info, + .get = snd_pcm_iec958_get, + .put = snd_pcm_iec958_put, +}; + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -619,6 +660,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) } } +static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai) +{ + unsigned char *cs = sai->iec958.status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_GENERAL; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID; +} + +static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai, + struct snd_pcm_runtime *runtime) +{ + if (!runtime) + return; + + /* Force the sample rate according to runtime rate */ + mutex_lock(&sai->ctrl_lock); + switch (runtime->rate) { + case 22050: + sai->iec958.status[3] = IEC958_AES3_CON_FS_22050; + break; + case 44100: + sai->iec958.status[3] = IEC958_AES3_CON_FS_44100; + break; + case 88200: + sai->iec958.status[3] = IEC958_AES3_CON_FS_88200; + break; + case 176400: + sai->iec958.status[3] = IEC958_AES3_CON_FS_176400; + break; + case 24000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_24000; + break; + case 48000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_48000; + break; + case 96000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_192000; + break; + case 32000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_32000; + break; + default: + sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID; + break; + } + mutex_unlock(&sai->ctrl_lock); +} + static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { @@ -709,7 +803,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream, sai->data_size = params_width(params); - if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + /* Rate not already set in runtime structure */ + substream->runtime->rate = params_rate(params); + stm32_sai_set_iec958_status(sai, substream->runtime); + } else { ret = stm32_sai_set_slots(cpu_dai); if (ret < 0) return ret; @@ -789,6 +887,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, sai->substream = NULL; } +static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__); + return snd_ctl_add(rtd->pcm->card, + snd_ctl_new1(&iec958_ctls, sai)); + } + + return 0; +} + static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); @@ -809,6 +921,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); + /* Next settings are not relevant for spdif mode */ + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + return 0; + cr1_mask = SAI_XCR1_RX_TX; if (STM_SAI_IS_CAPTURE(sai)) cr1 |= SAI_XCR1_RX_TX; @@ -820,10 +936,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) sai->synco, sai->synci); } - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - memcpy(sai->spdif_status_bits, default_status_bits, - sizeof(default_status_bits)); - cr1_mask |= SAI_XCR1_SYNCEN_MASK; cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync); @@ -861,7 +973,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, /* Set channel status bit */ byte = frm_cnt >> 3; mask = 1 << (frm_cnt - (byte << 3)); - if (sai->spdif_status_bits[byte] & mask) + if (sai->iec958.status[byte] & mask) *ptr |= 0x04000000; ptr++; @@ -888,6 +1000,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { { .probe = stm32_sai_dai_probe, + .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ .playback = { .channels_min = 1, @@ -998,6 +1111,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n"); return -EINVAL; } + stm32_sai_init_iec958_status(sai); sai->spdif = true; sai->master = true; } @@ -1114,6 +1228,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; + mutex_init(&sai->ctrl_lock); platform_set_drvdata(pdev, sai); sai->pdata = dev_get_drvdata(pdev->dev.parent); From a56df73ba5960848f60f609c68770d2638bf1dd5 Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sun, 10 Jun 2018 01:20:54 +0300 Subject: [PATCH 016/529] ASoC: rockchip: put device_node on remove snd_rk_mc_probe() gets a couple of device nodes with of_parse_phandle(), but there is no release of them. The patch adds remove handler and proper error handling in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_rt5645.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 4db4fd56db35..881c32498808 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (!rk_dailink.cpu_of_node) { dev_err(&pdev->dev, "Property 'rockchip,i2s-controller' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_codec_of_node; } rk_dailink.platform_of_node = rk_dailink.cpu_of_node; @@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Soc parse card name failed %d\n", ret); - return ret; + goto put_cpu_of_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "Soc register card failed %d\n", ret); - return ret; + goto put_cpu_of_node; } return ret; + +put_cpu_of_node: + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; +put_codec_of_node: + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return ret; +} + +static int snd_rk_mc_remove(struct platform_device *pdev) +{ + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return 0; } static const struct of_device_id rockchip_rt5645_of_match[] = { @@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, From 75b31192fe6ad20b42276b20ee3bdf1493216d63 Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Fri, 8 Jun 2018 16:31:09 +0800 Subject: [PATCH 017/529] ASoC: rockchip: add config for rockchip dmaengine pcm register This patch makes the rockchip i2s pcm configurable by adding rockchip pcm config for devm_snd_dmaengine_pcm_register. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/Makefile | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 3 ++- sound/soc/rockchip/rockchip_pcm.c | 45 +++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_pcm.h | 14 ++++++++++ 4 files changed, 63 insertions(+), 2 deletions(-) create mode 100644 sound/soc/rockchip/rockchip_pcm.c create mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 05b078e7b87f..65e814d46006 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,10 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 950823d69e9c..60d43d53a8f5 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -22,6 +22,7 @@ #include #include "rockchip_i2s.h" +#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); return ret; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c new file mode 100644 index 000000000000..f77538319221 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -0,0 +1,45 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include "rockchip_pcm.h" + +static const struct snd_pcm_hardware snd_rockchip_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, +}; + +static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { + .pcm_hardware = &snd_rockchip_hardware, + .prealloc_buffer_size = 32 * 1024, +}; + +int rockchip_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h new file mode 100644 index 000000000000..d6c36115c60a --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.h @@ -0,0 +1,14 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_PCM_H +#define _ROCKCHIP_PCM_H + +int rockchip_pcm_platform_register(struct device *dev); + +#endif From 4f29b663c08d369fe320a148179996c94cf7d01b Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Thu, 7 Jun 2018 15:50:48 +0200 Subject: [PATCH 018/529] ASoC: rt1305: Use ULL suffixes for 64-bit constants MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With gcc 4.1.2: sound/soc/codecs/rt1305.c: In function ‘rt1305_calibrate’: sound/soc/codecs/rt1305.c:1069: warning: integer constant is too large for ‘long’ type sound/soc/codecs/rt1305.c:1086: warning: integer constant is too large for ‘long’ type Add the missing "ULL" suffixes to fix this. Fixes: 29bc643ddd7efb74 ("ASoC: rt1305: Add RT1305/RT1306 amplifier driver") Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/codecs/rt1305.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index f4c8c45f4010..421b8fb2fa04 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0l = 562949953421312; + r0l = 562949953421312ULL; if (rhl != 0) do_div(r0l, rhl); pr_debug("Left_r0 = 0x%llx\n", r0l); @@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0r = 562949953421312; + r0r = 562949953421312ULL; if (rhl != 0) do_div(r0r, rhl); pr_debug("Right_r0 = 0x%llx\n", r0r); From d5a1826c32fa2ec2b161a89df904c6977f7ec44c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 8 Jun 2018 10:19:25 +0200 Subject: [PATCH 019/529] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Vi10 tablet Add a quirk for the Chuwi Vi10 tablet, this tablet uses IN1 for the internal mic rather then the default IN3 and it uses JD2 rather then JD1 for its jack-detect switch. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 33065ba294a9..5c4f9ea40f57 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -463,6 +463,22 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Vi10 (CWI505) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S165"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), From 62c2c9fcac4341d306dda4cf400b77e7e124480a Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:12 +0900 Subject: [PATCH 020/529] ASoC: simple-card-utils: move hp and mic detect gpios from simple-card This patch moves headphone and microphone jack detection gpios from simple-card driver. It is preparing for using this feature from other drivers. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 15 +++++++ sound/soc/generic/simple-card-utils.c | 59 ++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 64 --------------------------- 3 files changed, 74 insertions(+), 64 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 7e25afce6566..f82acef3b992 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -12,6 +12,11 @@ #include +#define asoc_simple_card_init_hp(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 1, prefix) +#define asoc_simple_card_init_mic(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 0, prefix) + struct asoc_simple_dai { const char *name; unsigned int sysclk; @@ -28,6 +33,12 @@ struct asoc_simple_card_data { u32 convert_channels; }; +struct asoc_simple_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_pin pin; + struct snd_soc_jack_gpio gpio; +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -107,4 +118,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, char *prefix); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3751a07de6aa..4398c9580929 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -8,9 +8,13 @@ * published by the Free Software Foundation. */ #include +#include +#include #include #include +#include #include +#include #include void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, @@ -419,6 +423,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) +{ + struct device *dev = card->dev; + enum of_gpio_flags flags; + char prop[128]; + char *pin_name; + char *gpio_name; + int mask; + int det; + + if (!prefix) + prefix = ""; + + sjack->gpio.gpio = -ENOENT; + + if (is_hp) { + snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + pin_name = "Headphones"; + gpio_name = "Headphone detection"; + mask = SND_JACK_HEADPHONE; + } else { + snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + pin_name = "Mic Jack"; + gpio_name = "Mic detection"; + mask = SND_JACK_MICROPHONE; + } + + det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); + if (det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (gpio_is_valid(det)) { + sjack->pin.pin = pin_name; + sjack->pin.mask = mask; + + sjack->gpio.name = gpio_name; + sjack->gpio.report = mask; + sjack->gpio.gpio = det; + sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.debounce_time = 150; + + snd_soc_card_jack_new(card, pin_name, mask, + &sjack->jack, + &sjack->pin, 1); + + snd_soc_jack_add_gpios(&sjack->jack, 1, + &sjack->gpio); + } + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b374af86a6e..a6477a022156 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,23 +10,14 @@ */ #include #include -#include #include #include -#include #include #include -#include #include #include #include -struct asoc_simple_jack { - struct snd_soc_jack jack; - struct snd_soc_jack_pin pin; - struct snd_soc_jack_gpio gpio; -}; - struct simple_card_data { struct snd_soc_card snd_card; struct simple_dai_props { @@ -49,61 +40,6 @@ struct simple_card_data { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -#define asoc_simple_card_init_hp(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 0, prefix) -static int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) -{ - struct device *dev = card->dev; - enum of_gpio_flags flags; - char prop[128]; - char *pin_name; - char *gpio_name; - int mask; - int det; - - sjack->gpio.gpio = -ENOENT; - - if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); - pin_name = "Headphones"; - gpio_name = "Headphone detection"; - mask = SND_JACK_HEADPHONE; - } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); - pin_name = "Mic Jack"; - gpio_name = "Mic detection"; - mask = SND_JACK_MICROPHONE; - } - - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; - - if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; - - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; - - snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); - - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); - } - - return 0; -} - static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; From 8d1bd113a194407f9ad083403ea1cf92108edf5c Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:13 +0900 Subject: [PATCH 021/529] ASoC: simple-card: move hp and mic detection to soc_card probe This patch moves headphone and microphone detection to probe() of snd_soc_card from init() of snd_soc_dai_link. This is because init() is called (and an input device /dev/input/eventX is created too) twice or above if simple card has two or more DAI links. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a6477a022156..c5b6e04cd926 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -149,14 +149,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); - if (ret < 0) - return ret; - return 0; } @@ -350,6 +342,22 @@ card_parse_end: return ret; } +static int asoc_simple_soc_card_probe(struct snd_soc_card *card) +{ + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -385,6 +393,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = num; + card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { From f6de35cc145fb55d842db94e74841ecd8382e012 Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:14 +0900 Subject: [PATCH 022/529] ASoC: audio-graph-card: add hp and mic detect gpios same as simple-card This patch adds headphone and microphone jack detection gpios as same as simple-card driver. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index d93bacacbd5b..a2a3e630f11c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -21,7 +21,6 @@ #include #include #include -#include #include struct graph_card_data { @@ -32,6 +31,8 @@ struct graph_card_data { unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct gpio_desc *pa_gpio; }; @@ -278,6 +279,22 @@ static int asoc_graph_get_dais_count(struct device *dev) return count; } +static int asoc_graph_soc_card_probe(struct snd_soc_card *card) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; @@ -319,6 +336,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->num_links = num; card->dapm_widgets = asoc_graph_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { From a0d847c380ba47886fcca4168698eef51c69f109 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:51:46 +0000 Subject: [PATCH 023/529] ASoC: rsnd: add rsnd_daidrv_get() rsnd priv has many parameters. On __rsnd_dai_probe() it uses rsnd_rdai_get() to get rdai pointer, but is using priv->daidrv directly to get daidrvhv, but it is confusable for reader. This patch adds rsnd_daidrv_get() to get daidrv from priv. Now reader can understand that rdai and daidrv are related. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eac22fef4543..6091e0916085 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -548,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +static struct snd_soc_dai_driver +*rsnd_daidrv_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= rsnd_rdai_nr(priv))) + return NULL; + + return priv->daidrv + id; +} + #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai) static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) { @@ -1033,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, int io_i; rdai = rsnd_rdai_get(priv, dai_i); - drv = priv->daidrv + dai_i; + drv = rsnd_daidrv_get(priv, dai_i); io_playback = &rdai->playback; io_capture = &rdai->capture; From d5c4e972d512ae0b59108ca92b9b35bc5cf5c14e Mon Sep 17 00:00:00 2001 From: Rohit Kumar Date: Wed, 6 Jun 2018 14:25:24 +0530 Subject: [PATCH 024/529] ASoC: qcom: apq8096: set card as device drvdata snd_soc_card is retrieved as device drvdata during unbind(). Set it as drvdata during bind() to avoid memory corruption during unbind(). Signed-off-by: Rohit kumar Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 561cd429e6f2..239b8cb77bdb 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -140,6 +140,7 @@ static int apq8096_bind(struct device *dev) component_bind_all(dev, card); card->dev = dev; + dev_set_drvdata(dev, card); ret = apq8096_sbc_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); From 510e419cb85798915c6426c496a164ec6328d1a7 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 10:35:04 +0100 Subject: [PATCH 025/529] ASoC: twl6040: make pointer dmic_codec_dev static The pointer dmic_codec_dev is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warning: warning: symbol 'dmic_codec_dev' was not declared. Should it be static? Signed-off-by: Colin Ian King Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 15ccbf479c96..d5ae9eb8c756 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -40,7 +40,7 @@ struct abe_twl6040 { int mclk_freq; /* MCLK frequency speed for twl6040 */ }; -struct platform_device *dmic_codec_dev; +static struct platform_device *dmic_codec_dev; static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) From e380be7c557c4ddb1cb71404c10e0bfb3daf4644 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 10:57:19 +0100 Subject: [PATCH 026/529] ASoC: ak5558: make two structures static The structure ak5558_pm and soc_codec_dev_ak5558 are local to the source and do not need to be in global scope, so make them static. Also make soc_codec_dev_ak5558 static. Cleans up sparse warnings: warning: symbol 'ak5558_pm' was not declared. Should it be static? warning: symbol 'soc_codec_dev_ak5558' was not declared. Should it be static? Signed-off-by: Colin Ian King Reviewed-by: Daniel Baluta Signed-off-by: Mark Brown --- sound/soc/codecs/ak5558.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index f4ed5cc40661..448bb90c9c8e 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return regcache_sync(ak5558->regmap); } -const struct dev_pm_ops ak5558_pm = { +static const struct dev_pm_ops ak5558_pm = { SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; -struct snd_soc_component_driver soc_codec_dev_ak5558 = { +static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .probe = ak5558_probe, .remove = ak5558_remove, .controls = ak5558_snd_controls, From 62624f72592b1d8e756e699cd6ade2be379f95a9 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 11:31:23 +0100 Subject: [PATCH 027/529] ASoC: ak4458: make structure soc_codec_dev_ak4458 static const The structure soc_codec_dev_ak4458 is local to the source and do not need to be in global scope and can be const, make it static const. Cleans up sparse warnings: warning: symbol 'soc_codec_dev_ak4458' was not declared. Should it be static? Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 31ec0ba2e639..299ada4dfaa0 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ -struct snd_soc_component_driver soc_codec_dev_ak4458 = { +static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { .probe = ak4458_probe, .remove = ak4458_remove, .controls = ak4458_snd_controls, From 58f7d470c85801f55bbae2d3c93fe1a3d34aa143 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:45:40 -0500 Subject: [PATCH 028/529] ASoC: TSCS42xx: Add mic bias boost control Add mic bias boost control Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d18ff17719cc..743194052fc4 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -644,6 +644,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Input Channel Map */ SOC_ENUM("Input Channel Map", ch_map_select_enum), + /* Mic Bias */ + SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), From 19d996cc3ad698379bcd8482dabe78abe3dc8d96 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:46:28 -0500 Subject: [PATCH 029/529] ASoC: TSCS42xx: Remove Playback/Capture in names These aren't needed and some userspace apps don't work consistently with them. Remove Playback/Capture from control names Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 743194052fc4..5d596e4dab0c 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -625,19 +625,19 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Volumes */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR, FB_HPVOLL, 0x7F, 0, hpvol_scale), - SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR, FB_SPKVOLL, 0x7F, 0, spkvol_scale), - SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR, FB_DACVOLL, 0xFF, 0, dacvol_scale), - SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR, FB_ADCVOLL, 0xFF, 0, adcvol_scale), - SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR, FB_INVOLL, 0x3F, 0, invol_scale), /* INSEL */ - SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR, FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, 0, mic_boost_scale), @@ -736,9 +736,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0), SOC_SINGLE("Comp Switch", R_CLECTL, FB_CLECTL_COMP_EN, 1, 0), - SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("CLE Make-Up Gain Volume", R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("Comp Thresh Playback Volume", + SOC_SINGLE_TLV("Comp Thresh Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), @@ -769,9 +769,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC1 Phase Invert Switch", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), @@ -781,9 +781,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC2 Phase Invert Switch", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), @@ -793,9 +793,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC3 Phase Invert Switch", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), From 53af408cd9f2d1c66bc081c1d82797bfe44af3e5 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:47:02 -0500 Subject: [PATCH 030/529] ASoC: TSCS42xx: Add headphone auto switching Add headphone auto switching controls Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 6 ++++++ sound/soc/codecs/tscs42xx.h | 8 ++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 5d596e4dab0c..7396a6e5277e 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -647,6 +647,12 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Mic Bias */ SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + /* Headphone Auto Switching */ + SOC_SINGLE("Headphone Auto Switching Switch", + R_CTL, FB_CTL_HPSWEN, 1, 0), + SOC_SINGLE("Headphone Detect Polarity Toggle Switch", + R_CTL, FB_CTL_HPSWPOL, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index 814c8f3c4a68..6b3a21081635 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -34,6 +34,7 @@ enum { #define R_DACSR 0x19 #define R_PWRM1 0x1A #define R_PWRM2 0x1B +#define R_CTL 0x1C #define R_CONFIG0 0x1F #define R_CONFIG1 0x20 #define R_DMICCTL 0x24 @@ -1110,6 +1111,13 @@ enum { #define RV_PWRM2_VREF_DISABLE \ RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) +/****************************** + * R_CTL (0x1C) * + ******************************/ + +/* Fiel Offsets */ +#define FB_CTL_HPSWEN 7 +#define FB_CTL_HPSWPOL 6 /****************************** * R_CONFIG0 (0x1F) * From 0ddce71c21f03fd19867c4939d3ca710f37cdf1a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 7 Jun 2018 16:37:38 +0800 Subject: [PATCH 031/529] ASoC: rt5682: add rt5682 codec driver This is the initial codec driver for rt5682. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rt5682.txt | 50 + include/sound/rt5682.h | 40 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682.c | 2682 +++++++++++++++++ sound/soc/codecs/rt5682.h | 1324 ++++++++ 6 files changed, 4104 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 include/sound/rt5682.h create mode 100644 sound/soc/codecs/rt5682.c create mode 100644 sound/soc/codecs/rt5682.h diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt new file mode 100644 index 000000000000..312e9a129530 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -0,0 +1,50 @@ +RT5682 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5682" or "realtek,rt5682i" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5682: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5682 { + compatible = "realtek,rt5682i"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = ; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h new file mode 100644 index 000000000000..0251797ab438 --- /dev/null +++ b/include/sound/rt5682.h @@ -0,0 +1,40 @@ +/* + * linux/sound/rt5682.h -- Platform data for RT5682 + * + * Copyright 2018 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5682_H +#define __LINUX_SND_RT5682_H + +enum rt5682_dmic1_data_pin { + RT5682_DMIC1_NULL, + RT5682_DMIC1_DATA_GPIO2, + RT5682_DMIC1_DATA_GPIO5, +}; + +enum rt5682_dmic1_clk_pin { + RT5682_DMIC1_CLK_GPIO1, + RT5682_DMIC1_CLK_GPIO3, +}; + +enum rt5682_jd_src { + RT5682_JD_NULL, + RT5682_JD1, +}; + +struct rt5682_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682_dmic1_data_pin dmic1_data_pin; + enum rt5682_dmic1_clk_pin dmic1_clk_pin; + enum rt5682_jd_src jd_src; +}; + +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63cf62e9c9aa..f6b8d4bf8796 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,6 +141,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER + select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC @@ -778,6 +779,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default y if SND_SOC_RT5682=y default y if SND_SOC_RT1305=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m @@ -791,6 +793,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m + default m if SND_SOC_RT5682=m default m if SND_SOC_RT1305=m config SND_SOC_RL6347A @@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI tristate default SND_SOC_RT5677 && SPI +config SND_SOC_RT5682 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e023fdf85221..e43d99a039d3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -146,6 +146,7 @@ snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o +snd-soc-rt5682-objs := rt5682.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -405,6 +406,7 @@ obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o +obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c new file mode 100644 index 000000000000..61a97301bcfa --- /dev/null +++ b/sound/soc/codecs/rt5682.c @@ -0,0 +1,2682 @@ +/* + * rt5682.c -- RT5682 ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rl6231.h" +#include "rt5682.h" + +#define RT5682_NUM_SUPPLIES 3 + +static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_sequence patch_list[] = { + {0x01c1, 0x1000}, +}; + +static const struct reg_default rt5682_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5682_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_CBJ_CTRL_2: + case RT5682_INT_ST_1: + case RT5682_4BTN_IL_CMD_1: + case RT5682_AJD1_CTRL: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_DEVICE_ID: + case RT5682_I2C_MODE: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_EFUSE_CTRL_2: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11: + case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5682_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_VERSION_ID: + case RT5682_VENDOR_ID: + case RT5682_DEVICE_ID: + case RT5682_HP_CTRL_1: + case RT5682_HP_CTRL_2: + case RT5682_HPL_GAIN: + case RT5682_HPR_GAIN: + case RT5682_I2C_CTRL: + case RT5682_CBJ_BST_CTRL: + case RT5682_CBJ_CTRL_1: + case RT5682_CBJ_CTRL_2: + case RT5682_CBJ_CTRL_3: + case RT5682_CBJ_CTRL_4: + case RT5682_CBJ_CTRL_5: + case RT5682_CBJ_CTRL_6: + case RT5682_CBJ_CTRL_7: + case RT5682_DAC1_DIG_VOL: + case RT5682_STO1_ADC_DIG_VOL: + case RT5682_STO1_ADC_BOOST: + case RT5682_HP_IMP_GAIN_1: + case RT5682_HP_IMP_GAIN_2: + case RT5682_SIDETONE_CTRL: + case RT5682_STO1_ADC_MIXER: + case RT5682_AD_DA_MIXER: + case RT5682_STO1_DAC_MIXER: + case RT5682_A_DAC1_MUX: + case RT5682_DIG_INF2_DATA: + case RT5682_REC_MIXER: + case RT5682_CAL_REC: + case RT5682_ALC_BACK_GAIN: + case RT5682_PWR_DIG_1: + case RT5682_PWR_DIG_2: + case RT5682_PWR_ANLG_1: + case RT5682_PWR_ANLG_2: + case RT5682_PWR_ANLG_3: + case RT5682_PWR_MIXER: + case RT5682_PWR_VOL: + case RT5682_CLK_DET: + case RT5682_RESET_LPF_CTRL: + case RT5682_RESET_HPF_CTRL: + case RT5682_DMIC_CTRL_1: + case RT5682_I2S1_SDP: + case RT5682_I2S2_SDP: + case RT5682_ADDA_CLK_1: + case RT5682_ADDA_CLK_2: + case RT5682_I2S1_F_DIV_CTRL_1: + case RT5682_I2S1_F_DIV_CTRL_2: + case RT5682_TDM_CTRL: + case RT5682_TDM_ADDA_CTRL_1: + case RT5682_TDM_ADDA_CTRL_2: + case RT5682_DATA_SEL_CTRL_1: + case RT5682_TDM_TCON_CTRL: + case RT5682_GLB_CLK: + case RT5682_PLL_CTRL_1: + case RT5682_PLL_CTRL_2: + case RT5682_PLL_TRACK_1: + case RT5682_PLL_TRACK_2: + case RT5682_PLL_TRACK_3: + case RT5682_PLL_TRACK_4: + case RT5682_PLL_TRACK_5: + case RT5682_PLL_TRACK_6: + case RT5682_PLL_TRACK_11: + case RT5682_SDW_REF_CLK: + case RT5682_DEPOP_1: + case RT5682_DEPOP_2: + case RT5682_HP_CHARGE_PUMP_1: + case RT5682_HP_CHARGE_PUMP_2: + case RT5682_MICBIAS_1: + case RT5682_MICBIAS_2: + case RT5682_PLL_TRACK_12: + case RT5682_PLL_TRACK_14: + case RT5682_PLL2_CTRL_1: + case RT5682_PLL2_CTRL_2: + case RT5682_PLL2_CTRL_3: + case RT5682_PLL2_CTRL_4: + case RT5682_RC_CLK_CTRL: + case RT5682_I2S_M_CLK_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_2: + case RT5682_EQ_CTRL_1: + case RT5682_EQ_CTRL_2: + case RT5682_IRQ_CTRL_1: + case RT5682_IRQ_CTRL_2: + case RT5682_IRQ_CTRL_3: + case RT5682_IRQ_CTRL_4: + case RT5682_INT_ST_1: + case RT5682_GPIO_CTRL_1: + case RT5682_GPIO_CTRL_2: + case RT5682_GPIO_CTRL_3: + case RT5682_HP_AMP_DET_CTRL_1: + case RT5682_HP_AMP_DET_CTRL_2: + case RT5682_MID_HP_AMP_DET: + case RT5682_LOW_HP_AMP_DET: + case RT5682_DELAY_BUF_CTRL: + case RT5682_SV_ZCD_1: + case RT5682_SV_ZCD_2: + case RT5682_IL_CMD_1: + case RT5682_IL_CMD_2: + case RT5682_IL_CMD_3: + case RT5682_IL_CMD_4: + case RT5682_IL_CMD_5: + case RT5682_IL_CMD_6: + case RT5682_4BTN_IL_CMD_1: + case RT5682_4BTN_IL_CMD_2: + case RT5682_4BTN_IL_CMD_3: + case RT5682_4BTN_IL_CMD_4: + case RT5682_4BTN_IL_CMD_5: + case RT5682_4BTN_IL_CMD_6: + case RT5682_4BTN_IL_CMD_7: + case RT5682_ADC_STO1_HP_CTRL_1: + case RT5682_ADC_STO1_HP_CTRL_2: + case RT5682_AJD1_CTRL: + case RT5682_JD1_THD: + case RT5682_JD2_THD: + case RT5682_JD_CTRL_1: + case RT5682_DUMMY_1: + case RT5682_DUMMY_2: + case RT5682_DUMMY_3: + case RT5682_DAC_ADC_DIG_VOL1: + case RT5682_BIAS_CUR_CTRL_2: + case RT5682_BIAS_CUR_CTRL_3: + case RT5682_BIAS_CUR_CTRL_4: + case RT5682_BIAS_CUR_CTRL_5: + case RT5682_BIAS_CUR_CTRL_6: + case RT5682_BIAS_CUR_CTRL_7: + case RT5682_BIAS_CUR_CTRL_8: + case RT5682_BIAS_CUR_CTRL_9: + case RT5682_BIAS_CUR_CTRL_10: + case RT5682_VREF_REC_OP_FB_CAP_CTRL: + case RT5682_CHARGE_PUMP_1: + case RT5682_DIG_IN_CTRL_1: + case RT5682_PAD_DRIVING_CTRL: + case RT5682_SOFT_RAMP_DEPOP: + case RT5682_CHOP_DAC: + case RT5682_CHOP_ADC: + case RT5682_CALIB_ADC_CTRL: + case RT5682_VOL_TEST: + case RT5682_SPKVDD_DET_STA: + case RT5682_TEST_MODE_CTRL_1: + case RT5682_TEST_MODE_CTRL_2: + case RT5682_TEST_MODE_CTRL_3: + case RT5682_TEST_MODE_CTRL_4: + case RT5682_TEST_MODE_CTRL_5: + case RT5682_PLL1_INTERNAL: + case RT5682_PLL2_INTERNAL: + case RT5682_STO_NG2_CTRL_1: + case RT5682_STO_NG2_CTRL_2: + case RT5682_STO_NG2_CTRL_3: + case RT5682_STO_NG2_CTRL_4: + case RT5682_STO_NG2_CTRL_5: + case RT5682_STO_NG2_CTRL_6: + case RT5682_STO_NG2_CTRL_7: + case RT5682_STO_NG2_CTRL_8: + case RT5682_STO_NG2_CTRL_9: + case RT5682_STO_NG2_CTRL_10: + case RT5682_STO1_DAC_SIL_DET: + case RT5682_SIL_PSV_CTRL1: + case RT5682_SIL_PSV_CTRL2: + case RT5682_SIL_PSV_CTRL3: + case RT5682_SIL_PSV_CTRL4: + case RT5682_SIL_PSV_CTRL5: + case RT5682_HP_IMP_SENS_CTRL_01: + case RT5682_HP_IMP_SENS_CTRL_02: + case RT5682_HP_IMP_SENS_CTRL_03: + case RT5682_HP_IMP_SENS_CTRL_04: + case RT5682_HP_IMP_SENS_CTRL_05: + case RT5682_HP_IMP_SENS_CTRL_06: + case RT5682_HP_IMP_SENS_CTRL_07: + case RT5682_HP_IMP_SENS_CTRL_08: + case RT5682_HP_IMP_SENS_CTRL_09: + case RT5682_HP_IMP_SENS_CTRL_10: + case RT5682_HP_IMP_SENS_CTRL_11: + case RT5682_HP_IMP_SENS_CTRL_12: + case RT5682_HP_IMP_SENS_CTRL_13: + case RT5682_HP_IMP_SENS_CTRL_14: + case RT5682_HP_IMP_SENS_CTRL_15: + case RT5682_HP_IMP_SENS_CTRL_16: + case RT5682_HP_IMP_SENS_CTRL_17: + case RT5682_HP_IMP_SENS_CTRL_18: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_HP_IMP_SENS_CTRL_20: + case RT5682_HP_IMP_SENS_CTRL_21: + case RT5682_HP_IMP_SENS_CTRL_22: + case RT5682_HP_IMP_SENS_CTRL_23: + case RT5682_HP_IMP_SENS_CTRL_24: + case RT5682_HP_IMP_SENS_CTRL_25: + case RT5682_HP_IMP_SENS_CTRL_26: + case RT5682_HP_IMP_SENS_CTRL_27: + case RT5682_HP_IMP_SENS_CTRL_28: + case RT5682_HP_IMP_SENS_CTRL_29: + case RT5682_HP_IMP_SENS_CTRL_30: + case RT5682_HP_IMP_SENS_CTRL_31: + case RT5682_HP_IMP_SENS_CTRL_32: + case RT5682_HP_IMP_SENS_CTRL_33: + case RT5682_HP_IMP_SENS_CTRL_34: + case RT5682_HP_IMP_SENS_CTRL_35: + case RT5682_HP_IMP_SENS_CTRL_36: + case RT5682_HP_IMP_SENS_CTRL_37: + case RT5682_HP_IMP_SENS_CTRL_38: + case RT5682_HP_IMP_SENS_CTRL_39: + case RT5682_HP_IMP_SENS_CTRL_40: + case RT5682_HP_IMP_SENS_CTRL_41: + case RT5682_HP_IMP_SENS_CTRL_42: + case RT5682_HP_IMP_SENS_CTRL_43: + case RT5682_HP_LOGIC_CTRL_1: + case RT5682_HP_LOGIC_CTRL_2: + case RT5682_HP_LOGIC_CTRL_3: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_HP_CALIB_CTRL_2: + case RT5682_HP_CALIB_CTRL_3: + case RT5682_HP_CALIB_CTRL_4: + case RT5682_HP_CALIB_CTRL_5: + case RT5682_HP_CALIB_CTRL_6: + case RT5682_HP_CALIB_CTRL_7: + case RT5682_HP_CALIB_CTRL_9: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_HP_CALIB_CTRL_11: + case RT5682_HP_CALIB_STA_1: + case RT5682_HP_CALIB_STA_2: + case RT5682_HP_CALIB_STA_3: + case RT5682_HP_CALIB_STA_4: + case RT5682_HP_CALIB_STA_5: + case RT5682_HP_CALIB_STA_6: + case RT5682_HP_CALIB_STA_7: + case RT5682_HP_CALIB_STA_8: + case RT5682_HP_CALIB_STA_9: + case RT5682_HP_CALIB_STA_10: + case RT5682_HP_CALIB_STA_11: + case RT5682_SAR_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_3: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_5: + case RT5682_SAR_IL_CMD_6: + case RT5682_SAR_IL_CMD_7: + case RT5682_SAR_IL_CMD_8: + case RT5682_SAR_IL_CMD_9: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_SAR_IL_CMD_12: + case RT5682_SAR_IL_CMD_13: + case RT5682_EFUSE_CTRL_1: + case RT5682_EFUSE_CTRL_2: + case RT5682_EFUSE_CTRL_3: + case RT5682_EFUSE_CTRL_4: + case RT5682_EFUSE_CTRL_5: + case RT5682_EFUSE_CTRL_6: + case RT5682_EFUSE_CTRL_7: + case RT5682_EFUSE_CTRL_8: + case RT5682_EFUSE_CTRL_9: + case RT5682_EFUSE_CTRL_10: + case RT5682_EFUSE_CTRL_11: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_DRC1_CTRL_0: + case RT5682_DRC1_CTRL_1: + case RT5682_DRC1_CTRL_2: + case RT5682_DRC1_CTRL_3: + case RT5682_DRC1_CTRL_4: + case RT5682_DRC1_CTRL_5: + case RT5682_DRC1_CTRL_6: + case RT5682_DRC1_HARD_LMT_CTRL_1: + case RT5682_DRC1_HARD_LMT_CTRL_2: + case RT5682_DRC1_PRIV_1: + case RT5682_DRC1_PRIV_2: + case RT5682_DRC1_PRIV_3: + case RT5682_DRC1_PRIV_4: + case RT5682_DRC1_PRIV_5: + case RT5682_DRC1_PRIV_6: + case RT5682_DRC1_PRIV_7: + case RT5682_DRC1_PRIV_8: + case RT5682_EQ_AUTO_RCV_CTRL1: + case RT5682_EQ_AUTO_RCV_CTRL2: + case RT5682_EQ_AUTO_RCV_CTRL3: + case RT5682_EQ_AUTO_RCV_CTRL4: + case RT5682_EQ_AUTO_RCV_CTRL5: + case RT5682_EQ_AUTO_RCV_CTRL6: + case RT5682_EQ_AUTO_RCV_CTRL7: + case RT5682_EQ_AUTO_RCV_CTRL8: + case RT5682_EQ_AUTO_RCV_CTRL9: + case RT5682_EQ_AUTO_RCV_CTRL10: + case RT5682_EQ_AUTO_RCV_CTRL11: + case RT5682_EQ_AUTO_RCV_CTRL12: + case RT5682_EQ_AUTO_RCV_CTRL13: + case RT5682_ADC_L_EQ_LPF1_A1: + case RT5682_R_EQ_LPF1_A1: + case RT5682_L_EQ_LPF1_H0: + case RT5682_R_EQ_LPF1_H0: + case RT5682_L_EQ_BPF1_A1: + case RT5682_R_EQ_BPF1_A1: + case RT5682_L_EQ_BPF1_A2: + case RT5682_R_EQ_BPF1_A2: + case RT5682_L_EQ_BPF1_H0: + case RT5682_R_EQ_BPF1_H0: + case RT5682_L_EQ_BPF2_A1: + case RT5682_R_EQ_BPF2_A1: + case RT5682_L_EQ_BPF2_A2: + case RT5682_R_EQ_BPF2_A2: + case RT5682_L_EQ_BPF2_H0: + case RT5682_R_EQ_BPF2_H0: + case RT5682_L_EQ_BPF3_A1: + case RT5682_R_EQ_BPF3_A1: + case RT5682_L_EQ_BPF3_A2: + case RT5682_R_EQ_BPF3_A2: + case RT5682_L_EQ_BPF3_H0: + case RT5682_R_EQ_BPF3_H0: + case RT5682_L_EQ_BPF4_A1: + case RT5682_R_EQ_BPF4_A1: + case RT5682_L_EQ_BPF4_A2: + case RT5682_R_EQ_BPF4_A2: + case RT5682_L_EQ_BPF4_H0: + case RT5682_R_EQ_BPF4_H0: + case RT5682_L_EQ_HPF1_A1: + case RT5682_R_EQ_HPF1_A1: + case RT5682_L_EQ_HPF1_H0: + case RT5682_R_EQ_HPF1_H0: + case RT5682_L_EQ_PRE_VOL: + case RT5682_R_EQ_PRE_VOL: + case RT5682_L_EQ_POST_VOL: + case RT5682_R_EQ_POST_VOL: + case RT5682_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5682_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum, + RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select); + +static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); + +static void rt5682_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5682_RESET, 0); + regmap_write(regmap, RT5682_I2C_MODE, 1); +} +/** + * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5682_CLK_SEL_SYS: + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src); + +static int rt5682_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + + return btn_type; +} + +static void rt5682_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, + RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE); + } +} + +/** + * rt5682_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2) + & RT5682_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5682->jack_type = SND_JACK_HEADSET; + rt5682_enable_push_button_irq(component, true); + break; + default: + rt5682->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5682_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5682->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); + return rt5682->jack_type; +} + +static irqreturn_t rt5682_irq(int irq, void *data) +{ + struct rt5682_priv *rt5682 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5682_jd_check_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK) { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682->jd_check_work, 500); + } +} + +static int rt5682_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5682->hs_jack = hs_jack; + + return 0; +} + +static void rt5682_jack_detect_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = + container_of(work, struct rt5682_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682->component) + usleep_range(10000, 15000); + + while (!rt5682->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682->calibrate_mutex); + + val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682->jack_type == 0) { + /* jack was out, report jack type */ + rt5682->jack_type = + rt5682_headset_detect(rt5682->component, 1); + } else { + /* jack is already in, report button event */ + rt5682->jack_type = SND_JACK_HEADSET; + btn_type = rt5682_button_detect(rt5682->component); + /** + * rt5682 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5682->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + } + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + + mutex_unlock(&rt5682->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5682_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, + RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, + RT5682_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, + RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5682_div_sel(struct rt5682_priv *rt5682, + int target, const int div[], int size) +{ + int i; + + if (rt5682->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5682->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682->sysclk) + return i; + if (target * div[i + 1] > rt5682->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5682->sysclk); + return i; + } + } + + if (target * div[i] < rt5682->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5682->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, + RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int ref, val, reg, sft, mask, idx = -EINVAL; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & + RT5682_GP4_PIN_MASK; + if (w->shift == RT5682_PWR_ADC_S1F_BIT && + val == RT5682_GP4_PIN_ADCDAT2) + ref = 256 * rt5682->lrck[RT5682_AIF2]; + else + ref = 256 * rt5682->lrck[RT5682_AIF1]; + + idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682_PWR_ADC_S1F_BIT) { + reg = RT5682_PLL_TRACK_3; + sft = RT5682_ADC_OSR_SFT; + mask = RT5682_ADC_OSR_MASK; + } else { + reg = RT5682_PLL_TRACK_2; + sft = RT5682_DAC_OSR_SFT; + mask = RT5682_DAC_OSR_MASK; + } + + snd_soc_component_update_bits(component, reg, + RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + mask, idx << sft); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682_ADC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_3; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + case RT5682_DAC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_2; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER, + RT5682_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5682_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum, + RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK, + rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACL1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACR1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_R_MUTE_SFT, 1, 1); + +static int rt5682_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5682_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, + RT5682_NG2_EN_MASK, RT5682_NG2_EN); + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, + RT5682_PWR_FV1); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, + RT5682_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5682_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5682_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, + RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK, + rt5682_adcdat_pin_select, rt5682_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, + ARRAY_SIZE(rt5682_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2, + RT5682_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC, + RT5682_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix, + ARRAY_SIZE(rt5682_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, + RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, + ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5682_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC, + RT5682_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, + RT5682_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, + RT5682_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, + RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET, + RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET, + RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET, + RT5682_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "CLKDET"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, RT5682_TDM_EN); + else + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682_TDM_TX_CH_4; + val |= RT5682_TDM_RX_CH_4; + break; + case 6: + val |= RT5682_TDM_TX_CH_6; + val |= RT5682_TDM_RX_CH_6; + break; + case 8: + val |= RT5682_TDM_TX_CH_8; + val |= RT5682_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_CTRL, + RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682_TDM_CL_16; + cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682_TDM_CL_20; + cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682_TDM_CL_24; + cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682_TDM_CL_32; + cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl); + + return 0; +} + + +static int rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5682->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5682->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682_I2S1_DL_20; + len_2 |= RT5682_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682_I2S1_DL_24; + len_2 |= RT5682_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682_I2S1_DL_32; + len_2 |= RT5682_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682_I2S2_DL_8; + len_2 |= RT5682_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_DL_MASK, len_1); + if (rt5682->master[RT5682_AIF1]) { + snd_soc_component_update_bits(component, + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, + pre_div << RT5682_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_DIS); + break; + case RT5682_AIF2: + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_DL_MASK, len_2); + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK, + pre_div << RT5682_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682_I2S_BP_INV; + tdm_ctrl |= RT5682_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV | + RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682_I2S_DF_LEFT; + tdm_ctrl |= RT5682_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682_I2S_DF_PCM_A; + tdm_ctrl |= RT5682_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682_I2S_DF_PCM_B; + tdm_ctrl |= RT5682_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK | + RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK | + RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK, + tdm_ctrl | rt5682->master[dai->id]); + break; + case RT5682_AIF2: + if (rt5682->master[dai->id] == 0) + reg_val |= RT5682_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK | + RT5682_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682_SCLK_S_MCLK: + reg_val |= RT5682_SCLK_SRC_MCLK; + src = RT5682_CLK_SRC_MCLK; + break; + case RT5682_SCLK_S_PLL1: + reg_val |= RT5682_SCLK_SRC_PLL1; + src = RT5682_CLK_SRC_PLL1; + break; + case RT5682_SCLK_S_PLL2: + reg_val |= RT5682_SCLK_SRC_PLL2; + src = RT5682_CLK_SRC_PLL2; + break; + case RT5682_SCLK_S_RCCLK: + reg_val |= RT5682_SCLK_SRC_RCCLK; + src = RT5682_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, reg_val); + + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK, + src << RT5682_I2S2_SRC_SFT); + } + + rt5682->sysclk = freq; + rt5682->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5682_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5682->pll_src && freq_in == rt5682->pll_in && + freq_out == rt5682->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5682->pll_in = 0; + rt5682->pll_out = 0; + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + + rt5682->pll_in = freq_in; + rt5682->pll_out = freq_out; + rt5682->pll_src = source; + + return 0; +} + +static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, + RT5682_PWR_MB | RT5682_PWR_BG); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5682_probe(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->component = component; + + return 0; +} + +static void rt5682_remove(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_PM +static int rt5682_suspend(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + return 0; +} + +static int rt5682_resume(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} +#else +#define rt5682_suspend NULL +#define rt5682_resume NULL +#endif + +#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_tdm_slot = rt5682_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_bclk_ratio = rt5682_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5682_dai[] = { + { + .name = "rt5682-aif1", + .id = RT5682_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif1_dai_ops, + }, + { + .name = "rt5682-aif2", + .id = RT5682_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5682 = { + .probe = rt5682_probe, + .remove = rt5682_remove, + .suspend = rt5682_suspend, + .resume = rt5682_resume, + .set_bias_level = rt5682_set_bias_level, + .controls = rt5682_snd_controls, + .num_controls = ARRAY_SIZE(rt5682_snd_controls), + .dapm_widgets = rt5682_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets), + .dapm_routes = rt5682_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes), + .set_sysclk = rt5682_set_component_sysclk, + .set_pll = rt5682_set_component_pll, + .set_jack = rt5682_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5682_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5682_i2c_id[] = { + {"rt5682", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); + +static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) +{ + + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682->pdata.jd_src); + + rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5682_calibrate(struct rt5682_priv *rt5682) +{ + int value, count; + + mutex_lock(&rt5682->calibrate_mutex); + + rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); + regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); + regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5682->calibrate_mutex); + +} + +static int rt5682_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682_priv *rt5682; + int i, ret; + unsigned int val; + + rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv), + GFP_KERNEL); + + if (rt5682 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682); + + if (pdata) + rt5682->pdata = *pdata; + else + rt5682_parse_dt(rt5682, &i2c->dev); + + rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++) + rt5682->supplies[i].supply = rt5682_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + rt5682_reset(rt5682->regmap); + + rt5682_calibrate(rt5682); + + ret = regmap_register_patch(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) { + switch (rt5682->pdata.dmic1_data_pin) { + case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA); + break; + + case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5682->pdata.dmic1_clk_pin) { + case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK); + break; + + case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, + RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682->jd_check_work, + rt5682_jd_check_handler); + + mutex_init(&rt5682->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5682", rt5682); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); +} + +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + +static void rt5682_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5682_of_match[] = { + {.compatible = "realtek,rt5682i"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5682_acpi_match[] = { + {"10EC5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); +#endif + +static struct i2c_driver rt5682_i2c_driver = { + .driver = { + .name = "rt5682", + .of_match_table = of_match_ptr(rt5682_of_match), + .acpi_match_table = ACPI_PTR(rt5682_acpi_match), + }, + .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, + .shutdown = rt5682_i2c_shutdown, + .id_table = rt5682_i2c_id, +}; +module_i2c_driver(rt5682_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h new file mode 100644 index 000000000000..8068140ebe3f --- /dev/null +++ b/sound/soc/codecs/rt5682.h @@ -0,0 +1,1324 @@ +/* + * rt5682.h -- RT5682/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5682_H__ +#define __RT5682_H__ + +#include + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5682_RESET 0x0000 +#define RT5682_VERSION_ID 0x00fd +#define RT5682_VENDOR_ID 0x00fe +#define RT5682_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682_HP_CTRL_1 0x0002 +#define RT5682_HP_CTRL_2 0x0003 +#define RT5682_HPL_GAIN 0x0005 +#define RT5682_HPR_GAIN 0x0006 + +#define RT5682_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682_CBJ_BST_CTRL 0x000b +#define RT5682_CBJ_CTRL_1 0x0010 +#define RT5682_CBJ_CTRL_2 0x0011 +#define RT5682_CBJ_CTRL_3 0x0012 +#define RT5682_CBJ_CTRL_4 0x0013 +#define RT5682_CBJ_CTRL_5 0x0014 +#define RT5682_CBJ_CTRL_6 0x0015 +#define RT5682_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5682_DAC1_DIG_VOL 0x0019 +#define RT5682_STO1_ADC_DIG_VOL 0x001c +#define RT5682_STO1_ADC_BOOST 0x001f +#define RT5682_HP_IMP_GAIN_1 0x0022 +#define RT5682_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682_SIDETONE_CTRL 0x0024 +#define RT5682_STO1_ADC_MIXER 0x0026 +#define RT5682_AD_DA_MIXER 0x0029 +#define RT5682_STO1_DAC_MIXER 0x002a +#define RT5682_A_DAC1_MUX 0x002b +#define RT5682_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682_REC_MIXER 0x003c +#define RT5682_CAL_REC 0x0044 +#define RT5682_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5682_PWR_DIG_1 0x0061 +#define RT5682_PWR_DIG_2 0x0062 +#define RT5682_PWR_ANLG_1 0x0063 +#define RT5682_PWR_ANLG_2 0x0064 +#define RT5682_PWR_ANLG_3 0x0065 +#define RT5682_PWR_MIXER 0x0066 +#define RT5682_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5682_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682_RESET_LPF_CTRL 0x006c +#define RT5682_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5682_I2S1_SDP 0x0070 +#define RT5682_I2S2_SDP 0x0071 +#define RT5682_ADDA_CLK_1 0x0073 +#define RT5682_ADDA_CLK_2 0x0074 +#define RT5682_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682_TDM_CTRL 0x0079 +#define RT5682_TDM_ADDA_CTRL_1 0x007a +#define RT5682_TDM_ADDA_CTRL_2 0x007b +#define RT5682_DATA_SEL_CTRL_1 0x007c +#define RT5682_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5682_GLB_CLK 0x0080 +#define RT5682_PLL_CTRL_1 0x0081 +#define RT5682_PLL_CTRL_2 0x0082 +#define RT5682_PLL_TRACK_1 0x0083 +#define RT5682_PLL_TRACK_2 0x0084 +#define RT5682_PLL_TRACK_3 0x0085 +#define RT5682_PLL_TRACK_4 0x0086 +#define RT5682_PLL_TRACK_5 0x0087 +#define RT5682_PLL_TRACK_6 0x0088 +#define RT5682_PLL_TRACK_11 0x008c +#define RT5682_SDW_REF_CLK 0x008d +#define RT5682_DEPOP_1 0x008e +#define RT5682_DEPOP_2 0x008f +#define RT5682_HP_CHARGE_PUMP_1 0x0091 +#define RT5682_HP_CHARGE_PUMP_2 0x0092 +#define RT5682_MICBIAS_1 0x0093 +#define RT5682_MICBIAS_2 0x0094 +#define RT5682_PLL_TRACK_12 0x0098 +#define RT5682_PLL_TRACK_14 0x009a +#define RT5682_PLL2_CTRL_1 0x009b +#define RT5682_PLL2_CTRL_2 0x009c +#define RT5682_PLL2_CTRL_3 0x009d +#define RT5682_PLL2_CTRL_4 0x009e +#define RT5682_RC_CLK_CTRL 0x009f +#define RT5682_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5682_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5682_EQ_CTRL_1 0x00ae +#define RT5682_EQ_CTRL_2 0x00af +#define RT5682_IRQ_CTRL_1 0x00b6 +#define RT5682_IRQ_CTRL_2 0x00b7 +#define RT5682_IRQ_CTRL_3 0x00b8 +#define RT5682_IRQ_CTRL_4 0x00b9 +#define RT5682_INT_ST_1 0x00be +#define RT5682_GPIO_CTRL_1 0x00c0 +#define RT5682_GPIO_CTRL_2 0x00c1 +#define RT5682_GPIO_CTRL_3 0x00c2 +#define RT5682_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5682_MID_HP_AMP_DET 0x00d2 +#define RT5682_LOW_HP_AMP_DET 0x00d3 +#define RT5682_DELAY_BUF_CTRL 0x00d4 +#define RT5682_SV_ZCD_1 0x00d9 +#define RT5682_SV_ZCD_2 0x00da +#define RT5682_IL_CMD_1 0x00db +#define RT5682_IL_CMD_2 0x00dc +#define RT5682_IL_CMD_3 0x00dd +#define RT5682_IL_CMD_4 0x00de +#define RT5682_IL_CMD_5 0x00df +#define RT5682_IL_CMD_6 0x00e0 +#define RT5682_4BTN_IL_CMD_1 0x00e2 +#define RT5682_4BTN_IL_CMD_2 0x00e3 +#define RT5682_4BTN_IL_CMD_3 0x00e4 +#define RT5682_4BTN_IL_CMD_4 0x00e5 +#define RT5682_4BTN_IL_CMD_5 0x00e6 +#define RT5682_4BTN_IL_CMD_6 0x00e7 +#define RT5682_4BTN_IL_CMD_7 0x00e8 + +#define RT5682_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682_AJD1_CTRL 0x00f0 +#define RT5682_JD1_THD 0x00f1 +#define RT5682_JD2_THD 0x00f2 +#define RT5682_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682_DUMMY_1 0x00fa +#define RT5682_DUMMY_2 0x00fb +#define RT5682_DUMMY_3 0x00fc + +#define RT5682_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682_BIAS_CUR_CTRL_2 0x010b +#define RT5682_BIAS_CUR_CTRL_3 0x010c +#define RT5682_BIAS_CUR_CTRL_4 0x010d +#define RT5682_BIAS_CUR_CTRL_5 0x010e +#define RT5682_BIAS_CUR_CTRL_6 0x010f +#define RT5682_BIAS_CUR_CTRL_7 0x0110 +#define RT5682_BIAS_CUR_CTRL_8 0x0111 +#define RT5682_BIAS_CUR_CTRL_9 0x0112 +#define RT5682_BIAS_CUR_CTRL_10 0x0113 +#define RT5682_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5682_CHARGE_PUMP_1 0x0125 +#define RT5682_DIG_IN_CTRL_1 0x0132 +#define RT5682_PAD_DRIVING_CTRL 0x0136 +#define RT5682_SOFT_RAMP_DEPOP 0x0138 +#define RT5682_CHOP_DAC 0x013a +#define RT5682_CHOP_ADC 0x013b +#define RT5682_CALIB_ADC_CTRL 0x013c +#define RT5682_VOL_TEST 0x013f +#define RT5682_SPKVDD_DET_STA 0x0142 +#define RT5682_TEST_MODE_CTRL_1 0x0145 +#define RT5682_TEST_MODE_CTRL_2 0x0146 +#define RT5682_TEST_MODE_CTRL_3 0x0147 +#define RT5682_TEST_MODE_CTRL_4 0x0148 +#define RT5682_TEST_MODE_CTRL_5 0x0149 +#define RT5682_PLL1_INTERNAL 0x0150 +#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_STO_NG2_CTRL_1 0x0160 +#define RT5682_STO_NG2_CTRL_2 0x0161 +#define RT5682_STO_NG2_CTRL_3 0x0162 +#define RT5682_STO_NG2_CTRL_4 0x0163 +#define RT5682_STO_NG2_CTRL_5 0x0164 +#define RT5682_STO_NG2_CTRL_6 0x0165 +#define RT5682_STO_NG2_CTRL_7 0x0166 +#define RT5682_STO_NG2_CTRL_8 0x0167 +#define RT5682_STO_NG2_CTRL_9 0x0168 +#define RT5682_STO_NG2_CTRL_10 0x0169 +#define RT5682_STO1_DAC_SIL_DET 0x0190 +#define RT5682_SIL_PSV_CTRL1 0x0194 +#define RT5682_SIL_PSV_CTRL2 0x0195 +#define RT5682_SIL_PSV_CTRL3 0x0197 +#define RT5682_SIL_PSV_CTRL4 0x0198 +#define RT5682_SIL_PSV_CTRL5 0x0199 +#define RT5682_HP_IMP_SENS_CTRL_01 0x01af +#define RT5682_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5682_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5682_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5682_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5682_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5682_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5682_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5682_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5682_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5682_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5682_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5682_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5682_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5682_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5682_HP_IMP_SENS_CTRL_16 0x01be +#define RT5682_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5682_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5682_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5682_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5682_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5682_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5682_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5682_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5682_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5682_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5682_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5682_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5682_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5682_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5682_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5682_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5682_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5682_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5682_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5682_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5682_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5682_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5682_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5682_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5682_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5682_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5682_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5682_HP_LOGIC_CTRL_1 0x01da +#define RT5682_HP_LOGIC_CTRL_2 0x01db +#define RT5682_HP_LOGIC_CTRL_3 0x01dc +#define RT5682_HP_CALIB_CTRL_1 0x01de +#define RT5682_HP_CALIB_CTRL_2 0x01df +#define RT5682_HP_CALIB_CTRL_3 0x01e0 +#define RT5682_HP_CALIB_CTRL_4 0x01e1 +#define RT5682_HP_CALIB_CTRL_5 0x01e2 +#define RT5682_HP_CALIB_CTRL_6 0x01e3 +#define RT5682_HP_CALIB_CTRL_7 0x01e4 +#define RT5682_HP_CALIB_CTRL_9 0x01e6 +#define RT5682_HP_CALIB_CTRL_10 0x01e7 +#define RT5682_HP_CALIB_CTRL_11 0x01e8 +#define RT5682_HP_CALIB_STA_1 0x01ea +#define RT5682_HP_CALIB_STA_2 0x01eb +#define RT5682_HP_CALIB_STA_3 0x01ec +#define RT5682_HP_CALIB_STA_4 0x01ed +#define RT5682_HP_CALIB_STA_5 0x01ee +#define RT5682_HP_CALIB_STA_6 0x01ef +#define RT5682_HP_CALIB_STA_7 0x01f0 +#define RT5682_HP_CALIB_STA_8 0x01f1 +#define RT5682_HP_CALIB_STA_9 0x01f2 +#define RT5682_HP_CALIB_STA_10 0x01f3 +#define RT5682_HP_CALIB_STA_11 0x01f4 +#define RT5682_SAR_IL_CMD_1 0x0210 +#define RT5682_SAR_IL_CMD_2 0x0211 +#define RT5682_SAR_IL_CMD_3 0x0212 +#define RT5682_SAR_IL_CMD_4 0x0213 +#define RT5682_SAR_IL_CMD_5 0x0214 +#define RT5682_SAR_IL_CMD_6 0x0215 +#define RT5682_SAR_IL_CMD_7 0x0216 +#define RT5682_SAR_IL_CMD_8 0x0217 +#define RT5682_SAR_IL_CMD_9 0x0218 +#define RT5682_SAR_IL_CMD_10 0x0219 +#define RT5682_SAR_IL_CMD_11 0x021a +#define RT5682_SAR_IL_CMD_12 0x021b +#define RT5682_SAR_IL_CMD_13 0x021c +#define RT5682_EFUSE_CTRL_1 0x0250 +#define RT5682_EFUSE_CTRL_2 0x0251 +#define RT5682_EFUSE_CTRL_3 0x0252 +#define RT5682_EFUSE_CTRL_4 0x0253 +#define RT5682_EFUSE_CTRL_5 0x0254 +#define RT5682_EFUSE_CTRL_6 0x0255 +#define RT5682_EFUSE_CTRL_7 0x0256 +#define RT5682_EFUSE_CTRL_8 0x0257 +#define RT5682_EFUSE_CTRL_9 0x0258 +#define RT5682_EFUSE_CTRL_10 0x0259 +#define RT5682_EFUSE_CTRL_11 0x025a +#define RT5682_JD_TOP_VC_VTRL 0x0270 +#define RT5682_DRC1_CTRL_0 0x02ff +#define RT5682_DRC1_CTRL_1 0x0300 +#define RT5682_DRC1_CTRL_2 0x0301 +#define RT5682_DRC1_CTRL_3 0x0302 +#define RT5682_DRC1_CTRL_4 0x0303 +#define RT5682_DRC1_CTRL_5 0x0304 +#define RT5682_DRC1_CTRL_6 0x0305 +#define RT5682_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5682_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5682_DRC1_PRIV_1 0x0310 +#define RT5682_DRC1_PRIV_2 0x0311 +#define RT5682_DRC1_PRIV_3 0x0312 +#define RT5682_DRC1_PRIV_4 0x0313 +#define RT5682_DRC1_PRIV_5 0x0314 +#define RT5682_DRC1_PRIV_6 0x0315 +#define RT5682_DRC1_PRIV_7 0x0316 +#define RT5682_DRC1_PRIV_8 0x0317 +#define RT5682_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5682_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5682_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5682_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5682_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5682_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5682_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5682_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5682_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5682_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5682_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5682_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5682_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5682_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5682_R_EQ_LPF1_A1 0x03d1 +#define RT5682_L_EQ_LPF1_H0 0x03d2 +#define RT5682_R_EQ_LPF1_H0 0x03d3 +#define RT5682_L_EQ_BPF1_A1 0x03d4 +#define RT5682_R_EQ_BPF1_A1 0x03d5 +#define RT5682_L_EQ_BPF1_A2 0x03d6 +#define RT5682_R_EQ_BPF1_A2 0x03d7 +#define RT5682_L_EQ_BPF1_H0 0x03d8 +#define RT5682_R_EQ_BPF1_H0 0x03d9 +#define RT5682_L_EQ_BPF2_A1 0x03da +#define RT5682_R_EQ_BPF2_A1 0x03db +#define RT5682_L_EQ_BPF2_A2 0x03dc +#define RT5682_R_EQ_BPF2_A2 0x03dd +#define RT5682_L_EQ_BPF2_H0 0x03de +#define RT5682_R_EQ_BPF2_H0 0x03df +#define RT5682_L_EQ_BPF3_A1 0x03e0 +#define RT5682_R_EQ_BPF3_A1 0x03e1 +#define RT5682_L_EQ_BPF3_A2 0x03e2 +#define RT5682_R_EQ_BPF3_A2 0x03e3 +#define RT5682_L_EQ_BPF3_H0 0x03e4 +#define RT5682_R_EQ_BPF3_H0 0x03e5 +#define RT5682_L_EQ_BPF4_A1 0x03e6 +#define RT5682_R_EQ_BPF4_A1 0x03e7 +#define RT5682_L_EQ_BPF4_A2 0x03e8 +#define RT5682_R_EQ_BPF4_A2 0x03e9 +#define RT5682_L_EQ_BPF4_H0 0x03ea +#define RT5682_R_EQ_BPF4_H0 0x03eb +#define RT5682_L_EQ_HPF1_A1 0x03ec +#define RT5682_R_EQ_HPF1_A1 0x03ed +#define RT5682_L_EQ_HPF1_H0 0x03ee +#define RT5682_R_EQ_HPF1_H0 0x03ef +#define RT5682_L_EQ_PRE_VOL 0x03f0 +#define RT5682_R_EQ_PRE_VOL 0x03f1 +#define RT5682_L_EQ_POST_VOL 0x03f2 +#define RT5682_R_EQ_POST_VOL 0x03f3 +#define RT5682_I2C_MODE 0xffff + + +/* global definition */ +#define RT5682_L_MUTE (0x1 << 15) +#define RT5682_L_MUTE_SFT 15 +#define RT5682_VOL_L_MUTE (0x1 << 14) +#define RT5682_VOL_L_SFT 14 +#define RT5682_R_MUTE (0x1 << 7) +#define RT5682_R_MUTE_SFT 7 +#define RT5682_VOL_R_MUTE (0x1 << 6) +#define RT5682_VOL_R_SFT 6 +#define RT5682_L_VOL_MASK (0x3f << 8) +#define RT5682_L_VOL_SFT 8 +#define RT5682_R_VOL_MASK (0x3f) +#define RT5682_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682_G_HP (0xf << 8) +#define RT5682_G_HP_SFT 8 +#define RT5682_G_STO_DA_DMIX (0xf) +#define RT5682_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5682_BST_CBJ_MASK (0xf << 8) +#define RT5682_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5682_EMB_JD_EN (0x1 << 15) +#define RT5682_EMB_JD_EN_SFT 15 +#define RT5682_EMB_JD_RST (0x1 << 14) +#define RT5682_JD_MODE (0x1 << 13) +#define RT5682_JD_MODE_SFT 13 +#define RT5682_DET_TYPE (0x1 << 12) +#define RT5682_DET_TYPE_SFT 12 +#define RT5682_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682_EXT_JD_DIG (0x1 << 9) +#define RT5682_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682_FAST_OFF_MASK (0x1 << 7) +#define RT5682_FAST_OFF_EN (0x1 << 7) +#define RT5682_FAST_OFF_DIS (0x0 << 7) +#define RT5682_VREF_POW_MASK (0x1 << 6) +#define RT5682_VREF_POW_FSM (0x0 << 6) +#define RT5682_VREF_POW_REG (0x1 << 6) +#define RT5682_MB1_PATH_MASK (0x1 << 5) +#define RT5682_CTRL_MB1_REG (0x1 << 5) +#define RT5682_CTRL_MB1_FSM (0x0 << 5) +#define RT5682_MB2_PATH_MASK (0x1 << 4) +#define RT5682_CTRL_MB2_REG (0x1 << 4) +#define RT5682_CTRL_MB2_FSM (0x0 << 4) +#define RT5682_TRIG_JD_MASK (0x1 << 3) +#define RT5682_TRIG_JD_HIGH (0x1 << 3) +#define RT5682_TRIG_JD_LOW (0x0 << 3) +#define RT5682_MIC_CAP_MASK (0x1 << 1) +#define RT5682_MIC_CAP_HS (0x1 << 1) +#define RT5682_MIC_CAP_HP (0x0 << 1) +#define RT5682_MIC_CAP_SRC_MASK (0x1) +#define RT5682_MIC_CAP_SRC_REG (0x1) +#define RT5682_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5682_EXT_JD_SRC (0x7 << 4) +#define RT5682_EXT_JD_SRC_SFT 4 +#define RT5682_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5682_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5682_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682_DAC_L1_VOL_SFT 8 +#define RT5682_DAC_R1_VOL_MASK (0xff) +#define RT5682_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682_ADC_L_VOL_SFT 8 +#define RT5682_ADC_R_VOL_MASK (0x7f) +#define RT5682_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682_STO1_ADC_L_BST_SFT 14 +#define RT5682_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682_ST_SRC_SEL (0x1 << 8) +#define RT5682_ST_SRC_SFT 8 +#define RT5682_ST_EN_MASK (0x1 << 6) +#define RT5682_ST_DIS (0x0 << 6) +#define RT5682_ST_EN (0x1 << 6) +#define RT5682_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682_M_STO1_ADC_L1_SFT 15 +#define RT5682_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682_M_STO1_ADC_L2_SFT 14 +#define RT5682_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682_STO1_ADC1L_SRC_SFT 13 +#define RT5682_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682_STO1_ADC2L_SRC_SFT 12 +#define RT5682_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682_STO1_ADCL_SRC_SFT 10 +#define RT5682_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5682_STO1_DD_L_SRC_SFT 9 +#define RT5682_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_SFT 8 +#define RT5682_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5682_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682_M_STO1_ADC_R1_SFT 7 +#define RT5682_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682_M_STO1_ADC_R2_SFT 6 +#define RT5682_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682_STO1_ADC1R_SRC_SFT 5 +#define RT5682_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682_STO1_ADC2R_SRC_SFT 4 +#define RT5682_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682_M_ADCMIX_L (0x1 << 15) +#define RT5682_M_ADCMIX_L_SFT 15 +#define RT5682_M_DAC1_L (0x1 << 14) +#define RT5682_M_DAC1_L_SFT 14 +#define RT5682_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5682_DAC1_R_SEL_SFT 10 +#define RT5682_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5682_DAC1_L_SEL_SFT 8 +#define RT5682_M_ADCMIX_R (0x1 << 7) +#define RT5682_M_ADCMIX_R_SFT 7 +#define RT5682_M_DAC1_R (0x1 << 6) +#define RT5682_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682_M_DAC_L1_STO_L_SFT 15 +#define RT5682_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682_G_DAC_L1_STO_L_SFT 14 +#define RT5682_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682_M_DAC_R1_STO_L_SFT 13 +#define RT5682_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682_G_DAC_R1_STO_L_SFT 12 +#define RT5682_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682_M_DAC_L1_STO_R_SFT 7 +#define RT5682_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682_G_DAC_L1_STO_R_SFT 6 +#define RT5682_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682_M_DAC_R1_STO_R_SFT 5 +#define RT5682_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682_M_ST_STO_L (0x1 << 9) +#define RT5682_M_ST_STO_L_SFT 9 +#define RT5682_M_ST_STO_R (0x1 << 8) +#define RT5682_M_ST_STO_R_SFT 8 +#define RT5682_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5682_A_DACL1_SFT 4 +#define RT5682_DAC_R1_SRC_MASK (0x3) +#define RT5682_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5682_G_CBJ_RM1_L (0x7 << 10) +#define RT5682_G_CBJ_RM1_L_SFT 10 +#define RT5682_M_CBJ_RM1_L (0x1 << 7) +#define RT5682_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682_PWR_I2S1 (0x1 << 15) +#define RT5682_PWR_I2S1_BIT 15 +#define RT5682_PWR_I2S2 (0x1 << 14) +#define RT5682_PWR_I2S2_BIT 14 +#define RT5682_PWR_DAC_L1 (0x1 << 11) +#define RT5682_PWR_DAC_L1_BIT 11 +#define RT5682_PWR_DAC_R1 (0x1 << 10) +#define RT5682_PWR_DAC_R1_BIT 10 +#define RT5682_PWR_LDO (0x1 << 8) +#define RT5682_PWR_LDO_BIT 8 +#define RT5682_PWR_ADC_L1 (0x1 << 4) +#define RT5682_PWR_ADC_L1_BIT 4 +#define RT5682_PWR_ADC_R1 (0x1 << 3) +#define RT5682_PWR_ADC_R1_BIT 3 +#define RT5682_DIG_GATE_CTRL (0x1 << 0) +#define RT5682_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682_PWR_ADC_S1F (0x1 << 15) +#define RT5682_PWR_ADC_S1F_BIT 15 +#define RT5682_PWR_DAC_S1F (0x1 << 10) +#define RT5682_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682_PWR_VREF1 (0x1 << 15) +#define RT5682_PWR_VREF1_BIT 15 +#define RT5682_PWR_FV1 (0x1 << 14) +#define RT5682_PWR_FV1_BIT 14 +#define RT5682_PWR_VREF2 (0x1 << 13) +#define RT5682_PWR_VREF2_BIT 13 +#define RT5682_PWR_FV2 (0x1 << 12) +#define RT5682_PWR_FV2_BIT 12 +#define RT5682_LDO1_DBG_MASK (0x3 << 10) +#define RT5682_PWR_MB (0x1 << 9) +#define RT5682_PWR_MB_BIT 9 +#define RT5682_PWR_BG (0x1 << 7) +#define RT5682_PWR_BG_BIT 7 +#define RT5682_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682_LDO1_BYPASS (0x1 << 6) +#define RT5682_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5682_PWR_MA_BIT 6 +#define RT5682_LDO1_DVO_MASK (0x3 << 4) +#define RT5682_LDO1_DVO_09 (0x0 << 4) +#define RT5682_LDO1_DVO_10 (0x1 << 4) +#define RT5682_LDO1_DVO_12 (0x2 << 4) +#define RT5682_LDO1_DVO_14 (0x3 << 4) +#define RT5682_HP_DRIVER_MASK (0x3 << 2) +#define RT5682_HP_DRIVER_1X (0x0 << 2) +#define RT5682_HP_DRIVER_3X (0x1 << 2) +#define RT5682_HP_DRIVER_5X (0x3 << 2) +#define RT5682_PWR_HA_L (0x1 << 1) +#define RT5682_PWR_HA_L_BIT 1 +#define RT5682_PWR_HA_R (0x1 << 0) +#define RT5682_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682_PWR_MB1 (0x1 << 11) +#define RT5682_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682_PWR_MB1_BIT 11 +#define RT5682_PWR_MB2 (0x1 << 10) +#define RT5682_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682_PWR_MB2_BIT 10 +#define RT5682_PWR_JDH (0x1 << 3) +#define RT5682_PWR_JDH_BIT 3 +#define RT5682_PWR_JDL (0x1 << 2) +#define RT5682_PWR_JDL_BIT 2 +#define RT5682_PWR_RM1_L (0x1 << 1) +#define RT5682_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682_PWR_CBJ (0x1 << 9) +#define RT5682_PWR_CBJ_BIT 9 +#define RT5682_PWR_PLL (0x1 << 6) +#define RT5682_PWR_PLL_BIT 6 +#define RT5682_PWR_PLL2B (0x1 << 5) +#define RT5682_PWR_PLL2B_BIT 5 +#define RT5682_PWR_PLL2F (0x1 << 4) +#define RT5682_PWR_PLL2F_BIT 4 +#define RT5682_PWR_LDO2 (0x1 << 2) +#define RT5682_PWR_LDO2_BIT 2 +#define RT5682_PWR_DET_SPKVDD (0x1 << 1) +#define RT5682_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5682_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682_PWR_STO1_DAC_L_BIT 5 +#define RT5682_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682_SYS_CLK_DET (0x1 << 15) +#define RT5682_SYS_CLK_DET_SFT 15 +#define RT5682_PLL1_CLK_DET (0x1 << 14) +#define RT5682_PLL1_CLK_DET_SFT 14 +#define RT5682_PLL2_CLK_DET (0x1 << 13) +#define RT5682_PLL2_CLK_DET_SFT 13 +#define RT5682_POW_CLK_DET2_SFT 8 +#define RT5682_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682_DMIC_1_EN_SFT 15 +#define RT5682_DMIC_1_DIS (0x0 << 15) +#define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682_DMIC_1_DP_SFT 4 +#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682_DMIC_CLK_MASK (0xf << 0) +#define RT5682_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682_SEL_ADCDAT_SFT 15 +#define RT5682_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682_I2S1_TX_CHL_SFT 12 +#define RT5682_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682_I2S1_RX_CHL_SFT 8 +#define RT5682_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682_I2S1_MONO_MASK (0x1 << 7) +#define RT5682_I2S1_MONO_EN (0x1 << 7) +#define RT5682_I2S1_MONO_DIS (0x0 << 7) +#define RT5682_I2S2_MONO_MASK (0x1 << 6) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S1_DL_MASK (0x7 << 4) +#define RT5682_I2S1_DL_SFT 4 +#define RT5682_I2S1_DL_16 (0x0 << 4) +#define RT5682_I2S1_DL_20 (0x1 << 4) +#define RT5682_I2S1_DL_24 (0x2 << 4) +#define RT5682_I2S1_DL_32 (0x3 << 4) +#define RT5682_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5682_I2S2_MS_MASK (0x1 << 15) +#define RT5682_I2S2_MS_SFT 15 +#define RT5682_I2S2_MS_M (0x0 << 15) +#define RT5682_I2S2_MS_S (0x1 << 15) +#define RT5682_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682_I2S2_PIN_CFG_SFT 14 +#define RT5682_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5682_I2S2_CLK_SEL_SFT 11 +#define RT5682_I2S2_OUT_MASK (0x1 << 9) +#define RT5682_I2S2_OUT_SFT 9 +#define RT5682_I2S2_OUT_UM (0x0 << 9) +#define RT5682_I2S2_OUT_M (0x1 << 9) +#define RT5682_I2S_BP_MASK (0x1 << 8) +#define RT5682_I2S_BP_SFT 8 +#define RT5682_I2S_BP_NOR (0x0 << 8) +#define RT5682_I2S_BP_INV (0x1 << 8) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S2_DL_MASK (0x3 << 4) +#define RT5682_I2S2_DL_SFT 4 +#define RT5682_I2S2_DL_16 (0x0 << 4) +#define RT5682_I2S2_DL_20 (0x1 << 4) +#define RT5682_I2S2_DL_24 (0x2 << 4) +#define RT5682_I2S2_DL_8 (0x3 << 4) +#define RT5682_I2S_DF_MASK (0x7) +#define RT5682_I2S_DF_SFT 0 +#define RT5682_I2S_DF_I2S (0x0) +#define RT5682_I2S_DF_LEFT (0x1) +#define RT5682_I2S_DF_PCM_A (0x2) +#define RT5682_I2S_DF_PCM_B (0x3) +#define RT5682_I2S_DF_PCM_A_N (0x6) +#define RT5682_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682_ADC_OSR_MASK (0xf << 12) +#define RT5682_ADC_OSR_SFT 12 +#define RT5682_ADC_OSR_D_1 (0x0 << 12) +#define RT5682_ADC_OSR_D_2 (0x1 << 12) +#define RT5682_ADC_OSR_D_4 (0x2 << 12) +#define RT5682_ADC_OSR_D_6 (0x3 << 12) +#define RT5682_ADC_OSR_D_8 (0x4 << 12) +#define RT5682_ADC_OSR_D_12 (0x5 << 12) +#define RT5682_ADC_OSR_D_16 (0x6 << 12) +#define RT5682_ADC_OSR_D_24 (0x7 << 12) +#define RT5682_ADC_OSR_D_32 (0x8 << 12) +#define RT5682_ADC_OSR_D_48 (0x9 << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_SFT 8 +#define RT5682_I2S_M_D_1 (0x0 << 8) +#define RT5682_I2S_M_D_2 (0x1 << 8) +#define RT5682_I2S_M_D_3 (0x2 << 8) +#define RT5682_I2S_M_D_4 (0x3 << 8) +#define RT5682_I2S_M_D_6 (0x4 << 8) +#define RT5682_I2S_M_D_8 (0x5 << 8) +#define RT5682_I2S_M_D_12 (0x6 << 8) +#define RT5682_I2S_M_D_16 (0x7 << 8) +#define RT5682_I2S_M_D_24 (0x8 << 8) +#define RT5682_I2S_M_D_32 (0x9 << 8) +#define RT5682_I2S_M_D_48 (0x10 << 8) +#define RT5682_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5682_I2S_CLK_SRC_SFT 4 +#define RT5682_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5682_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5682_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5682_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5682_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5682_DAC_OSR_MASK (0xf << 0) +#define RT5682_DAC_OSR_SFT 0 +#define RT5682_DAC_OSR_D_1 (0x0 << 0) +#define RT5682_DAC_OSR_D_2 (0x1 << 0) +#define RT5682_DAC_OSR_D_4 (0x2 << 0) +#define RT5682_DAC_OSR_D_6 (0x3 << 0) +#define RT5682_DAC_OSR_D_8 (0x4 << 0) +#define RT5682_DAC_OSR_D_12 (0x5 << 0) +#define RT5682_DAC_OSR_D_16 (0x6 << 0) +#define RT5682_DAC_OSR_D_24 (0x7 << 0) +#define RT5682_DAC_OSR_D_32 (0x8 << 0) +#define RT5682_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682_I2S2_BCLK_MS2_SFT 11 +#define RT5682_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682_TDM_TX_CH_2 (0x0 << 12) +#define RT5682_TDM_TX_CH_4 (0x1 << 12) +#define RT5682_TDM_TX_CH_6 (0x2 << 12) +#define RT5682_TDM_TX_CH_8 (0x3 << 12) +#define RT5682_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682_TDM_RX_CH_2 (0x0 << 8) +#define RT5682_TDM_RX_CH_4 (0x1 << 8) +#define RT5682_TDM_RX_CH_6 (0x2 << 8) +#define RT5682_TDM_RX_CH_8 (0x3 << 8) +#define RT5682_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5682_TDM_ADC_LCA_SFT 4 +#define RT5682_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682_IF1_ADC1_SEL_SFT 14 +#define RT5682_IF1_ADC2_SEL_SFT 12 +#define RT5682_IF1_ADC3_SEL_SFT 10 +#define RT5682_IF1_ADC4_SEL_SFT 8 +#define RT5682_TDM_ADC_SEL_SFT 4 + +/* TDM control 3 (0x007b) */ +#define RT5682_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682_TDM_S_BP_MASK (0x1 << 15) +#define RT5682_TDM_S_BP_SFT 15 +#define RT5682_TDM_S_BP_NOR (0x0 << 15) +#define RT5682_TDM_S_BP_INV (0x1 << 15) +#define RT5682_TDM_S_LP_MASK (0x1 << 14) +#define RT5682_TDM_S_LP_SFT 14 +#define RT5682_TDM_S_LP_NOR (0x0 << 14) +#define RT5682_TDM_S_LP_INV (0x1 << 14) +#define RT5682_TDM_DF_MASK (0x7 << 11) +#define RT5682_TDM_DF_SFT 11 +#define RT5682_TDM_DF_I2S (0x0 << 11) +#define RT5682_TDM_DF_LEFT (0x1 << 11) +#define RT5682_TDM_DF_PCM_A (0x2 << 11) +#define RT5682_TDM_DF_PCM_B (0x3 << 11) +#define RT5682_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_CL_MASK (0x3 << 4) +#define RT5682_TDM_CL_16 (0x0 << 4) +#define RT5682_TDM_CL_20 (0x1 << 4) +#define RT5682_TDM_CL_24 (0x2 << 4) +#define RT5682_TDM_CL_32 (0x3 << 4) +#define RT5682_TDM_M_BP_MASK (0x1 << 2) +#define RT5682_TDM_M_BP_SFT 2 +#define RT5682_TDM_M_BP_NOR (0x0 << 2) +#define RT5682_TDM_M_BP_INV (0x1 << 2) +#define RT5682_TDM_M_LP_MASK (0x1 << 1) +#define RT5682_TDM_M_LP_SFT 1 +#define RT5682_TDM_M_LP_NOR (0x0 << 1) +#define RT5682_TDM_M_LP_INV (0x1 << 1) +#define RT5682_TDM_MS_MASK (0x1 << 0) +#define RT5682_TDM_MS_SFT 0 +#define RT5682_TDM_MS_M (0x0 << 0) +#define RT5682_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682_SCLK_SRC_MASK (0x7 << 13) +#define RT5682_SCLK_SRC_SFT 13 +#define RT5682_SCLK_SRC_MCLK (0x0 << 13) +#define RT5682_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5682_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5682_SCLK_SRC_SDW (0x3 << 13) +#define RT5682_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5682_PLL1_SRC_MASK (0x3 << 10) +#define RT5682_PLL1_SRC_SFT 10 +#define RT5682_PLL1_SRC_MCLK (0x0 << 10) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL1_SRC_SDW (0x2 << 10) +#define RT5682_PLL1_SRC_RC (0x3 << 10) +#define RT5682_PLL2_SRC_MASK (0x3 << 8) +#define RT5682_PLL2_SRC_SFT 8 +#define RT5682_PLL2_SRC_MCLK (0x0 << 8) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL2_SRC_SDW (0x2 << 8) +#define RT5682_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5682_PLL_INP_MAX 40000000 +#define RT5682_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5682_PLL_N_MAX 0x001ff +#define RT5682_PLL_N_MASK (RT5682_PLL_N_MAX << 7) +#define RT5682_PLL_N_SFT 7 +#define RT5682_PLL_K_MAX 0x001f +#define RT5682_PLL_K_MASK (RT5682_PLL_K_MAX) +#define RT5682_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5682_PLL_M_MAX 0x00f +#define RT5682_PLL_M_MASK (RT5682_PLL_M_MAX << 12) +#define RT5682_PLL_M_SFT 12 +#define RT5682_PLL_M_BP (0x1 << 11) +#define RT5682_PLL_M_BP_SFT 11 +#define RT5682_PLL_K_BP (0x1 << 10) +#define RT5682_PLL_K_BP_SFT 10 +#define RT5682_PLL_RST (0x1 << 1) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682_DA_ASRC_MASK (0x1 << 13) +#define RT5682_DA_ASRC_SFT 13 +#define RT5682_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682_DAC_STO1_ASRC_SFT 12 +#define RT5682_AD_ASRC_MASK (0x1 << 8) +#define RT5682_AD_ASRC_SFT 8 +#define RT5682_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682_AD_ASRC_SEL_SFT 4 +#define RT5682_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682_DMIC_ASRC_SFT 3 +#define RT5682_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682_ADC_STO1_ASRC_SFT 2 +#define RT5682_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682_FILTER_CLK_SEL_SFT 12 +#define RT5682_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682_ASRCIN_FTK_N1_SFT 14 +#define RT5682_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682_ASRCIN_FTK_N2_SFT 12 +#define RT5682_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682_ASRCIN_FTK_M1_SFT 8 +#define RT5682_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5682_PLL2_OUT_MASK (0x1 << 8) +#define RT5682_PLL2_OUT_98M (0x0 << 8) +#define RT5682_PLL2_OUT_49M (0x1 << 8) +#define RT5682_SDW_REF_2_MASK (0xf << 4) +#define RT5682_SDW_REF_2_SFT 4 +#define RT5682_SDW_REF_2_48K (0x0 << 4) +#define RT5682_SDW_REF_2_96K (0x1 << 4) +#define RT5682_SDW_REF_2_192K (0x2 << 4) +#define RT5682_SDW_REF_2_32K (0x3 << 4) +#define RT5682_SDW_REF_2_24K (0x4 << 4) +#define RT5682_SDW_REF_2_16K (0x5 << 4) +#define RT5682_SDW_REF_2_12K (0x6 << 4) +#define RT5682_SDW_REF_2_8K (0x7 << 4) +#define RT5682_SDW_REF_2_44K (0x8 << 4) +#define RT5682_SDW_REF_2_88K (0x9 << 4) +#define RT5682_SDW_REF_2_176K (0xa << 4) +#define RT5682_SDW_REF_2_353K (0xb << 4) +#define RT5682_SDW_REF_2_22K (0xc << 4) +#define RT5682_SDW_REF_2_384K (0xd << 4) +#define RT5682_SDW_REF_2_11K (0xe << 4) +#define RT5682_SDW_REF_1_MASK (0xf << 0) +#define RT5682_SDW_REF_1_SFT 0 +#define RT5682_SDW_REF_1_48K (0x0 << 0) +#define RT5682_SDW_REF_1_96K (0x1 << 0) +#define RT5682_SDW_REF_1_192K (0x2 << 0) +#define RT5682_SDW_REF_1_32K (0x3 << 0) +#define RT5682_SDW_REF_1_24K (0x4 << 0) +#define RT5682_SDW_REF_1_16K (0x5 << 0) +#define RT5682_SDW_REF_1_12K (0x6 << 0) +#define RT5682_SDW_REF_1_8K (0x7 << 0) +#define RT5682_SDW_REF_1_44K (0x8 << 0) +#define RT5682_SDW_REF_1_88K (0x9 << 0) +#define RT5682_SDW_REF_1_176K (0xa << 0) +#define RT5682_SDW_REF_1_353K (0xb << 0) +#define RT5682_SDW_REF_1_22K (0xc << 0) +#define RT5682_SDW_REF_1_384K (0xd << 0) +#define RT5682_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682_PUMP_EN (0x1 << 3) +#define RT5682_PUMP_EN_SFT 3 +#define RT5682_CAPLESS_EN (0x1 << 0) +#define RT5682_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682_RAMP_MASK (0x1 << 12) +#define RT5682_RAMP_SFT 12 +#define RT5682_RAMP_DIS (0x0 << 12) +#define RT5682_RAMP_EN (0x1 << 12) +#define RT5682_BPS_MASK (0x1 << 11) +#define RT5682_BPS_SFT 11 +#define RT5682_BPS_DIS (0x0 << 11) +#define RT5682_BPS_EN (0x1 << 11) +#define RT5682_FAST_UPDN_MASK (0x1 << 10) +#define RT5682_FAST_UPDN_SFT 10 +#define RT5682_FAST_UPDN_DIS (0x0 << 10) +#define RT5682_FAST_UPDN_EN (0x1 << 10) +#define RT5682_VLO_MASK (0x1 << 7) +#define RT5682_VLO_SFT 7 +#define RT5682_VLO_3V (0x0 << 7) +#define RT5682_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682_OSW_L_MASK (0x1 << 11) +#define RT5682_OSW_L_SFT 11 +#define RT5682_OSW_L_DIS (0x0 << 11) +#define RT5682_OSW_L_EN (0x1 << 11) +#define RT5682_OSW_R_MASK (0x1 << 10) +#define RT5682_OSW_R_SFT 10 +#define RT5682_OSW_R_DIS (0x0 << 10) +#define RT5682_OSW_R_EN (0x1 << 10) +#define RT5682_PM_HP_MASK (0x3 << 8) +#define RT5682_PM_HP_SFT 8 +#define RT5682_PM_HP_LV (0x0 << 8) +#define RT5682_PM_HP_MV (0x1 << 8) +#define RT5682_PM_HP_HV (0x2 << 8) +#define RT5682_IB_HP_MASK (0x3 << 6) +#define RT5682_IB_HP_SFT 6 +#define RT5682_IB_HP_125IL (0x0 << 6) +#define RT5682_IB_HP_25IL (0x1 << 6) +#define RT5682_IB_HP_5IL (0x2 << 6) +#define RT5682_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5682_MIC1_OV_MASK (0x3 << 14) +#define RT5682_MIC1_OV_SFT 14 +#define RT5682_MIC1_OV_2V7 (0x0 << 14) +#define RT5682_MIC1_OV_2V4 (0x1 << 14) +#define RT5682_MIC1_OV_2V25 (0x3 << 14) +#define RT5682_MIC1_OV_1V8 (0x4 << 14) +#define RT5682_MIC1_CLK_MASK (0x1 << 13) +#define RT5682_MIC1_CLK_SFT 13 +#define RT5682_MIC1_CLK_DIS (0x0 << 13) +#define RT5682_MIC1_CLK_EN (0x1 << 13) +#define RT5682_MIC1_OVCD_MASK (0x1 << 12) +#define RT5682_MIC1_OVCD_SFT 12 +#define RT5682_MIC1_OVCD_DIS (0x0 << 12) +#define RT5682_MIC1_OVCD_EN (0x1 << 12) +#define RT5682_MIC1_OVTH_MASK (0x3 << 10) +#define RT5682_MIC1_OVTH_SFT 10 +#define RT5682_MIC1_OVTH_768UA (0x0 << 10) +#define RT5682_MIC1_OVTH_960UA (0x1 << 10) +#define RT5682_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5682_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5682_MIC2_OV_MASK (0x3 << 8) +#define RT5682_MIC2_OV_SFT 8 +#define RT5682_MIC2_OV_2V7 (0x0 << 8) +#define RT5682_MIC2_OV_2V4 (0x1 << 8) +#define RT5682_MIC2_OV_2V25 (0x3 << 8) +#define RT5682_MIC2_OV_1V8 (0x4 << 8) +#define RT5682_MIC2_CLK_MASK (0x1 << 7) +#define RT5682_MIC2_CLK_SFT 7 +#define RT5682_MIC2_CLK_DIS (0x0 << 7) +#define RT5682_MIC2_CLK_EN (0x1 << 7) +#define RT5682_MIC2_OVTH_MASK (0x3 << 4) +#define RT5682_MIC2_OVTH_SFT 4 +#define RT5682_MIC2_OVTH_768UA (0x0 << 4) +#define RT5682_MIC2_OVTH_960UA (0x1 << 4) +#define RT5682_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5682_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5682_PWR_MB_MASK (0x1 << 3) +#define RT5682_PWR_MB_SFT 3 +#define RT5682_PWR_MB_PD (0x0 << 3) +#define RT5682_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5682_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682_PWR_CLK25M_SFT 9 +#define RT5682_PWR_CLK25M_PD (0x0 << 9) +#define RT5682_PWR_CLK25M_PU (0x1 << 9) +#define RT5682_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682_PWR_CLK1M_SFT 8 +#define RT5682_PWR_CLK1M_PD (0x0 << 8) +#define RT5682_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5682_POW_IRQ (0x1 << 15) +#define RT5682_POW_JDH (0x1 << 14) +#define RT5682_POW_JDL (0x1 << 13) +#define RT5682_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5682_CLK_SRC_MCLK (0x0) +#define RT5682_CLK_SRC_PLL1 (0x1) +#define RT5682_CLK_SRC_PLL2 (0x2) +#define RT5682_CLK_SRC_SDW (0x3) +#define RT5682_CLK_SRC_RCCLK (0x4) +#define RT5682_I2S_PD_1 (0x0) +#define RT5682_I2S_PD_2 (0x1) +#define RT5682_I2S_PD_3 (0x2) +#define RT5682_I2S_PD_4 (0x3) +#define RT5682_I2S_PD_6 (0x4) +#define RT5682_I2S_PD_8 (0x5) +#define RT5682_I2S_PD_12 (0x6) +#define RT5682_I2S_PD_16 (0x7) +#define RT5682_I2S_PD_24 (0x8) +#define RT5682_I2S_PD_32 (0x9) +#define RT5682_I2S_PD_48 (0xa) +#define RT5682_I2S2_SRC_MASK (0x3 << 4) +#define RT5682_I2S2_SRC_SFT 4 +#define RT5682_I2S2_M_PD_MASK (0xf << 0) +#define RT5682_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682_JD1_PULSE_EN_SFT 10 +#define RT5682_JD1_PULSE_DIS (0x0 << 10) +#define RT5682_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682_JD1_EN_MASK (0x1 << 15) +#define RT5682_JD1_EN_SFT 15 +#define RT5682_JD1_DIS (0x0 << 15) +#define RT5682_JD1_EN (0x1 << 15) +#define RT5682_JD1_POL_MASK (0x1 << 13) +#define RT5682_JD1_POL_NOR (0x0 << 13) +#define RT5682_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682_IL_IRQ_MASK (0x1 << 7) +#define RT5682_IL_IRQ_DIS (0x0 << 7) +#define RT5682_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682_GP1_PIN_MASK (0x3 << 14) +#define RT5682_GP1_PIN_SFT 14 +#define RT5682_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682_GP1_PIN_IRQ (0x1 << 14) +#define RT5682_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682_GP2_PIN_MASK (0x3 << 12) +#define RT5682_GP2_PIN_SFT 12 +#define RT5682_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682_GP3_PIN_MASK (0x3 << 10) +#define RT5682_GP3_PIN_SFT 10 +#define RT5682_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682_GP4_PIN_MASK (0x3 << 8) +#define RT5682_GP4_PIN_SFT 8 +#define RT5682_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682_GP5_PIN_MASK (0x3 << 6) +#define RT5682_GP5_PIN_SFT 6 +#define RT5682_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682_GP6_PIN_MASK (0x1 << 5) +#define RT5682_GP6_PIN_SFT 5 +#define RT5682_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682_GP1_PF_MASK (0x1 << 15) +#define RT5682_GP1_PF_IN (0x0 << 15) +#define RT5682_GP1_PF_OUT (0x1 << 15) +#define RT5682_GP1_OUT_MASK (0x1 << 14) +#define RT5682_GP1_OUT_L (0x0 << 14) +#define RT5682_GP1_OUT_H (0x1 << 14) +#define RT5682_GP2_PF_MASK (0x1 << 13) +#define RT5682_GP2_PF_IN (0x0 << 13) +#define RT5682_GP2_PF_OUT (0x1 << 13) +#define RT5682_GP2_OUT_MASK (0x1 << 12) +#define RT5682_GP2_OUT_L (0x0 << 12) +#define RT5682_GP2_OUT_H (0x1 << 12) +#define RT5682_GP3_PF_MASK (0x1 << 11) +#define RT5682_GP3_PF_IN (0x0 << 11) +#define RT5682_GP3_PF_OUT (0x1 << 11) +#define RT5682_GP3_OUT_MASK (0x1 << 10) +#define RT5682_GP3_OUT_L (0x0 << 10) +#define RT5682_GP3_OUT_H (0x1 << 10) +#define RT5682_GP4_PF_MASK (0x1 << 9) +#define RT5682_GP4_PF_IN (0x0 << 9) +#define RT5682_GP4_PF_OUT (0x1 << 9) +#define RT5682_GP4_OUT_MASK (0x1 << 8) +#define RT5682_GP4_OUT_L (0x0 << 8) +#define RT5682_GP4_OUT_H (0x1 << 8) +#define RT5682_GP5_PF_MASK (0x1 << 7) +#define RT5682_GP5_PF_IN (0x0 << 7) +#define RT5682_GP5_PF_OUT (0x1 << 7) +#define RT5682_GP5_OUT_MASK (0x1 << 6) +#define RT5682_GP5_OUT_L (0x0 << 6) +#define RT5682_GP5_OUT_H (0x1 << 6) +#define RT5682_GP6_PF_MASK (0x1 << 5) +#define RT5682_GP6_PF_IN (0x0 << 5) +#define RT5682_GP6_PF_OUT (0x1 << 5) +#define RT5682_GP6_OUT_MASK (0x1 << 4) +#define RT5682_GP6_OUT_L (0x0 << 4) +#define RT5682_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5682_GP6_STA (0x1 << 6) +#define RT5682_GP5_STA (0x1 << 5) +#define RT5682_GP4_STA (0x1 << 4) +#define RT5682_GP3_STA (0x1 << 3) +#define RT5682_GP2_STA (0x1 << 2) +#define RT5682_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682_SV_MASK (0x1 << 15) +#define RT5682_SV_SFT 15 +#define RT5682_SV_DIS (0x0 << 15) +#define RT5682_SV_EN (0x1 << 15) +#define RT5682_ZCD_MASK (0x1 << 10) +#define RT5682_ZCD_SFT 10 +#define RT5682_ZCD_PD (0x0 << 10) +#define RT5682_ZCD_PU (0x1 << 10) +#define RT5682_SV_DLY_MASK (0xf) +#define RT5682_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5682_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5682_ZCD_BST1_CBJ_SFT 7 +#define RT5682_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5682_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5682_ZCD_RECMIX_MASK (0x1) +#define RT5682_ZCD_RECMIX_SFT 0 +#define RT5682_ZCD_RECMIX_DIS (0x0) +#define RT5682_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682_4BTN_IL_MASK (0x1 << 15) +#define RT5682_4BTN_IL_EN (0x1 << 15) +#define RT5682_4BTN_IL_DIS (0x0 << 15) +#define RT5682_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682_4BTN_IL_NOR (0x1 << 14) +#define RT5682_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5682_JDH_RS_MASK (0x1 << 4) +#define RT5682_JDH_NO_PLUG (0x1 << 4) +#define RT5682_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682_CKXEN_DAC1_SFT 13 +#define RT5682_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682_CKXEN_ADC1_SFT 13 +#define RT5682_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682_AD2DA_LB_MASK (0x1 << 10) +#define RT5682_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682_NG2_EN_MASK (0x1 << 15) +#define RT5682_NG2_EN (0x1 << 15) +#define RT5682_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5682_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5682_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5682_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5682_SAR_BUTDET_RST (0x0 << 13) +#define RT5682_SAR_POW_MASK (0x1 << 12) +#define RT5682_SAR_POW_EN (0x1 << 12) +#define RT5682_SAR_POW_DIS (0x0 << 12) +#define RT5682_SAR_RST_MASK (0x1 << 11) +#define RT5682_SAR_RST_NORMAL (0x1 << 11) +#define RT5682_SAR_RST (0x0 << 11) +#define RT5682_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682_SAR_BYPASS_EN (0x1 << 10) +#define RT5682_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5682_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5682_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5682_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5682_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5682_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5682_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5682_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682_SAR_SOUR_MASK (0x3f) +#define RT5682_SAR_SOUR_BTN (0x3f) +#define RT5682_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5682_SCLK_S_MCLK, + RT5682_SCLK_S_PLL1, + RT5682_SCLK_S_PLL2, + RT5682_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682_PLL1_S_MCLK, + RT5682_PLL1_S_BCLK1, + RT5682_PLL1_S_RCCLK, +}; + +enum { + RT5682_AIF1, + RT5682_AIF2, + RT5682_AIFS +}; + +/* filter mask */ +enum { + RT5682_DA_STEREO1_FILTER = 0x1, + RT5682_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682_CLK_SEL_SYS, + RT5682_CLK_SEL_I2S1_ASRC, + RT5682_CLK_SEL_I2S2_ASRC, +}; + +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682_H__ */ From a7dc662c6a7b9209df600c64b16d33d72dbf56b1 Mon Sep 17 00:00:00 2001 From: Stefan Agner Date: Sun, 17 Jun 2018 15:45:29 +0200 Subject: [PATCH 032/529] ASoC: codecs: PCM1789: unconditionally flush work Work is guaranteed to be initialized on exit. Drop the unnecessary if statement and always call flush_work. This fixes a warning seen with clang: sound/soc/codecs/pcm1789.c:265:13: warning: address of 'priv->work' will always evaluate to 'true' [-Wpointer-bool-conversion] if (&priv->work) ~~ ~~~~~~^~~~ Signed-off-by: Stefan Agner Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1789.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c index 21f15219b3ad..8df6447c76a6 100644 --- a/sound/soc/codecs/pcm1789.c +++ b/sound/soc/codecs/pcm1789.c @@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev) { struct pcm1789_private *priv = dev_get_drvdata(dev); - if (&priv->work) - flush_work(&priv->work); + flush_work(&priv->work); return 0; } From e2b35e468c89000cf3b8b5beb63fd73dfa8a5435 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 12:50:00 +0200 Subject: [PATCH 033/529] ASoC: pxa: add binding for pxa2xx-ac97 audio complex This adds a binding for the Marvell PXA audio complex, available in pxa2xx and pxa3xx variants. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- .../bindings/sound/marvell,pxa2xx-ac97.txt | 27 +++++++++++++++++++ 1 file changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt diff --git a/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt new file mode 100644 index 000000000000..2ea85d5be6a4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt @@ -0,0 +1,27 @@ +Marvell PXA2xx audio complex + +This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series. + +Required properties: + - compatible: should be one of the following: + "marvell,pxa250-ac97" + "marvell,pxa270-ac97" + "marvell,pxa300-ac97" + - reg: device MMIO address space + - interrupts: single interrupt generated by AC97 IP + - clocks: input clock of the AC97 IP, refer to clock-bindings.txt + +Optional properties: + - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt + - reset-gpios: gpio used for AC97 reset, refer to gpio.txt + +Example: + ac97: sound@40500000 { + compatible = "marvell,pxa250-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pmux_ac97_default >; + }; From a4519526ebbd261e36425fa1c269515ee0648ab2 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 12:50:01 +0200 Subject: [PATCH 034/529] ASoC: pxa: add devicetree support Add the devicetree support, so that the driver can be used in a devictree platform. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++++++++ sound/soc/pxa/pxa2xx-ac97.c | 12 ++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 5950a9e218d9..8eafd3d3dff6 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -19,6 +19,7 @@ #include #include #include +#include #include @@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) dev_err(&dev->dev, "Invalid reset GPIO %d\n", pdata->reset_gpio); } + } else if (!pdata && dev->dev.of_node) { + pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node, + "reset-gpios", 0); + if (pdata->reset_gpio == -ENOENT) + pdata->reset_gpio = -1; + else if (pdata->reset_gpio < 0) + return pdata->reset_gpio; + reset_gpio = pdata->reset_gpio; } else { if (cpu_is_pxa27x()) reset_gpio = 113; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..5738a0abcd6a 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -238,6 +238,17 @@ static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", }; +#ifdef CONFIG_OF +static const struct of_device_id pxa2xx_ac97_dt_ids[] = { + { .compatible = "marvell,pxa250-ac97", }, + { .compatible = "marvell,pxa270-ac97", }, + { .compatible = "marvell,pxa300-ac97", }, + { } +}; +MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); + +#endif + static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; @@ -296,6 +307,7 @@ static struct platform_driver pxa2xx_ac97_driver = { #ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif + .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; From 7c5dfd549617b87db8e891ff4ecaa4a582b6c4cc Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 16 Jun 2018 01:22:58 +0300 Subject: [PATCH 035/529] ASoC: tegra: fix device_node refcounting tegra_rt5677_probe() gets a couple of device nodes with of_parse_phandle(), but there is no release of them. The patch adds the release to tegra_rt5677_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Reviewed-by: Nicholas Mc Guire Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 0e4805c7b4ca..7081f15302cc 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + tegra_rt5677_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; err: return ret; } @@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + tegra_rt5677_dai.platform_of_node = NULL; + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + return 0; } From 6cea3590820819049df5945136b8a5acd72ed0f8 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 3 Jun 2018 15:42:31 +0200 Subject: [PATCH 036/529] ASoC: Intel: bytcr_rt5640: Add quirk for the Nuvison/TMax TM800W560 tablet Add a quirk for the Nuvison/TMax TM800W560 tablet, this tablet uses IN1 for the internal mic rather then the default IN3 and it uses JD2 rather then JD1 for its not-inverted jack-detect switch. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c4f9ea40f57..8571f41767ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -565,6 +565,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_DIFF_MIC | BYT_RT5640_MCLK_EN), }, + { /* Nuvison/TMax TM800W560 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), From f12a0a3c4cc6f594d7c2ea361f2396ae5c518d2c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 3 Jun 2018 15:42:32 +0200 Subject: [PATCH 037/529] ASoC: Intel: bytcr_rt5640: Fix Acer Iconia 8 over-current detect threshold Change the over-current detect threshold on the Acer Iconia 8 from 2000ua to 1500uA, this fixes headset button presses not being detected. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8571f41767ef..7456566c5648 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_TH_1500UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), From 01655193c2da12510af8a8b66b56da5e13ce1f91 Mon Sep 17 00:00:00 2001 From: Stefan Agner Date: Sun, 17 Jun 2018 15:46:29 +0200 Subject: [PATCH 038/529] ALSA: ice1724: remove unused array This fixes a warning seen with clang: sound/pci/ice1712/prodigy_hifi.c:321:28: warning: variable 'wm_vol' is not needed and will not be emitted [-Wunneeded-internal-declaration] static const unsigned char wm_vol[256] = { ^ Signed-off-by: Stefan Agner Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index d7366ade5a25..c97b5528e4b8 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -314,26 +314,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] = { /* --------------- */ -/* - * Logarithmic volume values for WM87*6 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 255 #define WM_VOL_MUTE 0x8000 From a753af301c616cd51dedb3b5a8b3ba6cac3e58b8 Mon Sep 17 00:00:00 2001 From: Stefan Agner Date: Sun, 17 Jun 2018 15:46:49 +0200 Subject: [PATCH 039/529] ALSA: ctxfi: use enum type CT_SUM_CTL where appropriate Currently a variable of type enum CT_AMIXER_CTL is used for enum CT_SUM_CTL values. This leads to warnings when using clang: sound/pci/ctxfi/ctmixer.c:945:32: warning: implicit conversion from enumeration type 'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL' [-Wenum-conversion] for (i = AMIXER_MASTER_F, j = SUM_IN_F; ~ ^~~~~~~~ sound/pci/ctxfi/ctmixer.c:975:29: warning: implicit conversion from enumeration type 'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL' [-Wenum-conversion] for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { ~ ^~~~~~~~ Introduce enum CT_SUM_CTL k and it instead. Signed-off-by: Stefan Agner Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctmixer.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index db710d0a609f..4777d50fbbf8 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -938,17 +938,18 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) struct sum *sum; struct amixer *amix_d, *amix_s; enum CT_AMIXER_CTL i, j; + enum CT_SUM_CTL k; /* Build topology from destination to source */ /* Set up Master mixer */ - for (i = AMIXER_MASTER_F, j = SUM_IN_F; - i <= AMIXER_MASTER_S; i++, j++) { + for (i = AMIXER_MASTER_F, k = SUM_IN_F; + i <= AMIXER_MASTER_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); } @@ -972,12 +973,12 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); /* Set up PCM-in mixer */ - for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { + for (i = AMIXER_PCM_F, k = SUM_IN_F; i <= AMIXER_PCM_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); } From a8eaad7b04eaab3df6b8db722d4418286815b46c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 18 Jun 2018 17:41:01 +0200 Subject: [PATCH 040/529] ALSA: line6: stop using get_seconds() The get_seconds() function is deprecated because it truncates the timestamp to 32 bits, so all users should change to ktime_get_seconds() or ktime_get_real_seconds(). The firmware interface for passing the timestamp is also limited to 32 bits, so this patch only has the cosmetic effect of avoiding the old interface. Signed-off-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/usb/line6/toneport.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 750467fb95db..f47ba94e6f4a 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -367,12 +367,13 @@ static bool toneport_has_source_select(struct usb_line6_toneport *toneport) */ static void toneport_setup(struct usb_line6_toneport *toneport) { - int ticks; + u32 ticks; struct usb_line6 *line6 = &toneport->line6; struct usb_device *usbdev = line6->usbdev; /* sync time on device with host: */ - ticks = (int)get_seconds(); + /* note: 32-bit timestamps overflow in year 2106 */ + ticks = (u32)ktime_get_real_seconds(); line6_write_data(line6, 0x80c6, &ticks, 4); /* enable device: */ From 420c0117db25db38b72b6230223f7a976d3070ea Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:04 +0200 Subject: [PATCH 041/529] dmaengine: pxa: use a dma slave map In order to remove the specific knowledge of the dma mapping from PXA drivers, add a default slave map for pxa architectures. This won't impact MMP architecture, but is aimed only at all PXA boards. This is the first step, and once all drivers are converted, pxad_filter_fn() will be made static, and the DMA resources removed from device.c. Signed-off-by: Robert Jarzmik Reported-by: Arnd Bergmann Acked-by: Vinod Koul --- drivers/dma/pxa_dma.c | 10 +++++++++- include/linux/platform_data/mmp_dma.h | 4 ++++ 2 files changed, 13 insertions(+), 1 deletion(-) diff --git a/drivers/dma/pxa_dma.c b/drivers/dma/pxa_dma.c index b53fb618bbf6..9505334f9c6e 100644 --- a/drivers/dma/pxa_dma.c +++ b/drivers/dma/pxa_dma.c @@ -179,6 +179,8 @@ static unsigned int pxad_drcmr(unsigned int line) return 0x1000 + line * 4; } +bool pxad_filter_fn(struct dma_chan *chan, void *param); + /* * Debug fs */ @@ -1396,9 +1398,10 @@ static int pxad_probe(struct platform_device *op) { struct pxad_device *pdev; const struct of_device_id *of_id; + const struct dma_slave_map *slave_map = NULL; struct mmp_dma_platdata *pdata = dev_get_platdata(&op->dev); struct resource *iores; - int ret, dma_channels = 0, nb_requestors = 0; + int ret, dma_channels = 0, nb_requestors = 0, slave_map_cnt = 0; const enum dma_slave_buswidth widths = DMA_SLAVE_BUSWIDTH_1_BYTE | DMA_SLAVE_BUSWIDTH_2_BYTES | DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -1429,6 +1432,8 @@ static int pxad_probe(struct platform_device *op) } else if (pdata && pdata->dma_channels) { dma_channels = pdata->dma_channels; nb_requestors = pdata->nb_requestors; + slave_map = pdata->slave_map; + slave_map_cnt = pdata->slave_map_cnt; } else { dma_channels = 32; /* default 32 channel */ } @@ -1440,6 +1445,9 @@ static int pxad_probe(struct platform_device *op) pdev->slave.device_prep_dma_memcpy = pxad_prep_memcpy; pdev->slave.device_prep_slave_sg = pxad_prep_slave_sg; pdev->slave.device_prep_dma_cyclic = pxad_prep_dma_cyclic; + pdev->slave.filter.map = slave_map; + pdev->slave.filter.mapcnt = slave_map_cnt; + pdev->slave.filter.fn = pxad_filter_fn; pdev->slave.copy_align = PDMA_ALIGNMENT; pdev->slave.src_addr_widths = widths; diff --git a/include/linux/platform_data/mmp_dma.h b/include/linux/platform_data/mmp_dma.h index d1397c8ed94e..6397b9c8149a 100644 --- a/include/linux/platform_data/mmp_dma.h +++ b/include/linux/platform_data/mmp_dma.h @@ -12,9 +12,13 @@ #ifndef MMP_DMA_H #define MMP_DMA_H +struct dma_slave_map; + struct mmp_dma_platdata { int dma_channels; int nb_requestors; + int slave_map_cnt; + const struct dma_slave_map *slave_map; }; #endif /* MMP_DMA_H */ From 1da10c17afd1109ae22d529c3b16d9f6de3fdbec Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:05 +0200 Subject: [PATCH 042/529] ARM: pxa: add dma slave map In order to remove the specific knowledge of the dma mapping from PXA drivers, add a default slave map for pxa architectures. This is the first step, and once all drivers are converted, pxad_filter_fn() will be made static, and the DMA resources removed from device.c. Signed-off-by: Robert Jarzmik Reported-by: Arnd Bergmann --- arch/arm/mach-pxa/devices.c | 12 +++-------- arch/arm/mach-pxa/devices.h | 6 +++++- arch/arm/mach-pxa/pxa25x.c | 38 +++++++++++++++++++++++++++++++++- arch/arm/mach-pxa/pxa27x.c | 39 ++++++++++++++++++++++++++++++++++- arch/arm/mach-pxa/pxa3xx.c | 41 ++++++++++++++++++++++++++++++++++++- 5 files changed, 123 insertions(+), 13 deletions(-) diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index d7c9a8476d57..1e8915fc340d 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -4,6 +4,7 @@ #include #include #include +#include #include #include @@ -1202,11 +1203,6 @@ void __init pxa2xx_set_spi_info(unsigned id, struct pxa2xx_spi_master *info) platform_device_add(pd); } -static struct mmp_dma_platdata pxa_dma_pdata = { - .dma_channels = 0, - .nb_requestors = 0, -}; - static struct resource pxa_dma_resource[] = { [0] = { .start = 0x40000000, @@ -1233,9 +1229,7 @@ static struct platform_device pxa2xx_pxa_dma = { .resource = pxa_dma_resource, }; -void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors) +void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata) { - pxa_dma_pdata.dma_channels = nb_channels; - pxa_dma_pdata.nb_requestors = nb_requestors; - pxa_register_device(&pxa2xx_pxa_dma, &pxa_dma_pdata); + pxa_register_device(&pxa2xx_pxa_dma, dma_pdata); } diff --git a/arch/arm/mach-pxa/devices.h b/arch/arm/mach-pxa/devices.h index 11263f7c455b..498b07bc6a3e 100644 --- a/arch/arm/mach-pxa/devices.h +++ b/arch/arm/mach-pxa/devices.h @@ -1,4 +1,8 @@ /* SPDX-License-Identifier: GPL-2.0 */ +#define PDMA_FILTER_PARAM(_prio, _requestor) (&(struct pxad_param) { \ + .prio = PXAD_PRIO_##_prio, .drcmr = _requestor }) +struct mmp_dma_platdata; + extern struct platform_device pxa_device_pmu; extern struct platform_device pxa_device_mci; extern struct platform_device pxa3xx_device_mci2; @@ -55,7 +59,7 @@ extern struct platform_device pxa3xx_device_gpio; extern struct platform_device pxa93x_device_gpio; void __init pxa_register_device(struct platform_device *dev, void *data); -void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors); +void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata); struct i2c_pxa_platform_data; extern void pxa_set_i2c_info(struct i2c_pxa_platform_data *info); diff --git a/arch/arm/mach-pxa/pxa25x.c b/arch/arm/mach-pxa/pxa25x.c index ba431fad5c47..ab8808ce7e21 100644 --- a/arch/arm/mach-pxa/pxa25x.c +++ b/arch/arm/mach-pxa/pxa25x.c @@ -16,6 +16,8 @@ * initialization stuff for PXA machines which can be overridden later if * need be. */ +#include +#include #include #include #include @@ -26,6 +28,7 @@ #include #include #include +#include #include #include @@ -201,6 +204,39 @@ static struct platform_device *pxa25x_devices[] __initdata = { &pxa_device_asoc_platform, }; +static const struct dma_slave_map pxa25x_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + + /* PXA25x specific map */ + { "pxa25x-ssp.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa25x-ssp.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa25x-nssp.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa25x-nssp.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa25x-nssp.2", "rx", PDMA_FILTER_PARAM(LOWEST, 23) }, + { "pxa25x-nssp.2", "tx", PDMA_FILTER_PARAM(LOWEST, 24) }, +}; + +static struct mmp_dma_platdata pxa25x_dma_pdata = { + .dma_channels = 16, + .nb_requestors = 40, + .slave_map = pxa25x_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa25x_slave_map), +}; + static int __init pxa25x_init(void) { int ret = 0; @@ -215,7 +251,7 @@ static int __init pxa25x_init(void) register_syscore_ops(&pxa2xx_mfp_syscore_ops); if (!of_have_populated_dt()) { - pxa2xx_set_dmac_info(16, 40); + pxa2xx_set_dmac_info(&pxa25x_dma_pdata); pxa_register_device(&pxa25x_device_gpio, &pxa25x_gpio_info); ret = platform_add_devices(pxa25x_devices, ARRAY_SIZE(pxa25x_devices)); diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c index 0c06f383ad52..5a8990a9313d 100644 --- a/arch/arm/mach-pxa/pxa27x.c +++ b/arch/arm/mach-pxa/pxa27x.c @@ -11,6 +11,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include +#include #include #include #include @@ -23,6 +25,7 @@ #include #include #include +#include #include #include @@ -297,6 +300,40 @@ static struct platform_device *devices[] __initdata = { &pxa27x_device_pwm1, }; +static const struct dma_slave_map pxa27x_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) }, + + /* PXA27x specific map */ + { "pxa2xx-i2s", "rx", PDMA_FILTER_PARAM(LOWEST, 2) }, + { "pxa2xx-i2s", "tx", PDMA_FILTER_PARAM(LOWEST, 3) }, + { "pxa27x-camera.0", "CI_Y", PDMA_FILTER_PARAM(HIGHEST, 68) }, + { "pxa27x-camera.0", "CI_U", PDMA_FILTER_PARAM(HIGHEST, 69) }, + { "pxa27x-camera.0", "CI_V", PDMA_FILTER_PARAM(HIGHEST, 70) }, +}; + +static struct mmp_dma_platdata pxa27x_dma_pdata = { + .dma_channels = 32, + .nb_requestors = 75, + .slave_map = pxa27x_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa27x_slave_map), +}; + static int __init pxa27x_init(void) { int ret = 0; @@ -313,7 +350,7 @@ static int __init pxa27x_init(void) if (!of_have_populated_dt()) { pxa_register_device(&pxa27x_device_gpio, &pxa27x_gpio_info); - pxa2xx_set_dmac_info(32, 75); + pxa2xx_set_dmac_info(&pxa27x_dma_pdata); ret = platform_add_devices(devices, ARRAY_SIZE(devices)); } diff --git a/arch/arm/mach-pxa/pxa3xx.c b/arch/arm/mach-pxa/pxa3xx.c index 8c64f93b669b..df9c8970adcf 100644 --- a/arch/arm/mach-pxa/pxa3xx.c +++ b/arch/arm/mach-pxa/pxa3xx.c @@ -12,6 +12,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include +#include #include #include #include @@ -24,6 +26,7 @@ #include #include #include +#include #include #include @@ -421,6 +424,42 @@ static struct platform_device *devices[] __initdata = { &pxa27x_device_pwm1, }; +static const struct dma_slave_map pxa3xx_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) }, + + /* PXA3xx specific map */ + { "pxa-ssp-dai.3", "rx", PDMA_FILTER_PARAM(LOWEST, 2) }, + { "pxa-ssp-dai.3", "tx", PDMA_FILTER_PARAM(LOWEST, 3) }, + { "pxa2xx-mci.1", "rx", PDMA_FILTER_PARAM(LOWEST, 93) }, + { "pxa2xx-mci.1", "tx", PDMA_FILTER_PARAM(LOWEST, 94) }, + { "pxa3xx-nand", "data", PDMA_FILTER_PARAM(LOWEST, 97) }, + { "pxa2xx-mci.2", "rx", PDMA_FILTER_PARAM(LOWEST, 100) }, + { "pxa2xx-mci.2", "tx", PDMA_FILTER_PARAM(LOWEST, 101) }, +}; + +static struct mmp_dma_platdata pxa3xx_dma_pdata = { + .dma_channels = 32, + .nb_requestors = 100, + .slave_map = pxa3xx_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa3xx_slave_map), +}; + static int __init pxa3xx_init(void) { int ret = 0; @@ -456,7 +495,7 @@ static int __init pxa3xx_init(void) if (of_have_populated_dt()) return 0; - pxa2xx_set_dmac_info(32, 100); + pxa2xx_set_dmac_info(&pxa3xx_dma_pdata); ret = platform_add_devices(devices, ARRAY_SIZE(devices)); if (ret) return ret; From 88a0513cf6114bbabbd3a158f039dbd03f49e0bf Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:06 +0200 Subject: [PATCH 043/529] dmaengine: pxa: add a default requestor policy As what former drcmr -1 value meant, add a this as a default to each channel, ie. that by default no requestor line is used. This is specifically used for network drivers smc91x and smc911x, and needed for their port to slave maps. Cc: Arnd Bergmann Signed-off-by: Robert Jarzmik Acked-by: Vinod Koul --- drivers/dma/pxa_dma.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/drivers/dma/pxa_dma.c b/drivers/dma/pxa_dma.c index 9505334f9c6e..b31c28b67ad3 100644 --- a/drivers/dma/pxa_dma.c +++ b/drivers/dma/pxa_dma.c @@ -762,6 +762,8 @@ static void pxad_free_chan_resources(struct dma_chan *dchan) dma_pool_destroy(chan->desc_pool); chan->desc_pool = NULL; + chan->drcmr = U32_MAX; + chan->prio = PXAD_PRIO_LOWEST; } static void pxad_free_desc(struct virt_dma_desc *vd) @@ -1386,6 +1388,9 @@ static int pxad_init_dmadev(struct platform_device *op, c = devm_kzalloc(&op->dev, sizeof(*c), GFP_KERNEL); if (!c) return -ENOMEM; + + c->drcmr = U32_MAX; + c->prio = PXAD_PRIO_LOWEST; c->vc.desc_free = pxad_free_desc; vchan_init(&c->vc, &pdev->slave); init_waitqueue_head(&c->wq_state); From 6b3348f9e6eb35d2c2d49ffa274039ef9a901adc Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:07 +0200 Subject: [PATCH 044/529] mmc: pxamci: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik Acked-by: Ulf Hansson --- drivers/mmc/host/pxamci.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) diff --git a/drivers/mmc/host/pxamci.c b/drivers/mmc/host/pxamci.c index c763b404510f..6c94474e36f4 100644 --- a/drivers/mmc/host/pxamci.c +++ b/drivers/mmc/host/pxamci.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include @@ -637,10 +636,8 @@ static int pxamci_probe(struct platform_device *pdev) { struct mmc_host *mmc; struct pxamci_host *host = NULL; - struct resource *r, *dmarx, *dmatx; - struct pxad_param param_rx, param_tx; + struct resource *r; int ret, irq, gpio_cd = -1, gpio_ro = -1, gpio_power = -1; - dma_cap_mask_t mask; ret = pxamci_of_init(pdev); if (ret) @@ -739,34 +736,14 @@ static int pxamci_probe(struct platform_device *pdev) platform_set_drvdata(pdev, mmc); - if (!pdev->dev.of_node) { - dmarx = platform_get_resource(pdev, IORESOURCE_DMA, 0); - dmatx = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx || !dmatx) { - ret = -ENXIO; - goto out; - } - param_rx.prio = PXAD_PRIO_LOWEST; - param_rx.drcmr = dmarx->start; - param_tx.prio = PXAD_PRIO_LOWEST; - param_tx.drcmr = dmatx->start; - } - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - - host->dma_chan_rx = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m_rx, &pdev->dev, "rx"); + host->dma_chan_rx = dma_request_slave_channel(&pdev->dev, "rx"); if (host->dma_chan_rx == NULL) { dev_err(&pdev->dev, "unable to request rx dma channel\n"); ret = -ENODEV; goto out; } - host->dma_chan_tx = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m_tx, &pdev->dev, "tx"); + host->dma_chan_tx = dma_request_slave_channel(&pdev->dev, "tx"); if (host->dma_chan_tx == NULL) { dev_err(&pdev->dev, "unable to request tx dma channel\n"); ret = -ENODEV; From f727b6cda449184188d8a64987f194687bf01782 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:08 +0200 Subject: [PATCH 045/529] media: pxa_camera: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik Acked-by: Hans Verkuil Acked-by: Mauro Carvalho Chehab --- drivers/media/platform/pxa_camera.c | 22 +++------------------- 1 file changed, 3 insertions(+), 19 deletions(-) diff --git a/drivers/media/platform/pxa_camera.c b/drivers/media/platform/pxa_camera.c index d85ffbfb7c1f..b6e9e93bde7a 100644 --- a/drivers/media/platform/pxa_camera.c +++ b/drivers/media/platform/pxa_camera.c @@ -2375,8 +2375,6 @@ static int pxa_camera_probe(struct platform_device *pdev) .src_maxburst = 8, .direction = DMA_DEV_TO_MEM, }; - dma_cap_mask_t mask; - struct pxad_param params; char clk_name[V4L2_CLK_NAME_SIZE]; int irq; int err = 0, i; @@ -2450,34 +2448,20 @@ static int pxa_camera_probe(struct platform_device *pdev) pcdev->base = base; /* request dma */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_PRIVATE, mask); - - params.prio = 0; - params.drcmr = 68; - pcdev->dma_chans[0] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_Y"); + pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y"); if (!pcdev->dma_chans[0]) { dev_err(&pdev->dev, "Can't request DMA for Y\n"); return -ENODEV; } - params.drcmr = 69; - pcdev->dma_chans[1] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_U"); + pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U"); if (!pcdev->dma_chans[1]) { dev_err(&pdev->dev, "Can't request DMA for Y\n"); err = -ENODEV; goto exit_free_dma_y; } - params.drcmr = 70; - pcdev->dma_chans[2] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_V"); + pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V"); if (!pcdev->dma_chans[2]) { dev_err(&pdev->dev, "Can't request DMA for V\n"); err = -ENODEV; From ac75a50b6de3b092d084fdd9818707d0d5073ad6 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:09 +0200 Subject: [PATCH 046/529] mtd: rawnand: marvell: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Daniel Mack Signed-off-by: Robert Jarzmik Acked-by: Miquel Raynal --- drivers/mtd/nand/raw/marvell_nand.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/drivers/mtd/nand/raw/marvell_nand.c b/drivers/mtd/nand/raw/marvell_nand.c index ebb1d141b900..00d9f29bbdb6 100644 --- a/drivers/mtd/nand/raw/marvell_nand.c +++ b/drivers/mtd/nand/raw/marvell_nand.c @@ -2612,8 +2612,6 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc) dev); struct dma_slave_config config = {}; struct resource *r; - dma_cap_mask_t mask; - struct pxad_param param; int ret; if (!IS_ENABLED(CONFIG_PXA_DMA)) { @@ -2626,20 +2624,7 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc) if (ret) return ret; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - dev_err(nfc->dev, "No resource defined for data DMA\n"); - return -ENXIO; - } - - param.drcmr = r->start; - param.prio = PXAD_PRIO_LOWEST; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - nfc->dma_chan = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m, nfc->dev, - "data"); + nfc->dma_chan = dma_request_slave_channel(nfc->dev, "data"); if (!nfc->dma_chan) { dev_err(nfc->dev, "Unable to request data DMA channel\n"); From 273340e8bf86de53eef7073993352ea11c563696 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:13 +0200 Subject: [PATCH 047/529] ata: pata_pxa: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik Acked-by: Bartlomiej Zolnierkiewicz --- drivers/ata/pata_pxa.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/drivers/ata/pata_pxa.c b/drivers/ata/pata_pxa.c index f6c46e9a4dc0..e8b6a2e464c9 100644 --- a/drivers/ata/pata_pxa.c +++ b/drivers/ata/pata_pxa.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include @@ -180,8 +179,6 @@ static int pxa_ata_probe(struct platform_device *pdev) struct resource *irq_res; struct pata_pxa_pdata *pdata = dev_get_platdata(&pdev->dev); struct dma_slave_config config; - dma_cap_mask_t mask; - struct pxad_param param; int ret = 0; /* @@ -278,10 +275,6 @@ static int pxa_ata_probe(struct platform_device *pdev) ap->private_data = data; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - param.prio = PXAD_PRIO_LOWEST; - param.drcmr = pdata->dma_dreq; memset(&config, 0, sizeof(config)); config.src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; config.dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; @@ -294,8 +287,7 @@ static int pxa_ata_probe(struct platform_device *pdev) * Request the DMA channel */ data->dma_chan = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m, &pdev->dev, "data"); + dma_request_slave_channel(&pdev->dev, "data"); if (!data->dma_chan) return -EBUSY; ret = dmaengine_slave_config(data->dma_chan, &config); From b6d1a17f4729e4fda5740a855da91d202db2c118 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:14 +0200 Subject: [PATCH 048/529] dmaengine: pxa: document pxad_param Add some documentation for the pxad_param structure, and describe the contract behind the minimal required priority of a DMA channel. Signed-off-by: Robert Jarzmik Acked-by: Vinod Koul --- include/linux/dma/pxa-dma.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/include/linux/dma/pxa-dma.h b/include/linux/dma/pxa-dma.h index e56ec7af4fd7..9fc594f69eff 100644 --- a/include/linux/dma/pxa-dma.h +++ b/include/linux/dma/pxa-dma.h @@ -9,6 +9,15 @@ enum pxad_chan_prio { PXAD_PRIO_LOWEST, }; +/** + * struct pxad_param - dma channel request parameters + * @drcmr: requestor line number + * @prio: minimal mandatory priority of the channel + * + * If a requested channel is granted, its priority will be at least @prio, + * ie. if PXAD_PRIO_LOW is required, the requested channel will be either + * PXAD_PRIO_LOW, PXAD_PRIO_NORMAL or PXAD_PRIO_HIGHEST. + */ struct pxad_param { unsigned int drcmr; enum pxad_chan_prio prio; From b77ed2e6d61d40117272be1b2377c5dfd101e9cd Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:16 +0200 Subject: [PATCH 049/529] ARM: pxa: remove the DMA IO resources As the last driver using the former mechanism to acquire the DMA requestor line has be converted to the dma_slave_map, remove all these resources from the PXA devices. Signed-off-by: Robert Jarzmik --- arch/arm/mach-pxa/devices.c | 136 ------------------------------------ 1 file changed, 136 deletions(-) diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index 1e8915fc340d..5a16ea74e28a 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -60,16 +60,6 @@ static struct resource pxamci_resources[] = { .end = IRQ_MMC, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 21, - .end = 21, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 22, - .end = 22, - .flags = IORESOURCE_DMA, - }, }; static u64 pxamci_dmamask = 0xffffffffUL; @@ -407,16 +397,6 @@ static struct resource pxa_ir_resources[] = { .end = 0x40700023, .flags = IORESOURCE_MEM, }, - [5] = { - .start = 17, - .end = 17, - .flags = IORESOURCE_DMA, - }, - [6] = { - .start = 18, - .end = 18, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa_device_ficp = { @@ -545,18 +525,6 @@ static struct resource pxa25x_resource_ssp[] = { .end = IRQ_SSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 13, - .end = 13, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 14, - .end = 14, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_ssp = { @@ -583,18 +551,6 @@ static struct resource pxa25x_resource_nssp[] = { .end = IRQ_NSSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 15, - .end = 15, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 16, - .end = 16, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_nssp = { @@ -621,18 +577,6 @@ static struct resource pxa25x_resource_assp[] = { .end = IRQ_ASSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 23, - .end = 23, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 24, - .end = 24, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_assp = { @@ -751,18 +695,6 @@ static struct resource pxa27x_resource_ssp1[] = { .end = IRQ_SSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 13, - .end = 13, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 14, - .end = 14, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp1 = { @@ -789,18 +721,6 @@ static struct resource pxa27x_resource_ssp2[] = { .end = IRQ_SSP2, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 15, - .end = 15, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 16, - .end = 16, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp2 = { @@ -827,18 +747,6 @@ static struct resource pxa27x_resource_ssp3[] = { .end = IRQ_SSP3, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 66, - .end = 66, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 67, - .end = 67, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp3 = { @@ -895,16 +803,6 @@ static struct resource pxa3xx_resources_mci2[] = { .end = IRQ_MMC2, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 93, - .end = 93, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 94, - .end = 94, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa3xx_device_mci2 = { @@ -934,16 +832,6 @@ static struct resource pxa3xx_resources_mci3[] = { .end = IRQ_MMC3, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 100, - .end = 100, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 101, - .end = 101, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa3xx_device_mci3 = { @@ -1021,18 +909,6 @@ static struct resource pxa3xx_resources_nand[] = { .end = IRQ_NAND, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for Data DMA */ - .start = 97, - .end = 97, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for Command DMA */ - .start = 99, - .end = 99, - .flags = IORESOURCE_DMA, - }, }; static u64 pxa3xx_nand_dma_mask = DMA_BIT_MASK(32); @@ -1066,18 +942,6 @@ static struct resource pxa3xx_resource_ssp4[] = { .end = IRQ_SSP4, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 2, - .end = 2, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 3, - .end = 3, - .flags = IORESOURCE_DMA, - }, }; /* From cd31b80736852d34bc1072f3e579a6fd73a244e7 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:17 +0200 Subject: [PATCH 050/529] ARM: pxa: change SSP DMA channels allocation Now the dma_slave_map is available for PXA architecture, switch the SSP device to it. This specifically means that : - for platform data based machines, the DMA requestor channels are extracted from the slave map, where pxa-ssp-dai. is a 1-1 match to ssp., and the channels are either "rx" or "tx". - for device tree platforms, the dma node should be hooked into the pxa2xx-ac97 or pxa-ssp-dai node. Signed-off-by: Robert Jarzmik Acked-by: Daniel Mack --- arch/arm/plat-pxa/ssp.c | 47 -------------------------------------- include/linux/pxa2xx_ssp.h | 2 -- sound/soc/pxa/pxa-ssp.c | 5 ++-- 3 files changed, 2 insertions(+), 52 deletions(-) diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index ba13f793fbce..ed36dcab80f1 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -127,53 +127,6 @@ static int pxa_ssp_probe(struct platform_device *pdev) if (IS_ERR(ssp->clk)) return PTR_ERR(ssp->clk); - if (dev->of_node) { - struct of_phandle_args dma_spec; - struct device_node *np = dev->of_node; - int ret; - - /* - * FIXME: we should allocate the DMA channel from this - * context and pass the channel down to the ssp users. - * For now, we lookup the rx and tx indices manually - */ - - /* rx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 0, &dma_spec); - - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_rx = dma_spec.args[0]; - of_node_put(dma_spec.np); - - /* tx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 1, &dma_spec); - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_tx = dma_spec.args[0]; - of_node_put(dma_spec.np); - } else { - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (res == NULL) { - dev_err(dev, "no SSP RX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_rx = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res == NULL) { - dev_err(dev, "no SSP TX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_tx = res->start; - } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { dev_err(dev, "no memory resource defined\n"); diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 8461b18e4608..03a7ca46735b 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -212,8 +212,6 @@ struct ssp_device { int type; int use_count; int irq; - int drcmr_rx; - int drcmr_tx; struct device_node *of_node; }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc986080130..0b441338bdd4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -105,9 +105,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - - dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - &ssp->drcmr_tx : &ssp->drcmr_rx; + dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "tx" : "rx"; snd_soc_dai_set_dma_data(cpu_dai, substream, dma); From 5fd46e649ee63259f2197625662477ac67a69e79 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Tue, 19 Jun 2018 14:16:06 +0800 Subject: [PATCH 051/529] ASoC: Intel: kbl_da7219_max98357a: add fe_ops for kbl Audio Capture Port platform support fixed constraint hw_prams as Stereo, 48KHz, 16 bits. This fixed the headset mic recorded noise due to mono capturing request from some apps. e.g. online Voice Recorder Signed-off-by: Louis Collard Signed-off-by: Mac Chiang Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 94294c27d1db..7961f1fd18bd 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .trigger = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", From 95555f580dca21fac5ea35c10fa92fa034bd403f Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Mon, 18 Jun 2018 13:29:35 -0500 Subject: [PATCH 052/529] ASoC: Intel: broxton: reduce machine name for bxt_da7219_max98357a Use truncated names in bxt id table and bxt_da7219_max98357a machine as platform device id table expects names to be less then 20chars. Signed-off-by: Naveen Manohar Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- sound/soc/intel/skylake/skl.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 40eb979d5ac1..3aba5bcf806a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -586,7 +586,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { - .name = "bxt_da7219_max98357a_i2s", + .name = "bxt_da7219_max98357a", .pm = &snd_soc_pm_ops, }, }; @@ -599,4 +599,4 @@ MODULE_AUTHOR("Rohit Ainapure "); MODULE_AUTHOR("Harsha Priya "); MODULE_AUTHOR("Conrad Cooke "); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s"); +MODULE_ALIAS("platform:bxt_da7219_max98357a"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f0d9793f872a..0a8f0768e987 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -1093,7 +1093,7 @@ static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { }, { .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a_i2s", + .drv_name = "bxt_da7219_max98357a", .fw_filename = "intel/dsp_fw_bxtn.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, From 5f15f267daf81a4c7c2a1cd2a0d6743ec7fc8b59 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:36 -0500 Subject: [PATCH 053/529] ASoC: Intel: Skylake: cleanup before moving ACPI tables There is no need to deal with DMICs if the DSP is not present and there is no ACPI machine ID found. Simplify before moving these ACPI tables to sound/soc/intel/common Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 0a8f0768e987..6dec748e8949 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -500,10 +500,12 @@ static int skl_find_machine(struct skl *skl, void *driver_data) skl->mach = mach; skl->fw_name = mach->fw_filename; - pdata = skl->mach->pdata; + pdata = mach->pdata; - if (mach->pdata) + if (pdata) { skl->use_tplg_pcm = pdata->use_tplg_pcm; + pdata->dmic_num = skl_get_dmic_geo(skl); + } return 0; } @@ -930,8 +932,6 @@ static int skl_probe(struct pci_dev *pci, pci_set_drvdata(skl->pci, ebus); - skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); - /* check if dsp is there */ if (bus->ppcap) { /* create device for dsp clk */ From cbaa7f0bdbee1969bb311c641abbd0d2af6ba861 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:37 -0500 Subject: [PATCH 054/529] ASoC: Intel: move SKL+ codec ACPI tables to common directory No functionality change, just move to common tables to make it easier to deal with SOF and share the same machine drivers - as done previously for BYT/CHT/HSW/BDW. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 5 + sound/soc/intel/common/Makefile | 6 +- .../intel/common/soc-acpi-intel-bxt-match.c | 35 ++++ .../intel/common/soc-acpi-intel-cnl-match.c | 29 ++++ .../intel/common/soc-acpi-intel-glk-match.c | 23 +++ .../intel/common/soc-acpi-intel-kbl-match.c | 91 ++++++++++ .../intel/common/soc-acpi-intel-skl-match.c | 47 +++++ sound/soc/intel/skylake/skl.c | 162 +----------------- 8 files changed, 241 insertions(+), 157 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-bxt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cnl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-glk-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-kbl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-skl-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 9da6388c20a1..917ddd0f2762 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -29,5 +29,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[]; #endif diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 7379d8830c39..915a34cdc8ac 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o -snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o +snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ + soc-acpi-intel-hsw-bdw-match.o \ + soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ + soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ + soc-acpi-intel-cnl-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c new file mode 100644 index 000000000000..569f1de97e82 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -0,0 +1,35 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include + +static struct snd_soc_acpi_codecs bxt_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { + { + .id = "INT343A", + .drv_name = "bxt_alc298s_i2s", + .fw_filename = "intel/dsp_fw_bxtn.bin", + }, + { + .id = "DLGS7219", + .drv_name = "bxt_da7219_max98357a", + .fw_filename = "intel/dsp_fw_bxtn.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &bxt_codecs, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c new file mode 100644 index 000000000000..b83ee053d417 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -0,0 +1,29 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata cnl_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { + { + .id = "INT34C2", + .drv_name = "cnl_rt274", + .fw_filename = "intel/dsp_fw_cnl.bin", + .pdata = &cnl_pdata, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c new file mode 100644 index 000000000000..dee09439e7cb --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -0,0 +1,23 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include + +struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { + { + .id = "INT343A", + .drv_name = "glk_alc298s_i2s", + .fw_filename = "intel/dsp_fw_glk.bin", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c new file mode 100644 index 000000000000..0ee173ca437d --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -0,0 +1,91 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs kbl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { + .num_codecs = 1, + .codecs = {"10EC5663"} +}; + +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { + .num_codecs = 2, + .codecs = {"10EC5663", "10EC5514"} +}; + +static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { + { + .id = "INT343A", + .drv_name = "kbl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "INT343B", + .drv_name = "kbl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "kbl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_r5514_5663_max", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_5663_5514_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_rt5663_m98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_poppy_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "10EC5663", + .drv_name = "kbl_rt5663", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98357_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c new file mode 100644 index 000000000000..0c9c0edd35b3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs skl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { + { + .id = "INT343A", + .drv_name = "skl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_release.bin", + }, + { + .id = "INT343B", + .drv_name = "skl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "skl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6dec748e8949..670ff9aaca55 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -36,8 +37,6 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" -static struct skl_machine_pdata skl_dmic_data; - /* * initialize the PCI registers */ @@ -1026,172 +1025,23 @@ static void skl_remove(struct pci_dev *pci) dev_set_drvdata(&pci->dev, NULL); } -static struct snd_soc_acpi_codecs skl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs kbl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs bxt_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct snd_soc_acpi_codecs kbl_poppy_codecs = { - .num_codecs = 1, - .codecs = {"10EC5663"} -}; - -static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { - .num_codecs = 2, - .codecs = {"10EC5663", "10EC5514"} -}; - -static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct skl_machine_pdata cnl_pdata = { - .use_tplg_pcm = true, -}; - -static struct snd_soc_acpi_mach sst_skl_devdata[] = { - { - .id = "INT343A", - .drv_name = "skl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_release.bin", - }, - { - .id = "INT343B", - .drv_name = "skl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "skl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - {} -}; - -static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { - { - .id = "INT343A", - .drv_name = "bxt_alc298s_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - }, - { - .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a", - .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &bxt_codecs, - }, - {} -}; - -static struct snd_soc_acpi_mach sst_kbl_devdata[] = { - { - .id = "INT343A", - .drv_name = "kbl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "INT343B", - .drv_name = "kbl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "kbl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_r5514_5663_max", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_5663_5514_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_rt5663_m98927", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_poppy_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "10EC5663", - .drv_name = "kbl_rt5663", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_7219_98357_codecs, - .pdata = &skl_dmic_data - }, - - {} -}; - -static struct snd_soc_acpi_mach sst_glk_devdata[] = { - { - .id = "INT343A", - .drv_name = "glk_alc298s_i2s", - .fw_filename = "intel/dsp_fw_glk.bin", - }, - {} -}; - -static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { - { - .id = "INT34C2", - .drv_name = "cnl_rt274", - .fw_filename = "intel/dsp_fw_cnl.bin", - .pdata = &cnl_pdata, - }, - {} -}; - /* PCI IDs */ static const struct pci_device_id skl_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&sst_skl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = (unsigned long)&sst_bxtp_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), - .driver_data = (unsigned long)&sst_kbl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, /* GLK */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = (unsigned long)&sst_glk_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = (unsigned long)&sst_cnl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); From 65a33883c778befcb85ef45285763fd8ac1b2ba3 Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Mon, 18 Jun 2018 13:29:38 -0500 Subject: [PATCH 055/529] ASoC: Intel: common: Add Geminilake Dialog+Maxim machine driver entry This patch adds da7219_max98357a machine driver entry into machine table Signed-off-by: Naveen Manohar Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index dee09439e7cb..5902aa1d0ee3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -9,12 +9,24 @@ #include #include +static struct snd_soc_acpi_codecs glk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { { .id = "INT343A", .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + { + .id = "DLGS7219", + .drv_name = "glk_da7219_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); From e6d298fd4a4454dd121343323e3f00a27f8819a4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:39 -0500 Subject: [PATCH 056/529] ASoC: Intel: common: add firmware/topology information for SOF No functionality change for Skylake driver, add relevant names needed by SOF for BXT/APL, GLK and CNL. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 +++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 3 +++ sound/soc/intel/common/soc-acpi-intel-glk-match.c | 6 ++++++ 3 files changed, 12 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 569f1de97e82..50869eddbb3b 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -26,6 +26,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .fw_filename = "intel/dsp_fw_bxtn.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index b83ee053d417..ec8e28e7b937 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -20,6 +20,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .drv_name = "cnl_rt274", .fw_filename = "intel/dsp_fw_cnl.bin", .pdata = &cnl_pdata, + .sof_fw_filename = "intel/sof-cnl.ri", + .sof_tplg_filename = "intel/sof-cnl-rt274.tplg", + .asoc_plat_name = "0000:00:1f.3", }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 5902aa1d0ee3..305875af71ca 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -19,6 +19,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .id = "INT343A", .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-alc298.tplg", + .asoc_plat_name = "0000:00:0e.0", }, { .id = "DLGS7219", @@ -26,6 +29,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .fw_filename = "intel/dsp_fw_glk.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &glk_codecs, + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", }, {}, }; From b45350135b9241b64cc91ccc8dddca2ee4dc25d7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:41 -0500 Subject: [PATCH 057/529] ASoC: Intel: common: add entries for SOF-based machine drivers While we are at it, add entries for machine drivers that are used on SOF-based platforms. The drivers will be submitted upstream after the core SOF patches, but there's no harm in adding these references now. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-bxt-match.c | 21 +++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 50869eddbb3b..f39386e540d3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -30,6 +30,27 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .sof_tplg_filename = "intel/sof-apl-da7219.tplg", .asoc_plat_name = "0000:00:0e.0", }, + { + .id = "104C5122", + .drv_name = "bxt-pcm512x", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-pcm512x.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "1AEC8804", + .drv_name = "bxt-wm8804", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-wm8804.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "INT34C3", + .drv_name = "bxt_tdf8532", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); From f0d9034b290d8bad590e843c2a1081eb47d813ad Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:42 -0500 Subject: [PATCH 058/529] ASoC: Intel: common: fix copy/paste issue with SOF/broadwell topology file There are two commercially-available Broadwell platforms based on I2S (Dell XPS13 and 'Samus' Pixel 2015 Chromebook). Fix a copy/paste issue to allow each platform to enable different features if needed when SOF is enabled Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index e0e8c8c27528..53e7acdfbb81 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -45,7 +45,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_tplg_filename = "intel/reef-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { From 244e293690a6e07cbdfa11af1977488d91931eed Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Jun 2018 16:22:09 +0100 Subject: [PATCH 059/529] ASoC: pcm: Tidy up open/hw_params handling Currently, the core will continue processing open/hw_params component callbacks after one has failed even though it will abort immediately afterwards. This is unnecessary and also has the issue that close/hw_free will be called on the component which failed open/hw_params which could result in issues if the driver doesn't expect this behaviour. Update the core to abort processing open/hw_params when an error is hit and only call close/hw_free for those components that were successfully opened. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 116 +++++++++++++++++++++++--------------------- 1 file changed, 62 insertions(+), 54 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..45b52f7b9690 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -448,6 +448,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->close) + continue; + + component->driver->ops->close(substream); + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -462,7 +485,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; const char *codec_dai_name = "multicodec"; - int i, ret = 0, __ret; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); for (i = 0; i < rtd->num_codecs; i++) @@ -486,7 +509,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -494,16 +516,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) !component->driver->ops->open) continue; - __ret = component->driver->ops->open(substream); - if (__ret < 0) { + ret = component->driver->ops->open(substream); + if (ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; @@ -612,15 +633,7 @@ codec_dai_err: } component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, component); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -714,15 +727,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, NULL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { @@ -874,6 +879,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->hw_free) + continue; + + component->driver->ops->hw_free(substream); + } + + return 0; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -886,7 +914,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret = 0, __ret; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops->hw_params) { @@ -944,7 +972,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto interface_err; - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -952,16 +979,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, !component->driver->ops->hw_params) continue; - __ret = component->driver->ops->hw_params(substream, params); - if (__ret < 0) { + ret = component->driver->ops->hw_params(substream, params); + if (ret < 0) { dev_err(component->dev, "ASoC: %s hw params failed: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); @@ -977,15 +1003,7 @@ out: return ret; component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, component); if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -1014,8 +1032,6 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; @@ -1052,15 +1068,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) rtd->dai_link->ops->hw_free(substream); /* free any component resources */ - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { From 9fba738a53dda20e748d6ee240b6c017c8146b4b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 19 Jun 2018 16:41:41 +0200 Subject: [PATCH 060/529] clk: add duty cycle support Add the possibility to apply and query the clock signal duty cycle ratio. This is useful when the duty cycle of the clock signal depends on some other parameters controlled by the clock framework. For example, the duty cycle of a divider may depends on the raw divider setting (ratio = N / div) , which is controlled by the CCF. In such case, going through the pwm framework to control the duty cycle ratio of this clock would be a burden. A clock provider is not required to implement the operation to set and get the duty cycle. If it does not implement .get_duty_cycle(), the ratio is assumed to be 50%. This change also adds a new flag, CLK_DUTY_CYCLE_PARENT. This flag should be used to indicate that a clock, such as gates and muxes, may inherit the duty cycle ratio of its parent clock. If a clock does not provide a get_duty_cycle() callback and has CLK_DUTY_CYCLE_PARENT, then the call will be directly forwarded to its parent clock, if any. For set_duty_cycle(), the clock should also have CLK_SET_RATE_PARENT for the call to be forwarded Signed-off-by: Jerome Brunet Signed-off-by: Michael Turquette Link: lkml.kernel.org/r/20180619144141.8506-1-jbrunet@baylibre.com --- drivers/clk/clk.c | 199 ++++++++++++++++++++++++++++++++++- include/linux/clk-provider.h | 26 +++++ include/linux/clk.h | 33 ++++++ include/trace/events/clk.h | 36 +++++++ 4 files changed, 289 insertions(+), 5 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index 9760b526ca31..b0a2719d86f3 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -68,6 +68,7 @@ struct clk_core { unsigned long max_rate; unsigned long accuracy; int phase; + struct clk_duty duty; struct hlist_head children; struct hlist_node child_node; struct hlist_head clks; @@ -2402,6 +2403,172 @@ int clk_get_phase(struct clk *clk) } EXPORT_SYMBOL_GPL(clk_get_phase); +static void clk_core_reset_duty_cycle_nolock(struct clk_core *core) +{ + /* Assume a default value of 50% */ + core->duty.num = 1; + core->duty.den = 2; +} + +static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core); + +static int clk_core_update_duty_cycle_nolock(struct clk_core *core) +{ + struct clk_duty *duty = &core->duty; + int ret = 0; + + if (!core->ops->get_duty_cycle) + return clk_core_update_duty_cycle_parent_nolock(core); + + ret = core->ops->get_duty_cycle(core->hw, duty); + if (ret) + goto reset; + + /* Don't trust the clock provider too much */ + if (duty->den == 0 || duty->num > duty->den) { + ret = -EINVAL; + goto reset; + } + + return 0; + +reset: + clk_core_reset_duty_cycle_nolock(core); + return ret; +} + +static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core) +{ + int ret = 0; + + if (core->parent && + core->flags & CLK_DUTY_CYCLE_PARENT) { + ret = clk_core_update_duty_cycle_nolock(core->parent); + memcpy(&core->duty, &core->parent->duty, sizeof(core->duty)); + } else { + clk_core_reset_duty_cycle_nolock(core); + } + + return ret; +} + +static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core, + struct clk_duty *duty); + +static int clk_core_set_duty_cycle_nolock(struct clk_core *core, + struct clk_duty *duty) +{ + int ret; + + lockdep_assert_held(&prepare_lock); + + if (clk_core_rate_is_protected(core)) + return -EBUSY; + + trace_clk_set_duty_cycle(core, duty); + + if (!core->ops->set_duty_cycle) + return clk_core_set_duty_cycle_parent_nolock(core, duty); + + ret = core->ops->set_duty_cycle(core->hw, duty); + if (!ret) + memcpy(&core->duty, duty, sizeof(*duty)); + + trace_clk_set_duty_cycle_complete(core, duty); + + return ret; +} + +static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core, + struct clk_duty *duty) +{ + int ret = 0; + + if (core->parent && + core->flags & (CLK_DUTY_CYCLE_PARENT | CLK_SET_RATE_PARENT)) { + ret = clk_core_set_duty_cycle_nolock(core->parent, duty); + memcpy(&core->duty, &core->parent->duty, sizeof(core->duty)); + } + + return ret; +} + +/** + * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal + * @clk: clock signal source + * @num: numerator of the duty cycle ratio to be applied + * @den: denominator of the duty cycle ratio to be applied + * + * Apply the duty cycle ratio if the ratio is valid and the clock can + * perform this operation + * + * Returns (0) on success, a negative errno otherwise. + */ +int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den) +{ + int ret; + struct clk_duty duty; + + if (!clk) + return 0; + + /* sanity check the ratio */ + if (den == 0 || num > den) + return -EINVAL; + + duty.num = num; + duty.den = den; + + clk_prepare_lock(); + + if (clk->exclusive_count) + clk_core_rate_unprotect(clk->core); + + ret = clk_core_set_duty_cycle_nolock(clk->core, &duty); + + if (clk->exclusive_count) + clk_core_rate_protect(clk->core); + + clk_prepare_unlock(); + + return ret; +} +EXPORT_SYMBOL_GPL(clk_set_duty_cycle); + +static int clk_core_get_scaled_duty_cycle(struct clk_core *core, + unsigned int scale) +{ + struct clk_duty *duty = &core->duty; + int ret; + + clk_prepare_lock(); + + ret = clk_core_update_duty_cycle_nolock(core); + if (!ret) + ret = mult_frac(scale, duty->num, duty->den); + + clk_prepare_unlock(); + + return ret; +} + +/** + * clk_get_scaled_duty_cycle - return the duty cycle ratio of a clock signal + * @clk: clock signal source + * @scale: scaling factor to be applied to represent the ratio as an integer + * + * Returns the duty cycle ratio of a clock node multiplied by the provided + * scaling factor, or negative errno on error. + */ +int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale) +{ + if (!clk) + return 0; + + return clk_core_get_scaled_duty_cycle(clk->core, scale); +} +EXPORT_SYMBOL_GPL(clk_get_scaled_duty_cycle); + /** * clk_is_match - check if two clk's point to the same hardware clock * @p: clk compared against q @@ -2455,12 +2622,13 @@ static void clk_summary_show_one(struct seq_file *s, struct clk_core *c, if (!c) return; - seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %-3d\n", + seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %5d %6d\n", level * 3 + 1, "", 30 - level * 3, c->name, c->enable_count, c->prepare_count, c->protect_count, clk_core_get_rate(c), clk_core_get_accuracy(c), - clk_core_get_phase(c)); + clk_core_get_phase(c), + clk_core_get_scaled_duty_cycle(c, 100000)); } static void clk_summary_show_subtree(struct seq_file *s, struct clk_core *c, @@ -2482,9 +2650,9 @@ static int clk_summary_show(struct seq_file *s, void *data) struct clk_core *c; struct hlist_head **lists = (struct hlist_head **)s->private; - seq_puts(s, " enable prepare protect \n"); - seq_puts(s, " clock count count count rate accuracy phase\n"); - seq_puts(s, "----------------------------------------------------------------------------------------\n"); + seq_puts(s, " enable prepare protect duty\n"); + seq_puts(s, " clock count count count rate accuracy phase cycle\n"); + seq_puts(s, "---------------------------------------------------------------------------------------------\n"); clk_prepare_lock(); @@ -2511,6 +2679,8 @@ static void clk_dump_one(struct seq_file *s, struct clk_core *c, int level) seq_printf(s, "\"rate\": %lu,", clk_core_get_rate(c)); seq_printf(s, "\"accuracy\": %lu,", clk_core_get_accuracy(c)); seq_printf(s, "\"phase\": %d", clk_core_get_phase(c)); + seq_printf(s, "\"duty_cycle\": %u", + clk_core_get_scaled_duty_cycle(c, 100000)); } static void clk_dump_subtree(struct seq_file *s, struct clk_core *c, int level) @@ -2572,6 +2742,7 @@ static const struct { ENTRY(CLK_SET_RATE_UNGATE), ENTRY(CLK_IS_CRITICAL), ENTRY(CLK_OPS_PARENT_ENABLE), + ENTRY(CLK_DUTY_CYCLE_PARENT), #undef ENTRY }; @@ -2610,6 +2781,17 @@ static int possible_parents_show(struct seq_file *s, void *data) } DEFINE_SHOW_ATTRIBUTE(possible_parents); +static int clk_duty_cycle_show(struct seq_file *s, void *data) +{ + struct clk_core *core = s->private; + struct clk_duty *duty = &core->duty; + + seq_printf(s, "%u/%u\n", duty->num, duty->den); + + return 0; +} +DEFINE_SHOW_ATTRIBUTE(clk_duty_cycle); + static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry) { struct dentry *root; @@ -2628,6 +2810,8 @@ static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry) debugfs_create_u32("clk_enable_count", 0444, root, &core->enable_count); debugfs_create_u32("clk_protect_count", 0444, root, &core->protect_count); debugfs_create_u32("clk_notifier_count", 0444, root, &core->notifier_count); + debugfs_create_file("clk_duty_cycle", 0444, root, core, + &clk_duty_cycle_fops); if (core->num_parents > 1) debugfs_create_file("clk_possible_parents", 0444, root, core, @@ -2845,6 +3029,11 @@ static int __clk_core_init(struct clk_core *core) else core->phase = 0; + /* + * Set clk's duty cycle. + */ + clk_core_update_duty_cycle_nolock(core); + /* * Set clk's rate. The preferred method is to use .recalc_rate. For * simple clocks and lazy developers the default fallback is to use the diff --git a/include/linux/clk-provider.h b/include/linux/clk-provider.h index b7cfa037e593..08b1aa70a38d 100644 --- a/include/linux/clk-provider.h +++ b/include/linux/clk-provider.h @@ -38,6 +38,8 @@ #define CLK_IS_CRITICAL BIT(11) /* do not gate, ever */ /* parents need enable during gate/ungate, set rate and re-parent */ #define CLK_OPS_PARENT_ENABLE BIT(12) +/* duty cycle call may be forwarded to the parent clock */ +#define CLK_DUTY_CYCLE_PARENT BIT(13) struct clk; struct clk_hw; @@ -66,6 +68,17 @@ struct clk_rate_request { struct clk_hw *best_parent_hw; }; +/** + * struct clk_duty - Struture encoding the duty cycle ratio of a clock + * + * @num: Numerator of the duty cycle ratio + * @den: Denominator of the duty cycle ratio + */ +struct clk_duty { + unsigned int num; + unsigned int den; +}; + /** * struct clk_ops - Callback operations for hardware clocks; these are to * be provided by the clock implementation, and will be called by drivers @@ -169,6 +182,15 @@ struct clk_rate_request { * by the second argument. Valid values for degrees are * 0-359. Return 0 on success, otherwise -EERROR. * + * @get_duty_cycle: Queries the hardware to get the current duty cycle ratio + * of a clock. Returned values denominator cannot be 0 and must be + * superior or equal to the numerator. + * + * @set_duty_cycle: Apply the duty cycle ratio to this clock signal specified by + * the numerator (2nd argurment) and denominator (3rd argument). + * Argument must be a valid ratio (denominator > 0 + * and >= numerator) Return 0 on success, otherwise -EERROR. + * * @init: Perform platform-specific initialization magic. * This is not not used by any of the basic clock types. * Please consider other ways of solving initialization problems @@ -218,6 +240,10 @@ struct clk_ops { unsigned long parent_accuracy); int (*get_phase)(struct clk_hw *hw); int (*set_phase)(struct clk_hw *hw, int degrees); + int (*get_duty_cycle)(struct clk_hw *hw, + struct clk_duty *duty); + int (*set_duty_cycle)(struct clk_hw *hw, + struct clk_duty *duty); void (*init)(struct clk_hw *hw); void (*debug_init)(struct clk_hw *hw, struct dentry *dentry); }; diff --git a/include/linux/clk.h b/include/linux/clk.h index 0dbd0885b2c2..4f750c481b82 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -141,6 +141,27 @@ int clk_set_phase(struct clk *clk, int degrees); */ int clk_get_phase(struct clk *clk); +/** + * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal + * @clk: clock signal source + * @num: numerator of the duty cycle ratio to be applied + * @den: denominator of the duty cycle ratio to be applied + * + * Adjust the duty cycle of a clock signal by the specified ratio. Returns 0 on + * success, -EERROR otherwise. + */ +int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den); + +/** + * clk_get_duty_cycle - return the duty cycle ratio of a clock signal + * @clk: clock signal source + * @scale: scaling factor to be applied to represent the ratio as an integer + * + * Returns the duty cycle ratio multiplied by the scale provided, otherwise + * returns -EERROR. + */ +int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale); + /** * clk_is_match - check if two clk's point to the same hardware clock * @p: clk compared against q @@ -183,6 +204,18 @@ static inline long clk_get_phase(struct clk *clk) return -ENOTSUPP; } +static inline int clk_set_duty_cycle(struct clk *clk, unsigned int num, + unsigned int den) +{ + return -ENOTSUPP; +} + +static inline unsigned int clk_get_scaled_duty_cycle(struct clk *clk, + unsigned int scale) +{ + return 0; +} + static inline bool clk_is_match(const struct clk *p, const struct clk *q) { return p == q; diff --git a/include/trace/events/clk.h b/include/trace/events/clk.h index 2cd449328aee..9004ffff7f32 100644 --- a/include/trace/events/clk.h +++ b/include/trace/events/clk.h @@ -192,6 +192,42 @@ DEFINE_EVENT(clk_phase, clk_set_phase_complete, TP_ARGS(core, phase) ); +DECLARE_EVENT_CLASS(clk_duty_cycle, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty), + + TP_STRUCT__entry( + __string( name, core->name ) + __field( unsigned int, num ) + __field( unsigned int, den ) + ), + + TP_fast_assign( + __assign_str(name, core->name); + __entry->num = duty->num; + __entry->den = duty->den; + ), + + TP_printk("%s %u/%u", __get_str(name), (unsigned int)__entry->num, + (unsigned int)__entry->den) +); + +DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty) +); + +DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle_complete, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty) +); + #endif /* _TRACE_CLK_H */ /* This part must be outside protection */ From 6c1549c4cc3c1b0d8623cde00e28f094b2db0d41 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 18 Jun 2018 21:07:51 +0900 Subject: [PATCH 061/529] ALSA: firewire-motu: suppless consumption for unused element of array in stack In MOTU firewire protocol, data block consists of 24 bit data chunks except for one quadlet for source packet header (SPH). The number of data chunk in a data block is different between three clock modes; low, middle and high. When unit supports ADAT on optical interface, the data block includes some chunks for ADAT channels. These ADAT chunks are unavailable at high mode. This driver has local functions to calculate the number of ADAT chunks. But They uses stack for three clock modes. This is useless for higher mode. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 525b746330be..a51fd196d884 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -176,7 +176,7 @@ static void calculate_differed_part(struct snd_motu_packet_format *formats, enum snd_motu_spec_flags flags, u32 data, u32 mask, u32 shift) { - unsigned char pcm_chunks[3] = {0, 0}; + unsigned char pcm_chunks[2] = {0, 0}; /* * When optical interfaces are configured for S/PDIF (TOSLINK), From 81720c6d49b7932d642e7dca736bef9a40c9b5f7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 18 Jun 2018 21:07:52 +0900 Subject: [PATCH 062/529] ALSA: firewire-motu: add a flag for chunks for main 1/2 out This driver explicitly assumes that all of supported models have main data chunk separated from chunk for analog ports. However, MOTU Traveler doesn't support the separated main data chunk. This commit adds a flag for the separated main data chunk. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 14 +++++++++----- sound/firewire/motu/motu-protocol-v3.c | 14 +++++++++----- sound/firewire/motu/motu.c | 3 +++ sound/firewire/motu/motu.h | 1 + 4 files changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index a51fd196d884..614f9b11e010 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -149,11 +149,15 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - /* - * Packets to v2 units transfer main-out-1/2 and phone-out-1/2. - */ - pcm_chunks[0] += 4; - pcm_chunks[1] += 4; + if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + // Packets to v2 units include 2 chunks for phone 1/2, except + // for 176.4/192.0 kHz. + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; } /* diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index c7cd9864dc4d..293353991591 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -188,11 +188,15 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - /* - * Packets to v2 units transfer main-out-1/2 and phone-out-1/2. - */ - pcm_chunks[0] += 4; - pcm_chunks[1] += 4; + if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + // Packets to v3 units include 2 chunks for phone 1/2, except + // for 176.4/192.0 kHz. + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; } /* diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 0d6b526105ab..445aa589582d 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -200,6 +200,7 @@ static const struct snd_motu_spec motu_828mk2 = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_2ND_Q, @@ -216,6 +217,7 @@ static const struct snd_motu_spec motu_828mk3 = { SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | SND_MOTU_SPEC_TX_REVERB_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_HAS_OPT_IFACE_B | SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -231,6 +233,7 @@ static const struct snd_motu_spec motu_audio_express = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_3RD_Q, .analog_in_ports = 2, diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 4b23cf337c4b..bced0407179e 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -86,6 +86,7 @@ enum snd_motu_spec_flags { SND_MOTU_SPEC_RX_MIDI_3RD_Q = 0x0200, SND_MOTU_SPEC_TX_MIDI_2ND_Q = 0x0400, SND_MOTU_SPEC_TX_MIDI_3RD_Q = 0x0800, + SND_MOTU_SPEC_RX_SEPARETED_MAIN = 0x1000, }; #define SND_MOTU_CLOCK_RATE_COUNT 6 From 06ac0b6f8f74e98d32f9dea5209bd26f3e7b50ba Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 18 Jun 2018 21:07:53 +0900 Subject: [PATCH 063/529] ALSA: firewire-motu: add a flag for AES/EBU on XLR interface MOTU Traveler supports AES/EBU on XLR interface and data block of rx/tx packet includes two chunk for the interface. This commit adds a flag for this purpose. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 5 +++++ sound/firewire/motu/motu-protocol-v3.c | 5 +++++ sound/firewire/motu/motu.h | 2 +- 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 614f9b11e010..f25b1ba118a2 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -160,6 +160,11 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } + if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + /* * All of v2 models have a pair of coaxial interfaces for digital in/out * port. At 44.1/48.0/88.2/96.0 kHz, packets includes PCM from these diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 293353991591..7cc80a05e91f 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -199,6 +199,11 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } + if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + /* * At least, packets have two data chunks for S/PDIF on coaxial * interface. diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index bced0407179e..2764bcaab327 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -79,7 +79,7 @@ enum snd_motu_spec_flags { SND_MOTU_SPEC_TX_MICINST_CHUNK = 0x0004, SND_MOTU_SPEC_TX_RETURN_CHUNK = 0x0008, SND_MOTU_SPEC_TX_REVERB_CHUNK = 0x0010, - SND_MOTU_SPEC_TX_AESEBU_CHUNK = 0x0020, + SND_MOTU_SPEC_HAS_AESEBU_IFACE = 0x0020, SND_MOTU_SPEC_HAS_OPT_IFACE_A = 0x0040, SND_MOTU_SPEC_HAS_OPT_IFACE_B = 0x0080, SND_MOTU_SPEC_RX_MIDI_2ND_Q = 0x0100, From 191ef57683aab1939d9b7afdc43f9213c21c5e1e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 18 Jun 2018 21:07:54 +0900 Subject: [PATCH 064/529] ALSA: firewire-motu: cancel chunk alignment for protocol version 2 For MOTU protocol version 2, this driver arranges the number of data chunks to align chunks to quadlet data channel. However, MOTU Traveler has padding bytes in the end of data block at high clock mode. This commit removes the arrangement. Fortunately, at low and middle clock mode, supported model for v2 protocol (828mkII) gets no influence from this change because all of combination for data chunks are just aligned to quadlet data channel. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index f25b1ba118a2..e5bd3ac02300 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -173,12 +173,9 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[0] += 2; pcm_chunks[1] += 2; - /* This part should be multiples of 4. */ - formats->fixed_part_pcm_chunks[0] = round_up(2 + pcm_chunks[0], 4) - 2; - formats->fixed_part_pcm_chunks[1] = round_up(2 + pcm_chunks[1], 4) - 2; - if (flags & SND_MOTU_SPEC_SUPPORT_CLOCK_X4) - formats->fixed_part_pcm_chunks[2] = - round_up(2 + pcm_chunks[2], 4) - 2; + formats->fixed_part_pcm_chunks[0] = pcm_chunks[0]; + formats->fixed_part_pcm_chunks[1] = pcm_chunks[1]; + formats->fixed_part_pcm_chunks[2] = pcm_chunks[2]; } static void calculate_differed_part(struct snd_motu_packet_format *formats, From 6c5e1ac0e144a8560cfa11bed8cdadab9491952f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 18 Jun 2018 21:07:55 +0900 Subject: [PATCH 065/529] ALSA: firewire-motu: add support for Motu Traveler This commit adds support for MOTU Traveler, launched in 2005, discontinued quite before. As a result, transmission of PCM frame and MIDI messages is available via ALSA PCM and RawMIDI/Sequencer interfaces. This model supports sampling transmission frequency up to 192.0 kHz, and AES/EBU on XLR interface and ADAT on optical interface. Unlike Motu 828MkII, Windows driver can switch fetching mode for DSP, like mute/unmute feature. Although this commit enables high sampling transmission frequency, actual sound from this model is not good. As long as I tested, it's silence at 176.4 kHz, and it includes hissing noise at 192.0 kHz. In my opinion, as I reported at 3526ce7f9ba7 ('ALSA: firewire-motu: add MOTU specific protocol layer'), timestamping on source packet header (SPH) may not still be good for this model as well. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04106505 bus_info_length 4, crc_length 16, crc 25861 404 31333934 bus_name "1394" 408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4) 40c 0001f200 company_id 0001f2 | 410 0001f32f device_id 000001f32f | EUI-64 0001f2000001f32f root directory ----------------------------------------------------------------- 414 0004c65c directory_length 4, crc 50780 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 00035955 directory_length 3, crc 22869 42c 120001f2 specifier id 430 13000009 version 434 17107800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 000206b2 leaf_length 2, crc 1714 43c 0001f200 company_id 0001f2 | 440 0001f32f device_id 000001f32f | EUI-64 0001f2000001f32f Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 34 ++++++++++++++++++++++++-- sound/firewire/motu/motu.c | 16 ++++++++++++ sound/firewire/motu/motu.h | 2 ++ 3 files changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index e5bd3ac02300..453fc29fade7 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -13,6 +13,8 @@ #define V2_CLOCK_RATE_SHIFT 3 #define V2_CLOCK_SRC_MASK 0x00000007 #define V2_CLOCK_SRC_SHIFT 0 +#define V2_CLOCK_TRAVELER_FETCH_DISABLE 0x04000000 +#define V2_CLOCK_TRAVELER_FETCH_ENABLE 0x03000000 #define V2_IN_OUT_CONF_OFFSET 0x0c04 #define V2_OPT_OUT_IFACE_MASK 0x00000c00 @@ -66,6 +68,11 @@ static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate) data &= ~V2_CLOCK_RATE_MASK; data |= i << V2_CLOCK_RATE_SHIFT; + if (motu->spec == &snd_motu_spec_traveler) { + data &= ~V2_CLOCK_TRAVELER_FETCH_ENABLE; + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + } + reg = cpu_to_be32(data); return snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, ®, sizeof(reg)); @@ -121,8 +128,31 @@ static int v2_get_clock_source(struct snd_motu *motu, static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) { - /* V2 protocol doesn't have this feature. */ - return 0; + __be32 reg; + u32 data; + int err = 0; + + if (motu->spec == &snd_motu_spec_traveler) { + err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, + ®, sizeof(reg)); + if (err < 0) + return err; + data = be32_to_cpu(reg); + + data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | + V2_CLOCK_TRAVELER_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; + else + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + + reg = cpu_to_be32(data); + err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, + ®, sizeof(reg)); + } + + return err; } static void calculate_fixed_part(struct snd_motu_packet_format *formats, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 445aa589582d..300d31b6f191 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -209,6 +209,21 @@ static const struct snd_motu_spec motu_828mk2 = { .analog_out_ports = 8, }; +const struct snd_motu_spec snd_motu_spec_traveler = { + .name = "Traveler", + .protocol = &snd_motu_protocol_v2, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_SUPPORT_CLOCK_X4 | + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_HAS_AESEBU_IFACE | + SND_MOTU_SPEC_HAS_OPT_IFACE_A | + SND_MOTU_SPEC_RX_MIDI_2ND_Q | + SND_MOTU_SPEC_TX_MIDI_2ND_Q, + + .analog_in_ports = 8, + .analog_out_ports = 8, +}; + static const struct snd_motu_spec motu_828mk3 = { .name = "828mk3", .protocol = &snd_motu_protocol_v3, @@ -253,6 +268,7 @@ static const struct snd_motu_spec motu_audio_express = { static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x101800, &motu_828mk2), + SND_MOTU_DEV_ENTRY(0x107800, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x106800, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x100800, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x104800, &motu_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 2764bcaab327..fd5327d30ab1 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -129,6 +129,8 @@ struct snd_motu_spec { extern const struct snd_motu_protocol snd_motu_protocol_v2; extern const struct snd_motu_protocol snd_motu_protocol_v3; +extern const struct snd_motu_spec snd_motu_spec_traveler; + int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, const struct snd_motu_protocol *const protocol); From 7f7cca08abf4c6c58ff1037cff5beec5a10a7da3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Jun 2018 11:56:21 +0100 Subject: [PATCH 066/529] ASoC: wm_adsp: Simplify handling of alg offset and length The current code that reads the algorithm list from the DSP is somewhat unclear, it converts directly from bytes to registers using a hard coded divide by 2. Most offsets are usually handled in DSP words within the driver and there is a function specifically for converting from words to register addresses. So update the handling to use these. This also removes the assumption that the registers are 16-bit word addressed, which will no longer be true on some of our newer parts. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 29 ++++++++++++++++++++--------- 1 file changed, 20 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2fcdd84021a5..07c17acc8a4f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1871,9 +1871,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, } static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, + const struct wm_adsp_region *mem, unsigned int pos, unsigned int len) { void *alg; + unsigned int reg; int ret; __be32 val; @@ -1888,7 +1890,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } /* Read the terminator first to validate the length */ - ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val)); + reg = wm_adsp_region_to_reg(mem, pos + len); + + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list end: %d\n", ret); @@ -1897,13 +1901,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, if (be32_to_cpu(val) != 0xbedead) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", - pos + len, be32_to_cpu(val)); + reg, be32_to_cpu(val)); + + /* Convert length from DSP words to bytes */ + len *= sizeof(u32); alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); - ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2); + reg = wm_adsp_region_to_reg(mem, pos); + + ret = regmap_raw_read(dsp->regmap, reg, alg, len); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list: %d\n", ret); kfree(alg); @@ -2002,10 +2011,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp1_id) / 2; - len = (sizeof(*adsp1_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp1_id) / sizeof(u32); + len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32); - adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp1_alg)) return PTR_ERR(adsp1_alg); @@ -2113,10 +2123,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp2_id) / 2; - len = (sizeof(*adsp2_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp2_id) / sizeof(u32); + len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32); - adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp2_alg)) return PTR_ERR(adsp2_alg); From 1b31de922e28de2bf2078b7a1e341ad4aee6aa03 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Jun 2018 11:56:20 +0100 Subject: [PATCH 067/529] ASoC: arizona: Set compressed IRQ to a wake source The current code is not setting the compressed IRQ as a wake source. Normally this doesn't cause any issues as the CODEC IRQ is set as a wake source by the jack detection code and the CODEC only produces a single IRQ line. However if the system is not using jack detection the compressed audio IRQ should still function as a wake source, as such directly set the compressed audio IRQ as a wake source. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 8 ++++++++ sound/soc/codecs/wm5102.c | 8 ++++++++ sound/soc/codecs/wm5110.c | 8 ++++++++ 3 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 196e9c343aeb..0da52ead91e0 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1283,6 +1283,12 @@ static int cs47l24_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -1306,6 +1312,7 @@ static int cs47l24_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return ret; @@ -1323,6 +1330,7 @@ static int cs47l24_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return 0; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1ac83388d1b8..a01a0c0e01eb 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2094,6 +2094,12 @@ static int wm5102_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2117,6 +2123,7 @@ static int wm5102_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return ret; @@ -2133,6 +2140,7 @@ static int wm5102_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fb9835dcd836..00c735c585d9 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2455,6 +2455,12 @@ static int wm5110_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2478,6 +2484,7 @@ static int wm5110_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return ret; @@ -2496,6 +2503,7 @@ static int wm5110_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return 0; From 4f722a6a736765310bc95dd3879f1545e5f64303 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 19 Jun 2018 14:00:37 -0500 Subject: [PATCH 068/529] ASoC: Intel: common: rename 'reef' to 'sof' in ACPI matching table Align with firmware tools, no functionality change Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-byt-match.c | 40 +++++++------- .../intel/common/soc-acpi-intel-cht-match.c | 52 +++++++++---------- .../common/soc-acpi-intel-hsw-bdw-match.c | 16 +++--- 3 files changed, 54 insertions(+), 54 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index bfe1ca68a542..4daa8a4f0c0c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5670.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -98,8 +98,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", .machine_quirk = byt_quirk, - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -107,8 +107,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -116,8 +116,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -125,8 +125,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5651.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -134,8 +134,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -143,8 +143,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ @@ -153,8 +153,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -162,8 +162,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* use CHT driver to Baytrail Chromebooks */ @@ -172,8 +172,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-max98090.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index ad1eb2d644be..3c3f2f8585d7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -68,8 +68,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -77,8 +77,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -86,8 +86,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -95,8 +95,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -104,8 +104,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -113,8 +113,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-max98090.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -131,8 +131,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -140,8 +140,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -149,8 +149,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-es8316.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-es8316.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ @@ -160,8 +160,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", .machine_quirk = cht_quirk, - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -169,8 +169,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5651, use Baytrail machine driver */ @@ -179,8 +179,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5651.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 53e7acdfbb81..494a0ea9b029 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST1.bin", - .sof_fw_filename = "intel/reef-hsw.ri", - .sof_tplg_filename = "intel/reef-hsw.tplg", + .sof_fw_filename = "intel/sof-hsw.ri", + .sof_tplg_filename = "intel/sof-hsw.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} @@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .id = "INT343A", .drv_name = "broadwell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt286.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5677.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5640.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5640.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} From f567b78851d49a4887b9bb1a8b3cfad37da515c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jun 2018 17:26:12 +0200 Subject: [PATCH 069/529] ALSA: hda - Move mic mute LED helper to the generic parser MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Move the code for setting up and controlling the mic mute LED hook from dell-wmi helper to the generic parser, so that it can be referred from the multiple driver codes. No functional change. Tested-by: Pali Rohár Signed-off-by: Takashi Iwai --- sound/pci/hda/dell_wmi_helper.c | 116 ++------------------------ sound/pci/hda/hda_generic.c | 140 ++++++++++++++++++++++++++++++++ sound/pci/hda/hda_generic.h | 16 ++++ 3 files changed, 162 insertions(+), 110 deletions(-) diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..8a7dbd1a7fbf 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -6,111 +6,18 @@ #if IS_ENABLED(CONFIG_DELL_LAPTOP) #include -enum { - MICMUTE_LED_ON, - MICMUTE_LED_OFF, - MICMUTE_LED_FOLLOW_CAPTURE, - MICMUTE_LED_FOLLOW_MUTE, -}; - -static int dell_led_mode = MICMUTE_LED_FOLLOW_MUTE; -static int dell_capture; -static int dell_led_value; static int (*dell_micmute_led_set_func)(int); -static void (*dell_old_cap_hook)(struct hda_codec *, - struct snd_kcontrol *, - struct snd_ctl_elem_value *); -static void call_micmute_led_update(void) +static void dell_micmute_update(struct hda_codec *codec) { - int val; + struct hda_gen_spec *spec = codec->spec; - switch (dell_led_mode) { - case MICMUTE_LED_ON: - val = 1; - break; - case MICMUTE_LED_OFF: - val = 0; - break; - case MICMUTE_LED_FOLLOW_CAPTURE: - val = dell_capture; - break; - case MICMUTE_LED_FOLLOW_MUTE: - default: - val = !dell_capture; - break; - } - - if (val == dell_led_value) - return; - dell_led_value = val; - dell_micmute_led_set_func(dell_led_value); + dell_micmute_led_set_func(spec->micmute_led.led_value); } -static void update_dell_wmi_micmute_led(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - if (dell_old_cap_hook) - dell_old_cap_hook(codec, kcontrol, ucontrol); - - if (!ucontrol || !dell_micmute_led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - dell_capture = (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - call_micmute_led_update(); - } -} - -static int dell_mic_mute_led_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "On", "Off", "Follow Capture", "Follow Mute", - }; - - return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); -} - -static int dell_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = dell_led_mode; - return 0; -} - -static int dell_mic_mute_led_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int mode; - - mode = ucontrol->value.enumerated.item[0]; - if (mode > MICMUTE_LED_FOLLOW_MUTE) - mode = MICMUTE_LED_FOLLOW_MUTE; - if (mode == dell_led_mode) - return 0; - dell_led_mode = mode; - call_micmute_led_update(); - return 1; -} - -static const struct snd_kcontrol_new dell_mic_mute_mode_ctls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Mute-LED Mode", - .info = dell_mic_mute_led_mode_info, - .get = dell_mic_mute_led_mode_get, - .put = dell_mic_mute_led_mode_put, - }, - {} -}; - static void alc_fixup_dell_wmi(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { @@ -121,25 +28,14 @@ static void alc_fixup_dell_wmi(struct hda_codec *codec, return; } - removefunc = true; - if (dell_micmute_led_set_func(false) >= 0) { - dell_led_value = 0; - if (spec->gen.num_adc_nids > 1 && !spec->gen.dyn_adc_switch) - codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); - else { - dell_old_cap_hook = spec->gen.cap_sync_hook; - spec->gen.cap_sync_hook = update_dell_wmi_micmute_led; - removefunc = false; - add_mixer(spec, dell_mic_mute_mode_ctls); - } - } - + removefunc = (dell_micmute_led_set_func(false) < 0) || + (snd_hda_gen_add_micmute_led(codec, + dell_micmute_update) <= 0); } if (dell_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { symbol_put(dell_micmute_led_set); dell_micmute_led_set_func = NULL; - dell_old_cap_hook = NULL; } } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index db773e219aaa..cdce9ce6b901 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3899,6 +3899,146 @@ static int parse_mic_boost(struct hda_codec *codec) return 0; } +/* + * mic mute LED hook helpers + */ +enum { + MICMUTE_LED_ON, + MICMUTE_LED_OFF, + MICMUTE_LED_FOLLOW_CAPTURE, + MICMUTE_LED_FOLLOW_MUTE, +}; + +static void call_micmute_led_update(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + unsigned int val; + + switch (spec->micmute_led.led_mode) { + case MICMUTE_LED_ON: + val = 1; + break; + case MICMUTE_LED_OFF: + val = 0; + break; + case MICMUTE_LED_FOLLOW_CAPTURE: + val = spec->micmute_led.capture; + break; + case MICMUTE_LED_FOLLOW_MUTE: + default: + val = !spec->micmute_led.capture; + break; + } + + if (val == spec->micmute_led.led_value) + return; + spec->micmute_led.led_value = val; + if (spec->micmute_led.update) + spec->micmute_led.update(codec); +} + +static void update_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_gen_spec *spec = codec->spec; + + if (spec->micmute_led.old_hook) + spec->micmute_led.old_hook(codec, kcontrol, ucontrol); + + if (!ucontrol) + return; + if (!strcmp("Capture Switch", ucontrol->id.name) && + !ucontrol->id.index) { + /* TODO: How do I verify if it's a mono or stereo here? */ + spec->micmute_led.capture = (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]); + call_micmute_led_update(codec); + } +} + +static int micmute_led_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "On", "Off", "Follow Capture", "Follow Mute", + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int micmute_led_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->micmute_led.led_mode; + return 0; +} + +static int micmute_led_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + unsigned int mode; + + mode = ucontrol->value.enumerated.item[0]; + if (mode > MICMUTE_LED_FOLLOW_MUTE) + mode = MICMUTE_LED_FOLLOW_MUTE; + if (mode == spec->micmute_led.led_mode) + return 0; + spec->micmute_led.led_mode = mode; + call_micmute_led_update(codec); + return 1; +} + +static const struct snd_kcontrol_new micmute_led_mode_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Mute-LED Mode", + .info = micmute_led_mode_info, + .get = micmute_led_mode_get, + .put = micmute_led_mode_put, +}; + +/** + * snd_hda_gen_add_micmute_led - helper for setting up mic mute LED hook + * @codec: the HDA codec + * @hook: the callback for updating LED + * + * Called from the codec drivers for offering the mic mute LED controls. + * Only valid for a single ADC (or a single input). When established, it + * sets up cap_sync_hook and triggers the callback at each time when the + * capture mixer switch changes. The callback is supposed to update the LED + * accordingly. + * + * Returns 1 if the hook is established, 0 if skipped (no valid config), or + * a negative error code. + */ +int snd_hda_gen_add_micmute_led(struct hda_codec *codec, + void (*hook)(struct hda_codec *)) +{ + struct hda_gen_spec *spec = codec->spec; + + if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) { + codec_dbg(codec, + "Skipping micmute LED control due to several ADCs"); + return 0; + } + + spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE; + spec->micmute_led.capture = 0; + spec->micmute_led.led_value = 0; + spec->micmute_led.old_hook = spec->cap_sync_hook; + spec->micmute_led.update = hook; + spec->cap_sync_hook = update_micmute_led; + if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) + return -ENOMEM; + return 1; +} +EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); + /* * parse digital I/Os and set up NIDs in BIOS auto-parse mode */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 61772317de46..10123664fa61 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -86,6 +86,16 @@ struct badness_table { extern const struct badness_table hda_main_out_badness; extern const struct badness_table hda_extra_out_badness; +struct hda_micmute_hook { + unsigned int led_mode; + unsigned int capture; + unsigned int led_value; + void (*update)(struct hda_codec *codec); + void (*old_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +}; + struct hda_gen_spec { char stream_name_analog[32]; /* analog PCM stream */ const struct hda_pcm_stream *stream_analog_playback; @@ -276,6 +286,9 @@ struct hda_gen_spec { struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + /* mic mute LED hook; called via cap_sync_hook */ + struct hda_micmute_hook micmute_led; + /* PCM hooks */ void (*pcm_playback_hook)(struct hda_pcm_stream *hinfo, struct hda_codec *codec, @@ -342,4 +355,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +int snd_hda_gen_add_micmute_led(struct hda_codec *codec, + void (*hook)(struct hda_codec *)); + #endif /* __SOUND_HDA_GENERIC_H */ From 69b2c6d7c0204e68185b694f9cfcb8f2636fed5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jun 2018 17:28:45 +0200 Subject: [PATCH 070/529] ALSA: hda - Use the common helper for thinkpad_acpi mic mute LED handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use the new common helper for setting up and controlling the mic mute LED over thinkpad_acpi. This also provides a new mixer enum "Mic Mute-LED Mode" (that was present only for Dell models), which allows user to choose the mic mute LED behavior. For example, if you want the mic mute LED turned on only while mic is on, choose "Follow Capture" there. Tested-by: Pali Rohár Signed-off-by: Takashi Iwai --- sound/pci/hda/thinkpad_helper.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 65bb3ac6af4c..97f49b751e6e 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -27,17 +27,11 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) led_set_func(TPACPI_LED_MUTE, !enabled); } -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void update_tpacpi_micmute(struct hda_codec *codec) { - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } + struct hda_gen_spec *spec = codec->spec; + + led_set_func(TPACPI_LED_MICMUTE, spec->micmute_led.led_value); } static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, @@ -63,15 +57,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - else { - spec->cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && + snd_hda_gen_add_micmute_led(codec, + update_tpacpi_micmute) > 0) + removefunc = false; } if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { From d03abecab5b4f4fa533f5971a4b205d05e9a4bc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 12:29:13 +0200 Subject: [PATCH 071/529] ALSA: hda/realtek - Use the mic-mute LED helper for HP and others Similar as the previous commit, convert to use the common helper for controlling the mic mute LED for HP and other machines in the Realtek codec driver, too. This will give the mic mute LED enum as gratis. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 49 +++++++++++++---------------------- 1 file changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 487ceb9fd038..32a7a72033ae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3718,16 +3718,12 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) } /* turn on/off mic-mute LED via GPIO per capture hook */ -static void alc_fixup_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void alc_gpio_micmute_update(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (ucontrol) - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); + alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->gen.micmute_led.led_value); } static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, @@ -3742,12 +3738,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->gpio_mic_led_mask = 0x10; snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); } } @@ -3763,39 +3759,30 @@ static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x02; spec->gpio_mic_led_mask = 0x20; snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); } } /* turn on/off mic-mute LED per capture hook */ -static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void alc_cap_micmute_update(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int pinval, enable, disable; + unsigned int pinval; + if (!spec->cap_mute_led_nid) + return; pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid); pinval &= ~AC_PINCTL_VREFEN; - enable = pinval | AC_PINCTL_VREF_80; - disable = pinval | AC_PINCTL_VREF_HIZ; - - if (!ucontrol) - return; - - if (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]) - pinval = disable; + if (spec->gen.micmute_led.led_value) + pinval |= AC_PINCTL_VREF_80; else - pinval = enable; - - if (spec->cap_mute_led_nid) - snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval); + pinval |= AC_PINCTL_VREF_HIZ; + snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval); } static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, @@ -3810,12 +3797,12 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } } @@ -3833,12 +3820,12 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } } @@ -3915,11 +3902,11 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, gpio2_mic_hotkey_event); spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->gpio_mic_led_mask = 0x10; + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); return; } @@ -3957,10 +3944,10 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, snd_hda_jack_detect_enable_callback(codec, 0x1b, gpio2_mic_hotkey_event); - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mic_led_mask = 0x04; + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); return; } @@ -3988,10 +3975,10 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; spec->mute_led_polarity = 0; spec->mute_led_nid = 0x1a; spec->cap_mute_led_nid = 0x18; + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); spec->gen.vmaster_mute_enum = 1; codec->power_filter = led_power_filter; } From 184e302b46c94bfb1d53fe3d4c925a45a6990430 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 12:36:36 +0200 Subject: [PATCH 072/529] ALSA: hda/conexant - Use the mic-mute LED helper Convert to use the common helper for controlling the mic mute LED for HP laptops, just as we've done for Realtek codecs. This will give the mic mute LED enum as gratis. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e7fcfc3b8885..a9fd0572d526 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -695,16 +695,12 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) } /* turn on/off mic-mute LED via GPIO per capture hook */ -static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void cxt_gpio_micmute_update(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - if (ucontrol) - cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, - ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); + cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->gen.micmute_led.led_value); } @@ -721,11 +717,11 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x01; spec->gpio_mic_led_mask = 0x02; + snd_hda_gen_add_micmute_led(codec, cxt_gpio_micmute_update); } snd_hda_add_verbs(codec, gpio_init); if (spec->gpio_led) From c647f806b8c227de05f7f91b0ba8450b58cb3dfe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 12:42:03 +0200 Subject: [PATCH 073/529] ALSA: hda - Allow multiple ADCs for mic mute LED controls Instead of refusing, allow the configuration with the multiple ADCs (thus multiple capture switches) for enabling the mic mute LED. This has been done for Sigmatel/IDT codecs, and we treat the OR-ed values from all capture switches as the boolean condition. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 32 ++++++++++++++------------------ 1 file changed, 14 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index cdce9ce6b901..942f96e184b6 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3922,7 +3922,7 @@ static void call_micmute_led_update(struct hda_codec *codec) val = 0; break; case MICMUTE_LED_FOLLOW_CAPTURE: - val = spec->micmute_led.capture; + val = !!spec->micmute_led.capture; break; case MICMUTE_LED_FOLLOW_MUTE: default: @@ -3942,17 +3942,21 @@ static void update_micmute_led(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol) { struct hda_gen_spec *spec = codec->spec; + unsigned int mask; if (spec->micmute_led.old_hook) spec->micmute_led.old_hook(codec, kcontrol, ucontrol); if (!ucontrol) return; - if (!strcmp("Capture Switch", ucontrol->id.name) && - !ucontrol->id.index) { + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + if (!strcmp("Capture Switch", ucontrol->id.name)) { /* TODO: How do I verify if it's a mono or stereo here? */ - spec->micmute_led.capture = (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->micmute_led.capture |= mask; + else + spec->micmute_led.capture &= ~mask; call_micmute_led_update(codec); } } @@ -4008,25 +4012,17 @@ static const struct snd_kcontrol_new micmute_led_mode_ctl = { * @hook: the callback for updating LED * * Called from the codec drivers for offering the mic mute LED controls. - * Only valid for a single ADC (or a single input). When established, it - * sets up cap_sync_hook and triggers the callback at each time when the - * capture mixer switch changes. The callback is supposed to update the LED - * accordingly. + * When established, it sets up cap_sync_hook and triggers the callback at + * each time when the capture mixer switch changes. The callback is supposed + * to update the LED accordingly. * - * Returns 1 if the hook is established, 0 if skipped (no valid config), or - * a negative error code. + * Returns 0 if the hook is established or a negative error code. */ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)) { struct hda_gen_spec *spec = codec->spec; - if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) { - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - return 0; - } - spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE; spec->micmute_led.capture = 0; spec->micmute_led.led_value = 0; @@ -4035,7 +4031,7 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, spec->cap_sync_hook = update_micmute_led; if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) return -ENOMEM; - return 1; + return 0; } EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); From 3bf29db731ce22480de748464031b4447b248c0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 12:44:35 +0200 Subject: [PATCH 074/529] ALSA: hda/sigmatel - Use common helper for mic mute LED To simplify the code and to get the mic-mute LED behavior control, use the new helper function for controlling the mic mute LED instead of open-codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 31 ++++++------------------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 63d15b545b33..046705b4691a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -332,33 +332,15 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, } /* hook for controlling mic-mute LED GPIO */ -static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_capture_led_update(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int mask; - bool cur_mute, prev_mute; - if (!kcontrol || !ucontrol) - return; - - mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - prev_mute = !spec->mic_enabled; - if (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]) - spec->mic_enabled |= mask; + if (spec->gen.micmute_led.led_value) + spec->gpio_data |= spec->mic_mute_led_gpio; else - spec->mic_enabled &= ~mask; - cur_mute = !spec->mic_enabled; - if (cur_mute != prev_mute) { - if (cur_mute) - spec->gpio_data |= spec->mic_mute_led_gpio; - else - spec->gpio_data &= ~spec->mic_mute_led_gpio; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } + spec->gpio_data &= ~spec->mic_mute_led_gpio; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } static int stac_vrefout_set(struct hda_codec *codec, @@ -4656,8 +4638,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->mic_mute_led_gpio; spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; - - spec->gen.cap_sync_hook = stac_capture_led_hook; + snd_hda_gen_add_micmute_led(codec, stac_capture_led_update); } } From 0bed2aa3ac5cbbd0b89bf5e94f165e2ef18180ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 12:45:53 +0200 Subject: [PATCH 075/529] ALSA: hda - Sanity check of unexpected cap_sync_hook override There are a couple of places setting cap_sync_hook in the codec drivers, and they just overwrite the value. Add a sanity check via WARN_ON() in case if an old non-NULL value is overridden and forgotten. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a9fd0572d526..b7339cb5c45b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -343,6 +343,7 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, snd_hdac_regmap_add_vendor_verb(&codec->core, 0x410); break; case HDA_FIXUP_ACT_PROBE: + WARN_ON(spec->gen.cap_sync_hook); spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; spec->gen.automute_hook = cxt_update_headset_mode; break; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 32a7a72033ae..d9461eebcfdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4828,6 +4828,7 @@ static void alc_probe_headset_mode(struct hda_codec *codec) spec->headphone_mic_pin = cfg->inputs[i].pin; } + WARN_ON(spec->gen.cap_sync_hook); spec->gen.cap_sync_hook = alc_update_headset_mode_hook; spec->gen.automute_hook = alc_update_headset_mode; spec->gen.hp_automute_hook = alc_update_headset_jack_cb; From 1bce62a6e0dd85c15cbab36a0f7b9d4766cef18e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 23:15:59 +0200 Subject: [PATCH 076/529] ALSA: hda/realtek - Simplify alc269_fixup_hp_line1_mic1_led() alc269_fixup_hp_line1_mic1_led() can be simplified more with the existing helper code. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d9461eebcfdf..7934c5df4d80 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3973,14 +3973,10 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1a); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x1a; spec->cap_mute_led_nid = 0x18; snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); - spec->gen.vmaster_mute_enum = 1; - codec->power_filter = led_power_filter; } } From bb450fa59c0772310ebcc704722dd8a0313bf8ed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 21 Jun 2018 15:47:35 -0500 Subject: [PATCH 077/529] ASoC: Intel: common: fix missing rename from 'reef' to 'sof' Somehow I missed the Nau8824 support which was added in 4.17. Oops Fixes: 4f722a6a736 ("ASoC: Intel: common: rename 'reef' to 'sof' in ACPI matching table") Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 3c3f2f8585d7..91bb99b69601 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -122,8 +122,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-nau8824", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-nau8824.tplg", .asoc_plat_name = "sst-mfld-platform", }, { From a98ec93d7e378db21a6cdbc6d55feaba9fb4b999 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 22 Jun 2018 02:23:24 +0000 Subject: [PATCH 078/529] ASoC: rt5682: use devm_snd_soc_register_component() Using devm_snd_soc_register_component() and drop all of the code related to .remove hook. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 61a97301bcfa..baad177908ab 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2630,17 +2630,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } -static int rt5682_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2671,7 +2665,6 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, - .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; From 2854a214f398f2c3315204a05efff11739fec062 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 22 Jun 2018 02:23:34 +0000 Subject: [PATCH 079/529] ASoC: rt1305: use devm_snd_soc_register_component() Using devm_snd_soc_register_component() and drop all of the code related to .remove hook. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/rt1305.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 421b8fb2fa04..c4452efc7970 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c, rt1305_reset(rt1305->regmap); rt1305_calibrate(rt1305); - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1305, rt1305_dai, ARRAY_SIZE(rt1305_dai)); } -static int rt1305_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt1305_i2c_shutdown(struct i2c_client *client) { struct rt1305_priv *rt1305 = i2c_get_clientdata(client); @@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = { #endif }, .probe = rt1305_i2c_probe, - .remove = rt1305_i2c_remove, .shutdown = rt1305_i2c_shutdown, .id_table = rt1305_i2c_id, }; From 366f074d047b2538cc5d7e4a820bb6c117a3ccec Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Thu, 21 Jun 2018 11:56:28 +0900 Subject: [PATCH 080/529] ASoC: uniphier: remove redundant check of PLL ID This patch removes redudant check of PLL ID. struct uniphier_aio_pll enable member has already been checked at is_valid_pll(). Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-cpu.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 2d9b7dde2ffa..ee90e6c3937c 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_out) { struct uniphier_aio *aio = uniphier_priv(dai); - struct device *dev = &aio->chip->pdev->dev; int ret; if (!is_valid_pll(aio->chip, pll_id)) return -EINVAL; - if (!aio->chip->plls[pll_id].enable) { - dev_err(dev, "PLL(%d) is not implemented\n", pll_id); - return -ENOTSUPP; - } ret = aio_chip_set_pll(aio->chip, pll_id, freq_out); if (ret < 0) From 3bec6fa3cdd338860d4b7d4110a49292646d801e Mon Sep 17 00:00:00 2001 From: "Agrawal, Akshu" Date: Thu, 21 Jun 2018 12:58:16 +0800 Subject: [PATCH 081/529] ASoC: AMD: Change codec to channel link as per hardware redesign This is a correction to match acutal hardware configuration. The hardware configuration looks like: I2S_BT -> SPK(Max) + DMIC(Adau) I2S_SP -> DA7219 Headset No actual products have been shipped with previous configuration. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c..566bd268be3a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -148,7 +148,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->i2s_instance = I2S_SP_INSTANCE; return da7219_clk_enable(substream); } @@ -163,7 +163,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -178,7 +178,7 @@ static int cz_dmic_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -204,16 +204,27 @@ static const struct snd_soc_ops cz_dmic_cap_ops = { static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play-cap", - .stream_name = "Playback and Capture", + .name = "amd-da7219-play", + .stream_name = "Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.3.auto", + .cpu_dai_name = "designware-i2s.1.auto", .codec_dai_name = "da7219-hifi", .codec_name = "i2c-DLGS7219:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .ops = &cz_da7219_cap_ops, + }, + { + .name = "amd-da7219-cap", + .stream_name = "Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "da7219-hifi", + .codec_name = "i2c-DLGS7219:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, @@ -221,7 +232,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.1.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "HiFi", .codec_name = "MX98357A:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -233,7 +244,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic", .stream_name = "DMIC Capture", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.2.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "adau7002-hifi", .codec_name = "ADAU7002:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF From 2718c89a233bf8549fdba0925947b2c3cb887a95 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Thu, 21 Jun 2018 12:58:17 +0800 Subject: [PATCH 082/529] ASoC: AMD: Configure channel 1 or channel 0 for capture ST/CZ SoC have 2 channels for capture in the I2SSP path. The DMA though these channels is done using the same dma descriptors. We configure the channel and enable it on the basis of channel selected by machine driver. Machine driver knows which codec sits on which channel and thus sends the information to dma driver. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 71 +++++++++++++++++++++++++++- sound/soc/amd/acp.h | 4 ++ 3 files changed, 111 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 566bd268be3a..f42606e5879e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -149,6 +149,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -172,7 +173,7 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } -static int cz_dmic_startup(struct snd_pcm_substream *substream) +static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -182,6 +183,17 @@ static int cz_dmic_startup(struct snd_pcm_substream *substream) return da7219_clk_enable(substream); } +static int cz_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return da7219_clk_enable(substream); +} + static void cz_dmic_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); @@ -197,8 +209,13 @@ static const struct snd_soc_ops cz_max_play_ops = { .shutdown = cz_max_shutdown, }; -static const struct snd_soc_ops cz_dmic_cap_ops = { - .startup = cz_dmic_startup, +static const struct snd_soc_ops cz_dmic0_cap_ops = { + .startup = cz_dmic0_startup, + .shutdown = cz_dmic_shutdown, +}; + +static const struct snd_soc_ops cz_dmic1_cap_ops = { + .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, }; @@ -241,8 +258,9 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .ops = &cz_max_play_ops, }, { - .name = "dmic", - .stream_name = "DMIC Capture", + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "adau7002-hifi", @@ -250,7 +268,20 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, - .ops = &cz_dmic_cap_ops, + .ops = &cz_dmic0_cap_ops, + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "adau7002-hifi", + .codec_name = "ADAU7002:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_dmic1_cap_ops, }, }; diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 1458b5048498..3c3d398d0d0b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -336,6 +336,61 @@ static void config_acp_dma(void __iomem *acp_mmio, rtd->dma_dscr_idx_2, asic_type); } +static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg, res_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + ch_reg = mmACP_I2SMICSP_RER1; + res_reg = mmACP_I2SMICSP_RCR1; + imr_reg = mmACP_I2SMICSP_IMR1; + break; + case CAP_CHANNEL0: + default: + ch_reg = mmACP_I2SMICSP_RER0; + res_reg = mmACP_I2SMICSP_RCR0; + imr_reg = mmACP_I2SMICSP_IMR0; + break; + } + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, ch_reg); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, res_reg); + } + val = acp_reg_read(acp_mmio, imr_reg); + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x1, acp_mmio, ch_reg); +} + +static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + imr_reg = mmACP_I2SMICSP_IMR1; + ch_reg = mmACP_I2SMICSP_RER1; + break; + case CAP_CHANNEL0: + default: + imr_reg = mmACP_I2SMICSP_IMR0; + ch_reg = mmACP_I2SMICSP_RER0; + break; + } + val = acp_reg_read(acp_mmio, imr_reg); + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x0, acp_mmio, ch_reg); +} + /* Start a given DMA channel transfer */ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) { @@ -773,8 +828,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - if (pinfo) + if (pinfo) { rtd->i2s_instance = pinfo->i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); @@ -990,6 +1047,18 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_start(rtd->acp_mmio, rtd->ch1); acp_dma_start(rtd->acp_mmio, rtd->ch2); } else { + if (rtd->capture_channel == CAP_CHANNEL0) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL1); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL0); + } + if (rtd->capture_channel == CAP_CHANNEL1) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL0); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL1); + } acp_dma_start(rtd->acp_mmio, rtd->ch2); acp_dma_start(rtd->acp_mmio, rtd->ch1); } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96c84d4..3190fdce6307 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,8 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define CAP_CHANNEL0 0x00 +#define CAP_CHANNEL1 0x01 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -125,6 +127,7 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 i2s_instance; + u16 capture_channel; u16 direction; u16 ch1; u16 ch2; @@ -155,6 +158,7 @@ struct audio_drv_data { */ struct acp_platform_info { u16 i2s_instance; + u16 capture_channel; }; union acp_dma_count { From a12f671b4241f53e7cd9dec8770d51549682453b Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Wed, 20 Jun 2018 10:59:59 +0900 Subject: [PATCH 083/529] ASoC: add hp-det-gpio and mic-det-gpio to audio graph card binding Add headphone and microphone detection GPIO support to audio graph card same as supported in simple card. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/audio-graph-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index d04ea3b1a1dd..7e63e53a901c 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -18,6 +18,8 @@ Below are same as Simple-Card. - bitclock-inversion - frame-inversion - mclk-fs +- hp-det-gpio +- mic-det-gpio - dai-tdm-slot-num - dai-tdm-slot-width - clocks / system-clock-frequency From f4c277b817cc9489fffabffb4e15d2f3b686056c Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Wed, 20 Jun 2018 18:25:20 +0900 Subject: [PATCH 084/529] ASoC: soc-pcm: DPCM cares BE channel constraint Current DPCM is caring only FE channel configuration. Sometimes it will be trouble if user selects channel which isn't supported by BE. This patch adds new .dpcm_merged_chan on struct snd_soc_dai_link. DPCM will use FE / BE merged channel if struct snd_soc_dai_link has it. Signed-off-by: Jiada Wang Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 46 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1378dcd2128a..f7579f82cc00 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -957,6 +957,8 @@ struct snd_soc_dai_link { /* DPCM used FE & BE merged format */ unsigned int dpcm_merged_format:1; + /* DPCM used FE & BE merged channel */ + unsigned int dpcm_merged_chan:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 45b52f7b9690..19ebfc958b9d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1715,6 +1715,46 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) return formats; } +static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_chan) + return; + + *channels_min = 0; + *channels_max = UINT_MAX; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + int i; + + for (i = 0; i < be->num_codecs; i++) { + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *channels_min = max(*channels_min, + codec_stream->channels_min); + *channels_max = min(*channels_max, + codec_stream->channels_max); + } + } +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1722,11 +1762,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; u64 format = dpcm_runtime_base_format(substream); + unsigned int channels_min = 0, channels_max = UINT_MAX; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); else dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + + dpcm_runtime_base_chan(substream, &channels_min, &channels_max); + + runtime->hw.channels_min = max(channels_min, runtime->hw.channels_min); + runtime->hw.channels_max = min(channels_max, runtime->hw.channels_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); From cc51574ad2632dde790241129617bb97f46f7c85 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Fri, 22 Jun 2018 21:28:33 +0200 Subject: [PATCH 085/529] ALSA: ac97: add bus binding for codecs Add the generic ac97 bus binding, especially for ac97 codecs discovered by ac97 hardware probing. Signed-off-by: Robert Jarzmik Signed-off-by: Takashi Iwai --- .../devicetree/bindings/sound/ac97-bus.txt | 32 +++++++++++++++++++ 1 file changed, 32 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ac97-bus.txt diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt new file mode 100644 index 000000000000..103c428f2595 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt @@ -0,0 +1,32 @@ +Generic AC97 Device Properties + +This documents describes the devicetree bindings for an ac97 controller child +node describing ac97 codecs. + +Required properties: +-compatible : Must be "ac97,vendor_id1,vendor_id2 + The ids shall be the 4 characters hexadecimal encoding, such as + given by "%04x" formatting of printf +-reg : Must be the ac97 codec number, between 0 and 3 + +Example: +ac97: sound@40500000 { + compatible = "marvell,pxa270-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pinctrl_ac97_default >; + clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; + clock-names = "AC97CLK", "AC97CONFCLK"; + + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + reg = <0>; + compatible = "ac97,574d,4c13"; + clocks = <&fixed_wm9713_clock>; + clock-names = "ac97_clk"; + } +}; From 2225a3e6af78e605119e0f178ae957d860882b1d Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Fri, 22 Jun 2018 21:28:34 +0200 Subject: [PATCH 086/529] ALSA: ac97: add codecs devicetree binding Add a devicetree binding for codecs. This is especially useful if the AC97 bitclk clock is provided by the codec, as it has to be described in the devicetree description for the ac97 bus code to aquire it. Signed-off-by: Robert Jarzmik Signed-off-by: Takashi Iwai --- sound/ac97/bus.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 31f858eceffc..7a0dfca03a57 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -68,6 +69,27 @@ ac97_codec_find(struct ac97_controller *ac97_ctrl, unsigned int codec_num) return ac97_ctrl->codecs[codec_num]; } +static struct device_node * +ac97_of_get_child_device(struct ac97_controller *ac97_ctrl, int idx, + unsigned int vendor_id) +{ + struct device_node *node; + u32 reg; + char compat[] = "ac97,0000,0000"; + + snprintf(compat, sizeof(compat), "ac97,%04x,%04x", + vendor_id >> 16, vendor_id & 0xffff); + + for_each_child_of_node(ac97_ctrl->parent->of_node, node) { + if ((idx != of_property_read_u32(node, "reg", ®)) || + !of_device_is_compatible(node, compat)) + continue; + return of_node_get(node); + } + + return NULL; +} + static void ac97_codec_release(struct device *dev) { struct ac97_codec_device *adev; @@ -76,6 +98,7 @@ static void ac97_codec_release(struct device *dev) adev = to_ac97_device(dev); ac97_ctrl = adev->ac97_ctrl; ac97_ctrl->codecs[adev->num] = NULL; + of_node_put(dev->of_node); kfree(adev); } @@ -98,6 +121,8 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, device_initialize(&codec->dev); dev_set_name(&codec->dev, "%s:%u", dev_name(ac97_ctrl->parent), idx); + codec->dev.of_node = ac97_of_get_child_device(ac97_ctrl, idx, + vendor_id); ret = device_add(&codec->dev); if (ret) @@ -105,6 +130,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, return 0; err_free_codec: + of_node_put(codec->dev.of_node); put_device(&codec->dev); kfree(codec); ac97_ctrl->codecs[idx] = NULL; From 1c76aa5fb48d8357f38fb2f1d2cef742a617d695 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Jun 2018 16:37:54 +0200 Subject: [PATCH 087/529] ALSA: hda/realtek - Allow skipping spec->init_amp detection Some devices have the overrides of spec->init_amp at HDA_FIXUP_ACT_PROBE just because alc_ssid_check() gives the false-positive values from the SSID. For more consistent behavior, define the logic in the following way: - Define ALC_INIT_UNDEFINED as the default value before calling alc_ssid_check() - Each fixup may set up spec->init_amp with another value at HDA_FIXUP_ACT_PRE_PROBE - At detection, check whether spec->init_amp is ALC_INIT_UNDEFINED or not; if it's different, we skip the detection Also, it turned out that ASUS TX300 requires the spec->init_amp override, too; currently it ignores the GPIO bits implicitly by its static init verb, but this will be changed in the later patchset. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 44 +++++++++++++++++------------------ 1 file changed, 21 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7934c5df4d80..1f054e5ae2b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -43,6 +43,7 @@ /* extra amp-initialization sequence types */ enum { + ALC_INIT_UNDEFINED, ALC_INIT_NONE, ALC_INIT_DEFAULT, ALC_INIT_GPIO1, @@ -656,20 +657,22 @@ do_sku: * 7~6 : Reserved */ tmp = (ass & 0x38) >> 3; /* external Amp control */ - switch (tmp) { - case 1: - spec->init_amp = ALC_INIT_GPIO1; - break; - case 3: - spec->init_amp = ALC_INIT_GPIO2; - break; - case 7: - spec->init_amp = ALC_INIT_GPIO3; - break; - case 5: - default: - spec->init_amp = ALC_INIT_DEFAULT; - break; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + switch (tmp) { + case 1: + spec->init_amp = ALC_INIT_GPIO1; + break; + case 3: + spec->init_amp = ALC_INIT_GPIO2; + break; + case 7: + spec->init_amp = ALC_INIT_GPIO3; + break; + case 5: + default: + spec->init_amp = ALC_INIT_DEFAULT; + break; + } } /* is laptop or Desktop and enable the function "Mute internal speaker @@ -1589,8 +1592,6 @@ static void alc260_fixup_kn1(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_apply_pincfgs(codec, pincfgs); - break; - case HDA_FIXUP_ACT_PROBE: spec->init_amp = ALC_INIT_NONE; break; } @@ -1600,7 +1601,7 @@ static void alc260_fixup_fsc_s7020(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) spec->init_amp = ALC_INIT_NONE; } @@ -3892,6 +3893,7 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; @@ -3914,9 +3916,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, return; switch (action) { - case HDA_FIXUP_ACT_PROBE: - spec->init_amp = ALC_INIT_DEFAULT; - break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; @@ -3937,6 +3936,7 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; @@ -3955,9 +3955,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, return; switch (action) { - case HDA_FIXUP_ACT_PROBE: - spec->init_amp = ALC_INIT_DEFAULT; - break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; @@ -5227,6 +5224,7 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; snd_hda_add_verbs(codec, gpio2_verbs); snd_hda_apply_pincfgs(codec, dock_pins); spec->gen.auto_mute_via_amp = 1; From 5579cd6f6629bc1e09b3b2d13ab7a1ed371c5ac2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 22:22:41 +0200 Subject: [PATCH 088/529] ALSA: hda/realtek - Manage GPIO bits commonly Currently the GPIO bits are managed by individual verbs in some cases while toggled dynamically in other cases. For simplifying the GPIO management, define the GPIO mask, dir and data bits in alc_spec fields, and refer to / set them consistently from all places. As a first step, along with the definition of the new gpio_* fields, this patch replaces the static verbs that are used at initialization and fixups with the common helper functions. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 129 +++++++++++++++++++++------------- 1 file changed, 81 insertions(+), 48 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f054e5ae2b3..489075b23652 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -46,9 +46,6 @@ enum { ALC_INIT_UNDEFINED, ALC_INIT_NONE, ALC_INIT_DEFAULT, - ALC_INIT_GPIO1, - ALC_INIT_GPIO2, - ALC_INIT_GPIO3, }; enum { @@ -93,6 +90,11 @@ struct alc_spec { struct alc_customize_define cdefine; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + /* GPIO bits */ + unsigned int gpio_mask; + unsigned int gpio_dir; + unsigned int gpio_data; + /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; hda_nid_t mute_led_nid; @@ -220,27 +222,63 @@ static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) /* * GPIO setup tables, used in initialization */ + /* Enable GPIO mask and set output */ -static const struct hda_verb alc_gpio1_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - { } -}; +static void alc_setup_gpio(struct hda_codec *codec, unsigned int mask) +{ + struct alc_spec *spec = codec->spec; -static const struct hda_verb alc_gpio2_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, - { } -}; + spec->gpio_mask |= mask; + spec->gpio_dir |= mask; + spec->gpio_data |= mask; +} -static const struct hda_verb alc_gpio3_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, - { } -}; +static void alc_write_gpio_data(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); +} + +static void alc_write_gpio(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->gpio_mask) + return; + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_MASK, spec->gpio_mask); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DIRECTION, spec->gpio_dir); + alc_write_gpio_data(codec); +} + +static void alc_fixup_gpio(struct hda_codec *codec, int action, + unsigned int mask) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + alc_setup_gpio(codec, mask); +} + +static void alc_fixup_gpio1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x01); +} + +static void alc_fixup_gpio2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x02); +} + +static void alc_fixup_gpio3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x03); +} /* * Fix hardware PLL issue @@ -448,16 +486,8 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) { alc_fill_eapd_coef(codec); alc_auto_setup_eapd(codec, true); + alc_write_gpio(codec); switch (type) { - case ALC_INIT_GPIO1: - snd_hda_sequence_write(codec, alc_gpio1_init_verbs); - break; - case ALC_INIT_GPIO2: - snd_hda_sequence_write(codec, alc_gpio2_init_verbs); - break; - case ALC_INIT_GPIO3: - snd_hda_sequence_write(codec, alc_gpio3_init_verbs); - break; case ALC_INIT_DEFAULT: switch (codec->core.vendor_id) { case 0x10ec0260: @@ -660,13 +690,13 @@ do_sku: if (spec->init_amp == ALC_INIT_UNDEFINED) { switch (tmp) { case 1: - spec->init_amp = ALC_INIT_GPIO1; + alc_setup_gpio(codec, 0x01); break; case 3: - spec->init_amp = ALC_INIT_GPIO2; + alc_setup_gpio(codec, 0x02); break; case 7: - spec->init_amp = ALC_INIT_GPIO3; + alc_setup_gpio(codec, 0x03); break; case 5: default: @@ -1107,12 +1137,12 @@ static void alc880_fixup_vol_knob(struct hda_codec *codec, static const struct hda_fixup alc880_fixups[] = { [ALC880_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC880_FIXUP_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio2_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, }, [ALC880_FIXUP_MEDION_RIM] = { .type = HDA_FIXUP_VERBS, @@ -1565,7 +1595,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->gen.autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable_callback(codec, 0x0f, snd_hda_gen_hp_automute); - snd_hda_add_verbs(codec, alc_gpio1_init_verbs); + alc_setup_gpio(codec, 0x01); } } @@ -1639,8 +1669,8 @@ static const struct hda_fixup alc260_fixups[] = { }, }, [ALC260_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC260_FIXUP_GPIO1_TOGGLE] = { .type = HDA_FIXUP_FUNC, @@ -2144,20 +2174,20 @@ static const struct hda_fixup alc882_fixups[] = { } }, [ALC882_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC882_FIXUP_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio2_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, }, [ALC882_FIXUP_GPIO3] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio3_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio3, }, [ALC882_FIXUP_ASUS_W2JC] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, .chained = true, .chain_id = ALC882_FIXUP_EAPD, }, @@ -5232,6 +5262,9 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, snd_hda_jack_detect_enable_callback(codec, 0x1b, snd_hda_gen_hp_automute); break; + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; case HDA_FIXUP_ACT_BUILD: /* this is a bit tricky; give more sane names for the main * (tablet) speaker and the dock speaker, respectively From aaf312de4eb915e5b45c65c2da7304bf34b5ab47 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 22:28:22 +0200 Subject: [PATCH 089/529] ALSA: hda/realtek - Add GPIO data update helper For updating GPIO bits dynamically, provide a new helper, and use it from the alc260 automute hook. This helper will be used by other places in future, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 489075b23652..bd54d9e25440 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -241,6 +241,20 @@ static void alc_write_gpio_data(struct hda_codec *codec) spec->gpio_data); } +static void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask, + bool on) +{ + struct alc_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_data; + + if (on) + spec->gpio_data |= mask; + else + spec->gpio_data &= ~mask; + if (oldval != spec->gpio_data) + alc_write_gpio_data(codec); +} + static void alc_write_gpio(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1577,8 +1591,8 @@ enum { static void alc260_gpio1_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gen.hp_jack_present); + + alc_update_gpio_data(codec, 0x01, spec->gen.hp_jack_present); } static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, From d261eec80ca987a2415dd26f982ca25844e4497c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 22:32:29 +0200 Subject: [PATCH 090/529] ALSA: hda/realtek - Consolidate gpio_data and gpio_led Until now, two fields, gpio_data and gpio_led, coexist in alc_spec although basically both of them serve for the same purpose -- the GPIO data bits. This patch consolidates both usages and eliminates the superfluous gpio_led field. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++------------------------ 1 file changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bd54d9e25440..0db8329aa114 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -100,7 +100,6 @@ struct alc_spec { hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; - unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; @@ -3497,9 +3496,8 @@ static int alc269_resume(struct hda_codec *codec) * suspend, and won't restore the data after resume, so we restore it * in the driver. */ - if (spec->gpio_led) - snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); + if (spec->gpio_data) + alc_write_gpio_data(codec); if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -3739,18 +3737,10 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, bool enabled) { struct alc_spec *spec = codec->spec; - unsigned int oldval = spec->gpio_led; if (spec->mute_led_polarity) enabled = !enabled; - - if (enabled) - spec->gpio_led &= ~mask; - else - spec->gpio_led |= mask; - if (spec->gpio_led != oldval) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); + alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } /* turn on/off mute LED via GPIO per vmaster hook */ @@ -3783,7 +3773,6 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->gpio_mic_led_mask = 0x10; @@ -3804,7 +3793,6 @@ static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x02; spec->gpio_mic_led_mask = 0x20; @@ -3842,7 +3830,6 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; @@ -3865,7 +3852,6 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; @@ -3948,7 +3934,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, gpio2_mic_hotkey_event); spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x08; spec->gpio_mic_led_mask = 0x10; @@ -3988,7 +3973,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, snd_hda_jack_detect_enable_callback(codec, 0x1b, gpio2_mic_hotkey_event); - spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mic_led_mask = 0x04; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); @@ -5365,9 +5349,6 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; spec->gen.hp_automute_hook = alc280_hp_gpio4_automute_hook; - /* The GPIOs are currently off */ - spec->gpio_led = 0; - /* GPIO3 is connected to the output mute LED, * high is on, low is off */ @@ -7592,7 +7573,7 @@ static unsigned int gpio_led_power_filter(struct hda_codec *codec, unsigned int power_state) { struct alc_spec *spec = codec->spec; - if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_led) + if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_data) return AC_PWRST_D0; return power_state; } @@ -7609,7 +7590,6 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 1; spec->gpio_mute_led_mask = 0x01; snd_hda_add_verbs(codec, gpio_init); From 215c850cf205fa502aa45c2540b8d8bc70bd0f1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 22:34:26 +0200 Subject: [PATCH 091/529] ALSA: hda/realtek - Simplify alc885_fixup_macpro_gpio() The fixup for Macbook Pro is nothing but setting the GPIO bits as usual but with one exception: it adds some delay at writing the GPIO bits. Add a flag to put the conditional delay in the common helper, and clean up alc885_fixup_macpro_gpio() with the new flag. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 44 ++++++----------------------------- 1 file changed, 7 insertions(+), 37 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db8329aa114..786b29eb2ba9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,6 +94,7 @@ struct alc_spec { unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; + bool gpio_write_delay; /* add a delay before writing gpio_data */ /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; @@ -265,6 +266,8 @@ static void alc_write_gpio(struct hda_codec *codec) AC_VERB_SET_GPIO_MASK, spec->gpio_mask); snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DIRECTION, spec->gpio_dir); + if (spec->gpio_write_delay) + msleep(1); alc_write_gpio_data(codec); } @@ -1868,47 +1871,14 @@ static void alc889_fixup_coef(struct hda_codec *codec, alc_update_coef_idx(codec, 7, 0, 0x2030); } -/* toggle speaker-output according to the hp-jack state */ -static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) -{ - unsigned int gpiostate, gpiomask, gpiodir; - - gpiostate = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - - if (!muted) - gpiostate |= (1 << pin); - else - gpiostate &= ~(1 << pin); - - gpiomask = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= (1 << pin); - - gpiodir = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= (1 << pin); - - - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); - - msleep(1); - - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); -} - /* set up GPIO at initialization */ static void alc885_fixup_macpro_gpio(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action != HDA_FIXUP_ACT_INIT) - return; - alc882_gpio_mute(codec, 0, 0); - alc882_gpio_mute(codec, 1, 0); + struct alc_spec *spec = codec->spec; + + spec->gpio_write_delay = true; + alc_fixup_gpio3(codec, fix, action); } /* Fix the connection of some pins for ALC889: From 01e4a275e93bc955d0b79520156da6367505658f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 22:47:30 +0200 Subject: [PATCH 092/529] ALSA: hda/realtek - Simplify mute LED GPIO handling Now we can simplify the mute LED GPIO handling as well. Each fixup dealing with GPIO for the mute LED controls defined the static init verbs, and they are converted to the common GPIO bit fields with the new helper, alc_fixup_hp_gpio_led(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 155 +++++++++++----------------------- 1 file changed, 49 insertions(+), 106 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 786b29eb2ba9..a0881ffcd9d7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3731,44 +3731,38 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) spec->gen.micmute_led.led_value); } +/* setup mute and mic-mute GPIO bits, add hooks appropriately */ +static void alc_fixup_hp_gpio_led(struct hda_codec *codec, + int action, + unsigned int mute_mask, + unsigned int micmute_mask) +{ + struct alc_spec *spec = codec->spec; + + alc_fixup_gpio(codec, action, mute_mask | micmute_mask); + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + if (mute_mask) { + spec->gpio_mute_led_mask = mute_mask; + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + } + if (micmute_mask) { + spec->gpio_mic_led_mask = micmute_mask; + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); + } +} + static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; - - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; - spec->gpio_mic_led_mask = 0x10; - snd_hda_add_verbs(codec, gpio_init); - snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - } + alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); } static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x22 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x22 }, - {} - }; - - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x02; - spec->gpio_mic_led_mask = 0x20; - snd_hda_add_verbs(codec, gpio_init); - snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - } + alc_fixup_hp_gpio_led(codec, action, 0x02, 0x20); } /* turn on/off mic-mute LED per capture hook */ @@ -3792,18 +3786,15 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x08 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x08 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; + /* Like hp_gpio_mic1_led, but also needs GPIO4 low to + * enable headphone amp + */ + spec->gpio_mask |= 0x10; + spec->gpio_dir |= 0x10; spec->cap_mute_led_nid = 0x18; - snd_hda_add_verbs(codec, gpio_init); snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } @@ -3812,20 +3803,11 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, static void alc280_fixup_hp_gpio4(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* Like hp_gpio_mic1_led, but also needs GPIO4 low to enable headphone amp */ struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; - snd_hda_add_verbs(codec, gpio_init); snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } @@ -3876,38 +3858,29 @@ static int alc_register_micmute_input_device(struct hda_codec *codec) return 0; } +/* GPIO1 = set according to SKU external amp + * GPIO2 = mic mute hotkey + * GPIO3 = mute LED + * GPIO4 = mic mute LED + */ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* GPIO1 = set according to SKU external amp - GPIO2 = mic mute hotkey - GPIO3 = mute LED - GPIO4 = mic mute LED */ - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x1e }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x1a }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x02 }, - {} - }; - struct alc_spec *spec = codec->spec; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; - snd_hda_add_verbs(codec, gpio_init); + spec->gpio_mask |= 0x06; + spec->gpio_dir |= 0x02; + spec->gpio_data |= 0x02; snd_hda_codec_write_cache(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x04); snd_hda_jack_detect_enable_callback(codec, codec->core.afg, gpio2_mic_hotkey_event); - - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; - spec->gpio_mic_led_mask = 0x10; - snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); return; } @@ -3921,31 +3894,22 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } } +/* Line2 = mic mute hotkey + * GPIO2 = mic mute LED + */ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* Line2 = mic mute hotkey - GPIO2 = mic mute LED */ - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - {} - }; - struct alc_spec *spec = codec->spec; + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; - snd_hda_add_verbs(codec, gpio_init); snd_hda_jack_detect_enable_callback(codec, 0x1b, gpio2_mic_hotkey_event); - - spec->mute_led_polarity = 0; - spec->gpio_mic_led_mask = 0x04; - snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); return; } @@ -5306,27 +5270,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - /* Set the hooks to turn the headphone amp on/off - * as needed - */ - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + /* amp at GPIO4; toggled via alc280_hp_gpio4_automute_hook() */ + spec->gpio_mask |= 0x10; + spec->gpio_dir |= 0x10; spec->gen.hp_automute_hook = alc280_hp_gpio4_automute_hook; - - /* GPIO3 is connected to the output mute LED, - * high is on, low is off - */ - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; - - /* Initialize GPIO configuration */ - snd_hda_add_verbs(codec, gpio_init); } } @@ -7552,17 +7502,10 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x01, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; spec->mute_led_polarity = 1; - spec->gpio_mute_led_mask = 0x01; - snd_hda_add_verbs(codec, gpio_init); codec->power_filter = gpio_led_power_filter; } } From ae065f1ce07c09600af0e1dbb6b071991f7cb6e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 23:00:03 +0200 Subject: [PATCH 093/529] ALSA: hda/realtek - Convert some manual GPIO setups This patch converts the remaining static init verbs for GPIO bits with the common gpio_* fields management. Only the verbs setting the GPIO data bits are targeted in this patch. The rest will be changed in later patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 49 +++++++++++++++++++---------------- 1 file changed, 26 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0881ffcd9d7..f5cc506504c0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -296,6 +296,12 @@ static void alc_fixup_gpio3(struct hda_codec *codec, alc_fixup_gpio(codec, action, 0x03); } +static void alc_fixup_gpio4(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x04); +} + /* * Fix hardware PLL issue * On some codecs, the analog PLL gating control must be off while @@ -5172,13 +5178,6 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - /* TX300 needs to set up GPIO2 for the speaker amp */ - static const struct hda_verb gpio2_verbs[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, - {} - }; static const struct hda_pintbl dock_pins[] = { { 0x1b, 0x21114000 }, /* dock speaker pin */ {} @@ -5187,7 +5186,8 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->init_amp = ALC_INIT_DEFAULT; - snd_hda_add_verbs(codec, gpio2_verbs); + /* TX300 needs to set up GPIO2 for the speaker amp */ + alc_setup_gpio(codec, 0x04); snd_hda_apply_pincfgs(codec, dock_pins); spec->gen.auto_mute_via_amp = 1; spec->gen.automute_hook = asus_tx300_automute; @@ -5280,6 +5280,19 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, } } +static void alc275_fixup_gpio4_off(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_mask |= 0x04; + spec->gpio_dir |= 0x04; + /* set data bit low */ + } +} + static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5468,13 +5481,8 @@ static const struct hda_fixup alc269_fixups[] = { } }, [ALC275_FIXUP_SONY_VAIO_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - { } - }, + .type = HDA_FIXUP_FUNC, + .v.func = alc275_fixup_gpio4_off, .chained = true, .chain_id = ALC269_FIXUP_SONY_VAIO }, @@ -6219,14 +6227,9 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, [ALC256_FIXUP_ASUS_AIO_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* Set up GPIO2 for the speaker amp */ - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, - {} - }, + .type = HDA_FIXUP_FUNC, + /* Set up GPIO2 for the speaker amp */ + .v.func = alc_fixup_gpio4, }, [ALC233_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, From d44a68640668544891fc4b468284d1eaeb49c1d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 23:04:03 +0200 Subject: [PATCH 094/529] ALSA: hda/realtek - Simplify Dell XPS13 GPIO handling Dell XPS13 has multi-step fixups, and one of them (ALC288_FIXUP_DELL_XPS_13_GPIO6) corresponds to the management of GPIO bit6 (0x40). It used to be a static init verbs (to turn *off* the bit6). In this patch, we convert it as the gpio_mask and gpio_dir initializations folded in the existing fixup function. With this change, ALC288_FIXUP_DELL_XPS_13_GPIO6 becomes superfluous, thus it's removed. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f5cc506504c0..2cda4a614435 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4881,13 +4881,10 @@ static void alc288_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_callback *jack) { struct alc_spec *spec = codec->spec; - int present; alc_update_headset_jack_cb(codec, jack); /* Headset Mic enable or disable, only for Dell Dino */ - present = spec->gen.hp_jack_present ? 0x40 : 0; - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - present); + alc_update_gpio_data(codec, 0x40, spec->gen.hp_jack_present); } static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec, @@ -4896,6 +4893,9 @@ static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); if (action == HDA_FIXUP_ACT_PROBE) { struct alc_spec *spec = codec->spec; + /* toggled via hp_automute_hook */ + spec->gpio_mask |= 0x40; + spec->gpio_dir |= 0x40; spec->gen.hp_automute_hook = alc288_update_headset_jack_cb; } } @@ -5433,7 +5433,6 @@ enum { ALC280_FIXUP_HP_9480M, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC288_FIXUP_DELL_XPS_13_GPIO6, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, @@ -6049,22 +6048,11 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC288_FIXUP_DELL_HEADSET_MODE }, - [ALC288_FIXUP_DELL_XPS_13_GPIO6] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x40}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - { } - }, - .chained = true, - .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE - }, [ALC288_FIXUP_DISABLE_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, .chained = true, - .chain_id = ALC288_FIXUP_DELL_XPS_13_GPIO6 + .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC288_FIXUP_DELL_XPS_13] = { .type = HDA_FIXUP_FUNC, @@ -6902,7 +6890,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6, + SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), From df73d83fad97237b68949058e632b48d55533c09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2018 23:05:47 +0200 Subject: [PATCH 095/529] ALSA: hda/realtek - Use common GPIO mask for ALC660VD ASUS fixup The ALC660VD_FIX_ASUS_GPIO1 quirk requires to set up GPIO bit0 ON while bit 1 OFF. Implement the fixup function and convert from the static init verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2cda4a614435..1ee086bcefcf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7354,16 +7354,21 @@ static void alc861vd_fixup_dallas(struct hda_codec *codec, } } +/* reset GPIO1 */ +static void alc660vd_fixup_asus_gpio1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_mask |= 0x02; + alc_fixup_gpio(codec, action, 0x01); +} + static const struct hda_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* reset GPIO1 */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - { } - } + .type = HDA_FIXUP_FUNC, + .v.func = alc660vd_fixup_asus_gpio1, }, [ALC861VD_FIX_DALLAS] = { .type = HDA_FIXUP_FUNC, From a5cb463a81737dde1ef3f1b1cf3e17bf69f20669 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 12:50:11 +0200 Subject: [PATCH 096/529] ALSA: hda/realtek - Use common helper for creating ALC268 beep controls The beep mixer controls are the only remaining stuff that uses spec->mixers[] array, and they can be well converted to the standard helper in the generic parser, snd_hda_gen_add_kctl(). This simplifies the code, especially the superfluous mixers and num_mixers fields can be now removed from alc_spec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 33 +++++++++------------------------ 1 file changed, 9 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ee086bcefcf..42d3b0f77577 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -83,8 +83,6 @@ struct alc_spec { struct hda_gen_spec gen; /* must be at head */ /* codec parameterization */ - const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ - unsigned int num_mixers; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ struct alc_customize_define cdefine; @@ -207,18 +205,6 @@ static void alc_process_coef_fw(struct hda_codec *codec, } } -/* - * Append the given mixer and verb elements for the later use - * The mixer array is referred in build_controls(), and init_verbs are - * called in init(). - */ -static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) -{ - if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers))) - return; - spec->mixers[spec->num_mixers++] = mix; -} - /* * GPIO setup tables, used in initialization */ @@ -789,18 +775,12 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = { static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; err = snd_hda_gen_build_controls(codec); if (err < 0) return err; - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - #ifdef CONFIG_SND_HDA_INPUT_BEEP /* create beep controls if needed */ if (spec->beep_amp) { @@ -2663,7 +2643,6 @@ static const struct snd_kcontrol_new alc268_beep_mixer[] = { .put = alc268_beep_switch_put, .private_value = HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT) }, - { } }; /* set PCBEEP vol = 0, mute connections */ @@ -2731,7 +2710,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; - int err; + int i, err; /* ALC268 has no aa-loopback mixer */ err = alc_alloc_spec(codec, 0); @@ -2753,7 +2732,13 @@ static int patch_alc268(struct hda_codec *codec) if (err > 0 && !spec->gen.no_analog && spec->gen.autocfg.speaker_pins[0] != 0x1d) { - add_mixer(spec, alc268_beep_mixer); + for (i = 0; i < ARRAY_SIZE(alc268_beep_mixer); i++) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, + &alc268_beep_mixer[i])) { + err = -ENOMEM; + goto error; + } + } snd_hda_add_verbs(codec, alc268_beep_init_verbs); if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) /* override the amp caps for beep generator */ From fea80fae552c428b67591cb0aacca56c11c3eeaf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 12:52:46 +0200 Subject: [PATCH 097/529] ALSA: hda/realtek - Use common helper for creating beep controls In the Realtek codec driver, we used to build kctl elements for beep mixer in the own build_controls callback. This is an open-code and can be covered by the standard feature of the generic parser with snd_hda_gen_add_kctl() instead. Also, after the conversion, spec->beep_amp becomes superfluous; hence it's removed along with the conversion. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 114 +++++++++++++++++++--------------- 1 file changed, 65 insertions(+), 49 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42d3b0f77577..0e84d9cdd42d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -83,8 +83,6 @@ struct alc_spec { struct hda_gen_spec gen; /* must be at head */ /* codec parameterization */ - unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - struct alc_customize_define cdefine; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ @@ -763,41 +761,14 @@ static void alc_fixup_inv_dmic(struct hda_codec *codec, } -#ifdef CONFIG_SND_HDA_INPUT_BEEP -/* additional beep mixers; the actual parameters are overwritten at build */ -static const struct snd_kcontrol_new alc_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), - HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), - { } /* end */ -}; -#endif - static int alc_build_controls(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; int err; err = snd_hda_gen_build_controls(codec); if (err < 0) return err; -#ifdef CONFIG_SND_HDA_INPUT_BEEP - /* create beep controls if needed */ - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = alc_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } - } -#endif - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_BUILD); return 0; } @@ -1008,8 +979,30 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) * Digital-beep handlers */ #ifdef CONFIG_SND_HDA_INPUT_BEEP -#define set_beep_amp(spec, nid, idx, dir) \ - ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* additional beep mixers; private_value will be overwritten */ +static const struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), +}; + +/* set up and create beep controls */ +static int set_beep_amp(struct alc_spec *spec, hda_nid_t nid, + int idx, int dir) +{ + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir); + int i; + + for (i = 0; i < ARRAY_SIZE(alc_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &alc_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; + } + return 0; +} static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1), @@ -1034,7 +1027,7 @@ static inline int has_cdefine_beep(struct hda_codec *codec) return spec->cdefine.enable_pcbeep; } #else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#define set_beep_amp(spec, nid, idx, dir) 0 #define has_cdefine_beep(codec) 0 #endif @@ -1536,8 +1529,11 @@ static int patch_alc880(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -1784,8 +1780,11 @@ static int patch_alc260(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -2434,8 +2433,11 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -2596,8 +2598,11 @@ static int patch_alc262(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7168,8 +7173,11 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) - set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) { + err = set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7298,8 +7306,11 @@ static int patch_alc861(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7392,8 +7403,11 @@ static int patch_alc861vd(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -8103,18 +8117,20 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) { switch (codec->core.vendor_id) { case 0x10ec0662: - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); break; case 0x10ec0272: case 0x10ec0663: case 0x10ec0665: case 0x10ec0668: - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); break; case 0x10ec0273: - set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); break; } + if (err < 0) + goto error; } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); From 51e19ca5f7555802687324d3bee3bc6b7df240d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 17:06:58 +0200 Subject: [PATCH 098/529] ALSA: hda/conexant - Clean up beep code Like the previous commit for Realtek codec, the similar cleanup work can be applied to Conexant codec, too. A slight difference is that the call of cx_auto_parse_beep() is moved after snd_hda_gen_parse_auto_config(). It's not strictly needed, but it'd be good to make the creation of such beep mixers at the end, which matches with the former situation. Along with this conversion, cx_auto_build_controls() becomes just calling snd_hda_gen_build_controls(), so it's simply replaced with snd_hda_gen_build_controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 85 +++++++++++----------------------- 1 file changed, 27 insertions(+), 58 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b7339cb5c45b..75ba66eb4ccd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -37,8 +37,6 @@ struct conexant_spec { struct hda_gen_spec gen; - unsigned int beep_amp; - /* extra EAPD pins */ unsigned int num_eapds; hda_nid_t eapds[4]; @@ -62,65 +60,48 @@ struct conexant_spec { #ifdef CONFIG_SND_HDA_INPUT_BEEP -static inline void set_beep_amp(struct conexant_spec *spec, hda_nid_t nid, - int idx, int dir) -{ - spec->gen.beep_nid = nid; - spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); -} -/* additional beep mixers; the actual parameters are overwritten at build */ +/* additional beep mixers; private_value will be overwritten */ static const struct snd_kcontrol_new cxt_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), - { } /* end */ }; -/* create beep controls if needed */ -static int add_beep_ctls(struct hda_codec *codec) +static int set_beep_amp(struct conexant_spec *spec, hda_nid_t nid, + int idx, int dir) { - struct conexant_spec *spec = codec->spec; - int err; + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); + int i; - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = cxt_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } + spec->gen.beep_nid = nid; + for (i = 0; i < ARRAY_SIZE(cxt_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &cxt_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; } return 0; } -#else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ -#define add_beep_ctls(codec) 0 -#endif -/* - * Automatic parser for CX20641 & co - */ - -#ifdef CONFIG_SND_HDA_INPUT_BEEP -static void cx_auto_parse_beep(struct hda_codec *codec) +static int cx_auto_parse_beep(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; hda_nid_t nid; for_each_hda_codec_node(nid, codec) - if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { - set_beep_amp(spec, nid, 0, HDA_OUTPUT); - break; - } + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) + return set_beep_amp(spec, nid, 0, HDA_OUTPUT); + return 0; } #else -#define cx_auto_parse_beep(codec) +#define cx_auto_parse_beep(codec) 0 #endif +/* + * Automatic parser for CX20641 & co + */ + /* parse EAPDs */ static void cx_auto_parse_eapd(struct hda_codec *codec) { @@ -179,21 +160,6 @@ static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled) enabled ? 0x00 : 0x02); } -static int cx_auto_build_controls(struct hda_codec *codec) -{ - int err; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - return 0; -} - static int cx_auto_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -234,7 +200,7 @@ static void cx_auto_free(struct hda_codec *codec) } static const struct hda_codec_ops cx_auto_patch_ops = { - .build_controls = cx_auto_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, .reboot_notify = cx_auto_reboot_notify, @@ -1033,7 +999,6 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = cx_auto_patch_ops; - cx_auto_parse_beep(codec); cx_auto_parse_eapd(codec); spec->gen.own_eapd_ctl = 1; if (spec->dynamic_eapd) @@ -1093,6 +1058,10 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; + err = cx_auto_parse_beep(codec); + if (err < 0) + goto error; + /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. * Better to make reset, then. From 0785b0ecb8fa960b4f49010d0679d174efad423c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 17:10:51 +0200 Subject: [PATCH 099/529] ALSA: hda/cirrus - Simplify creation of new controls This patch moves the mixer creation code in Cirrus codec driver from its own build_controls callback to snd_hda_gen_add_kctl() for simplification. As a bonus, this allows us to remove the cs421x_build_controls as it becomes identical with snd_hda_gen_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 29 +++++++++-------------------- 1 file changed, 9 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d6e079f4ec09..a7f91be45194 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1096,25 +1096,6 @@ static int cs421x_init(struct hda_codec *codec) return 0; } -static int cs421x_build_controls(struct hda_codec *codec) -{ - struct cs_spec *spec = codec->spec; - int err; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - if (spec->gen.autocfg.speaker_outs && - spec->vendor_nid == CS4210_VENDOR_NID) { - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&cs421x_speaker_boost_ctl, codec)); - if (err < 0) - return err; - } - return 0; -} - static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) { unsigned int caps; @@ -1144,6 +1125,14 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) return err; parse_cs421x_digital(codec); + + if (spec->gen.autocfg.speaker_outs && + spec->vendor_nid == CS4210_VENDOR_NID) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, + &cs421x_speaker_boost_ctl)) + return -ENOMEM; + } + return 0; } @@ -1175,7 +1164,7 @@ static int cs421x_suspend(struct hda_codec *codec) #endif static const struct hda_codec_ops cs421x_patch_ops = { - .build_controls = cs421x_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cs421x_init, .free = cs_free, From fcbdcc1a93dd49ccd0e0f34224ce1ba4203b75de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 22:36:17 +0200 Subject: [PATCH 100/529] ALSA: hda/via - Rewrite with error goto Currently VIA codec driver invokes via_free() at each place of the error path. Move the error handling to the end of each function commonly and do goto-error as a standard idiom. This is a preliminary patch for the further cleanups, and no functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 100 +++++++++++++++++++++++--------------- 1 file changed, 60 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fc30d1e8aa76..76e47d088a41 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -686,10 +686,8 @@ static int patch_vt1708(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; /* add jack detect on/off control */ spec->mixers[spec->num_mixers++] = vt1708_jack_detect_ctl; @@ -700,6 +698,10 @@ static int patch_vt1708(struct hda_codec *codec) codec->jackpoll_interval = 0; return 0; + + error: + via_free(codec); + return err; } static int patch_vt1709(struct hda_codec *codec) @@ -715,12 +717,14 @@ static int patch_vt1709(struct hda_codec *codec) spec->gen.mixer_nid = 0x18; err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } static int patch_vt1708S(struct hda_codec *codec); @@ -741,12 +745,14 @@ static int patch_vt1708B(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1708S */ @@ -793,14 +799,16 @@ static int patch_vt1708S(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1702 */ @@ -834,14 +842,16 @@ static int patch_vt1702(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1718S */ @@ -906,14 +916,16 @@ static int patch_vt1718S(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1716S */ @@ -1002,10 +1014,8 @@ static int patch_vt1716S(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt1716S_init_verbs; @@ -1013,6 +1023,10 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; return 0; + + error: + via_free(codec); + return err; } /* for vt2002P */ @@ -1109,10 +1123,8 @@ static int patch_vt2002P(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; if (spec->codec_type == VT1802) spec->init_verbs[spec->num_iverbs++] = vt1802_init_verbs; @@ -1120,6 +1132,10 @@ static int patch_vt2002P(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* for vt1812 */ @@ -1150,14 +1166,16 @@ static int patch_vt1812(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* patch for vt3476 */ @@ -1187,14 +1205,16 @@ static int patch_vt3476(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; return 0; + + error: + via_free(codec); + return err; } /* From 0e8f9862493a55d85d3351cb4517f2e4d95c9600 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 17:20:42 +0200 Subject: [PATCH 101/529] ALSA: hda/via - Simplify control management This patch replaces the control element creations in VIA codec driver with the standard snd_hda_gen_add_kctl() calls as a cleanup. There are two major fields targeted by this patch: the beep controls and static init controls. The former is converted just like other codec drivers do. The spec->beep_amp field can be eliminated by this change as well. The latter, static init controls, are replaced simply with explicit snd_hda_gen_add_kctl() calls. After these conversions, via_build_controls() becomes superfluous and replaced with snd_hda_gen_build_controls(), too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 130 +++++++++++++------------------------- 1 file changed, 44 insertions(+), 86 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 76e47d088a41..dc4961f0dfd1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -91,9 +91,6 @@ struct via_spec { struct hda_gen_spec gen; /* codec parameterization */ - const struct snd_kcontrol_new *mixers[6]; - unsigned int num_mixers; - const struct hda_verb *init_verbs[5]; unsigned int num_iverbs; @@ -107,8 +104,6 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; - - unsigned int beep_amp; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -262,69 +257,51 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new via_pin_power_ctl_enum[] = { - { +static const struct snd_kcontrol_new via_pin_power_ctl_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Dynamic Power-Control", .info = via_pin_power_ctl_info, .get = via_pin_power_ctl_get, .put = via_pin_power_ctl_put, - }, - {} /* terminator */ }; #ifdef CONFIG_SND_HDA_INPUT_BEEP -static inline void set_beep_amp(struct via_spec *spec, hda_nid_t nid, - int idx, int dir) -{ - spec->gen.beep_nid = nid; - spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); -} - /* additional beep mixers; the actual parameters are overwritten at build */ -static const struct snd_kcontrol_new cxt_beep_mixer[] = { +static const struct snd_kcontrol_new via_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), - { } /* end */ }; -/* create beep controls if needed */ -static int add_beep_ctls(struct hda_codec *codec) +static int set_beep_amp(struct via_spec *spec, hda_nid_t nid, + int idx, int dir) { - struct via_spec *spec = codec->spec; - int err; + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); + int i; - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = cxt_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } + spec->gen.beep_nid = nid; + for (i = 0; i < ARRAY_SIZE(via_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &via_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; } return 0; } -static void auto_parse_beep(struct hda_codec *codec) +static int auto_parse_beep(struct hda_codec *codec) { struct via_spec *spec = codec->spec; hda_nid_t nid; for_each_hda_codec_node(nid, codec) - if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { - set_beep_amp(spec, nid, 0, HDA_OUTPUT); - break; - } + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) + return set_beep_amp(spec, nid, 0, HDA_OUTPUT); + return 0; } #else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ -#define add_beep_ctls(codec) 0 -#define auto_parse_beep(codec) +#define auto_parse_beep(codec) 0 #endif /* check AA path's mute status */ @@ -403,30 +380,6 @@ static void analog_low_current_mode(struct hda_codec *codec) return __analog_low_current_mode(codec, false); } -static int via_build_controls(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int err, i; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - - return 0; -} - static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream, @@ -481,7 +434,7 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) static int via_init(struct hda_codec *codec); static const struct hda_codec_ops via_patch_ops = { - .build_controls = via_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = via_init, .free = via_free, @@ -545,16 +498,13 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = { - { +static const struct snd_kcontrol_new vt1708_jack_detect_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Jack Detect", .count = 1, .info = snd_ctl_boolean_mono_info, .get = vt1708_jack_detect_get, .put = vt1708_jack_detect_put, - }, - {} /* terminator */ }; static const struct badness_table via_main_out_badness = { @@ -586,12 +536,17 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - auto_parse_beep(codec); - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) return err; + err = auto_parse_beep(codec); + if (err < 0) + return err; + + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &via_pin_power_ctl_enum)) + return -ENOMEM; + /* disable widget PM at start for compatibility */ codec->power_save_node = 0; spec->gen.power_down_unused = 0; @@ -623,7 +578,7 @@ static int vt1708_build_controls(struct hda_codec *codec) int err; int old_interval = codec->jackpoll_interval; codec->jackpoll_interval = msecs_to_jiffies(100); - err = via_build_controls(codec); + err = snd_hda_gen_build_controls(codec); codec->jackpoll_interval = old_interval; return err; } @@ -690,7 +645,10 @@ static int patch_vt1708(struct hda_codec *codec) goto error; /* add jack detect on/off control */ - spec->mixers[spec->num_mixers++] = vt1708_jack_detect_ctl; + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1708_jack_detect_ctl)) { + err = -ENOMEM; + goto error; + } spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; @@ -967,9 +925,9 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = { - HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), - { +static const struct snd_kcontrol_new vt1716s_dmic_mixer_vol = + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT); +static const struct snd_kcontrol_new vt1716s_dmic_mixer_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mic Capture Switch", .subdevice = HDA_SUBDEV_NID_FLAG | 0x26, @@ -977,16 +935,12 @@ static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = { .info = vt1716s_dmic_info, .get = vt1716s_dmic_get, .put = vt1716s_dmic_put, - }, - {} /* end */ }; /* mono-out mixer elements */ -static const struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { - HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), - { } /* end */ -}; +static const struct snd_kcontrol_new vt1716S_mono_out_mixer = + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT); static const struct hda_verb vt1716S_init_verbs[] = { /* Enable Boost Volume backdoor */ @@ -1019,8 +973,12 @@ static int patch_vt1716S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1716S_init_verbs; - spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer; - spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_vol) || + !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_sw) || + !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716S_mono_out_mixer)) { + err = -ENOMEM; + goto error; + } return 0; From f8bfc628f73c95c242dd49efa16d59005d8558fc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2018 17:27:50 +0200 Subject: [PATCH 102/529] ALSA: hda/via - Use standard verb containers In this patch, the remaining static init verbs in VIA codec driver are converted to the standard snd_hda_add_verbs() calls. The conversion is straightforward, but one change to be noted is the place of calls: since these verbs are supposed to be executed at the beginning of the init / resume procedure, we need to add snd_hda_add_verbs() calls before calling the other parsers. This is merely a cleanup, no functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 64 +++++++++++++++++++++------------------ 1 file changed, 35 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index dc4961f0dfd1..6b9617aee0e6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -90,10 +90,6 @@ enum VIA_HDA_CODEC { struct via_spec { struct hda_gen_spec gen; - /* codec parameterization */ - const struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - /* HP mode source */ unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; @@ -555,12 +551,6 @@ static int via_parse_auto_config(struct hda_codec *codec) static int via_init(struct hda_codec *codec) { - struct via_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_iverbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - /* init power states */ __analog_low_current_mode(codec, true); @@ -639,6 +629,10 @@ static int patch_vt1708(struct hda_codec *codec) vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID); vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID); + err = snd_hda_add_verbs(codec, vt1708_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) @@ -650,8 +644,6 @@ static int patch_vt1708(struct hda_codec *codec) goto error; } - spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; - /* clear jackpoll_interval again; it's set dynamically */ codec->jackpoll_interval = 0; @@ -755,13 +747,15 @@ static int patch_vt1708S(struct hda_codec *codec) if (codec->core.vendor_id == 0x11064397) snd_hda_codec_set_name(codec, "VT1705"); + err = snd_hda_add_verbs(codec, vt1708S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; - return 0; error: @@ -798,13 +792,15 @@ static int patch_vt1702(struct hda_codec *codec) (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | (1 << AC_AMPCAP_MUTE_SHIFT)); + err = snd_hda_add_verbs(codec, vt1702_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; - return 0; error: @@ -872,13 +868,15 @@ static int patch_vt1718S(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt1718S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; - return 0; error: @@ -966,13 +964,15 @@ static int patch_vt1716S(struct hda_codec *codec) override_mic_boost(codec, 0x1a, 0, 3, 40); override_mic_boost(codec, 0x1e, 0, 3, 40); + err = snd_hda_add_verbs(codec, vt1716S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt1716S_init_verbs; - if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_vol) || !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_sw) || !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716S_mono_out_mixer)) { @@ -1079,16 +1079,18 @@ static int patch_vt2002P(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (spec->codec_type == VT1802) + err = snd_hda_add_verbs(codec, vt1802_init_verbs); + else + err = snd_hda_add_verbs(codec, vt2002P_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - if (spec->codec_type == VT1802) - spec->init_verbs[spec->num_iverbs++] = vt1802_init_verbs; - else - spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; - return 0; error: @@ -1122,13 +1124,15 @@ static int patch_vt1812(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt1812_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; - return 0; error: @@ -1161,13 +1165,15 @@ static int patch_vt3476(struct hda_codec *codec) spec->gen.mixer_nid = 0x3f; add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt3476_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) goto error; - spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; - return 0; error: From efe557320ab6cea12205794fd4062e6e850b7e1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Jun 2018 11:55:02 +0200 Subject: [PATCH 103/529] ALSA: hda/realtek - Apply PRE_PROBE fixup after ALC269 codec variant setups Currently patch_alc269() calls the fixup with HDA_FIXUP_ACT_PRE_PROBE before setting up the codec model-specific setups (e.g. setting codec_variant or mixer_nid setup). This is rather confusing as others do call the *_PRE_PROBE fixup after such a setup. Due to this disorder, we have to override spec->shutup not at the usual HDA_FIXUP_ACT_PRE_PROBE but the unusual HDA_FIXUP_ACT_PROBE time. This patch corrects the fixup call orders in patch_alc269(), and also corrects the action to set up spec->shutup accordingly. No functional changes but just refactoring. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 ++++++++++++++--------------- 1 file changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e84d9cdd42d..c93e09f9c109 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4906,7 +4906,7 @@ static void alc_no_shutup(struct hda_codec *codec) static void alc_fixup_no_shutup(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PROBE) { + if (action == HDA_FIXUP_ACT_PRE_PROBE) { struct alc_spec *spec = codec->spec; spec->shutup = alc_no_shutup; } @@ -4988,10 +4988,9 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, * it causes a click noise at start up */ snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); + spec->shutup = alc_shutup_dell_xps13; break; case HDA_FIXUP_ACT_PROBE: - spec->shutup = alc_shutup_dell_xps13; - /* Make the internal mic the default input source. */ for (i = 0; i < imux->num_items; i++) { if (spec->gen.imux_pins[i] == 0x12) { @@ -7037,18 +7036,6 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc_default_shutup; spec->init_hook = alc_default_init; - snd_hda_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); - snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, - alc269_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - - if (has_cdefine_beep(codec)) - spec->gen.beep_nid = 0x01; - switch (codec->core.vendor_id) { case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; @@ -7168,6 +7155,18 @@ static int patch_alc269(struct hda_codec *codec) spec->init_hook = alc5505_dsp_init; } + snd_hda_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, + alc269_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) From 50c678772a0b3f9dc6b04262f27e2c373fe03a4d Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 24 Jun 2018 20:14:41 +0200 Subject: [PATCH 104/529] ASoC: cx20442: Don't ignore regulator_get() errors. In its current shape, the driver just ignores errors returned by regulator_get() at component_probe(). This doesn't hurt on Amstrad Delta board as long as it registers the codec device at late_initcall, when the regulator which depends on basic-mmio-gpio device (probed as late as at dev_initcall) is already available. Otherwise the driver may end up trying to control a codec which is not powered up. Remove that dependency on initialization order by handling the error. If the regulator is not yet available and -ENODEV is returned, convert it to -EPROBE_DEFER to get another chance. Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 07dd33b09596..ab174b5114dc 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component) return -ENOMEM; cx20442->por = regulator_get(component->dev, "POR"); - if (IS_ERR(cx20442->por)) - dev_warn(component->dev, "failed to get the regulator"); + if (IS_ERR(cx20442->por)) { + int err = PTR_ERR(cx20442->por); + + dev_warn(component->dev, "failed to get POR supply (%d)", err); + /* + * When running on a non-dt platform and requested regulator + * is not available, regulator_get() never returns + * -EPROBE_DEFER as it is not able to justify if the regulator + * may still appear later. On the other hand, the board can + * still set full constraints flag at late_initcall in order + * to instruct regulator_get() to return a dummy one if + * sufficient. Hence, if we get -ENODEV here, let's convert + * it to -EPROBE_DEFER and wait for the board to decide or + * let Deferred Probe infrastructure handle this error. + */ + if (err == -ENODEV) + err = -EPROBE_DEFER; + kfree(cx20442); + return err; + } + cx20442->tty = NULL; snd_soc_component_set_drvdata(component, cx20442); From b66c9b911fe6ca188002b342b05c43deab4491a3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 26 Jun 2018 08:58:35 -0300 Subject: [PATCH 105/529] ASoC: soc-utils: Fix unregistration order The unregistration should happen in the opposite order of the registration, so change it accordingly. No real issue has been noticed, but it is good practice to keep the correct unregistration order. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..a863bb3f66c2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -381,6 +381,6 @@ int __init snd_soc_util_init(void) void __exit snd_soc_util_exit(void) { - platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); } From c486a185744c593417a126aebb119771c1bbe670 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:05 +0100 Subject: [PATCH 106/529] ASoC: q6adm: dt-bindings: add compatible string to routing Add compatible string to routing so that it can support DT based module autoloading. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,q6adm.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt index cb709e5dbc44..bbae426cdfb1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt @@ -18,6 +18,11 @@ used by the apr service device. = ADM routing "routing" subnode of the ADM node represents adm routing specific configuration +- compatible: + Usage: required + Value type: + Definition: must be "qcom,q6adm-routing". + - #sound-dai-cells Usage: required Value type: @@ -28,6 +33,7 @@ q6adm@8 { compatible = "qcom,q6adm"; reg = ; q6routing: routing { + compatible = "qcom,q6adm-routing"; #sound-dai-cells = <0>; }; }; From 9618b706672db1b595c7076389a833078335417c Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:06 +0100 Subject: [PATCH 107/529] ASoC: q6asm: dt-bindings: add compatible string to dais Add compatible string to dais so that it can support DT based module autoloading. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 2178eb91146f..f9c7bd8c1bc0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -17,6 +17,11 @@ used by the apr service device. = ASM DAIs (Digial Audio Interface) "dais" subnode of the ASM node represents dai specific configuration +- compatible: + Usage: required + Value type: + Definition: must be "qcom,q6asm-dais". + - #sound-dai-cells Usage: required Value type: @@ -28,6 +33,7 @@ q6asm@7 { compatible = "qcom,q6asm"; reg = ; q6asmdai: dais { + compatible = "qcom,q6asm-dais"; #sound-dai-cells = <1>; }; }; From e43792c6e5027ad3f0280f5a2e6952b0d436b19b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:07 +0100 Subject: [PATCH 108/529] ASoC: q6afe: dt-bindings: add compatible string to dais Add compatible string to dais so that it can support DT based module autoloading. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index bdbf87df8c0b..a8179409c194 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -17,6 +17,11 @@ used by all apr services. Must contain the following properties. subnode of "dais" representing board specific dai setup. "dais" node should have following properties followed by dai children. +- compatible: + Usage: required + Value type: + Definition: must be "qcom,q6afe-dais" + - #sound-dai-cells Usage: required Value type: @@ -100,6 +105,7 @@ q6afe@4 { reg = ; dais { + compatible = "qcom,q6afe-dais"; #sound-dai-cells = <1>; #address-cells = <1>; #size-cells = <0>; From f614c9b070ed149bbaac4edefb2b5fcb7755c4b0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:08 +0100 Subject: [PATCH 109/529] ASoC: qdsp6: q6adm: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 9983c665a941..932c3ebfd252 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -64,7 +64,6 @@ struct q6adm { struct aprv2_ibasic_rsp_result_t result; struct mutex lock; wait_queue_head_t matrix_map_wait; - struct platform_device *pdev_routing; }; struct q6adm_cmd_device_open_v5 { @@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close); static int q6adm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6adm *adm; adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); @@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - dais_np = of_get_child_by_name(dev->of_node, "routing"); - if (dais_np) { - adm->pdev_routing = of_platform_device_create(dais_np, - "q6routing", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6adm_remove(struct apr_device *adev) { - struct q6adm *adm = dev_get_drvdata(&adev->dev); - - if (adm->pdev_routing) - of_platform_device_destroy(&adm->pdev_routing->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } From 4aac7e2773030d667491fbb6d97c9f467fdcbc05 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:09 +0100 Subject: [PATCH 110/529] ASoC: qdsp6: q6asm: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 530852385cad..c4fd28f168d5 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -177,7 +177,6 @@ struct q6asm { struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; - struct platform_device *pdev_dais; }; struct audio_client { @@ -1344,7 +1343,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); static int q6asm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6asm *q6asm; q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL); @@ -1359,22 +1357,12 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - q6asm->pdev_dais = of_platform_device_create(dais_np, - "q6asm-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6asm_remove(struct apr_device *adev) { - struct q6asm *q6asm = dev_get_drvdata(&adev->dev); - - if (q6asm->pdev_dais) - of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } From 01afbd45f78cb0557db18c3ba768eea3e9576cfd Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:10 +0100 Subject: [PATCH 111/529] ASoC: qdsp6: q6afe: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..621b67b34db9 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -316,7 +316,6 @@ struct q6afe { struct mutex lock; struct list_head port_list; spinlock_t port_list_lock; - struct platform_device *pdev_dais; }; struct afe_port_cmd_device_start { @@ -1438,7 +1437,6 @@ static int q6afe_probe(struct apr_device *adev) { struct q6afe *afe; struct device *dev = &adev->dev; - struct device_node *dais_np; afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); if (!afe) @@ -1453,22 +1451,12 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - afe->pdev_dais = of_platform_device_create(dais_np, - "q6afe-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6afe_remove(struct apr_device *adev) { - struct q6afe *afe = dev_get_drvdata(&adev->dev); - - if (afe->pdev_dais) - of_platform_device_destroy(&afe->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } From eb7cc9be6e9ce17e69252a6fc00e75a5b08201ab Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:11 +0100 Subject: [PATCH 112/529] ASoC: qdsp6: q6afe-dai: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..1d2e5013c121 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1290,9 +1290,16 @@ static int q6afe_dai_dev_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6afe_dai_device_id[] = { + { .compatible = "qcom,q6afe-dais" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6afe_dai_device_id); + static struct platform_driver q6afe_dai_platform_driver = { .driver = { .name = "q6afe-dai", + .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, .remove = q6afe_dai_dev_remove, From 1ce09ef36fb190fa207fdb3e31fc1c8caa292125 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:12 +0100 Subject: [PATCH 113/529] ASoC: qdsp6: q6asm-dai: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..1196dc7483d2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -611,9 +611,16 @@ static int q6asm_dai_dev_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6asm_dai_device_id[] = { + { .compatible = "qcom,q6asm-dais" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); + static struct platform_driver q6asm_dai_platform_driver = { .driver = { .name = "q6asm-dai", + .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, .remove = q6asm_dai_dev_remove, From f48bde4bfbcf434d6aef604c1c50d68b12a4bc45 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:13 +0100 Subject: [PATCH 114/529] ASoC: qdsp6: q6routing: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..ab696bf8d1d3 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -993,9 +993,16 @@ static int q6pcm_routing_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6pcm_routing_device_id[] = { + { .compatible = "qcom,q6adm-routing" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id); + static struct platform_driver q6pcm_routing_platform_driver = { .driver = { .name = "q6routing", + .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, .remove = q6pcm_routing_remove, From 2d12c20b98ad610892151da37367f2d018181455 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:14 +0100 Subject: [PATCH 115/529] ASoC: qcom: apq8096: remove redundant owner assignment module owner is already set in platform_driver_register(), so remove this redundant assignment. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 239b8cb77bdb..cab8c4ff7c00 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -246,7 +246,6 @@ static struct platform_driver msm_snd_apq8096_driver = { .remove = apq8096_platform_remove, .driver = { .name = "msm-snd-apq8096", - .owner = THIS_MODULE, .of_match_table = msm_snd_apq8096_dt_match, }, }; From 972562f7aaaa261ec6e1ac14fed0c5bdd0dfb1cb Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:15 +0100 Subject: [PATCH 116/529] ASoC: qdsp6: q6routing: add proper error check q6adm_open can return error pointer or a null in error cases. Fix the return handling. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index ab696bf8d1d3..c80fdbc2442e 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -310,7 +310,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode, session->channels, topology, perf_mode, session->bits_per_sample, 0, 0); - if (!copp) { + if (IS_ERR_OR_NULL(copp)) { mutex_unlock(&routing_data->lock); return -EINVAL; } From f339155a4063cf3177bb38f6e83760951148e86b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:16 +0100 Subject: [PATCH 117/529] ASoC: qdsp6: q6asm: remove unused struct q6asm member pcmdev in struct q6asm seems be left over and unused, so just remove it. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index c4fd28f168d5..2b2c7233bb5f 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -174,7 +174,6 @@ struct q6asm { struct device *dev; struct q6core_svc_api_info ainfo; wait_queue_head_t mem_wait; - struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; }; From 5c10ed433da26ff3509ae11c6b22d21e131484d8 Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Tue, 26 Jun 2018 06:24:37 -0300 Subject: [PATCH 118/529] sound: restore MultiSound script This script is mentioned at multisound Kconfig and files. As the driver still exists, it probably makes sense to restore it. Fixes: 727dede0ba8a ("sound: Retire OSS") Signed-off-by: Mauro Carvalho Chehab Signed-off-by: Takashi Iwai --- Documentation/sound/cards/multisound.sh | 1137 +++++++++++++++++++++++ 1 file changed, 1137 insertions(+) create mode 100755 Documentation/sound/cards/multisound.sh diff --git a/Documentation/sound/cards/multisound.sh b/Documentation/sound/cards/multisound.sh new file mode 100755 index 000000000000..7a7a88256dfd --- /dev/null +++ b/Documentation/sound/cards/multisound.sh @@ -0,0 +1,1137 @@ +#! /bin/sh +# +# Turtle Beach MultiSound Driver Notes +# -- Andrew Veliath +# +# Last update: September 10, 1998 +# Corresponding msnd driver: 0.8.3 +# +# ** This file is a README (top part) and shell archive (bottom part). +# The corresponding archived utility sources can be unpacked by +# running `sh MultiSound' (the utilities are only needed for the +# Pinnacle and Fiji cards). ** +# +# +# -=-=- Getting Firmware -=-=- +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# See the section `Obtaining and Creating Firmware Files' in this +# document for instructions on obtaining the necessary firmware +# files. +# +# +# Supported Features +# ~~~~~~~~~~~~~~~~~~ +# +# Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is +# not currently available) and mixer functionality (/dev/mixer) are +# supported (memory mapped digital audio is not yet supported). +# Digital transfers and monitoring can be done as well if you have +# the digital daughterboard (see the section on using the S/PDIF port +# for more information). +# +# Support for the Turtle Beach MultiSound Hurricane architecture is +# composed of the following modules (these can also operate compiled +# into the kernel): +# +# msnd - MultiSound base (requires soundcore) +# +# msnd_classic - Base audio/mixer support for Classic, Monetery and +# Tahiti cards +# +# msnd_pinnacle - Base audio/mixer support for Pinnacle and Fiji cards +# +# +# Important Notes - Read Before Using +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# The firmware files are not included (may change in future). You +# must obtain these images from Turtle Beach (they are included in +# the MultiSound Development Kits), and place them in /etc/sound for +# example, and give the full paths in the Linux configuration. If +# you are compiling in support for the MultiSound driver rather than +# using it as a module, these firmware files must be accessible +# during kernel compilation. +# +# Please note these files must be binary files, not assembler. See +# the section later in this document for instructions to obtain these +# files. +# +# +# Configuring Card Resources +# ~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# ** This section is very important, as your card may not work at all +# or your machine may crash if you do not do this correctly. ** +# +# * Classic/Monterey/Tahiti +# +# These cards are configured through the driver msnd_classic. You must +# know the io port, then the driver will select the irq and memory resources +# on the card. It is up to you to know if these are free locations or now, +# a conflict can lock the machine up. +# +# * Pinnacle/Fiji +# +# The Pinnacle and Fiji cards have an extra config port, either +# 0x250, 0x260 or 0x270. This port can be disabled to have the card +# configured strictly through PnP, however you lose the ability to +# access the IDE controller and joystick devices on this card when +# using PnP. The included pinnaclecfg program in this shell archive +# can be used to configure the card in non-PnP mode, and in PnP mode +# you can use isapnptools. These are described briefly here. +# +# pinnaclecfg is not required; you can use the msnd_pinnacle module +# to fully configure the card as well. However, pinnaclecfg can be +# used to change the resource values of a particular device after the +# msnd_pinnacle module has been loaded. If you are compiling the +# driver into the kernel, you must set these values during compile +# time, however other peripheral resource values can be changed with +# the pinnaclecfg program after the kernel is loaded. +# +# +# *** PnP mode +# +# Use pnpdump to obtain a sample configuration if you can; I was able +# to obtain one with the command `pnpdump 1 0x203' -- this may vary +# for you (running pnpdump by itself did not work for me). Then, +# edit this file and use isapnp to uncomment and set the card values. +# Use these values when inserting the msnd_pinnacle module. Using +# this method, you can set the resources for the DSP and the Kurzweil +# synth (Pinnacle). Since Linux does not directly support PnP +# devices, you may have difficulty when using the card in PnP mode +# when it the driver is compiled into the kernel. Using non-PnP mode +# is preferable in this case. +# +# Here is an example mypinnacle.conf for isapnp that sets the card to +# io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil +# synth to 0x330 and irq 9 (may need editing for your system): +# +# (READPORT 0x0203) +# (CSN 2) +# (IDENTIFY *) +# +# # DSP +# (CONFIGURE BVJ0440/-1 (LD 0 +# (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000)) +# (ACT Y))) +# +# # Kurzweil Synth (Pinnacle Only) +# (CONFIGURE BVJ0440/-1 (LD 1 +# (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E))) +# (ACT Y))) +# +# (WAITFORKEY) +# +# +# *** Non-PnP mode +# +# The second way is by running the card in non-PnP mode. This +# actually has some advantages in that you can access some other +# devices on the card, such as the joystick and IDE controller. To +# configure the card, unpack this shell archive and build the +# pinnaclecfg program. Using this program, you can assign the +# resource values to the card's devices, or disable the devices. As +# an alternative to using pinnaclecfg, you can specify many of the +# configuration values when loading the msnd_pinnacle module (or +# during kernel configuration when compiling the driver into the +# kernel). +# +# If you specify cfg=0x250 for the msnd_pinnacle module, it +# automatically configure the card to the given io, irq and memory +# values using that config port (the config port is jumper selectable +# on the card to 0x250, 0x260 or 0x270). +# +# See the `msnd_pinnacle Additional Options' section below for more +# information on these parameters (also, if you compile the driver +# directly into the kernel, these extra parameters can be useful +# here). +# +# +# ** It is very easy to cause problems in your machine if you choose a +# resource value which is incorrect. ** +# +# +# Examples +# ~~~~~~~~ +# +# * MultiSound Classic/Monterey/Tahiti: +# +# modprobe soundcore +# insmod msnd +# insmod msnd_classic io=0x290 irq=7 mem=0xd0000 +# +# * MultiSound Pinnacle in PnP mode: +# +# modprobe soundcore +# insmod msnd +# isapnp mypinnacle.conf +# insmod msnd_pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values +# +# * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port, +# one of 0x250, 0x260 or 0x270): +# +# insmod soundcore +# insmod msnd +# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP +# mode, add the following (assumes you did `isapnp mypinnacle.conf'): +# +# insmod sound +# insmod mpu401 io=0x330 irq=9 <-- match mypinnacle.conf values +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP +# mode, add the following. Note how we first configure the peripheral's +# resources, _then_ install a Linux driver for it: +# +# insmod sound +# pinnaclecfg 0x250 mpu 0x330 9 +# insmod mpu401 io=0x330 irq=9 +# +# -- OR you can use the following sequence without pinnaclecfg in non-PnP mode: +# +# insmod soundcore +# insmod msnd +# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9 +# insmod sound +# insmod mpu401 io=0x330 irq=9 +# +# * To setup the joystick port on the Pinnacle in non-PnP mode (though +# you have to find the actual Linux joystick driver elsewhere), you +# can use pinnaclecfg: +# +# pinnaclecfg 0x250 joystick 0x200 +# +# -- OR you can configure this using msnd_pinnacle with the following: +# +# insmod soundcore +# insmod msnd +# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200 +# +# +# msnd_classic, msnd_pinnacle Required Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If the following options are not given, the module will not load. +# Examine the kernel message log for informative error messages. +# WARNING--probing isn't supported so try to make sure you have the +# correct shared memory area, otherwise you may experience problems. +# +# io I/O base of DSP, e.g. io=0x210 +# irq IRQ number, e.g. irq=5 +# mem Shared memory area, e.g. mem=0xd8000 +# +# +# msnd_classic, msnd_pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# fifosize The digital audio FIFOs, in kilobytes. If not +# specified, the default will be used. Increasing +# this value will reduce the chance of a FIFO +# underflow at the expense of increasing overall +# latency. For example, fifosize=512 will +# allocate 512kB read and write FIFOs (1MB total). +# While this may reduce dropouts, a heavy machine +# load will undoubtedly starve the FIFO of data +# and you will eventually get dropouts. One +# option is to alter the scheduling priority of +# the playback process, using `nice' or some form +# of POSIX soft real-time scheduling. +# +# calibrate_signal Setting this to one calibrates the ADCs to the +# signal, zero calibrates to the card (defaults +# to zero). +# +# +# msnd_pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# digital Specify digital=1 to enable the S/PDIF input +# if you have the digital daughterboard +# adapter. This will enable access to the +# DIGITAL1 input for the soundcard in the mixer. +# Some mixer programs might have trouble setting +# the DIGITAL1 source as an input. If you have +# trouble, you can try the setdigital.c program +# at the bottom of this document. +# +# cfg Non-PnP configuration port for the Pinnacle +# and Fiji (typically 0x250, 0x260 or 0x270, +# depending on the jumper configuration). If +# this option is omitted, then it is assumed +# that the card is in PnP mode, and that the +# specified DSP resource values are already +# configured with PnP (i.e. it won't attempt to +# do any sort of configuration). +# +# When the Pinnacle is in non-PnP mode, you can use the following +# options to configure particular devices. If a full specification +# for a device is not given, then the device is not configured. Note +# that you still must use a Linux driver for any of these devices +# once their resources are setup (such as the Linux joystick driver, +# or the MPU401 driver from OSS for the Kurzweil synth). +# +# mpu_io I/O port of MPU (on-board Kurzweil synth) +# mpu_irq IRQ of MPU (on-board Kurzweil synth) +# ide_io0 First I/O port of IDE controller +# ide_io1 Second I/O port of IDE controller +# ide_irq IRQ IDE controller +# joystick_io I/O port of joystick +# +# +# Obtaining and Creating Firmware Files +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# For the Classic/Tahiti/Monterey +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach: +# +# ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip +# +# When unzipped, unzip the file named MsndFiles.zip. Then copy the +# following firmware files to /etc/sound (note the file renaming): +# +# cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin +# cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin +# +# When configuring the Linux kernel, specify /etc/sound/msndinit.bin and +# /etc/sound/msndperm.bin for the two firmware files (Linux kernel +# versions older than 2.2 do not ask for firmware paths, and are +# hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# For the Pinnacle/Fiji +# ~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach (be +# sure to use the entire URL; some have had trouble navigating to the +# URL): +# +# ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip +# +# Unpack this shell archive, and run make in the created directory +# (you need a C compiler and flex to build the utilities). This +# should give you the executables conv, pinnaclecfg and setdigital. +# conv is only used temporarily here to create the firmware files, +# while pinnaclecfg is used to configure the Pinnacle or Fiji card in +# non-PnP mode, and setdigital can be used to set the S/PDIF input on +# the mixer (pinnaclecfg and setdigital should be copied to a +# convenient place, possibly run during system initialization). +# +# To generating the firmware files with the `conv' program, we create +# the binary firmware files by doing the following conversion +# (assuming the archive unpacked into a directory named PINNDDK): +# +# ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin +# ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin +# +# The conv (and conv.l) program is not needed after conversion and can +# be safely deleted. Then, when configuring the Linux kernel, specify +# /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two +# firmware files (Linux kernel versions older than 2.2 do not ask for +# firmware paths, and are hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# Using Digital I/O with the S/PDIF Port +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If you have a Pinnacle or Fiji with the digital daughterboard and +# want to set it as the input source, you can use this program if you +# have trouble trying to do it with a mixer program (be sure to +# insert the module with the digital=1 option, or say Y to the option +# during compiled-in kernel operation). Upon selection of the S/PDIF +# port, you should be able monitor and record from it. +# +# There is something to note about using the S/PDIF port. Digital +# timing is taken from the digital signal, so if a signal is not +# connected to the port and it is selected as recording input, you +# will find PCM playback to be distorted in playback rate. Also, +# attempting to record at a sampling rate other than the DAT rate may +# be problematic (i.e. trying to record at 8000Hz when the DAT signal +# is 44100Hz). If you have a problem with this, set the recording +# input to analog if you need to record at a rate other than that of +# the DAT rate. +# +# +# -- Shell archive attached below, just run `sh MultiSound' to extract. +# Contains Pinnacle/Fiji utilities to convert firmware, configure +# in non-PnP mode, and select the DIGITAL1 input for the mixer. +# +# +#!/bin/sh +# This is a shell archive (produced by GNU sharutils 4.2). +# To extract the files from this archive, save it to some FILE, remove +# everything before the `!/bin/sh' line above, then type `sh FILE'. +# +# Made on 1998-12-04 10:07 EST by . +# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'. +# +# Existing files will *not* be overwritten unless `-c' is specified. +# +# This shar contains: +# length mode name +# ------ ---------- ------------------------------------------ +# 2046 -rw-rw-r-- MultiSound.d/setdigital.c +# 10235 -rw-rw-r-- MultiSound.d/pinnaclecfg.c +# 106 -rw-rw-r-- MultiSound.d/Makefile +# 141 -rw-rw-r-- MultiSound.d/conv.l +# 1472 -rw-rw-r-- MultiSound.d/msndreset.c +# +save_IFS="${IFS}" +IFS="${IFS}:" +gettext_dir=FAILED +locale_dir=FAILED +first_param="$1" +for dir in $PATH +do + if test "$gettext_dir" = FAILED && test -f $dir/gettext \ + && ($dir/gettext --version >/dev/null 2>&1) + then + set `$dir/gettext --version 2>&1` + if test "$3" = GNU + then + gettext_dir=$dir + fi + fi + if test "$locale_dir" = FAILED && test -f $dir/shar \ + && ($dir/shar --print-text-domain-dir >/dev/null 2>&1) + then + locale_dir=`$dir/shar --print-text-domain-dir` + fi +done +IFS="$save_IFS" +if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED +then + echo=echo +else + TEXTDOMAINDIR=$locale_dir + export TEXTDOMAINDIR + TEXTDOMAIN=sharutils + export TEXTDOMAIN + echo="$gettext_dir/gettext -s" +fi +touch -am 1231235999 $$.touch >/dev/null 2>&1 +if test ! -f 1231235999 && test -f $$.touch; then + shar_touch=touch +else + shar_touch=: + echo + $echo 'WARNING: not restoring timestamps. Consider getting and' + $echo "installing GNU \`touch', distributed in GNU File Utilities..." + echo +fi +rm -f 1231235999 $$.touch +# +if mkdir _sh01426; then + $echo 'x -' 'creating lock directory' +else + $echo 'failed to create lock directory' + exit 1 +fi +# ============= MultiSound.d/setdigital.c ============== +if test ! -d 'MultiSound.d'; then + $echo 'x -' 'creating directory' 'MultiSound.d' + mkdir 'MultiSound.d' +fi +if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' && +/********************************************************************* +X * +X * setdigital.c - sets the DIGITAL1 input for a mixer +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include +#include +#include +#include +#include +#include +#include +X +int main(int argc, char *argv[]) +{ +X int fd; +X unsigned long recmask, recsrc; +X +X if (argc != 2) { +X fprintf(stderr, "usage: setdigital \n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) { +X fprintf(stderr, "error: ioctl read recording mask failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X if (!(recmask & SOUND_MASK_DIGITAL1)) { +X fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n"); +X close(fd); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl read recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X recsrc |= SOUND_MASK_DIGITAL1; +X +X if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl write recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' && + chmod 0664 'MultiSound.d/setdigital.c' || + $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed' +e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`" + test 2046 -eq "$shar_count" || + $echo 'MultiSound.d/setdigital.c:' 'original size' '2046,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/pinnaclecfg.c ============== +if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' && +/********************************************************************* +X * +X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program +X * +X * This is for NON-PnP mode only. For PnP mode, use isapnptools. +X * +X * This is Linux-specific, and must be run with root permissions. +X * +X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include +#include +#include +#include +#include +#include +#include +X +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 +X +#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((BYTE)(w)) +#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8))) +X +typedef __u8 BYTE; +typedef __u16 USHORT; +typedef __u16 WORD; +X +static int config_port = -1; +X +static int msnd_write_cfg(int cfg, int reg, int value) +{ +X outb(reg, cfg); +X outb(value, cfg + 1); +X if (value != inb(cfg + 1)) { +X fprintf(stderr, "error: msnd_write_cfg: I/O error\n"); +X return -EIO; +X } +X return 0; +} +X +static int msnd_read_cfg(int cfg, int reg) +{ +X outb(reg, cfg); +X return inb(cfg + 1); +} +X +static int msnd_write_cfg_io0(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io0(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO), +X msnd_read_cfg(cfg, IREG_IO0_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_io1(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io1(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO), +X msnd_read_cfg(cfg, IREG_IO1_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_irq(int cfg, int num, WORD irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_irq(int cfg, int num, WORD *irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER); +X +X return 0; +} +X +static int msnd_write_cfg_mem(int cfg, int num, int mem) +{ +X WORD wmem; +X +X mem >>= 8; +X mem &= 0xfff; +X wmem = (WORD)mem; +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) +X return -EIO; +X if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_mem(int cfg, int num, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO), +X msnd_read_cfg(cfg, IREG_MEMBASEHI)); +X *mem <<= 8; +X +X return 0; +} +X +static int msnd_activate_logical(int cfg, int num) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) +X return -EIO; +X return 0; +} +X +static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_write_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_write_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_write_cfg_mem(cfg, num, mem)) +X return -EIO; +X if (msnd_activate_logical(cfg, num)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_read_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_read_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_read_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_read_cfg_mem(cfg, num, mem)) +X return -EIO; +X return 0; +} +X +static void usage(void) +{ +X fprintf(stderr, +X "\n" +X "pinnaclecfg 1.0\n" +X "\n" +X "usage: pinnaclecfg [device config]\n" +X "\n" +X "This is for use with the card in NON-PnP mode only.\n" +X "\n" +X "Available devices (not all available for Fiji):\n" +X "\n" +X " Device Description\n" +X " -------------------------------------------------------------------\n" +X " reset Reset all devices (i.e. disable)\n" +X " show Display current device configurations\n" +X "\n" +X " dsp Audio device\n" +X " mpu Internal Kurzweil synth\n" +X " ide On-board IDE controller\n" +X " joystick Joystick port\n" +X "\n"); +X exit(1); +} +X +static int cfg_reset(void) +{ +X int i; +X +X for (i = 0; i < 4; ++i) +X msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0); +X +X return 0; +} +X +static int cfg_show(void) +{ +X int i; +X int count = 0; +X +X for (i = 0; i < 4; ++i) { +X WORD io0, io1, irq; +X int mem; +X msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem); +X switch (i) { +X case 0: +X if (io0 || irq || mem) { +X printf("dsp 0x%x %d 0x%x\n", io0, irq, mem); +X ++count; +X } +X break; +X case 1: +X if (io0 || irq) { +X printf("mpu 0x%x %d\n", io0, irq); +X ++count; +X } +X break; +X case 2: +X if (io0 || io1 || irq) { +X printf("ide 0x%x 0x%x %d\n", io0, io1, irq); +X ++count; +X } +X break; +X case 3: +X if (io0) { +X printf("joystick 0x%x\n", io0); +X ++count; +X } +X break; +X } +X } +X +X if (count == 0) +X fprintf(stderr, "no devices configured\n"); +X +X return 0; +} +X +static int cfg_dsp(int argc, char *argv[]) +{ +X int io, irq, mem; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1 || +X sscanf(argv[2], "0x%x", &mem) != 1) +X usage(); +X +X if (!(io == 0x290 || +X io == 0x260 || +X io == 0x250 || +X io == 0x240 || +X io == 0x230 || +X io == 0x220 || +X io == 0x210 || +X io == 0x3e0)) { +X fprintf(stderr, "error: io must be one of " +X "210, 220, 230, 240, 250, 260, 290, or 3E0\n"); +X usage(); +X } +X +X if (!(irq == 5 || +X irq == 7 || +X irq == 9 || +X irq == 10 || +X irq == 11 || +X irq == 12)) { +X fprintf(stderr, "error: irq must be one of " +X "5, 7, 9, 10, 11 or 12\n"); +X usage(); +X } +X +X if (!(mem == 0xb0000 || +X mem == 0xc8000 || +X mem == 0xd0000 || +X mem == 0xd8000 || +X mem == 0xe0000 || +X mem == 0xe8000)) { +X fprintf(stderr, "error: mem must be one of " +X "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); +X usage(); +X } +X +X return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem); +} +X +static int cfg_mpu(int argc, char *argv[]) +{ +X int io, irq; +X +X if (argc < 2 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0); +} +X +static int cfg_ide(int argc, char *argv[]) +{ +X int io0, io1, irq; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io0) != 1 || +X sscanf(argv[0], "0x%x", &io1) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0); +} +X +static int cfg_joystick(int argc, char *argv[]) +{ +X int io; +X +X if (argc < 1 || +X sscanf(argv[0], "0x%x", &io) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0); +} +X +int main(int argc, char *argv[]) +{ +X char *device; +X int rv = 0; +X +X --argc; ++argv; +X +X if (argc < 2) +X usage(); +X +X sscanf(argv[0], "0x%x", &config_port); +X if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) { +X fprintf(stderr, "error: must be 0x250, 0x260 or 0x270\n"); +X exit(1); +X } +X if (ioperm(config_port, 2, 1)) { +X perror("ioperm"); +X fprintf(stderr, "note: pinnaclecfg must be run as root\n"); +X exit(1); +X } +X device = argv[1]; +X +X argc -= 2; argv += 2; +X +X if (strcmp(device, "reset") == 0) +X rv = cfg_reset(); +X else if (strcmp(device, "show") == 0) +X rv = cfg_show(); +X else if (strcmp(device, "dsp") == 0) +X rv = cfg_dsp(argc, argv); +X else if (strcmp(device, "mpu") == 0) +X rv = cfg_mpu(argc, argv); +X else if (strcmp(device, "ide") == 0) +X rv = cfg_ide(argc, argv); +X else if (strcmp(device, "joystick") == 0) +X rv = cfg_joystick(argc, argv); +X else { +X fprintf(stderr, "error: unknown device %s\n", device); +X usage(); +X } +X +X if (rv) +X fprintf(stderr, "error: device configuration failed\n"); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' && + chmod 0664 'MultiSound.d/pinnaclecfg.c' || + $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed' +366bdf27f0db767a3c7921d0a6db20fe MultiSound.d/pinnaclecfg.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`" + test 10235 -eq "$shar_count" || + $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10235,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/Makefile ============== +if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' && +CC = gcc +CFLAGS = -O +PROGS = setdigital msndreset pinnaclecfg conv +X +all: $(PROGS) +X +clean: +X rm -f $(PROGS) +SHAR_EOF + $shar_touch -am 1204092398 'MultiSound.d/Makefile' && + chmod 0664 'MultiSound.d/Makefile' || + $echo 'restore of' 'MultiSound.d/Makefile' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/Makefile:' 'MD5 check failed' +76ca8bb44e3882edcf79c97df6c81845 MultiSound.d/Makefile +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`" + test 106 -eq "$shar_count" || + $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/conv.l ============== +if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' && +%% +[ \n\t,\r] +\;.* +DB +[0-9A-Fa-f]+H { int n; sscanf(yytext, "%xH", &n); printf("%c", n); } +%% +int yywrap() { return 1; } +main() { yylex(); } +SHAR_EOF + $shar_touch -am 0828231798 'MultiSound.d/conv.l' && + chmod 0664 'MultiSound.d/conv.l' || + $echo 'restore of' 'MultiSound.d/conv.l' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/conv.l:' 'MD5 check failed' +d2411fc32cd71a00dcdc1f009e858dd2 MultiSound.d/conv.l +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`" + test 141 -eq "$shar_count" || + $echo 'MultiSound.d/conv.l:' 'original size' '141,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/msndreset.c ============== +if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' && +/********************************************************************* +X * +X * msndreset.c - resets the MultiSound card +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include +#include +#include +#include +#include +#include +#include +X +int main(int argc, char *argv[]) +{ +X int fd; +X +X if (argc != 2) { +X fprintf(stderr, "usage: msndreset \n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) { +X fprintf(stderr, "error: msnd ioctl reset failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' && + chmod 0664 'MultiSound.d/msndreset.c' || + $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed' +c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`" + test 1472 -eq "$shar_count" || + $echo 'MultiSound.d/msndreset.c:' 'original size' '1472,' 'current size' "$shar_count!" + fi +fi +rm -fr _sh01426 +exit 0 From 06501a6d2d2912b9e357702327c0249a5ecd7c9e Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Tue, 26 Jun 2018 06:24:38 -0300 Subject: [PATCH 119/529] ALSA:: multisound.sh: fix script to make it build with modern tools The script is old and produce some warnings and errors, because it lacks including stdlib.h and io.h is at sys/io.h. Fix it to run with the tools found on modern Linux distros. Tested building it on Fedora 28. Signed-off-by: Mauro Carvalho Chehab Signed-off-by: Takashi Iwai --- Documentation/sound/cards/multisound.sh | 30 +++++++++++++------------ 1 file changed, 16 insertions(+), 14 deletions(-) diff --git a/Documentation/sound/cards/multisound.sh b/Documentation/sound/cards/multisound.sh index 7a7a88256dfd..0e0cb029421b 100755 --- a/Documentation/sound/cards/multisound.sh +++ b/Documentation/sound/cards/multisound.sh @@ -381,11 +381,11 @@ # This shar contains: # length mode name # ------ ---------- ------------------------------------------ -# 2046 -rw-rw-r-- MultiSound.d/setdigital.c -# 10235 -rw-rw-r-- MultiSound.d/pinnaclecfg.c +# 2064 -rw-rw-r-- MultiSound.d/setdigital.c +# 10224 -rw-rw-r-- MultiSound.d/pinnaclecfg.c # 106 -rw-rw-r-- MultiSound.d/Makefile -# 141 -rw-rw-r-- MultiSound.d/conv.l -# 1472 -rw-rw-r-- MultiSound.d/msndreset.c +# 146 -rw-rw-r-- MultiSound.d/conv.l +# 1491 -rw-rw-r-- MultiSound.d/msndreset.c # save_IFS="${IFS}" IFS="${IFS}:" @@ -471,6 +471,7 @@ X * X ********************************************************************/ X #include +#include #include #include #include @@ -538,8 +539,8 @@ e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c SHAR_EOF else shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`" - test 2046 -eq "$shar_count" || - $echo 'MultiSound.d/setdigital.c:' 'original size' '2046,' 'current size' "$shar_count!" + test 2064 -eq "$shar_count" || + $echo 'MultiSound.d/setdigital.c:' 'original size' '2064,' 'current size' "$shar_count!" fi fi # ============= MultiSound.d/pinnaclecfg.c ============== @@ -581,8 +582,8 @@ X #include #include #include -#include #include +#include X #define IREG_LOGDEVICE 0x07 #define IREG_ACTIVATE 0x30 @@ -992,8 +993,8 @@ SHAR_EOF SHAR_EOF else shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`" - test 10235 -eq "$shar_count" || - $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10235,' 'current size' "$shar_count!" + test 10224 -eq "$shar_count" || + $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10224,' 'current size' "$shar_count!" fi fi # ============= MultiSound.d/Makefile ============== @@ -1039,7 +1040,7 @@ DB [0-9A-Fa-f]+H { int n; sscanf(yytext, "%xH", &n); printf("%c", n); } %% int yywrap() { return 1; } -main() { yylex(); } +void main() { yylex(); } SHAR_EOF $shar_touch -am 0828231798 'MultiSound.d/conv.l' && chmod 0664 'MultiSound.d/conv.l' || @@ -1052,8 +1053,8 @@ d2411fc32cd71a00dcdc1f009e858dd2 MultiSound.d/conv.l SHAR_EOF else shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`" - test 141 -eq "$shar_count" || - $echo 'MultiSound.d/conv.l:' 'original size' '141,' 'current size' "$shar_count!" + test 146 -eq "$shar_count" || + $echo 'MultiSound.d/conv.l:' 'original size' '146,' 'current size' "$shar_count!" fi fi # ============= MultiSound.d/msndreset.c ============== @@ -1085,6 +1086,7 @@ X * X ********************************************************************/ X #include +#include #include #include #include @@ -1129,8 +1131,8 @@ c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c SHAR_EOF else shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`" - test 1472 -eq "$shar_count" || - $echo 'MultiSound.d/msndreset.c:' 'original size' '1472,' 'current size' "$shar_count!" + test 1491 -eq "$shar_count" || + $echo 'MultiSound.d/msndreset.c:' 'original size' '1491,' 'current size' "$shar_count!" fi fi rm -fr _sh01426 From 513f930667beca7f67e3aa4558196c143864083f Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Tue, 26 Jun 2018 06:24:39 -0300 Subject: [PATCH 120/529] ALSA: multisound.sh: update module namespace The modules for the cards described here changed their names. Update accordingly. Signed-off-by: Mauro Carvalho Chehab Signed-off-by: Takashi Iwai --- Documentation/sound/cards/multisound.sh | 72 ++++++++++++------------- 1 file changed, 36 insertions(+), 36 deletions(-) diff --git a/Documentation/sound/cards/multisound.sh b/Documentation/sound/cards/multisound.sh index 0e0cb029421b..a915a1affcde 100755 --- a/Documentation/sound/cards/multisound.sh +++ b/Documentation/sound/cards/multisound.sh @@ -34,12 +34,12 @@ # composed of the following modules (these can also operate compiled # into the kernel): # -# msnd - MultiSound base (requires soundcore) +# snd-msnd-lib - MultiSound base (requires snd) # -# msnd_classic - Base audio/mixer support for Classic, Monetery and -# Tahiti cards +# snd-msnd-classic - Base audio/mixer support for Classic, Monetery and +# Tahiti cards # -# msnd_pinnacle - Base audio/mixer support for Pinnacle and Fiji cards +# snd-msnd-pinnacle - Base audio/mixer support for Pinnacle and Fiji cards # # # Important Notes - Read Before Using @@ -66,7 +66,7 @@ # # * Classic/Monterey/Tahiti # -# These cards are configured through the driver msnd_classic. You must +# These cards are configured through the driver snd-msnd-classic. You must # know the io port, then the driver will select the irq and memory resources # on the card. It is up to you to know if these are free locations or now, # a conflict can lock the machine up. @@ -81,10 +81,10 @@ # can be used to configure the card in non-PnP mode, and in PnP mode # you can use isapnptools. These are described briefly here. # -# pinnaclecfg is not required; you can use the msnd_pinnacle module +# pinnaclecfg is not required; you can use the snd-msnd-pinnacle module # to fully configure the card as well. However, pinnaclecfg can be # used to change the resource values of a particular device after the -# msnd_pinnacle module has been loaded. If you are compiling the +# snd-msnd-pinnacle module has been loaded. If you are compiling the # driver into the kernel, you must set these values during compile # time, however other peripheral resource values can be changed with # the pinnaclecfg program after the kernel is loaded. @@ -96,7 +96,7 @@ # to obtain one with the command `pnpdump 1 0x203' -- this may vary # for you (running pnpdump by itself did not work for me). Then, # edit this file and use isapnp to uncomment and set the card values. -# Use these values when inserting the msnd_pinnacle module. Using +# Use these values when inserting the snd-msnd-pinnacle module. Using # this method, you can set the resources for the DSP and the Kurzweil # synth (Pinnacle). Since Linux does not directly support PnP # devices, you may have difficulty when using the card in PnP mode @@ -133,16 +133,16 @@ # pinnaclecfg program. Using this program, you can assign the # resource values to the card's devices, or disable the devices. As # an alternative to using pinnaclecfg, you can specify many of the -# configuration values when loading the msnd_pinnacle module (or +# configuration values when loading the snd-msnd-pinnacle module (or # during kernel configuration when compiling the driver into the # kernel). # -# If you specify cfg=0x250 for the msnd_pinnacle module, it +# If you specify cfg=0x250 for the snd-msnd-pinnacle module, it # automatically configure the card to the given io, irq and memory # values using that config port (the config port is jumper selectable # on the card to 0x250, 0x260 or 0x270). # -# See the `msnd_pinnacle Additional Options' section below for more +# See the `snd-msnd-pinnacle Additional Options' section below for more # information on these parameters (also, if you compile the driver # directly into the kernel, these extra parameters can be useful # here). @@ -157,44 +157,44 @@ # # * MultiSound Classic/Monterey/Tahiti: # -# modprobe soundcore -# insmod msnd -# insmod msnd_classic io=0x290 irq=7 mem=0xd0000 +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-classic io=0x290 irq=7 mem=0xd0000 # # * MultiSound Pinnacle in PnP mode: # -# modprobe soundcore -# insmod msnd +# modprobe snd +# insmod snd-msnd-lib # isapnp mypinnacle.conf -# insmod msnd_pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values +# insmod snd-msnd-pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values # # * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port, # one of 0x250, 0x260 or 0x270): # -# insmod soundcore -# insmod msnd -# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 # # * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP # mode, add the following (assumes you did `isapnp mypinnacle.conf'): # -# insmod sound +# insmod snd # insmod mpu401 io=0x330 irq=9 <-- match mypinnacle.conf values # # * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP # mode, add the following. Note how we first configure the peripheral's # resources, _then_ install a Linux driver for it: # -# insmod sound +# insmod snd # pinnaclecfg 0x250 mpu 0x330 9 # insmod mpu401 io=0x330 irq=9 # # -- OR you can use the following sequence without pinnaclecfg in non-PnP mode: # -# insmod soundcore -# insmod msnd -# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9 -# insmod sound +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9 +# insmod snd # insmod mpu401 io=0x330 irq=9 # # * To setup the joystick port on the Pinnacle in non-PnP mode (though @@ -203,15 +203,15 @@ # # pinnaclecfg 0x250 joystick 0x200 # -# -- OR you can configure this using msnd_pinnacle with the following: +# -- OR you can configure this using snd-msnd-pinnacle with the following: # -# insmod soundcore -# insmod msnd -# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200 +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200 # # -# msnd_classic, msnd_pinnacle Required Options -# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# snd-msnd-classic, snd-msnd-pinnacle Required Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ # # If the following options are not given, the module will not load. # Examine the kernel message log for informative error messages. @@ -223,8 +223,8 @@ # mem Shared memory area, e.g. mem=0xd8000 # # -# msnd_classic, msnd_pinnacle Additional Options -# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# snd-msnd-classic, snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ # # fifosize The digital audio FIFOs, in kilobytes. If not # specified, the default will be used. Increasing @@ -244,8 +244,8 @@ # to zero). # # -# msnd_pinnacle Additional Options -# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ # # digital Specify digital=1 to enable the S/PDIF input # if you have the digital daughterboard From 0e6995e3b37bac69b4a815a116f9992d31a7bba0 Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Tue, 26 Jun 2018 06:24:40 -0300 Subject: [PATCH 121/529] ALSA: Fix references to Documentation/.*/MultiSound Now that the documentation/script file got restored, fix the references within the Kernel tree. Signed-off-by: Mauro Carvalho Chehab Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 2 +- sound/isa/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 4d83c1c0ca04..4a3cecc8ad38 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1568,7 +1568,7 @@ joystick_io The driver requires firmware files ``turtlebeach/msndinit.bin`` and ``turtlebeach/msndperm.bin`` in the proper firmware directory. -See Documentation/sound/oss/MultiSound for important information +See Documentation/sound/cards/multisound.sh for important information about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 43b35a873d78..d7db1eeebc84 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -459,7 +459,7 @@ config SND_MSND_CLASSIC Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or Monterey (not for the Pinnacle or Fiji). - See for important information + See for important information about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at . From aa3841b56b3b4f68d1e0a64941189ec932c4881b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jun 2018 07:42:40 +0200 Subject: [PATCH 122/529] ALSA: hda/realtek - Comprehensive model list for ALC662 & co ALC662 and others have far more fixup entries than the model table. Let's add more model string entries so that user can test / debug without compiling kernels at each time. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 36 +++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 18 +++++++++++++ 2 files changed, 54 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 7c2d37571af0..f2f8402ff899 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -87,8 +87,14 @@ alc700-ref ALC66x/67x/892 ============== +aspire + Subwoofer pin fixup for Aspire laptops +ideapad + Subwoofer pin fixup for Ideapad laptops mario Chromebook mario model fixup +hp-rp5800 + Headphone pin fixup for HP RP5800 asus-mode1 ASUS asus-mode2 @@ -105,10 +111,40 @@ asus-mode7 ASUS asus-mode8 ASUS +zotac-z68 + Front HP fixup for Zotac Z68 inv-dmic Inverted internal mic workaround +alc662-headset-multi + Dell headset jack, which can also be used as mic-in (ALC662) dell-headset-multi Headset jack, which can also be used as mic-in +alc662-headset + Headset mode support on ALC662 +alc668-headset + Headset mode support on ALC668 +bass16 + Bass speaker fixup on pin 0x16 +bass1a + Bass speaker fixup on pin 0x1a +automute + Auto-mute fixups for ALC668 +dell-xps13 + Dell XPS13 fixups +asus-nx50 + ASUS Nx50 fixups +asus-nx51 + ASUS Nx51 fixups +alc891-headset + Headset mode support on ALC891 +alc891-headset-multi + Dell headset jack, which can also be used as mic-in (ALC891) +acer-veriton + Acer Veriton speaker pin fixup +asrock-mobo + Fix invalid 0x15 / 0x16 pins +usi-headset + Headset support on USI machines dual-codecs Lenovo laptops with dual codecs diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c93e09f9c109..c7c9b47c2d5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8012,7 +8012,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { }; static const struct hda_model_fixup alc662_fixup_models[] = { + {.id = ALC662_FIXUP_ASPIRE, .name = "aspire"}, + {.id = ALC662_FIXUP_IDEAPAD, .name = "ideapad"}, {.id = ALC272_FIXUP_MARIO, .name = "mario"}, + {.id = ALC662_FIXUP_HP_RP5800, .name = "hp-rp5800"}, {.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"}, {.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"}, {.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"}, @@ -8021,8 +8024,23 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"}, {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, + {.id = ALC662_FIXUP_ZOTAC_Z68, .name = "zotac-z68"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC662_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc662-headset-multi"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC662_FIXUP_HEADSET_MODE, .name = "alc662-headset"}, + {.id = ALC668_FIXUP_HEADSET_MODE, .name = "alc668-headset"}, + {.id = ALC662_FIXUP_BASS_16, .name = "bass16"}, + {.id = ALC662_FIXUP_BASS_1A, .name = "bass1a"}, + {.id = ALC668_FIXUP_AUTO_MUTE, .name = "automute"}, + {.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"}, + {.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"}, + {.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"}, + {.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"}, + {.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"}, + {.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"}, + {.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"}, + {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {} }; From a26d96c7802e09473d5997791829774ba4f7e867 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jun 2018 15:09:25 +0200 Subject: [PATCH 123/529] ALSA: hda/realtek - Comprehensive model list for ALC259 & co Like the previous commit for ALC662, let's give more comprehensive list of model entries for ALC269 & co as well. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 156 ++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 82 +++++++++++++ 2 files changed, 238 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index f2f8402ff899..a403a7f34a78 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -70,6 +70,10 @@ dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O +dell-headset3 + Headset jack (without mic-in), and also dock I/O, variant 3 +dell-headset4 + Headset jack (without mic-in), and also dock I/O, variant 4 alc283-dac-wcaps Fixups for Chromebook with ALC283 alc283-sense-combo @@ -80,10 +84,162 @@ tpt440 Lenovo Thinkpad T440s setup tpt460 Lenovo Thinkpad T460/560 setup +tpt470-dock + Lenovo Thinkpad T470 dock setup dual-codecs Lenovo laptops with dual codecs alc700-ref Intel reference board with ALC700 codec +vaio + Pin fixups for Sony VAIO laptops +dell-m101z + COEF setup for Dell M101z +asus-g73jw + Subwoofer pin fixup for ASUS G73JW +lenovo-eapd + Inversed EAPD setup for Lenovo laptops +sony-hweq + H/W EQ COEF setup for Sony laptops +pcm44k + Fixed PCM 44kHz constraints (for buggy devices) +lifebook + Dock pin fixups for Fujitsu Lifebook +lifebook-extmic + Headset mic fixup for Fujitsu Lifebook +lifebook-hp-pin + Headphone pin fixup for Fujitsu Lifebook +lifebook-u7x7 + Lifebook U7x7 fixups +alc269vb-amic + ALC269VB analog mic pin fixups +alc269vb-dmic + ALC269VB digital mic pin fixups +hp-mute-led-mic1 + Mute LED via Mic1 pin on HP +hp-mute-led-mic2 + Mute LED via Mic2 pin on HP +hp-mute-led-mic3 + Mute LED via Mic3 pin on HP +hp-gpio-mic1 + GPIO + Mic1 pin LED on HP +hp-line1-mic1 + Mute LED via Line1 + Mic1 pins on HP +noshutup + Skip shutup callback +sony-nomic + Headset mic fixup for Sony laptops +aspire-headset-mic + Headset pin fixup for Acer Aspire +asus-x101 + ASUS X101 fixups +acer-ao7xx + Acer AO7xx fixups +acer-aspire-e1 + Acer Aspire E1 fixups +acer-ac700 + Acer AC700 fixups +limit-mic-boost + Limit internal mic boost on Lenovo machines +asus-zenbook + ASUS Zenbook fixups +asus-zenbook-ux31a + ASUS Zenbook UX31A fixups +ordissimo + Ordissimo EVE2 (or Malata PC-B1303) fixups +asus-tx300 + ASUS TX300 fixups +alc283-int-mic + ALC283 COEF setup for Lenovo machines +mono-speakers + Subwoofer and headset fixupes for Dell Inspiron +alc290-subwoofer + Subwoofer fixups for Dell Vostro +thinkpad + Binding with thinkpad_acpi driver for Lenovo machines +dmic-thinkpad + thinkpad_acpi binding + digital mic support +alc255-acer + ALC255 fixups on Acer machines +alc255-asus + ALC255 fixups on ASUS machines +alc255-dell1 + ALC255 fixups on Dell machines +alc255-dell2 + ALC255 fixups on Dell machines, variant 2 +alc293-dell1 + ALC293 fixups on Dell machines +alc283-headset + Headset pin fixups on ALC283 +aspire-v5 + Acer Aspire V5 fixups +hp-gpio4 + GPIO and Mic1 pin mute LED fixups for HP +hp-gpio-led + GPIO mute LEDs on HP +hp-gpio2-hotkey + GPIO mute LED with hot key handling on HP +hp-dock-pins + GPIO mute LEDs and dock support on HP +hp-dock-gpio-mic + GPIO, Mic mute LED and dock support on HP +hp-9480m + HP 9480m fixups +alc288-dell1 + ALC288 fixups on Dell machines +alc288-dell-xps13 + ALC288 fixups on Dell XPS13 +dell-e7x + Dell E7x fixups +alc293-dell + ALC293 fixups on Dell machines +alc298-dell1 + ALC298 fixups on Dell machines +alc298-dell-aio + ALC298 fixups on Dell AIO machines +alc275-dell-xps + ALC275 fixups on Dell XPS models +alc256-dell-xps13 + ALC256 fixups on Dell XPS13 +lenovo-spk-noise + Workaround for speaker noise on Lenovo machines +lenovo-hotkey + Hot-key support via Mic2 pin on Lenovo machines +dell-spk-noise + Workaround for speaker noise on Dell machines +alc255-dell1 + ALC255 fixups on Dell machines +alc295-disable-dac3 + Disable DAC3 routing on ALC295 +alc280-hp-headset + HP Elitebook fixups +alc221-hp-mic + Front mic pin fixup on HP machines +alc298-spk-volume + Speaker pin routing workaround on ALC298 +dell-inspiron-7559 + Dell Inspiron 7559 fixups +ativ-book + Samsung Ativ book 8 fixups +alc221-hp-mic + ALC221 headset fixups on HP machines +alc256-asus-mic + ALC256 fixups on ASUS machines +alc256-asus-aio + ALC256 fixups on ASUS AIO machines +alc233-eapd + ALC233 fixups on ASUS machines +alc294-lenovo-mic + ALC294 Mic pin fixup for Lenovo AIO machines +alc225-wyse + Dell Wyse fixups +alc274-dell-aio + ALC274 fixups on Dell AIO machines +alc255-dummy-lineout + Dell Precision 3930 fixups +alc255-dell-headset"}, + Dell Precision 3630 fixups +alc295-hp-x360 + HP Spectre X360 fixups ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7c9b47c2d5d..a0828befe62a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6620,13 +6620,95 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, .name = "dell-headset3"}, + {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, .name = "dell-headset4"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, + {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, + {.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"}, + {.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"}, + {.id = ALC269_FIXUP_ASUS_G73JW, .name = "asus-g73jw"}, + {.id = ALC269_FIXUP_LENOVO_EAPD, .name = "lenovo-eapd"}, + {.id = ALC275_FIXUP_SONY_HWEQ, .name = "sony-hweq"}, + {.id = ALC269_FIXUP_PCM_44K, .name = "pcm44k"}, + {.id = ALC269_FIXUP_LIFEBOOK, .name = "lifebook"}, + {.id = ALC269_FIXUP_LIFEBOOK_EXTMIC, .name = "lifebook-extmic"}, + {.id = ALC269_FIXUP_LIFEBOOK_HP_PIN, .name = "lifebook-hp-pin"}, + {.id = ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC, .name = "lifebook-u7x7"}, + {.id = ALC269VB_FIXUP_AMIC, .name = "alc269vb-amic"}, + {.id = ALC269VB_FIXUP_DMIC, .name = "alc269vb-dmic"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC1, .name = "hp-mute-led-mic1"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC2, .name = "hp-mute-led-mic2"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC3, .name = "hp-mute-led-mic3"}, + {.id = ALC269_FIXUP_HP_GPIO_MIC1_LED, .name = "hp-gpio-mic1"}, + {.id = ALC269_FIXUP_HP_LINE1_MIC1_LED, .name = "hp-line1-mic1"}, + {.id = ALC269_FIXUP_NO_SHUTUP, .name = "noshutup"}, + {.id = ALC286_FIXUP_SONY_MIC_NO_PRESENCE, .name = "sony-nomic"}, + {.id = ALC269_FIXUP_ASPIRE_HEADSET_MIC, .name = "aspire-headset-mic"}, + {.id = ALC269_FIXUP_ASUS_X101, .name = "asus-x101"}, + {.id = ALC271_FIXUP_HP_GATE_MIC_JACK, .name = "acer-ao7xx"}, + {.id = ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, .name = "acer-aspire-e1"}, + {.id = ALC269_FIXUP_ACER_AC700, .name = "acer-ac700"}, + {.id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST, .name = "limit-mic-boost"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK, .name = "asus-zenbook"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, .name = "asus-zenbook-ux31a"}, + {.id = ALC269VB_FIXUP_ORDISSIMO_EVE2, .name = "ordissimo"}, + {.id = ALC282_FIXUP_ASUS_TX300, .name = "asus-tx300"}, + {.id = ALC283_FIXUP_INT_MIC, .name = "alc283-int-mic"}, + {.id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK, .name = "mono-speakers"}, + {.id = ALC290_FIXUP_SUBWOOFER_HSJACK, .name = "alc290-subwoofer"}, + {.id = ALC269_FIXUP_THINKPAD_ACPI, .name = "thinkpad"}, + {.id = ALC269_FIXUP_DMIC_THINKPAD_ACPI, .name = "dmic-thinkpad"}, + {.id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, .name = "alc255-acer"}, + {.id = ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc255-asus"}, + {.id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "alc255-dell2"}, + {.id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc293-dell1"}, + {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, + {.id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, .name = "alc255-dell-mute"}, + {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, + {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, + {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"}, + {.id = ALC280_FIXUP_HP_DOCK_PINS, .name = "hp-dock-pins"}, + {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic"}, + {.id = ALC280_FIXUP_HP_9480M, .name = "hp-9480m"}, + {.id = ALC288_FIXUP_DELL_HEADSET_MODE, .name = "alc288-dell-headset"}, + {.id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc288-dell1"}, + {.id = ALC288_FIXUP_DELL_XPS_13, .name = "alc288-dell-xps13"}, + {.id = ALC292_FIXUP_DELL_E7X, .name = "dell-e7x"}, + {.id = ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, .name = "alc293-dell"}, + {.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"}, + {.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"}, + {.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"}, + {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"}, + {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, + {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, + {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, + {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, + {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, + {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, + {.id = ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, .name = "dell-inspiron-7559"}, + {.id = ALC269_FIXUP_ATIV_BOOK_8, .name = "ativ-book"}, + {.id = ALC221_FIXUP_HP_MIC_NO_PRESENCE, .name = "alc221-hp-mic"}, + {.id = ALC256_FIXUP_ASUS_HEADSET_MODE, .name = "alc256-asus-headset"}, + {.id = ALC256_FIXUP_ASUS_MIC, .name = "alc256-asus-mic"}, + {.id = ALC256_FIXUP_ASUS_AIO_GPIO2, .name = "alc256-asus-aio"}, + {.id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc233-asus"}, + {.id = ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, .name = "alc233-eapd"}, + {.id = ALC294_FIXUP_LENOVO_MIC_LOCATION, .name = "alc294-lenovo-mic"}, + {.id = ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, .name = "alc225-wyse"}, + {.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"}, + {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, + {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, + {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {} }; #define ALC225_STANDARD_PINS \ From 03bf11c934c3ced43aa9be3cfb93962ab15d737b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jun 2018 16:56:41 +0200 Subject: [PATCH 124/529] ALSA: hda/realtek - Comprehensive model list for ALC268 Add the missing entry for ALC268 model strings. Only "spdif" was missing, and that's it. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 2 ++ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index a403a7f34a78..323c25c53eaf 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -41,6 +41,8 @@ inv-dmic Inverted internal mic workaround hp-eapd Disable HP EAPD on NID 0x15 +spdif + Enable SPDIF output on NID 0x1e ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) =================================================================== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0828befe62a..74540e0fa19d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2688,6 +2688,7 @@ static const struct hda_fixup alc268_fixups[] = { static const struct hda_model_fixup alc268_fixup_models[] = { {.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC268_FIXUP_HP_EAPD, .name = "hp-eapd"}, + {.id = ALC268_FIXUP_SPDIF, .name = "spdif"}, {} }; From e43c44d62dbb4d8e2f217198142c97fce1a25ac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jun 2018 17:02:08 +0200 Subject: [PATCH 125/529] ALSA: hda/realtek - Comprehensive model list for ALC262 Added a few missing entries for ALC262 model strings. All about specific hardwares. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 16 ++++++++++++++++ sound/pci/hda/patch_realtek.c | 8 ++++++++ 2 files changed, 24 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 323c25c53eaf..560235519555 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -34,6 +34,22 @@ ALC262 ====== inv-dmic Inverted internal mic workaround +fsc-h270 + Fixups for Fujitsu-Siemens Celsius H270 +fsc-s7110 + Fixups for Fujitsu-Siemens Lifebook S7110 +hp-z200 + Fixups for HP Z200 +tyan + Fixups for Tyan Thunder n6650W +lenovo-3000 + Fixups for Lenovo 3000 +benq + Fixups for Benq ED8 +benq-t31 + Fixups for Benq T31 +bayleybay + Fixups for Intel BayleyBay ALC267/268 ========== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 74540e0fa19d..fc21e2851b01 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2557,6 +2557,14 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { static const struct hda_model_fixup alc262_fixup_models[] = { {.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC262_FIXUP_FSC_H270, .name = "fsc-h270"}, + {.id = ALC262_FIXUP_FSC_S7110, .name = "fsc-s7110"}, + {.id = ALC262_FIXUP_HP_Z200, .name = "hp-z200"}, + {.id = ALC262_FIXUP_TYAN, .name = "tyan"}, + {.id = ALC262_FIXUP_LENOVO_3000, .name = "lenovo-3000"}, + {.id = ALC262_FIXUP_BENQ, .name = "benq"}, + {.id = ALC262_FIXUP_BENQ_T31, .name = "benq-t31"}, + {.id = ALC262_FIXUP_INTEL_BAYLEYBAY, .name = "bayleybay"}, {} }; From 772c2917ff4e3b15c38f74e77062360e5ffd1308 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jun 2018 17:17:53 +0200 Subject: [PATCH 126/529] ALSA: hda/realtek - Comprehensive model list for ALC882 & co More comprehensive list of model strings for ALC882 & co. Also corrected the subsection in models.rst, too. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 54 ++++++++++++++++++++++++- sound/pci/hda/patch_realtek.c | 25 ++++++++++++ 2 files changed, 77 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 560235519555..e06238131f77 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -326,20 +326,70 @@ ALC680 ====== N/A -ALC88x/898/1150 -====================== +ALC88x/898/1150/1220 +==================== +abit-aw9d + Pin fixups for Abit AW9D-MAX +lenovo-y530 + Pin fixups for Lenovo Y530 +acer-aspire-7736 + Fixup for Acer Aspire 7736 +asus-w90v + Pin fixup for ASUS W90V +cd + Enable audio CD pin NID 0x1c +no-front-hp + Disable front HP pin NID 0x1b +vaio-tt + Pin fixup for VAIO TT +eee1601 + COEF setups for ASUS Eee 1601 +alc882-eapd + Change EAPD COEF mode on ALC882 +alc883-eapd + Change EAPD COEF mode on ALC883 +gpio1 + Enable GPIO1 +gpio2 + Enable GPIO2 +gpio3 + Enable GPIO3 +alc889-coef + Setup ALC889 COEF +asus-w2jc + Fixups for ASUS W2JC acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G acer-aspire-8930g Acer Aspire 8330G/6935G acer-aspire Acer Aspire others +macpro-gpio + GPIO setup for Mac Pro +dac-route + Workaround for DAC routing on Acer Aspire +mbp-vref + Vref setup for Macbook Pro +imac91-vref + Vref setup for iMac 9,1 +mba11-vref + Vref setup for MacBook Air 1,1 +mba21-vref + Vref setup for MacBook Air 2,1 +mp11-vref + Vref setup for Mac Pro 1,1 +mp41-vref + Vref setup for Mac Pro 4,1 inv-dmic Inverted internal mic workaround no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) +asus-bass + Bass speaker setup for ASUS ET2700 dual-codecs ALC1220 dual codecs for Gaming mobos +clevo-p950 + Fixups for Clevo P950 ALC861/660 ========== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc21e2851b01..003e2c90e792 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2374,12 +2374,37 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { }; static const struct hda_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ABIT_AW9D_MAX, .name = "abit-aw9d"}, + {.id = ALC882_FIXUP_LENOVO_Y530, .name = "lenovo-y530"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_7736, .name = "acer-aspire-7736"}, + {.id = ALC882_FIXUP_ASUS_W90V, .name = "asus-w90v"}, + {.id = ALC889_FIXUP_CD, .name = "cd"}, + {.id = ALC889_FIXUP_FRONT_HP_NO_PRESENCE, .name = "no-front-hp"}, + {.id = ALC889_FIXUP_VAIO_TT, .name = "vaio-tt"}, + {.id = ALC888_FIXUP_EEE1601, .name = "eee1601"}, + {.id = ALC882_FIXUP_EAPD, .name = "alc882-eapd"}, + {.id = ALC883_FIXUP_EAPD, .name = "alc883-eapd"}, + {.id = ALC882_FIXUP_GPIO1, .name = "gpio1"}, + {.id = ALC882_FIXUP_GPIO2, .name = "gpio2"}, + {.id = ALC882_FIXUP_GPIO3, .name = "gpio3"}, + {.id = ALC889_FIXUP_COEF, .name = "alc889-coef"}, + {.id = ALC882_FIXUP_ASUS_W2JC, .name = "asus-w2jc"}, {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {.id = ALC885_FIXUP_MACPRO_GPIO, .name = "macpro-gpio"}, + {.id = ALC889_FIXUP_DAC_ROUTE, .name = "dac-route"}, + {.id = ALC889_FIXUP_MBP_VREF, .name = "mbp-vref"}, + {.id = ALC889_FIXUP_IMAC91_VREF, .name = "imac91-vref"}, + {.id = ALC889_FIXUP_MBA11_VREF, .name = "mba11-vref"}, + {.id = ALC889_FIXUP_MBA21_VREF, .name = "mba21-vref"}, + {.id = ALC889_FIXUP_MP11_VREF, .name = "mp11-vref"}, + {.id = ALC889_FIXUP_MP41_VREF, .name = "mp41-vref"}, {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, + {.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"}, {.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"}, + {.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"}, {} }; From 401caff70cd3d3ef3c96e1a1cbf6f973c24ba899 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jun 2018 11:43:09 +0200 Subject: [PATCH 127/529] ALSA: hda - Kill snd_hda_codec_update_cache() snd_hda_codec_update_cache() used to serve for a slightly different purpose from snd_hdac_write_cache(), but now both of them became identical. Let's unify and replace with the latter one consistently. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/hda_codec.h | 3 --- sound/pci/hda/hda_generic.c | 8 ++++---- sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- 5 files changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d91c87e41756..f42af88bd2de 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3842,7 +3842,7 @@ EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); * This function is a helper to set a pin ctl value more safely. * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the * value in pin target array via snd_hda_codec_set_pin_target(), then - * actually writes the value via either snd_hda_codec_update_cache() or + * actually writes the value via either snd_hda_codec_write_cache() or * snd_hda_codec_write() depending on @cached flag. */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, @@ -3851,7 +3851,7 @@ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, val = snd_hda_correct_pin_ctl(codec, pin, val); snd_hda_codec_set_pin_target(codec, pin, val); if (cached) - return snd_hda_codec_update_cache(codec, pin, 0, + return snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); else return snd_hda_codec_write(codec, pin, 0, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 681c360f29f9..8a095c16ee65 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -381,9 +381,6 @@ snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, return snd_hdac_regmap_write(&codec->core, nid, verb, parm); } -#define snd_hda_codec_update_cache(codec, nid, flags, verb, parm) \ - snd_hda_codec_write_cache(codec, nid, flags, verb, parm) - /* the struct for codec->pin_configs */ struct hda_pincfg { hda_nid_t nid; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 942f96e184b6..579984ecdec3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -209,7 +209,7 @@ static void parse_user_hints(struct hda_codec *codec) */ #define update_pin_ctl(codec, pin, val) \ - snd_hda_codec_update_cache(codec, pin, 0, \ + snd_hda_codec_write_cache(codec, pin, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) /* restore the pinctl based on the cached value */ @@ -898,7 +898,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, hda_nid_t nid = path->path[i]; if (enable && path->multi[i]) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, path->idx[i]); if (has_amp_in(codec, path, i)) @@ -930,7 +930,7 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) return; if (codec->inv_eapd) enable = !enable; - snd_hda_codec_update_cache(codec, pin, 0, + snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); } @@ -5973,7 +5973,7 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec) hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, 0); } } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 757857313426..fd476fb40e1b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -148,7 +148,7 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) return; if (codec->inv_eapd) enabled = !enabled; - snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, + snd_hda_codec_write_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); } @@ -991,7 +991,7 @@ static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) if (spec->eapd_nid) ad_vmaster_eapd_hook(private_data, enabled); - snd_hda_codec_update_cache(codec, 0x01, 0, + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, enabled ? 0x00 : 0x02); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75ba66eb4ccd..8b48352689b4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -341,7 +341,7 @@ static void cxt_fixup_headset_mic(struct hda_codec *codec, * control. */ #define update_mic_pin(codec, nid, val) \ - snd_hda_codec_update_cache(codec, nid, 0, \ + snd_hda_codec_write_cache(codec, nid, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) static const struct hda_input_mux olpc_xo_dc_bias = { From fc7c460fbb4003bf3f5d2b435079c85888644663 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:26 +0200 Subject: [PATCH 128/529] ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_DEFAULT_QUIRKS define Almost all boards use the mclk and use the same jack-detect settings, add a BYT_RT5651_DEFAULT_QUIRKS define for this. This shaves of some lines and makes it easier to see which settings are unique to a certain model. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 987720e203f9..735b8312e275 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -77,6 +77,11 @@ enum { #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ + BYT_RT5651_JD1_1 | \ + BYT_RT5651_OVCD_TH_2000UA | \ + BYT_RT5651_OVCD_SF_0P75) + /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */ #define MAX_NO_PROPS 5 @@ -86,10 +91,7 @@ struct byt_rt5651_private { }; /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ -static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | +static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_HS_IN3_MAP; static void log_quirks(struct device *dev) @@ -379,10 +381,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, { @@ -392,10 +391,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_HS_IN3_MAP), }, { @@ -405,10 +401,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_1P0 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, {} From 10876d24eb40c6bfaa0aabd97e3e143258176c53 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:27 +0200 Subject: [PATCH 129/529] ASoC: Intel: bytcr_rt5651: Change default input map from in2 to in1 Further testing on all 6 model x86 tablets with a rt5651 which I have access to for testing has shown that their single (mono) microphone is connected to both IN1 *and* IN2. The previous default mapping of IN2 was based on testing on the same 6 tablets, where the internal mic works fine with a mapping of IN2. But it works fine too with a mapping of IN1. This commit changes the default input mapping to to use IN1 instead of IN2, to match the mapping used for the other mono devices in the DMI quirk table. So that we need less different mappings. The same change is made to the Chuwi Vi8 Plus quirks, which is one of the 6 models tested. This is a preparation patch for simplifying the maps in a follow-up commit. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 735b8312e275..910890de38b0 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -90,9 +90,9 @@ struct byt_rt5651_private { struct snd_soc_jack jack; }; -/* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ +/* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN2_HS_IN3_MAP; + BYT_RT5651_IN1_HS_IN3_MAP; static void log_quirks(struct device *dev) { @@ -392,7 +392,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN2_HS_IN3_MAP), + BYT_RT5651_IN1_HS_IN3_MAP), }, { /* VIOS LTH17 */ From 366780df3e2d40533cc95a4bf25ddd3b934b5fd3 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:28 +0200 Subject: [PATCH 130/529] ASoC: Intel: bytcr_rt5651: Fix IN1_IN2_MAP quirk not being logged Fix the quirk logging code not logging the IN1_IN2_MAP quirk. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 910890de38b0..7cc6e36b7c47 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -102,6 +102,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) dev_info(dev, "quirk IN2_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) + dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) From fcdf1391caa6f7f01de56eea63e070555771fac7 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:29 +0200 Subject: [PATCH 131/529] ASoC: Intel: bytcr_rt5651: Remove IN2 input mappings BYT_RT5651_IN2_MAP was introduced in commit 39712db878a4 ("SoC: intel: byt: Introduce new custom IN2 map"), uses in commit 2fe30129b0a6 ("ASoC: intel: byt: Enable IN2 map quirk for a KIANO laptop"), only to be replaced by a new BYT_RT5651_IN1_IN2_MAP quirk in commit ea261bd02a67 ("ASoC: intel: byt: Introduce new map for dual mics") quickly afterwards, because the KIANO laptop has 2 internal mics on IN1 and IN2 and the headset mic is not in IN1 where the BYT_RT5651_IN2_MAP maps it, but on IN3. Now that the KIANO quirk entry uses BYT_RT5651_IN1_IN2_MAP, there are no users of BYT_RT5651_IN2_MAP left. This makes sense since the headset mic seems to always be connected to IN3, so BYT_RT5651_IN2_MAP is not useful. To deal with BYT_RT5651_IN2_MAP wrongly mapping the headset mic to IN1, BYT_RT5651_IN2_HS_IN3_MAP was added in commit f026e0631780 ("ASoC: Intel: bytcr_rt5651: Add new IN2_HS_IN3 input map and a quirk using it"). This was based on the assumption then some devices have the internal mic connected to IN2 only. Further testing has shown that this is wrong and the internal mic is always connected to IN1 and sometimes to both IN1 and IN2. TL;DR: Both BYT_RT5651_IN2_MAP and BYT_RT5651_IN2_HS_IN3_MAP are based on on wrong assumptions from the past and are no longer useful now, so they can both be removed. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 30 ++------------------------- 1 file changed, 2 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 7cc6e36b7c47..b8ca7137d8b2 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -40,10 +40,8 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, - BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, BYT_RT5651_IN1_HS_IN3_MAP, - BYT_RT5651_IN2_HS_IN3_MAP, }; enum { @@ -100,14 +98,10 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) - dev_info(dev, "quirk IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) - dev_info(dev, "quirk IN2_HS_IN3_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -256,12 +250,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"IN2P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Headset Mic"}, - {"IN2P", NULL, "Internal Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, @@ -275,12 +263,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN2P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_ssp0_aif1_map[] = { {"ssp0 Tx", NULL, "modem_out"}, {"modem_in", NULL, "ssp0 Rx"}, @@ -462,10 +444,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; - case BYT_RT5651_IN2_MAP: - custom_map = byt_rt5651_intmic_in2_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); - break; case BYT_RT5651_IN1_IN2_MAP: custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); @@ -474,10 +452,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_hs_in3_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); break; - case BYT_RT5651_IN2_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in2_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -723,9 +697,9 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { const char * const intmic_name[] = - { "dmic", "in1", "in2", "in12", "in1", "in2" }; + { "dmic", "in1", "in12", "in1" }; const char * const hsmic_name[] = - { "in2", "in2", "in1", "in3", "in3", "in3" }; + { "in2", "in2", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; From de23147983013591bc4d6812ce441f351dec6b9d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:30 +0200 Subject: [PATCH 132/529] ASoC: Intel: bytcr_rt5651: Fix IN1 map headsetmic mapping The initial bytcr_rt5651 machine driver commit mapped IN2 as the headset mic. In retrospect this is not correct as all known boards have the headset mic on IN3. To workaround this special IN?_HS_IN3 mappings were added. This commit fixes the original IN1 mapping to correctly have the headset mic on IN3, moves all users of the IN1_HS_IN3 mapping over to the fixed IN1_MAP and drops the now no longer needed IN1_HS_IN3 mapping. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 29 +++++++-------------------- 1 file changed, 7 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b8ca7137d8b2..bf2adb36f455 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -41,7 +41,6 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, BYT_RT5651_IN1_IN2_MAP, - BYT_RT5651_IN1_HS_IN3_MAP, }; enum { @@ -90,7 +89,7 @@ struct byt_rt5651_private { /* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_HS_IN3_MAP; + BYT_RT5651_IN1_MAP; static void log_quirks(struct device *dev) { @@ -100,8 +99,6 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) - dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -247,7 +244,7 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN2P", NULL, "Headset Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { @@ -257,12 +254,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_ssp0_aif1_map[] = { {"ssp0 Tx", NULL, "modem_out"}, {"modem_in", NULL, "ssp0 Rx"}, @@ -348,7 +339,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, - .driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_IN1_MAP), }, { .callback = byt_rt5651_quirk_cb, @@ -357,7 +348,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), }, .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { .callback = byt_rt5651_quirk_cb, @@ -376,7 +367,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { /* VIOS LTH17 */ @@ -448,10 +439,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); break; - case BYT_RT5651_IN1_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in1_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -696,10 +683,8 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = - { "dmic", "in1", "in12", "in1" }; - const char * const hsmic_name[] = - { "in2", "in2", "in3", "in3" }; + const char * const intmic_name[] = { "dmic", "in1", "in12" }; + const char * const hsmic_name[] = { "in2", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; From 37c7401e8c1f583d197c096152fc87a58f460277 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:31 +0200 Subject: [PATCH 133/529] ASoC: Intel: bytcr_rt5651: Fix DMIC map headsetmic mapping The initial bytcr_rt5651 machine driver commit mapped IN2 as the headset mic. In retrospect this is not correct as all known boards have the headset mic on IN3. This commit fixes the original DMIC mapping to correctly have the headset mic on IN3. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index bf2adb36f455..042334d9be32 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -236,9 +236,9 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { - {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { @@ -684,7 +684,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { const char * const intmic_name[] = { "dmic", "in1", "in12" }; - const char * const hsmic_name[] = { "in2", "in3", "in3" }; + const char * const hsmic_name[] = { "in3", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; From 8e69cd640097fa7af53fb476dbd3597608f32b10 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:32 +0200 Subject: [PATCH 134/529] ASoC: Intel: bytcr_rt5651: Simplify card long-name Now that the headset-mic is always IN3 there is no reason to have the headset-mic mapping in the long-name. This commit simplifies the long name to "bytcr-rt5651--mic". We can safely do this without causing regressions (UCM profile not found due to the longname change) as the UCM profiles are not in upstream alsa-lib yet. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 042334d9be32..e778142b8a6e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -662,7 +662,7 @@ static struct snd_soc_card byt_rt5651_card = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic" */ static bool is_valleyview(void) { @@ -683,8 +683,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = { "dmic", "in1", "in12" }; - const char * const hsmic_name[] = { "in3", "in3", "in3" }; + const char * const mic_name[] = { "dmic", "in1", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; @@ -831,9 +830,8 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-intmic-%s-hsmic", - intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], - hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-mic", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); From 8f250e7009d71e6f3f3aeb95a540c36fc9c03398 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:33 +0200 Subject: [PATCH 135/529] ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_HP_LR_SWAPPED quirk One some models (Chuwi Vi8 Plus, Chuwi Hi8 Pro) the headphone output has left and right swapped. This can be fixed in with special mixer settings in the UCM profile, bit this requires these devices loading a different UCM profile. This commit adds a BYT_RT5651_HP_LR_SWAPPED quirk for this and postfixes the longname with "-hp-swapped" if set, so that a different UCM profile will be loaded. We can safely do this without causing regressions (UCM profile not found due to the longname change) as the UCM profiles are not in upstream alsa-lib yet. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e778142b8a6e..ffd62eb5c266 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -73,6 +73,7 @@ enum { #define BYT_RT5651_SSP2_AIF2 BIT(19) /* default is using AIF1 */ #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_HP_LR_SWAPPED BIT(22) #define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ BYT_RT5651_JD1_1 | \ @@ -359,6 +360,17 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, + { + /* Chuwi Hi8 Pro (CWI513) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), + }, { /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, @@ -367,7 +379,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP), + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), }, { /* VIOS LTH17 */ @@ -662,7 +675,7 @@ static struct snd_soc_card byt_rt5651_card = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ static bool is_valleyview(void) { @@ -687,6 +700,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; + const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -829,9 +843,14 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } + if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) + hp_swapped = "-hp-swapped"; + else + hp_swapped = ""; + snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-mic", - mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-mic%s", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); From 55d69c0309acea65fb3dd99a05a665b51630362d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:34 +0200 Subject: [PATCH 136/529] ASoC: Intel: bytcr_rt5651: Sort DMI table entries alphabetically As we get more entries in the DMI quirk table it is nice to have some sort of ordering in the table, sort it alphabetically. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 55 ++++++++++++++------------- 1 file changed, 29 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index ffd62eb5c266..ba2753e0e12a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -334,32 +334,6 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id) } static const struct dmi_system_id byt_rt5651_quirk_table[] = { - { - .callback = byt_rt5651_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), - }, - .driver_data = (void *)(BYT_RT5651_IN1_MAP), - }, - { - .callback = byt_rt5651_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "ADI"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), - }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_MAP), - }, - { - .callback = byt_rt5651_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), - DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), - }, - .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_IN2_MAP), - }, { /* Chuwi Hi8 Pro (CWI513) */ .callback = byt_rt5651_quirk_cb, @@ -382,6 +356,35 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_IN1_MAP | BYT_RT5651_HP_LR_SWAPPED), }, + { + /* KIANO SlimNote 14.2 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), + DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN1_IN2_MAP), + }, + { + /* Minnowboard Max B3 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_MAP), + }, + { + /* Minnowboard Turbot */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ADI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), + }, + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN1_MAP), + }, { /* VIOS LTH17 */ .callback = byt_rt5651_quirk_cb, From 0ed03e6dc288c50b2e7d71523b99df1afd81cea1 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 14:11:26 +0200 Subject: [PATCH 137/529] ASoC: simple-amplifier: remame dio2125 documentation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The dio2125 is simple enough that we can make it a generic component. Rename the the dio2125 documentation to simple-amplifier to prepare this change. Suggested-by: Nicolò Veronese Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/dioo,dio2125.txt | 12 ------------ .../devicetree/bindings/sound/simple-amplifier.txt | 12 ++++++++++++ 2 files changed, 12 insertions(+), 12 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/dioo,dio2125.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt diff --git a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt b/Documentation/devicetree/bindings/sound/dioo,dio2125.txt deleted file mode 100644 index 63dbfe0f11d0..000000000000 --- a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt +++ /dev/null @@ -1,12 +0,0 @@ -DIO2125 Audio Driver - -Required properties: -- compatible : "dioo,dio2125" -- enable-gpios : the gpio connected to the enable pin of the dio2125 - -Example: - -amp: analog-amplifier { - compatible = "dioo,dio2125"; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt new file mode 100644 index 000000000000..8647edae7af0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -0,0 +1,12 @@ +Simple Amplifier Audio Driver + +Required properties: +- compatible : "dioo,dio2125" or "simple-audio-amplifier" +- enable-gpios : the gpio connected to the enable pin of the simple amplifier + +Example: + +amp: analog-amplifier { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; +}; From 8d881bb6216d28896112bdc0b42f1f21eb6cd4ee Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 14:11:27 +0200 Subject: [PATCH 138/529] ASoC: simple-amplifier: rename dio2125 to simple-amplifer MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The dio2125 is simple enough that we can make it a generic component. Just rename and sed the dio2125 amplifier driver to simple_amplifier. Suggested-by: Nicolò Veronese Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 ++--- sound/soc/codecs/Makefile | 4 +- .../codecs/{dio2125.c => simple-amplifier.c} | 41 ++++++++++--------- 3 files changed, 28 insertions(+), 27 deletions(-) rename sound/soc/codecs/{dio2125.c => simple-amplifier.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f6b8d4bf8796..53c2d726bedf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,7 +74,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DIO2125 select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER @@ -144,6 +143,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIMPLE_AMPLIFIER select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF select SND_SOC_SSM2305 @@ -573,10 +573,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DIO2125 - tristate "Dioo DIO2125 Amplifier" - select GPIOLIB - config SND_SOC_DMIC tristate @@ -897,6 +893,10 @@ config SND_SOC_SIGMADSP_REGMAP tristate select SND_SOC_SIGMADSP +config SND_SOC_SIMPLE_AMPLIFIER + tristate "Simple Audio Amplifier" + select GPIOLIB + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e43d99a039d3..f26ded89a1e5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -250,9 +250,9 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp -snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o @@ -509,7 +509,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp -obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o +obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c similarity index 69% rename from sound/soc/codecs/dio2125.c rename to sound/soc/codecs/simple-amplifier.c index 09451cd44f9b..6c27d4afaf3a 100644 --- a/sound/soc/codecs/dio2125.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -21,9 +21,9 @@ #include #include -#define DRV_NAME "dio2125" +#define DRV_NAME "simple-amplifier" -struct dio2125 { +struct simple_amp { struct gpio_desc *gpiod_enable; }; @@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); - struct dio2125 *priv = snd_soc_component_get_drvdata(c); + struct simple_amp *priv = snd_soc_component_get_drvdata(c); int val; switch (event) { @@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, @@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTR"), }; -static const struct snd_soc_dapm_route dio2125_dapm_routes[] = { +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -static const struct snd_soc_component_driver dio2125_component_driver = { - .dapm_widgets = dio2125_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets), - .dapm_routes = dio2125_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes), +static const struct snd_soc_component_driver simple_amp_component_driver = { + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), }; -static int dio2125_probe(struct platform_device *pdev) +static int simple_amp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct dio2125 *priv; + struct simple_amp *priv; int err; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -93,28 +93,29 @@ static int dio2125_probe(struct platform_device *pdev) return err; } - return devm_snd_soc_register_component(dev, &dio2125_component_driver, + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, NULL, 0); } #ifdef CONFIG_OF -static const struct of_device_id dio2125_ids[] = { +static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, { } }; -MODULE_DEVICE_TABLE(of, dio2125_ids); +MODULE_DEVICE_TABLE(of, simple_amp_ids); #endif -static struct platform_driver dio2125_driver = { +static struct platform_driver simple_amp_driver = { .driver = { .name = DRV_NAME, - .of_match_table = of_match_ptr(dio2125_ids), + .of_match_table = of_match_ptr(simple_amp_ids), }, - .probe = dio2125_probe, + .probe = simple_amp_probe, }; -module_platform_driver(dio2125_driver); +module_platform_driver(simple_amp_driver); -MODULE_DESCRIPTION("ASoC DIO2125 output driver"); +MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver"); MODULE_AUTHOR("Jerome Brunet "); MODULE_LICENSE("GPL"); From 8ed237e83ce9ff4b3964a6b096beb1cbd3397d5a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 14:11:28 +0200 Subject: [PATCH 139/529] ASoC: simple-amplifer: add simple-amplifier compatible MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add simple-audio-amplifier to the list of available compatible Suggested-by: Nicolò Veronese Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/simple-amplifier.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index 6c27d4afaf3a..85524acf3e9c 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -101,6 +101,7 @@ static int simple_amp_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, + { .compatible = "simple-audio-amplifier", }, { } }; MODULE_DEVICE_TABLE(of, simple_amp_ids); From 599eb9060c7ccaf6a8f0386ab89e3cb5c1f1fea4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 09:39:36 +0200 Subject: [PATCH 140/529] ASoC: tas571x: add tas5707 compatible Add the tas5707 to the available compatibles of the tas571x driver Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tas571x.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index b4959f10b74b..7c8fd37c2f9e 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -7,6 +7,7 @@ powerdown (optional). Required properties: - compatible: should be one of the following: + - "ti,tas5707" - "ti,tas5711", - "ti,tas5717", - "ti,tas5719", From f516d32262a4c0fef3fccdf2a82671f54f5c1e33 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 09:39:37 +0200 Subject: [PATCH 141/529] ASoC: tas517x: add tas5707 support Add support for the tas5707 audio power amplifier. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/tas571x.c | 110 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 16 ++++++ 3 files changed, 130 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 53c2d726bedf..6d1674699385 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -959,8 +959,11 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + tristate "Texas Instruments TAS571x power amplifiers" depends on I2C + help + Enable support for Texas Instruments TAS5707, TAS5711, TAS5717, + TAS5719 and TAS5721 power amplifiers config SND_SOC_TAS5720 tristate "Texas Instruments TAS5720 Mono Audio amplifier" diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 52f34c94ec25..ca2dfe12344e 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -7,6 +7,9 @@ * TAS5721 support: * Copyright (C) 2016 Petr Kulhavy, Barix AG * + * TAS5707 support: + * Copyright (C) 2018 Jerome Brunet, Baylibre SAS + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = { .vol_reg_size = 1, }; +static const struct regmap_range tas5707_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), + regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG), +}; + +static const struct regmap_access_table tas5707_volatile_regs = { + .yes_ranges = tas5707_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas5707_volatile_regs_range), + +}; + +static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1); + +static const char * const tas5707_volume_slew_step_txt[] = { + "256", "512", "1024", "2048", +}; + +static const unsigned int tas5707_volume_slew_step_values[] = { + 3, 0, 1, 2, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum, + TAS571X_VOL_CFG_REG, 0, 0x3, + tas5707_volume_slew_step_txt, + tas5707_volume_slew_step_values); + +static const struct snd_kcontrol_new tas5707_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), + + SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum), + + BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG), + BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG), + BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG), + BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG), + BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG), + BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG), + BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG), + + BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG), + BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG), + BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG), + BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG), + BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG), + BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG), + BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG), +}; + +static const struct reg_default tas5707_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x70}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5707_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5707_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5707_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas5707_volatile_regs, +}; + +static const struct tas571x_chip tas5707_chip = { + .supply_names = tas5711_supply_names, + .num_supply_names = ARRAY_SIZE(tas5711_supply_names), + .controls = tas5707_controls, + .num_controls = ARRAY_SIZE(tas5707_controls), + .regmap_config = &tas5707_regmap_config, + .vol_reg_size = 1, +}; + static const char *const tas5717_supply_names[] = { "AVDD", "DVDD", @@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client) } static const struct of_device_id tas571x_of_match[] = { + { .compatible = "ti,tas5707", .data = &tas5707_chip, }, { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, @@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = { MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { + { "tas5707", (kernel_ulong_t) &tas5707_chip }, { "tas5711", (kernel_ulong_t) &tas5711_chip }, { "tas5717", (kernel_ulong_t) &tas5717_chip }, { "tas5719", (kernel_ulong_t) &tas5717_chip }, diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index c45677bc26ad..bd23e89cfe79 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -53,6 +53,22 @@ #define TAS571X_PWM_MUX_REG 0x25 /* 20-byte biquad registers */ +#define TAS5707_CH1_BQ0_REG 0x29 +#define TAS5707_CH1_BQ1_REG 0x2a +#define TAS5707_CH1_BQ2_REG 0x2b +#define TAS5707_CH1_BQ3_REG 0x2c +#define TAS5707_CH1_BQ4_REG 0x2d +#define TAS5707_CH1_BQ5_REG 0x2e +#define TAS5707_CH1_BQ6_REG 0x2f + +#define TAS5707_CH2_BQ0_REG 0x30 +#define TAS5707_CH2_BQ1_REG 0x31 +#define TAS5707_CH2_BQ2_REG 0x32 +#define TAS5707_CH2_BQ3_REG 0x33 +#define TAS5707_CH2_BQ4_REG 0x34 +#define TAS5707_CH2_BQ5_REG 0x35 +#define TAS5707_CH2_BQ6_REG 0x36 + #define TAS5717_CH1_BQ0_REG 0x26 #define TAS5717_CH1_BQ1_REG 0x27 #define TAS5717_CH1_BQ2_REG 0x28 From c288248f5b26cd5563112fcdc077bf44964a942d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jun 2018 14:59:00 +0200 Subject: [PATCH 142/529] ALSA: intel_hdmi: Use strlcpy() instead of strncpy() hdmi_lpe_audio_probe() copies the pcm name string via strncpy(), but as a gcc8 warning suggests, it misses a NUL terminator, and unlikely the expected result. Use the proper one, strlcpy() instead. Signed-off-by: Takashi Iwai --- sound/x86/intel_hdmi_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 4ed9d0c41843..edc9f5a34eff 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1854,7 +1854,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* setup private data which can be retrieved when required */ pcm->private_data = ctx; pcm->info_flags = 0; - strncpy(pcm->name, card->shortname, strlen(card->shortname)); + strlcpy(pcm->name, card->shortname, strlen(card->shortname)); /* setup the ops for playabck */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops); From feb20faec73ba0b30f949d54c4153cf0ad3807c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jun 2018 09:03:51 +0200 Subject: [PATCH 143/529] ALSA: hda - Move in_pm accessors to HDA core The in_pm atomic in hdac_device is an important field used as a flag as well as a refcount for PM. The existing snd_hdac_power_up/down helpers already refer to it in the HD-audio core code, while the code to actually setting the value (atomic_inc() / _dec()) is open-coded in HDA legacy side, which is hard to find. This patch adds the helper functions to set/reset the in_pm counter to HDA core and use them in HDA legacy side, for making it clearer who / where the PM is managed. There is no functional changes, just code refactoring. Reviewed-by: Chris Wilson Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 27 +++++++++++++++++++++++++++ sound/pci/hda/hda_codec.c | 21 ++++++--------------- sound/pci/hda/patch_hdmi.c | 2 +- 3 files changed, 34 insertions(+), 16 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c052afc27547..294a5a21937b 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -8,6 +8,7 @@ #include #include +#include #include #include #include @@ -171,12 +172,38 @@ int snd_hdac_power_down(struct hdac_device *codec); int snd_hdac_power_up_pm(struct hdac_device *codec); int snd_hdac_power_down_pm(struct hdac_device *codec); int snd_hdac_keep_power_up(struct hdac_device *codec); + +/* call this at entering into suspend/resume callbacks in codec driver */ +static inline void snd_hdac_enter_pm(struct hdac_device *codec) +{ + atomic_inc(&codec->in_pm); +} + +/* call this at leaving from suspend/resume callbacks in codec driver */ +static inline void snd_hdac_leave_pm(struct hdac_device *codec) +{ + atomic_dec(&codec->in_pm); +} + +static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) +{ + return atomic_read(&codec->in_pm); +} + +static inline bool snd_hdac_is_power_on(struct hdac_device *codec) +{ + return !pm_runtime_suspended(&codec->dev); +} #else static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } +static inline void snd_hdac_enter_pm(struct hdac_device *codec) {} +static inline void snd_hdac_leave_pm(struct hdac_device *codec) {} +static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; } +static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; } #endif /* diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f42af88bd2de..decd46b51087 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -37,15 +37,8 @@ #include "hda_jack.h" #include -#ifdef CONFIG_PM -#define codec_in_pm(codec) atomic_read(&(codec)->core.in_pm) -#define hda_codec_is_power_on(codec) \ - (!pm_runtime_suspended(hda_codec_dev(codec))) -#else -#define codec_in_pm(codec) 0 -#define hda_codec_is_power_on(codec) 1 -#endif - +#define codec_in_pm(codec) snd_hdac_is_in_pm(&codec->core) +#define hda_codec_is_power_on(codec) snd_hdac_is_power_on(&codec->core) #define codec_has_epss(codec) \ ((codec)->core.power_caps & AC_PWRST_EPSS) #define codec_has_clkstop(codec) \ @@ -2846,14 +2839,13 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - atomic_inc(&codec->core.in_pm); - + snd_hdac_enter_pm(&codec->core); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); update_power_acct(codec, true); - atomic_dec(&codec->core.in_pm); + snd_hdac_leave_pm(&codec->core); return state; } @@ -2862,8 +2854,7 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - atomic_inc(&codec->core.in_pm); - + snd_hdac_enter_pm(&codec->core); if (codec->core.regmap) regcache_mark_dirty(codec->core.regmap); @@ -2886,7 +2877,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - atomic_dec(&codec->core.in_pm); + snd_hdac_leave_pm(&codec->core); } static int hda_codec_runtime_suspend(struct device *dev) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8840daf9c6a3..ed2318f79e3c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2489,7 +2489,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) return; /* ditto during suspend/resume process itself */ - if (atomic_read(&(codec)->core.in_pm)) + if (snd_hdac_is_in_pm(&codec->core)) return; snd_hdac_i915_set_bclk(&codec->bus->core); From 3787a39852b0d6a9e67336f8fb5815c13ab78bb6 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:49 -0500 Subject: [PATCH 144/529] ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead This patch removes the hdac_ext_device structure. The legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Follow-up patches in this series handle the bus and driver definitions. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 36 +--- sound/hda/ext/hdac_ext_bus.c | 25 +-- sound/soc/codecs/hdac_hdmi.c | 396 +++++++++++++++++------------------ 3 files changed, 204 insertions(+), 253 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 9c14e21dda85..c1a5ad0e6e39 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -181,48 +181,20 @@ struct hda_dai_map { u32 maxbps; }; -#define HDA_MAX_NIDS 16 - -/** - * struct hdac_ext_device - HDAC Ext device - * - * @hdac: hdac core device - * @nid_list - the dai map which matches the dai-name with the nid - * @map_cur_idx - the idx in use in dai_map - * @ops - the hda codec ops common to all codec drivers - * @pvt_data - private data, for asoc contains asoc codec object - */ -struct hdac_ext_device { - struct hdac_device hdev; - struct hdac_ext_bus *ebus; - - /* soc-dai to nid map */ - struct hda_dai_map nid_list[HDA_MAX_NIDS]; - unsigned int map_cur_idx; - - /* codec ops */ - struct hdac_ext_codec_ops ops; - - struct snd_card *card; - void *scodec; - void *private_data; -}; - struct hdac_ext_dma_params { u32 format; u8 stream_tag; }; -#define to_ehdac_device(dev) (container_of((dev), \ - struct hdac_ext_device, hdev)) + /* * HD-audio codec base driver */ struct hdac_ext_driver { struct hdac_driver hdac; - int (*probe)(struct hdac_ext_device *dev); - int (*remove)(struct hdac_ext_device *dev); - void (*shutdown)(struct hdac_ext_device *dev); + int (*probe)(struct hdac_device *dev); + int (*remove)(struct hdac_device *dev); + void (*shutdown)(struct hdac_device *dev); }; int snd_hda_ext_driver_register(struct hdac_ext_driver *drv); diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 0daf31383084..0e4823fdd411 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -137,17 +137,16 @@ static void default_release(struct device *dev) */ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) { - struct hdac_ext_device *edev; struct hdac_device *hdev = NULL; struct hdac_bus *bus = ebus_to_hbus(ebus); char name[15]; int ret; - edev = kzalloc(sizeof(*edev), GFP_KERNEL); - if (!edev) + hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + if (!hdev) return -ENOMEM; - hdev = &edev->hdev; - edev->ebus = ebus; + + hdev->bus = bus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); @@ -176,10 +175,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init); */ void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); - snd_hdac_device_exit(hdev); - kfree(edev); + kfree(hdev); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); @@ -212,27 +209,25 @@ static inline struct hdac_ext_driver *get_edrv(struct device *dev) return edrv; } -static inline struct hdac_ext_device *get_edev(struct device *dev) +static inline struct hdac_device *get_hdev(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); - struct hdac_ext_device *edev = to_ehdac_device(hdev); - - return edev; + return hdev; } static int hda_ext_drv_probe(struct device *dev) { - return (get_edrv(dev))->probe(get_edev(dev)); + return (get_edrv(dev))->probe(get_hdev(dev)); } static int hdac_ext_drv_remove(struct device *dev) { - return (get_edrv(dev))->remove(get_edev(dev)); + return (get_edrv(dev))->remove(get_hdev(dev)); } static void hdac_ext_drv_shutdown(struct device *dev) { - return (get_edrv(dev))->shutdown(get_edev(dev)); + return (get_edrv(dev))->shutdown(get_hdev(dev)); } /** diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..f1e235817a65 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -85,7 +85,7 @@ struct hdac_hdmi_pin { bool mst_capable; struct hdac_hdmi_port *ports; int num_ports; - struct hdac_ext_device *edev; + struct hdac_device *hdev; }; struct hdac_hdmi_port { @@ -126,6 +126,9 @@ struct hdac_hdmi_drv_data { }; struct hdac_hdmi_priv { + struct hdac_device *hdev; + struct snd_soc_component *component; + struct snd_card *card; struct hdac_hdmi_dai_port_map dai_map[HDA_MAX_CVTS]; struct list_head pin_list; struct list_head cvt_list; @@ -139,7 +142,7 @@ struct hdac_hdmi_priv { struct snd_soc_dai_driver *dai_drv; }; -#define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +#define hdev_to_hdmi_priv(_hdev) dev_get_drvdata(&(_hdev)->dev) static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, @@ -158,7 +161,7 @@ hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { - struct hdac_ext_device *edev = port->pin->edev; + struct hdac_device *hdev = port->pin->hdev; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -172,7 +175,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, * ports. */ if (pcm->jack_event == 0) { - dev_dbg(&edev->hdev.dev, + dev_dbg(&hdev->dev, "jack report for pcm=%d\n", pcm->pcm_id); snd_soc_jack_report(pcm->jack, SND_JACK_AVOUT, @@ -198,19 +201,18 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, /* * Get the no devices that can be connected to a port on the Pin widget. */ -static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_get_port_len(struct hdac_device *hdev, hda_nid_t nid) { unsigned int caps; unsigned int type, param; - caps = get_wcaps(&edev->hdev, nid); + caps = get_wcaps(hdev, nid); type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL) || (type != AC_WID_PIN)) return 0; - param = snd_hdac_read_parm_uncached(&edev->hdev, nid, - AC_PAR_DEVLIST_LEN); + param = snd_hdac_read_parm_uncached(hdev, nid, AC_PAR_DEVLIST_LEN); if (param == -1) return param; @@ -222,10 +224,10 @@ static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) * id selected on the pin. Return 0 means the first port entry * is selected or MST is not supported. */ -static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_get(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - return snd_hdac_codec_read(&edev->hdev, port->pin->nid, + return snd_hdac_codec_read(hdev, port->pin->nid, 0, AC_VERB_GET_DEVICE_SEL, 0); } @@ -233,7 +235,7 @@ static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, * Sets the selected port entry for the configuring Pin widget verb. * returns error if port set is not equal to port get otherwise success */ -static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_set(struct hdac_device *hdev, struct hdac_hdmi_port *port) { int num_ports; @@ -242,8 +244,7 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, return 0; /* AC_PAR_DEVLIST_LEN is 0 based. */ - num_ports = hdac_hdmi_get_port_len(edev, port->pin->nid); - + num_ports = hdac_hdmi_get_port_len(hdev, port->pin->nid); if (num_ports < 0) return -EIO; /* @@ -253,13 +254,13 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, if (num_ports + 1 < port->id) return 0; - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_DEVICE_SEL, port->id); - if (port->id != hdac_hdmi_port_select_get(edev, port)) + if (port->id != hdac_hdmi_port_select_get(hdev, port)) return -EIO; - dev_dbg(&edev->hdev.dev, "Selected the port=%d\n", port->id); + dev_dbg(&hdev->dev, "Selected the port=%d\n", port->id); return 0; } @@ -277,13 +278,6 @@ static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, return NULL; } -static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - return to_ehdac_device(hdev); -} - static unsigned int sad_format(const u8 *sad) { return ((sad[0] >> 0x3) & 0x1f); @@ -324,15 +318,13 @@ format_constraint: } static void -hdac_hdmi_set_dip_index(struct hdac_ext_device *edev, hda_nid_t pin_nid, +hdac_hdmi_set_dip_index(struct hdac_device *hdev, hda_nid_t pin_nid, int packet_index, int byte_index) { int val; val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hdac_codec_write(&edev->hdev, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_INDEX, val); + snd_hdac_codec_write(hdev, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); } struct dp_audio_infoframe { @@ -347,14 +339,14 @@ struct dp_audio_infoframe { u8 LFEPBL01_LSV36_DM_INH7; }; -static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, +static int hdac_hdmi_setup_audio_infoframe(struct hdac_device *hdev, struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port) { uint8_t buffer[HDMI_INFOFRAME_HEADER_SIZE + HDMI_AUDIO_INFOFRAME_SIZE]; struct hdmi_audio_infoframe frame; struct hdac_hdmi_pin *pin = port->pin; struct dp_audio_infoframe dp_ai; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt = pcm->cvt; u8 *dip; int ret; @@ -363,11 +355,11 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, u8 conn_type; int channels, ca; - ca = snd_hdac_channel_allocation(&edev->hdev, port->eld.info.spk_alloc, + ca = snd_hdac_channel_allocation(hdev, port->eld.info.spk_alloc, pcm->channels, pcm->chmap_set, true, pcm->chmap); channels = snd_hdac_get_active_channels(ca); - hdmi->chmap.ops.set_channel_count(&edev->hdev, cvt->nid, channels); + hdmi->chmap.ops.set_channel_count(hdev, cvt->nid, channels); snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, pcm->channels, pcm->chmap, pcm->chmap_set); @@ -400,32 +392,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, break; default: - dev_err(&edev->hdev.dev, "Invalid connection type: %d\n", - conn_type); + dev_err(&hdev->dev, "Invalid connection type: %d\n", conn_type); return -EIO; } /* stop infoframe transmission */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); /* Fill infoframe. Index auto-incremented */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { for (i = 0; i < sizeof(buffer); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, dip[i]); } /* Start infoframe */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); return 0; @@ -435,12 +426,12 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: strm_tag: %d\n", __func__, tx_mask); + dev_dbg(&hdev->dev, "%s: strm_tag: %d\n", __func__, tx_mask); dai_map = &hdmi->dai_map[dai->id]; @@ -455,8 +446,8 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; @@ -469,7 +460,7 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return -ENODEV; if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "device is not configured for this pin:port%d:%d\n", port->pin->nid, port->id); return -ENODEV; @@ -489,28 +480,28 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return 0; } -static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, +static int hdac_hdmi_query_port_connlist(struct hdac_device *hdev, struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - if (!(get_wcaps(&edev->hdev, pin->nid) & AC_WCAP_CONN_LIST)) { - dev_warn(&edev->hdev.dev, + if (!(get_wcaps(hdev, pin->nid) & AC_WCAP_CONN_LIST)) { + dev_warn(&hdev->dev, "HDMI: pin %d wcaps %#x does not support connection list\n", - pin->nid, get_wcaps(&edev->hdev, pin->nid)); + pin->nid, get_wcaps(hdev, pin->nid)); return -EINVAL; } - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; - port->num_mux_nids = snd_hdac_get_connections(&edev->hdev, pin->nid, + port->num_mux_nids = snd_hdac_get_connections(hdev, pin->nid, port->mux_nids, HDA_MAX_CONNECTIONS); if (port->num_mux_nids == 0) - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "No connections found for pin:port %d:%d\n", pin->nid, port->id); - dev_dbg(&edev->hdev.dev, "num_mux_nids %d for pin:port %d:%d\n", + dev_dbg(&hdev->dev, "num_mux_nids %d for pin:port %d:%d\n", port->num_mux_nids, pin->nid, port->id); return port->num_mux_nids; @@ -526,7 +517,7 @@ static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, * connected. */ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( - struct hdac_ext_device *edev, + struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { @@ -541,7 +532,7 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( list_for_each_entry(port, &pcm->port_list, head) { mutex_lock(&pcm->lock); - ret = hdac_hdmi_query_port_connlist(edev, + ret = hdac_hdmi_query_port_connlist(hdev, port->pin, port); mutex_unlock(&pcm->lock); if (ret < 0) @@ -568,8 +559,8 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; struct hdac_hdmi_port *port; @@ -578,7 +569,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, dai_map = &hdmi->dai_map[dai->id]; cvt = dai_map->cvt; - port = hdac_hdmi_get_port_from_cvt(edev, hdmi, cvt); + port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt); /* * To make PA and other userland happy. @@ -589,7 +580,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Failed: present?:%d ELD valid?:%d pin:port: %d:%d\n", port->eld.monitor_present, port->eld.eld_valid, port->pin->nid, port->id); @@ -611,8 +602,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; @@ -695,10 +685,10 @@ static void hdac_hdmi_fill_route(struct snd_soc_dapm_route *route, route->connected = handler; } -static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, +static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; struct hdac_hdmi_port *p; @@ -715,33 +705,32 @@ static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, return NULL; } -static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, +static void hdac_hdmi_set_power_state(struct hdac_device *hdev, hda_nid_t nid, unsigned int pwr_state) { int count; unsigned int state; - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) { - if (!snd_hdac_check_power_state(&edev->hdev, nid, pwr_state)) { + if (get_wcaps(hdev, nid) & AC_WCAP_POWER) { + if (!snd_hdac_check_power_state(hdev, nid, pwr_state)) { for (count = 0; count < 10; count++) { - snd_hdac_codec_read(&edev->hdev, nid, 0, + snd_hdac_codec_read(hdev, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); - state = snd_hdac_sync_power_state(&edev->hdev, + state = snd_hdac_sync_power_state(hdev, nid, pwr_state); if (!(state & AC_PWRST_ERROR)) break; } } - } } -static void hdac_hdmi_set_amp(struct hdac_ext_device *edev, +static void hdac_hdmi_set_amp(struct hdac_device *hdev, hda_nid_t nid, int val) { - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_OUT_AMP) - snd_hdac_codec_write(&edev->hdev, nid, 0, + if (get_wcaps(hdev, nid) & AC_WCAP_OUT_AMP) + snd_hdac_codec_write(hdev, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); } @@ -750,40 +739,40 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!pcm) return -EIO; /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D0); /* Enable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_UNMUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_UNMUTE); - return hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + return hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); case SND_SOC_DAPM_POST_PMD: - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_MUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_MUTE); /* Disable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D3); break; } @@ -795,11 +784,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_cvt *cvt = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); pcm = hdac_hdmi_get_pcm_from_cvt(hdmi, cvt); @@ -808,29 +797,29 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0); /* Enable transmission */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_1, 1); /* Category Code (CC) to zero */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, pcm->format); break; case SND_SOC_DAPM_POST_PMD: - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D3); break; } @@ -842,10 +831,10 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); int mux_idx; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); if (!kc) @@ -854,11 +843,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, mux_idx = dapm_kcontrol_get_value(kc); /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; if (mux_idx > 0) { - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_CONNECT_SEL, (mux_idx - 1)); } @@ -877,8 +866,8 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); struct snd_soc_dapm_context *dapm = w->dapm; struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; const char *cvt_name = e->texts[ucontrol->value.enumerated.item[0]]; @@ -931,12 +920,12 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, * care of selecting the right one and leaving all other inputs selected to * "NONE" */ -static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, +static int hdac_hdmi_create_pin_port_muxs(struct hdac_device *hdev, struct hdac_hdmi_port *port, struct snd_soc_dapm_widget *widget, const char *widget_name) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = port->pin; struct snd_kcontrol_new *kc; struct hdac_hdmi_cvt *cvt; @@ -948,17 +937,17 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, int i = 0; int num_items = hdmi->num_cvt + 1; - kc = devm_kzalloc(&edev->hdev.dev, sizeof(*kc), GFP_KERNEL); + kc = devm_kzalloc(&hdev->dev, sizeof(*kc), GFP_KERNEL); if (!kc) return -ENOMEM; - se = devm_kzalloc(&edev->hdev.dev, sizeof(*se), GFP_KERNEL); + se = devm_kzalloc(&hdev->dev, sizeof(*se), GFP_KERNEL); if (!se) return -ENOMEM; snprintf(kc_name, NAME_SIZE, "Pin %d port %d Input", pin->nid, port->id); - kc->name = devm_kstrdup(&edev->hdev.dev, kc_name, GFP_KERNEL); + kc->name = devm_kstrdup(&hdev->dev, kc_name, GFP_KERNEL); if (!kc->name) return -ENOMEM; @@ -976,35 +965,35 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, se->mask = roundup_pow_of_two(se->items) - 1; sprintf(mux_items, "NONE"); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; list_for_each_entry(cvt, &hdmi->cvt_list, head) { i++; sprintf(mux_items, "cvt %d", cvt->nid); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; } - se->texts = devm_kmemdup(&edev->hdev.dev, items, + se->texts = devm_kmemdup(&hdev->dev, items, (num_items * sizeof(char *)), GFP_KERNEL); if (!se->texts) return -ENOMEM; - return hdac_hdmi_fill_widget_info(&edev->hdev.dev, widget, + return hdac_hdmi_fill_widget_info(&hdev->dev, widget, snd_soc_dapm_mux, port, widget_name, NULL, kc, 1, hdac_hdmi_pin_mux_widget_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_REG); } /* Add cvt <- input <- mux route map */ -static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_ext_device *edev, +static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_device *hdev, struct snd_soc_dapm_widget *widgets, struct snd_soc_dapm_route *route, int rindex) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); const struct snd_kcontrol_new *kc; struct soc_enum *se; int mux_index = hdmi->num_cvt + hdmi->num_ports; @@ -1046,8 +1035,8 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct snd_soc_dai_driver *dai_drv = hdmi->dai_drv; char widget_name[NAME_SIZE]; struct hdac_hdmi_cvt *cvt; @@ -1099,7 +1088,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) for (j = 0; j < pin->num_ports; j++) { sprintf(widget_name, "Pin%d-Port%d Mux", pin->nid, pin->ports[j].id); - ret = hdac_hdmi_create_pin_port_muxs(edev, + ret = hdac_hdmi_create_pin_port_muxs(hdev, &pin->ports[j], &widgets[i], widget_name); if (ret < 0) @@ -1134,7 +1123,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } } - hdac_hdmi_add_pinmux_cvt_route(edev, widgets, route, i); + hdac_hdmi_add_pinmux_cvt_route(hdev, widgets, route, i); snd_soc_dapm_new_controls(dapm, widgets, ((2 * hdmi->num_ports) + hdmi->num_cvt)); @@ -1146,9 +1135,9 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } -static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) +static int hdac_hdmi_init_dai_map(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; int dai_id = 0; @@ -1164,7 +1153,7 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) dai_id++; if (dai_id == HDA_MAX_CVTS) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Max dais supported: %d\n", dai_id); break; } @@ -1173,9 +1162,9 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) return 0; } -static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt; char name[NAME_SIZE]; @@ -1190,10 +1179,10 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) list_add_tail(&cvt->head, &hdmi->cvt_list); hdmi->num_cvt++; - return hdac_hdmi_query_cvt_params(&edev->hdev, cvt); + return hdac_hdmi_query_cvt_params(hdev, cvt); } -static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_eld(struct hdac_device *hdev, struct hdac_hdmi_port *port) { unsigned int ver, mnl; @@ -1202,7 +1191,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, >> DRM_ELD_VER_SHIFT; if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) { - dev_err(&edev->hdev.dev, "HDMI: Unknown ELD version %d\n", ver); + dev_err(&hdev->dev, "HDMI: Unknown ELD version %d\n", ver); return -EINVAL; } @@ -1210,7 +1199,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT; if (mnl > ELD_MAX_MNL) { - dev_err(&edev->hdev.dev, "HDMI: MNL Invalid %d\n", mnl); + dev_err(&hdev->dev, "HDMI: MNL Invalid %d\n", mnl); return -EINVAL; } @@ -1222,8 +1211,8 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - struct hdac_ext_device *edev = pin->edev; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = pin->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; @@ -1241,14 +1230,14 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, if (pin->mst_capable) port_id = port->id; - size = snd_hdac_acomp_get_eld(&edev->hdev, pin->nid, port_id, + size = snd_hdac_acomp_get_eld(hdev, pin->nid, port_id, &port->eld.monitor_present, port->eld.eld_buffer, ELD_MAX_SIZE); if (size > 0) { size = min(size, ELD_MAX_SIZE); - if (hdac_hdmi_parse_eld(edev, port) < 0) + if (hdac_hdmi_parse_eld(hdev, port) < 0) size = -EINVAL; } @@ -1260,11 +1249,11 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { - dev_err(&edev->hdev.dev, "%s: disconnect for pin:port %d:%d\n", + dev_err(&hdev->dev, "%s: disconnect for pin:port %d:%d\n", __func__, pin->nid, port->id); /* @@ -1316,9 +1305,9 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, return 0; } -static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin; int ret; @@ -1328,7 +1317,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) pin->nid = nid; pin->mst_capable = false; - pin->edev = edev; + pin->hdev = hdev; ret = hdac_hdmi_add_ports(hdmi, pin); if (ret < 0) return ret; @@ -1459,15 +1448,14 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, * Parse all nodes and store the cvt/pin nids in array * Add one time initialization for pin and cvt widgets */ -static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, struct snd_soc_dai_driver **dais, int *num_dais) { hda_nid_t nid; int i, num_nodes; struct hdac_hdmi_cvt *temp_cvt, *cvt_next; struct hdac_hdmi_pin *temp_pin, *pin_next; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int ret; hdac_hdmi_skl_enable_all_pins(hdev); @@ -1492,13 +1480,13 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, switch (type) { case AC_WID_AUD_OUT: - ret = hdac_hdmi_add_cvt(edev, nid); + ret = hdac_hdmi_add_cvt(hdev, nid); if (ret < 0) goto free_widgets; break; case AC_WID_PIN: - ret = hdac_hdmi_add_pin(edev, nid); + ret = hdac_hdmi_add_pin(hdev, nid); if (ret < 0) goto free_widgets; break; @@ -1518,7 +1506,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, } *num_dais = hdmi->num_cvt; - ret = hdac_hdmi_init_dai_map(edev); + ret = hdac_hdmi_init_dai_map(hdev); if (ret < 0) goto free_widgets; @@ -1544,17 +1532,17 @@ free_widgets: static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { - struct hdac_ext_device *edev = aptr; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = NULL; struct hdac_hdmi_port *hport = NULL; - struct snd_soc_component *component = edev->scodec; + struct snd_soc_component *component = hdmi->component; int i; /* Don't know how this mapping is derived */ hda_nid_t pin_nid = port + 0x04; - dev_dbg(&edev->hdev.dev, "%s: for pin:%d port=%d\n", __func__, + dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); /* @@ -1567,7 +1555,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) SNDRV_CTL_POWER_D0) return; - if (atomic_read(&edev->hdev.in_pm)) + if (atomic_read(&hdev->in_pm)) return; list_for_each_entry(pin, &hdmi->pin_list, head) { @@ -1614,15 +1602,15 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, /* create jack pin kcontrols */ static int create_fill_jack_kcontrols(struct snd_soc_card *card, - struct hdac_ext_device *edev) + struct hdac_device *hdev) { struct hdac_hdmi_pin *pin; struct snd_kcontrol_new *kc; char kc_name[NAME_SIZE], xname[NAME_SIZE]; char *name; int i = 0, j; - struct snd_soc_component *component = edev->scodec; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + struct snd_soc_component *component = hdmi->component; kc = devm_kcalloc(component->dev, hdmi->num_ports, sizeof(*kc), GFP_KERNEL); @@ -1659,8 +1647,8 @@ static int create_fill_jack_kcontrols(struct snd_soc_card *card, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pin *pin; struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; @@ -1715,7 +1703,7 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, return ret; /* Add Jack Pin switch Kcontrol */ - ret = create_fill_jack_kcontrols(dapm->card, edev); + ret = create_fill_jack_kcontrols(dapm->card, hdev); if (ret < 0) return ret; @@ -1735,8 +1723,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, struct snd_soc_jack *jack) { struct snd_soc_component *component = dai->component; - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pcm *pcm; struct snd_pcm *snd_pcm; int err; @@ -1758,7 +1746,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, if (snd_pcm) { err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); if (err < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "chmap control add failed with err: %d for pcm: %d\n", err, device); kfree(pcm); @@ -1772,7 +1760,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); -static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, +static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { int i; @@ -1781,7 +1769,7 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, list_for_each_entry(pin, &hdmi->pin_list, head) { if (detect_pin_caps) { - if (hdac_hdmi_get_port_len(edev, pin->nid) == 0) + if (hdac_hdmi_get_port_len(hdev, pin->nid) == 0) pin->mst_capable = false; else pin->mst_capable = true; @@ -1798,68 +1786,68 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, static int hdmi_codec_probe(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct hdac_ext_link *hlink = NULL; int ret; - edev->scodec = component; + hdmi->component = component; /* * hold the ref while we probe, also no need to drop the ref on * exit, we call pm_runtime_suspend() so that will do for us */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hbus_to_ebus(hdev->bus), + dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hbus_to_ebus(hdev->bus), hlink); ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; - aops.audio_ptr = edev; + aops.audio_ptr = hdev; ret = snd_hdac_i915_register_notifier(&aops); if (ret < 0) { - dev_err(&edev->hdev.dev, "notifier register failed: err: %d\n", - ret); + dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret); return ret; } - hdac_hdmi_present_sense_all_pins(edev, hdmi, true); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, true); /* Imp: Store the card pointer in hda_codec */ - edev->card = dapm->card->snd_card; + hdmi->card = dapm->card->snd_card; /* * hdac_device core already sets the state to active and calls * get_noresume. So enable runtime and set the device to suspend. */ - pm_runtime_enable(&edev->hdev.dev); - pm_runtime_put(&edev->hdev.dev); - pm_runtime_suspend(&edev->hdev.dev); + pm_runtime_enable(&hdev->dev); + pm_runtime_put(&hdev->dev); + pm_runtime_suspend(&hdev->dev); return 0; } static void hdmi_codec_remove(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; - pm_runtime_disable(&edev->hdev.dev); + pm_runtime_disable(&hdev->dev); } #ifdef CONFIG_PM static int hdmi_codec_prepare(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); - pm_runtime_get_sync(&edev->hdev.dev); + pm_runtime_get_sync(&hdev->dev); /* * Power down afg. @@ -1876,16 +1864,15 @@ static int hdmi_codec_prepare(struct device *dev) static void hdmi_codec_complete(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* * As the ELD notify callback request is not entertained while the @@ -1893,9 +1880,9 @@ static void hdmi_codec_complete(struct device *dev) * all pins here. pin capablity change is not support, so use the * already set pin caps. */ - hdac_hdmi_present_sense_all_pins(edev, hdmi, false); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - pm_runtime_put_sync(&edev->hdev.dev); + pm_runtime_put_sync(&hdev->dev); } #else #define hdmi_codec_prepare NULL @@ -1922,7 +1909,6 @@ static void hdac_hdmi_get_chmap(struct hdac_device *hdev, int pcm_idx, static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, unsigned char *chmap, int prepared) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_port *port; @@ -1938,7 +1924,7 @@ static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, memcpy(pcm->chmap, chmap, ARRAY_SIZE(pcm->chmap)); list_for_each_entry(port, &pcm->port_list, head) if (prepared) - hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); mutex_unlock(&pcm->lock); } @@ -1987,10 +1973,9 @@ static struct hdac_hdmi_drv_data intel_drv_data = { .vendor_nid = INTEL_VENDOR_NID, }; -static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_device *hdev = &edev->hdev; - struct hdac_hdmi_priv *hdmi_priv; + struct hdac_hdmi_priv *hdmi_priv = NULL; struct snd_soc_dai_driver *hdmi_dais = NULL; struct hdac_ext_link *hlink = NULL; int num_dais = 0; @@ -1999,24 +1984,25 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); /* hold the ref while we probe */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hbus_to_ebus(hdev->bus), + dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hbus_to_ebus(hdev->bus), hlink); hdmi_priv = devm_kzalloc(&hdev->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; - edev->private_data = hdmi_priv; snd_hdac_register_chmap_ops(hdev, &hdmi_priv->chmap); hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; + hdmi_priv->hdev = hdev; if (!hdac_id) return -ENODEV; @@ -2027,7 +2013,7 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) else hdmi_priv->drv_data = &intel_drv_data; - dev_set_drvdata(&hdev->dev, edev); + dev_set_drvdata(&hdev->dev, hdmi_priv); INIT_LIST_HEAD(&hdmi_priv->pin_list); INIT_LIST_HEAD(&hdmi_priv->cvt_list); @@ -2038,15 +2024,15 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) * Turned off in the runtime_suspend during the first explicit * pm_runtime_suspend call. */ - ret = snd_hdac_display_power(edev->hdev.bus, true); + ret = snd_hdac_display_power(hdev->bus, true); if (ret < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "Cannot turn on display power on i915 err: %d\n", ret); return ret; } - ret = hdac_hdmi_parse_and_map_nid(edev, &hdmi_dais, &num_dais); + ret = hdac_hdmi_parse_and_map_nid(hdev, &hdmi_dais, &num_dais); if (ret < 0) { dev_err(&hdev->dev, "Failed in parse and map nid with err: %d\n", ret); @@ -2058,14 +2044,14 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec, hdmi_dais, num_dais); - snd_hdac_ext_bus_link_put(edev->ebus, hlink); + snd_hdac_ext_bus_link_put(hbus_to_ebus(hdev->bus), hlink); return ret; } -static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin, *pin_next; struct hdac_hdmi_cvt *cvt, *cvt_next; struct hdac_hdmi_pcm *pcm, *pcm_next; @@ -2105,8 +2091,7 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) #ifdef CONFIG_PM static int hdac_hdmi_runtime_suspend(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; @@ -2129,7 +2114,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) AC_PWRST_D3); err = snd_hdac_display_power(bus, false); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } @@ -2146,8 +2131,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) static int hdac_hdmi_runtime_resume(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; @@ -2169,12 +2153,12 @@ static int hdac_hdmi_runtime_resume(struct device *dev) err = snd_hdac_display_power(bus, true); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, From 76f56fae1cf9040325a58d1375291baf71dfaf03 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:50 -0500 Subject: [PATCH 145/529] ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead This patch removes the hdac_ext_bus structure. The legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Follow-up patches in this series handle the driver definition. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 15 ++ include/sound/hdaudio_ext.h | 74 ++++------ sound/hda/ext/hdac_ext_bus.c | 27 ++-- sound/hda/ext/hdac_ext_controller.c | 55 ++++---- sound/hda/ext/hdac_ext_stream.c | 104 ++++++-------- sound/soc/codecs/hdac_hdmi.c | 22 ++- sound/soc/intel/skylake/skl-messages.c | 50 +++---- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 112 ++++++++------- sound/soc/intel/skylake/skl-topology.c | 20 ++- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 184 +++++++++++-------------- sound/soc/intel/skylake/skl.h | 7 +- 13 files changed, 308 insertions(+), 376 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c052afc27547..9735b51aef08 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -250,6 +250,11 @@ struct hdac_rb { * @mlcap: MultiLink capabilities pointer * @gtscap: gts capabilities pointer * @drsmcap: dma resume capabilities pointer + * @num_streams: streams supported + * @idx: HDA link index + * @hlink_list: link list of HDA links + * @lock: lock for link mgmt + * @cmd_dma_state: state of cmd DMAs: CORB and RIRB */ struct hdac_bus { struct device *dev; @@ -317,6 +322,16 @@ struct hdac_bus { /* i915 component interface */ struct i915_audio_component *audio_component; int i915_power_refcount; + + /* parameters required for enhanced capabilities */ + int num_streams; + int idx; + + struct list_head hlink_list; + + struct mutex lock; + bool cmd_dma_state; + }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index c1a5ad0e6e39..e5b0cd1ade19 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -4,38 +4,14 @@ #include -/** - * hdac_ext_bus: HDAC extended bus for extended HDA caps - * - * @bus: hdac bus - * @num_streams: streams supported - * @hlink_list: link list of HDA links - * @lock: lock for link mgmt - * @cmd_dma_state: state of cmd DMAs: CORB and RIRB - */ -struct hdac_ext_bus { - struct hdac_bus bus; - int num_streams; - int idx; - - struct list_head hlink_list; - - struct mutex lock; - bool cmd_dma_state; -}; - -int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev, +int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, const struct hdac_io_ops *io_ops); -void snd_hdac_ext_bus_exit(struct hdac_ext_bus *sbus); -int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *sbus, int addr); +void snd_hdac_ext_bus_exit(struct hdac_bus *bus); +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr); void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); -void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); - -#define ebus_to_hbus(ebus) (&(ebus)->bus) -#define hbus_to_ebus(_bus) \ - container_of(_bus, struct hdac_ext_bus, bus) +void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \ { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ @@ -44,14 +20,14 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); #define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \ HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data) -void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable); -void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable); +void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *chip, bool enable); +void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *chip, bool enable); -void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *chip, +void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *chip, bool enable, int index); -int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *bus); -struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *bus, +int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus); +struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, const char *codec_name); enum hdac_ext_stream_type { @@ -100,28 +76,28 @@ struct hdac_ext_stream { #define stream_to_hdac_ext_stream(s) \ container_of(s, struct hdac_ext_stream, hstream) -void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus, +void snd_hdac_ext_stream_init(struct hdac_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag); -int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, +int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir); -void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus); -void snd_hdac_link_free_all(struct hdac_ext_bus *ebus); -struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus, +void snd_hdac_stream_free_all(struct hdac_bus *bus); +void snd_hdac_link_free_all(struct hdac_bus *bus); +struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type); void snd_hdac_ext_stream_release(struct hdac_ext_stream *azx_dev, int type); -void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *bus, +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *azx_dev, bool decouple); -void snd_hdac_ext_stop_streams(struct hdac_ext_bus *sbus); +void snd_hdac_ext_stop_streams(struct hdac_bus *bus); -int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value); -int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_ext_stream *stream); -void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index); -int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value); int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value); @@ -144,17 +120,15 @@ struct hdac_ext_link { int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link); int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link); -int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus); -int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus); +int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus); +int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus); void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, int stream); void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream); -int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, - struct hdac_ext_link *link); -int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, - struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link); /* update register macro */ #define snd_hdac_updatel(addr, reg, mask, val) \ diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 0e4823fdd411..77547ede9ae8 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -87,7 +87,7 @@ static const struct hdac_io_ops hdac_ext_default_io = { * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, +int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, const struct hdac_io_ops *io_ops) { @@ -98,15 +98,15 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, if (io_ops == NULL) io_ops = &hdac_ext_default_io; - ret = snd_hdac_bus_init(&ebus->bus, dev, ops, io_ops); + ret = snd_hdac_bus_init(bus, dev, ops, io_ops); if (ret < 0) return ret; - INIT_LIST_HEAD(&ebus->hlink_list); - ebus->idx = idx++; + INIT_LIST_HEAD(&bus->hlink_list); + bus->idx = idx++; - mutex_init(&ebus->lock); - ebus->cmd_dma_state = true; + mutex_init(&bus->lock); + bus->cmd_dma_state = true; return 0; } @@ -116,10 +116,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus * @ebus: the pointer to extended bus object */ -void snd_hdac_ext_bus_exit(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_exit(struct hdac_bus *bus) { - snd_hdac_bus_exit(&ebus->bus); - WARN_ON(!list_empty(&ebus->hlink_list)); + snd_hdac_bus_exit(bus); + WARN_ON(!list_empty(&bus->hlink_list)); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit); @@ -135,10 +135,9 @@ static void default_release(struct device *dev) * * Returns zero for success or a negative error code. */ -int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr) { struct hdac_device *hdev = NULL; - struct hdac_bus *bus = ebus_to_hbus(ebus); char name[15]; int ret; @@ -148,7 +147,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) hdev->bus = bus; - snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); + snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr); ret = snd_hdac_device_init(hdev, bus, name, addr); if (ret < 0) { @@ -185,14 +184,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); * * @ebus: HD-audio extended bus */ -void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus) { struct hdac_device *codec, *__codec; /* * we need to remove all the codec devices objects created in the * snd_hdac_ext_bus_device_init */ - list_for_each_entry_safe(codec, __codec, &ebus->bus.codec_list, list) { + list_for_each_entry_safe(codec, __codec, &bus->codec_list, list) { snd_hdac_device_unregister(codec); put_device(&codec->dev); } diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 84f3b8168716..72774119dd11 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -39,9 +39,8 @@ * @ebus: HD-audio extended core bus * @enable: flag to turn on/off the capability */ -void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL"); @@ -60,9 +59,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); * @ebus: HD-audio extended core bus * @enable: flag to enable/disable interrupt */ -void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL\n"); @@ -89,12 +87,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); * in hlink_list of extended hdac bus * Note: this will be freed on bus exit by driver */ -int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus) { int idx; u32 link_count; struct hdac_ext_link *hlink; - struct hdac_bus *bus = &ebus->bus; link_count = readl(bus->mlcap + AZX_REG_ML_MLCD) + 1; @@ -114,7 +111,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) /* since link in On, update the ref */ hlink->ref_count = 1; - list_add_tail(&hlink->list, &ebus->hlink_list); + list_add_tail(&hlink->list, &bus->hlink_list); } return 0; @@ -127,12 +124,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_get_ml_capabilities); * @ebus: HD-audio ext core bus */ -void snd_hdac_link_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_link_free_all(struct hdac_bus *bus) { struct hdac_ext_link *l; - while (!list_empty(&ebus->hlink_list)) { - l = list_first_entry(&ebus->hlink_list, struct hdac_ext_link, list); + while (!list_empty(&bus->hlink_list)) { + l = list_first_entry(&bus->hlink_list, struct hdac_ext_link, list); list_del(&l->list); kfree(l); } @@ -144,7 +141,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); * @ebus: HD-audio extended core bus * @codec_name: codec name */ -struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, +struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, const char *codec_name) { int i; @@ -153,10 +150,10 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, if (sscanf(codec_name, "ehdaudio%dD%d", &bus_idx, &addr) != 2) return NULL; - if (ebus->idx != bus_idx) + if (bus->idx != bus_idx) return NULL; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { if (hlink->lsdiid & (0x1 << addr)) return hlink; @@ -219,12 +216,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down); * snd_hdac_ext_bus_link_power_up_all -power up all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); @@ -240,12 +237,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up_all); * snd_hdac_ext_bus_link_power_down_all -power down all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) @@ -256,39 +253,39 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); -int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 0 to 1, count will be 1 so power up this link * as well, also check the dma status and trigger that */ if (++link->ref_count == 1) { - if (!ebus->cmd_dma_state) { - snd_hdac_bus_init_cmd_io(&ebus->bus); - ebus->cmd_dma_state = true; + if (!bus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(bus); + bus->cmd_dma_state = true; } ret = snd_hdac_ext_bus_link_power_up(link); } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); -int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; struct hdac_ext_link *hlink; bool link_up = false; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 1 to 0, count will be 0 @@ -301,7 +298,7 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, * now check if all links are off, if so turn off * cmd dma as well */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) { link_up = true; break; @@ -309,12 +306,12 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, } if (!link_up) { - snd_hdac_bus_stop_cmd_io(&ebus->bus); - ebus->cmd_dma_state = false; + snd_hdac_bus_stop_cmd_io(bus); + bus->cmd_dma_state = false; } } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c96d7a7a36af..1bd27576db98 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -25,7 +25,7 @@ /** * snd_hdac_ext_stream_init - initialize each stream (aka device) - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @idx: stream index number * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE) @@ -34,18 +34,16 @@ * initialize the stream, if ppcap is enabled then init those and then * invoke hdac stream initialization routine */ -void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_init(struct hdac_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag) { - struct hdac_bus *bus = &ebus->bus; - if (bus->ppcap) { stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + AZX_PPHC_INTERVAL * idx; stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE + - AZX_PPLC_MULTI * ebus->num_streams + + AZX_PPLC_MULTI * bus->num_streams + AZX_PPLC_INTERVAL * idx; } @@ -71,12 +69,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** * snd_hdac_ext_stream_init_all - create and initialize the stream objects * for an extended hda bus - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @start_idx: start index for streams * @num_stream: number of streams to initialize * @dir: direction of streams */ -int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, +int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir) { int stream_tag = 0; @@ -88,7 +86,7 @@ int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, if (!stream) return -ENOMEM; tag = ++stream_tag; - snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag); + snd_hdac_ext_stream_init(bus, stream, idx, dir, tag); idx++; } @@ -100,17 +98,16 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); /** * snd_hdac_stream_free_all - free hdac extended stream objects * - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_stream_free_all(struct hdac_bus *bus) { struct hdac_stream *s, *_s; struct hdac_ext_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); list_for_each_entry_safe(s, _s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); list_del(&s->list); kfree(stream); } @@ -119,15 +116,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); /** * snd_hdac_ext_stream_decouple - decouple the hdac stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @decouple: flag to decouple */ -void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *stream, bool decouple) { struct hdac_stream *hstream = &stream->hstream; - struct hdac_bus *bus = &ebus->bus; u32 val; int mask = AZX_PPCTL_PROCEN(hstream->index); @@ -251,19 +247,18 @@ void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); static struct hdac_ext_stream * -hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_link_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -277,34 +272,33 @@ hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, } if (!hstream->link_locked) { - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; } static struct hdac_ext_stream * -hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_host_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -313,17 +307,17 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, if (!stream->opened) { if (!hstream->decoupled) - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->hstream.opened = 1; res->hstream.running = 0; res->hstream.substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; @@ -331,7 +325,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, /** * snd_hdac_ext_stream_assign - assign a stream for the PCM - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @substream: PCM substream to assign * @type: type of stream (coupled, host or link stream) * @@ -346,27 +340,26 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, * the same stream object when it's used beforehand. when a stream is * decoupled, it becomes a host stream and link stream. */ -struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *ebus, +struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type) { struct hdac_ext_stream *hstream = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: - stream = snd_hdac_stream_assign(hbus, substream); + stream = snd_hdac_stream_assign(bus, substream); if (stream) hstream = container_of(stream, struct hdac_ext_stream, hstream); return hstream; case HDAC_EXT_STREAM_TYPE_HOST: - return hdac_ext_host_stream_assign(ebus, substream); + return hdac_ext_host_stream_assign(bus, substream); case HDAC_EXT_STREAM_TYPE_LINK: - return hdac_ext_link_stream_assign(ebus, substream); + return hdac_ext_link_stream_assign(bus, substream); default: return NULL; @@ -384,7 +377,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_assign); void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) { struct hdac_bus *bus = stream->hstream.bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: @@ -393,13 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) case HDAC_EXT_STREAM_TYPE_HOST: if (stream->decoupled && !stream->link_locked) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); snd_hdac_stream_release(&stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: if (stream->decoupled && !stream->hstream.opened) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); spin_lock_irq(&bus->reg_lock); stream->link_locked = 0; stream->link_substream = NULL; @@ -415,16 +407,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_release); /** * snd_hdac_ext_stream_spbcap_enable - enable SPIB for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable SPIB * @index: stream index for which SPIB need to be enabled */ -void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -446,14 +437,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable); /** * snd_hdac_ext_stream_set_spib - sets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: spib value to set */ -int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -468,15 +458,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_spib); /** * snd_hdac_ext_stream_get_spbmaxfifo - gets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * * Return maxfifo for the stream */ -int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_ext_stream *stream) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -490,11 +479,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo); /** * snd_hdac_ext_stop_streams - stop all stream if running - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_ext_stop_streams(struct hdac_ext_bus *ebus) +void snd_hdac_ext_stop_streams(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); struct hdac_stream *stream; if (bus->chip_init) { @@ -507,16 +495,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stop_streams); /** * snd_hdac_ext_stream_drsm_enable - enable DMA resume for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable DRSM * @index: stream index for which DRSM need to be enabled */ -void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -538,14 +525,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable); /** * snd_hdac_ext_stream_set_dpibr - sets the dpibr value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: dpib value to set */ -int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -560,7 +546,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr); /** * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: lpib value to set */ diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f1e235817a65..c3ccc8d9c91d 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1799,14 +1799,13 @@ static int hdmi_codec_probe(struct snd_soc_component *component) * hold the ref while we probe, also no need to drop the ref on * exit, we call pm_runtime_suspend() so that will do for us */ - hlink = snd_hdac_ext_bus_get_link(hbus_to_ebus(hdev->bus), - dev_name(&hdev->dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(hbus_to_ebus(hdev->bus), hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); ret = create_fill_widget_route_map(dapm); if (ret < 0) @@ -1984,14 +1983,13 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); /* hold the ref while we probe */ - hlink = snd_hdac_ext_bus_get_link(hbus_to_ebus(hdev->bus), - dev_name(&hdev->dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(hbus_to_ebus(hdev->bus), hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); hdmi_priv = devm_kzalloc(&hdev->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) @@ -2044,7 +2042,7 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec, hdmi_dais, num_dais); - snd_hdac_ext_bus_link_put(hbus_to_ebus(hdev->bus), hlink); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); return ret; } @@ -2093,7 +2091,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2118,13 +2115,13 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return err; } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_put(ebus, hlink); + snd_hdac_ext_bus_link_put(bus, hlink); return 0; } @@ -2133,7 +2130,6 @@ static int hdac_hdmi_runtime_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2143,13 +2139,13 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(ebus, hlink); + snd_hdac_ext_bus_link_get(bus, hlink); err = snd_hdac_display_power(bus, true); if (err < 0) { diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index d5f9c30eba32..8bfb8b0fa3d5 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -33,8 +33,7 @@ static int skl_alloc_dma_buf(struct device *dev, struct snd_dma_buffer *dmab, size_t size) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -44,8 +43,7 @@ static int skl_alloc_dma_buf(struct device *dev, static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -89,8 +87,7 @@ void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) static int skl_dsp_setup_spib(struct device *dev, unsigned int size, int stream_tag, int enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream = snd_hdac_get_stream(bus, SNDRV_PCM_STREAM_PLAYBACK, stream_tag); struct hdac_ext_stream *estream; @@ -100,10 +97,10 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, estream = stream_to_hdac_ext_stream(stream); /* enable/disable SPIB for this hdac stream */ - snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + snd_hdac_ext_stream_spbcap_enable(bus, enable, stream->index); /* set the spib value */ - snd_hdac_ext_stream_set_spib(ebus, estream, size); + snd_hdac_ext_stream_set_spib(bus, estream, size); return 0; } @@ -111,8 +108,7 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, static int skl_dsp_prepare(struct device *dev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_ext_stream *estream; struct hdac_stream *stream; struct snd_pcm_substream substream; @@ -124,7 +120,7 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, memset(&substream, 0, sizeof(substream)); substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - estream = snd_hdac_ext_stream_assign(ebus, &substream, + estream = snd_hdac_ext_stream_assign(bus, &substream, HDAC_EXT_STREAM_TYPE_HOST); if (!estream) return -ENODEV; @@ -143,9 +139,8 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -163,10 +158,9 @@ static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) static int skl_dsp_cleanup(struct device *dev, struct snd_dma_buffer *dmab, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; struct hdac_ext_stream *estream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -270,8 +264,7 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) int skl_init_dsp(struct skl *skl) { void __iomem *mmio_base; - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_dsp_loader_ops loader_ops; int irq = bus->irq; const struct skl_dsp_ops *ops; @@ -279,8 +272,8 @@ int skl_init_dsp(struct skl *skl) int ret; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* read the BAR of the ADSP MMIO */ mmio_base = pci_ioremap_bar(skl->pci, 4); @@ -335,12 +328,11 @@ unmap_mmio: int skl_free_dsp(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_sst *ctx = skl->skl_sst; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); ctx->dsp_ops->cleanup(bus->dev, ctx); @@ -383,10 +375,11 @@ int skl_suspend_late_dsp(struct skl *skl) int skl_suspend_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; ret = skl_dsp_sleep(ctx->dsp); @@ -394,8 +387,8 @@ int skl_suspend_dsp(struct skl *skl) return ret; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); + snd_hdac_ext_bus_ppcap_enable(bus, false); return 0; } @@ -403,15 +396,16 @@ int skl_suspend_dsp(struct skl *skl) int skl_resume_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* check if DSP 1st boot is done */ if (skl->skl_sst->is_first_boot == true) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index b9b140275be0..01a050cf8775 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -141,7 +141,7 @@ struct nhlt_specific_cfg { struct nhlt_fmt *fmt; struct nhlt_endpoint *epnt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; struct nhlt_acpi_table *nhlt = skl->nhlt; @@ -228,7 +228,7 @@ static void skl_nhlt_trim_space(char *trim) int skl_nhlt_update_topology_bin(struct skl *skl) { struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", @@ -248,8 +248,8 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; char platform_id[32]; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..d7fc3b2d3e68 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -67,16 +67,15 @@ struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream) return substream->runtime->private_data; } -static struct hdac_ext_bus *get_bus_ctx(struct snd_pcm_substream *substream) +static struct hdac_bus *get_bus_ctx(struct snd_pcm_substream *substream) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct hdac_stream *hstream = hdac_stream(stream); struct hdac_bus *bus = hstream->bus; - - return hbus_to_ebus(bus); + return bus; } -static int skl_substream_alloc_pages(struct hdac_ext_bus *ebus, +static int skl_substream_alloc_pages(struct hdac_bus *bus, struct snd_pcm_substream *substream, size_t size) { @@ -95,7 +94,7 @@ static int skl_substream_free_pages(struct hdac_bus *bus, return snd_pcm_lib_free_pages(substream); } -static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, +static void skl_set_pcm_constrains(struct hdac_bus *bus, struct snd_pcm_runtime *runtime) { snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -105,9 +104,9 @@ static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, 20, 178000000); } -static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *ebus) +static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_bus *bus) { - if ((ebus_to_hbus(ebus))->ppcap) + if (bus->ppcap) return HDAC_EXT_STREAM_TYPE_HOST; else return HDAC_EXT_STREAM_TYPE_COUPLED; @@ -123,9 +122,9 @@ static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *e static void skl_set_suspend_active(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, bool enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_dapm_widget *w; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -140,8 +139,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream, int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -153,7 +151,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->host_bps, 0); @@ -177,8 +175,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -190,7 +187,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->link_bps, 0); @@ -201,7 +198,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) snd_hdac_ext_link_stream_setup(stream, format_val); - list_for_each_entry(link, &ebus->hlink_list, list) { + list_for_each_entry(link, &bus->hlink_list, list) { if (link->index == params->link_index) snd_hdac_ext_link_set_stream_id(link, hstream->stream_tag); @@ -215,7 +212,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) static int skl_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream; struct snd_pcm_runtime *runtime = substream->runtime; struct skl_dma_params *dma_params; @@ -224,12 +221,12 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - stream = snd_hdac_ext_stream_assign(ebus, substream, - skl_get_host_stream_type(ebus)); + stream = snd_hdac_ext_stream_assign(bus, substream, + skl_get_host_stream_type(bus)); if (stream == NULL) return -EBUSY; - skl_set_pcm_constrains(ebus, runtime); + skl_set_pcm_constrains(bus, runtime); /* * disable WALLCLOCK timestamps for capture streams @@ -301,7 +298,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct skl_pipe_params p_params = {0}; @@ -309,7 +306,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, int ret, dma_id; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - ret = skl_substream_alloc_pages(ebus, substream, + ret = skl_substream_alloc_pages(bus, substream, params_buffer_bytes(params)); if (ret < 0) return ret; @@ -343,14 +340,14 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct skl_dma_params *dma_params = NULL; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(ebus)); + snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(bus)); dma_params = snd_soc_dai_get_dma_data(dai, substream); /* @@ -380,7 +377,7 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, static int skl_pcm_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct skl *skl = get_skl_ctx(dai->dev); struct skl_module_cfg *mconfig; @@ -400,7 +397,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream, snd_hdac_stream_cleanup(hdac_stream(stream)); hdac_stream(stream)->prepared = 0; - return skl_substream_free_pages(ebus_to_hbus(ebus), substream); + return skl_substream_free_pages(bus, substream); } static int skl_be_hw_params(struct snd_pcm_substream *substream, @@ -420,8 +417,7 @@ static int skl_be_hw_params(struct snd_pcm_substream *substream, static int skl_decoupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; int start; unsigned long cookie; @@ -470,7 +466,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl *skl = get_skl_ctx(dai->dev); struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_soc_dapm_widget *w; int ret; @@ -492,9 +488,9 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, * dpib & lpib position to resume before starting the * DMA */ - snd_hdac_ext_stream_drsm_enable(ebus, true, + snd_hdac_ext_stream_drsm_enable(bus, true, hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, + snd_hdac_ext_stream_set_dpibr(bus, stream, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } @@ -528,14 +524,14 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, ret = skl_decoupled_trigger(substream, cmd); if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ - stream->dpib = readl(ebus->bus.remap_addr + + stream->dpib = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(stream)->index)); stream->lpib = snd_hdac_stream_get_pos_lpib( hdac_stream(stream)); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } break; @@ -546,11 +542,12 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } + static int skl_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_soc_dai *codec_dai = rtd->codec_dai; @@ -558,14 +555,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_link *link; int stream_tag; - link_dev = snd_hdac_ext_stream_assign(ebus, substream, + link_dev = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_LINK); if (!link_dev) return -EBUSY; snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); if (!link) return -EINVAL; @@ -610,7 +607,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); @@ -626,7 +623,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: snd_hdac_ext_link_stream_clear(link_dev); if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); break; default: @@ -638,7 +635,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, static int skl_link_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); @@ -648,7 +645,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); if (!link) return -EINVAL; @@ -1041,8 +1038,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream) static int skl_coupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; struct snd_pcm_substream *s; bool start; @@ -1115,9 +1111,9 @@ static int skl_coupled_trigger(struct snd_pcm_substream *substream, static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); - if (!(ebus_to_hbus(ebus))->ppcap) + if (!bus->ppcap) return skl_coupled_trigger(substream, cmd); return 0; @@ -1127,7 +1123,7 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); unsigned int pos; /* @@ -1152,12 +1148,12 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + + pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); } else { udelay(20); - readl(ebus->bus.remap_addr + + readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); @@ -1242,11 +1238,11 @@ static void skl_pcm_free(struct snd_pcm *pcm) static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_pcm *pcm = rtd->pcm; unsigned int size; int retval = 0; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { @@ -1356,19 +1352,19 @@ static int skl_populate_modules(struct skl *skl) static int skl_platform_soc_probe(struct snd_soc_component *component) { - struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(component->dev); + struct skl *skl = bus_to_skl(bus); const struct skl_dsp_ops *ops; int ret; pm_runtime_get_sync(component->dev); - if ((ebus_to_hbus(ebus))->ppcap) { + if (bus->ppcap) { skl->component = component; /* init debugfs */ skl->debugfs = skl_debugfs_init(skl); - ret = skl_tplg_init(component, ebus); + ret = skl_tplg_init(component, bus); if (ret < 0) { dev_err(component->dev, "Failed to init topology!\n"); return ret; @@ -1425,10 +1421,10 @@ static const struct snd_soc_component_driver skl_component = { int skl_platform_register(struct device *dev) { int ret; - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); struct snd_soc_dai_driver *dais; int num_dais = ARRAY_SIZE(skl_platform_dai); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); INIT_LIST_HEAD(&skl->ppl_list); INIT_LIST_HEAD(&skl->bind_list); @@ -1464,8 +1460,8 @@ err: int skl_platform_unregister(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); struct skl_module_deferred_bind *modules, *tmp; if (!list_empty(&skl->bind_list)) { diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index fcdc716754b6..abfdb67c05cc 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -934,7 +934,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl, struct soc_bytes_ext *sb = (void *) k->private_value; struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private; struct skl_kpb_params *uuid_params, *params; - struct hdac_bus *bus = ebus_to_hbus(skl_to_ebus(skl)); + struct hdac_bus *bus = skl_to_bus(skl); int i, size, module_id; if (bc->set_params == SKL_PARAM_BIND && bc->max) { @@ -3029,9 +3029,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_dapm_widget *tplg_w) { int ret; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; if (!tplg_w->priv.size) @@ -3137,8 +3136,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, struct soc_bytes_ext *sb; struct snd_soc_tplg_bytes_control *tplg_bc; struct snd_soc_tplg_enum_control *tplg_ec; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); struct soc_enum *se; switch (hdr->ops.info) { @@ -3622,9 +3620,8 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); /* proceed only if we have private data defined */ if (manifest->priv.size == 0) @@ -3713,12 +3710,11 @@ static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe) /* * SKL topology init routine */ -int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus) +int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) { int ret; const struct firmware *fw; - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_pipeline *ppl; ret = request_firmware(&fw, skl->tplg_name, bus->dev); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..daeb6d2bb7fc 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -458,9 +458,9 @@ enum skl_channel { static inline struct skl *get_skl_ctx(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); - return ebus_to_skl(ebus); + return bus_to_skl(bus); } int skl_tplg_be_update_params(struct snd_soc_dai *dai, @@ -470,7 +470,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps, void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, struct skl_pipe_params *params, int stream); int skl_tplg_init(struct snd_soc_component *component, - struct hdac_ext_bus *ebus); + struct hdac_bus *ebus); struct skl_module_cfg *skl_tplg_fe_get_cpr_module( struct snd_soc_dai *dai, int stream); int skl_tplg_update_pipe_params(struct device *dev, diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f0d9793f872a..9c5a701d68ac 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -54,7 +54,7 @@ static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg, static void skl_init_pci(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = skl_to_bus(skl); /* * Clear bits 0-2 of PCI register TCSEL (at offset 0x44) @@ -63,7 +63,7 @@ static void skl_init_pci(struct skl *skl) * codecs. * The PCI register TCSEL is defined in the Intel manuals. */ - dev_dbg(ebus_to_hbus(ebus)->dev, "Clearing TCSEL\n"); + dev_dbg(bus->dev, "Clearing TCSEL\n"); skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); } @@ -103,8 +103,7 @@ static void skl_enable_miscbdcge(struct device *dev, bool enable) static void skl_clock_power_gating(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u32 val; /* Update PDCGE bit of CGCTL register */ @@ -127,7 +126,6 @@ static void skl_clock_power_gating(struct device *dev, bool enable) */ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink; int ret; @@ -135,7 +133,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) ret = snd_hdac_bus_init_chip(bus, full_reset); /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &ebus->hlink_list, list) + list_for_each_entry(hlink, &bus->hlink_list, list) bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); @@ -146,8 +144,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) void skl_update_d0i3c(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u8 reg; int timeout = 50; @@ -197,8 +194,7 @@ static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr) static irqreturn_t skl_interrupt(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; if (!pm_runtime_active(bus->dev)) @@ -227,8 +223,7 @@ static irqreturn_t skl_interrupt(int irq, void *dev_id) static irqreturn_t skl_threaded_handler(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; status = snd_hdac_chip_readl(bus, INTSTS); @@ -238,16 +233,15 @@ static irqreturn_t skl_threaded_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) +static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); int ret; ret = request_threaded_irq(skl->pci->irq, skl_interrupt, skl_threaded_handler, IRQF_SHARED, - KBUILD_MODNAME, ebus); + KBUILD_MODNAME, bus); if (ret) { dev_err(bus->dev, "unable to grab IRQ %d, disabling device\n", @@ -264,21 +258,20 @@ static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) static int skl_suspend_late(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); return skl_suspend_late_dsp(skl); } #ifdef CONFIG_PM -static int _skl_suspend(struct hdac_ext_bus *ebus) +static int _skl_suspend(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = to_pci_dev(bus->dev); int ret; - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); ret = skl_suspend_dsp(skl); if (ret < 0) @@ -295,10 +288,9 @@ static int _skl_suspend(struct hdac_ext_bus *ebus) return 0; } -static int _skl_resume(struct hdac_ext_bus *ebus) +static int _skl_resume(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl_init_pci(skl); skl_init_chip(bus, true); @@ -314,9 +306,8 @@ static int _skl_resume(struct hdac_ext_bus *ebus) static int skl_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); int ret = 0; /* @@ -325,15 +316,15 @@ static int skl_suspend(struct device *dev) */ if (skl->supend_active) { /* turn off the links and stop the CORB/RIRB DMA if it is On */ - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); - if (ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); enable_irq_wake(bus->irq); pci_save_state(pci); } else { - ret = _skl_suspend(ebus); + ret = _skl_suspend(bus); if (ret < 0) return ret; skl->skl_sst->fw_loaded = false; @@ -352,9 +343,8 @@ static int skl_suspend(struct device *dev) static int skl_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct hdac_ext_link *hlink = NULL; int ret; @@ -374,32 +364,32 @@ static int skl_resume(struct device *dev) */ if (skl->supend_active) { pci_restore_state(pci); - snd_hdac_ext_bus_link_power_up_all(ebus); + snd_hdac_ext_bus_link_power_up_all(bus); disable_irq_wake(bus->irq); /* * turn On the links which are On before active suspend * and start the CORB/RIRB DMA if On before * active suspend. */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) snd_hdac_ext_bus_link_power_up(hlink); } - if (ebus->cmd_dma_state) - snd_hdac_bus_init_cmd_io(&ebus->bus); ret = 0; + if (bus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(bus); } else { - ret = _skl_resume(ebus); + ret = _skl_resume(bus); /* turn off the links which are off before suspend */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (!hlink->ref_count) snd_hdac_ext_bus_link_power_down(hlink); } - if (!ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (!bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); } return ret; @@ -410,23 +400,21 @@ static int skl_resume(struct device *dev) static int skl_runtime_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_suspend(ebus); + return _skl_suspend(bus); } static int skl_runtime_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_resume(ebus); + return _skl_resume(bus); } #endif /* CONFIG_PM */ @@ -439,20 +427,19 @@ static const struct dev_pm_ops skl_pm = { /* * destructor */ -static int skl_free(struct hdac_ext_bus *ebus) +static int skl_free(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl->init_done = 0; /* to be sure */ - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); if (bus->irq >= 0) - free_irq(bus->irq, (void *)ebus); + free_irq(bus->irq, (void *)bus); snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(ebus); - snd_hdac_link_free_all(ebus); + snd_hdac_stream_free_all(bus); + snd_hdac_link_free_all(bus); if (bus->remap_addr) iounmap(bus->remap_addr); @@ -460,11 +447,11 @@ static int skl_free(struct hdac_ext_bus *ebus) pci_release_regions(skl->pci); pci_disable_device(skl->pci); - snd_hdac_ext_bus_exit(ebus); + snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_i915_exit(&ebus->bus); + snd_hdac_i915_exit(bus); return 0; } @@ -488,8 +475,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static int skl_find_machine(struct skl *skl, void *driver_data) { + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = driver_data; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); @@ -510,7 +497,7 @@ static int skl_find_machine(struct skl *skl, void *driver_data) static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; struct platform_device *pdev; int ret; @@ -544,7 +531,7 @@ static void skl_machine_device_unregister(struct skl *skl) static int skl_dmic_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct platform_device *pdev; int ret; @@ -643,9 +630,8 @@ static void skl_clock_device_unregister(struct skl *skl) /* * Probe the given codec address */ -static int probe_codec(struct hdac_ext_bus *ebus, int addr) +static int probe_codec(struct hdac_bus *bus, int addr) { - struct hdac_bus *bus = ebus_to_hbus(ebus); unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; @@ -658,13 +644,12 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK\n", addr); - return snd_hdac_ext_bus_device_init(ebus, addr); + return snd_hdac_ext_bus_device_init(bus, addr); } /* Codec initialization */ -static void skl_codec_create(struct hdac_ext_bus *ebus) +static void skl_codec_create(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); int c, max_slots; max_slots = HDA_MAX_CODECS; @@ -672,7 +657,7 @@ static void skl_codec_create(struct hdac_ext_bus *ebus) /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { if ((bus->codec_mask & (1 << c))) { - if (probe_codec(ebus, c) < 0) { + if (probe_codec(bus, c) < 0) { /* * Some BIOSen give you wrong codec addresses * that don't exist @@ -722,8 +707,7 @@ static int skl_i915_init(struct hdac_bus *bus) static void skl_probe_work(struct work_struct *work) { struct skl *skl = container_of(work, struct skl, probe_work); - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct hdac_ext_link *hlink = NULL; int err; @@ -744,7 +728,7 @@ static void skl_probe_work(struct work_struct *work) dev_info(bus->dev, "no hda codecs found!\n"); /* create codec instances */ - skl_codec_create(ebus); + skl_codec_create(bus); /* register platform dai and controls */ err = skl_platform_register(bus->dev); @@ -773,8 +757,8 @@ static void skl_probe_work(struct work_struct *work) /* * we are done probing so decrement link counts */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); @@ -796,7 +780,7 @@ static int skl_create(struct pci_dev *pci, struct skl **rskl) { struct skl *skl; - struct hdac_ext_bus *ebus; + struct hdac_bus *bus; int err; @@ -811,23 +795,22 @@ static int skl_create(struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - ebus = &skl->ebus; - snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); - ebus->bus.use_posbuf = 1; + + bus = skl_to_bus(skl); + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops); + bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - - ebus->bus.bdl_pos_adj = 0; + bus->bdl_pos_adj = 0; *rskl = skl; return 0; } -static int skl_first_init(struct hdac_ext_bus *ebus) +static int skl_first_init(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = skl->pci; int err; unsigned short gcap; @@ -848,7 +831,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) snd_hdac_bus_parse_capabilities(bus); - if (skl_acquire_irq(ebus, 0) < 0) + if (skl_acquire_irq(bus, 0) < 0) return -EBUSY; pci_set_master(pci); @@ -872,14 +855,14 @@ static int skl_first_init(struct hdac_ext_bus *ebus) if (!pb_streams && !cp_streams) return -EIO; - ebus->num_streams = cp_streams + pb_streams; + bus->num_streams = cp_streams + pb_streams; /* initialize streams */ snd_hdac_ext_stream_init_all - (ebus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); + (bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); start_idx = cp_streams; snd_hdac_ext_stream_init_all - (ebus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); + (bus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); err = snd_hdac_bus_alloc_stream_pages(bus); if (err < 0) @@ -895,7 +878,6 @@ static int skl_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct skl *skl; - struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; int err; @@ -904,10 +886,9 @@ static int skl_probe(struct pci_dev *pci, if (err < 0) return err; - ebus = &skl->ebus; - bus = ebus_to_hbus(ebus); + bus = skl_to_bus(skl); - err = skl_first_init(ebus); + err = skl_first_init(bus); if (err < 0) goto out_free; @@ -928,7 +909,7 @@ static int skl_probe(struct pci_dev *pci, skl_nhlt_update_topology_bin(skl); - pci_set_drvdata(skl->pci, ebus); + pci_set_drvdata(skl->pci, bus); skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); @@ -952,7 +933,7 @@ static int skl_probe(struct pci_dev *pci, skl->skl_sst->clock_power_gating = skl_clock_power_gating; } if (bus->mlcap) - snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_ext_bus_get_ml_capabilities(bus); snd_hdac_bus_stop_chip(bus); @@ -972,31 +953,30 @@ out_clk_free: out_nhlt_free: skl_nhlt_free(skl->nhlt); out_free: - skl_free(ebus); + skl_free(bus); return err; } static void skl_shutdown(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); struct hdac_stream *s; struct hdac_ext_stream *stream; struct skl *skl; - if (ebus == NULL) + if (!bus) return; - skl = ebus_to_skl(ebus); + skl = bus_to_skl(bus); if (!skl->init_done) return; - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); list_for_each_entry(s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } snd_hdac_bus_stop_chip(bus); @@ -1004,15 +984,15 @@ static void skl_shutdown(struct pci_dev *pci) static void skl_remove(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); release_firmware(skl->tplg); pm_runtime_get_noresume(&pci->dev); /* codec removal, invoke bus_device_remove */ - snd_hdac_ext_bus_device_remove(ebus); + snd_hdac_ext_bus_device_remove(bus); skl->debugfs = NULL; skl_platform_unregister(&pci->dev); @@ -1022,7 +1002,7 @@ static void skl_remove(struct pci_dev *pci) skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); skl_nhlt_free(skl->nhlt); - skl_free(ebus); + skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 0d5375cbcf6e..78aa8bdcb619 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -71,7 +71,7 @@ struct skl_fw_config { }; struct skl { - struct hdac_ext_bus ebus; + struct hdac_bus hbus; struct pci_dev *pci; unsigned int init_done:1; /* delayed init status */ @@ -105,9 +105,8 @@ struct skl { struct snd_soc_acpi_mach *mach; }; -#define skl_to_ebus(s) (&(s)->ebus) -#define ebus_to_skl(sbus) \ - container_of(sbus, struct skl, sbus) +#define skl_to_bus(s) (&(s)->hbus) +#define bus_to_skl(bus) container_of(bus, struct skl, hbus) /* to pass dai dma data */ struct skl_dma_params { From e1df9317cbb192582ed7aa88c5f294c2336a3c75 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:51 -0500 Subject: [PATCH 146/529] ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead This patch removes the hdac_ext_driver structure. The legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Signed-off-by: Rakesh Ughreja Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 5 +++++ include/sound/hdaudio_ext.h | 17 ++--------------- sound/hda/ext/hdac_ext_bus.c | 30 ++++++++++++++---------------- sound/soc/codecs/hdac_hdmi.c | 12 +++++------- 4 files changed, 26 insertions(+), 38 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 9735b51aef08..59ffe63cf194 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -188,6 +188,11 @@ struct hdac_driver { const struct hda_device_id *id_table; int (*match)(struct hdac_device *dev, struct hdac_driver *drv); void (*unsol_event)(struct hdac_device *dev, unsigned int event); + + /* fields used by ext bus APIs */ + int (*probe)(struct hdac_device *dev); + int (*remove)(struct hdac_device *dev); + void (*shutdown)(struct hdac_device *dev); }; #define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index e5b0cd1ade19..3c302477750b 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -160,20 +160,7 @@ struct hdac_ext_dma_params { u8 stream_tag; }; -/* - * HD-audio codec base driver - */ -struct hdac_ext_driver { - struct hdac_driver hdac; - - int (*probe)(struct hdac_device *dev); - int (*remove)(struct hdac_device *dev); - void (*shutdown)(struct hdac_device *dev); -}; - -int snd_hda_ext_driver_register(struct hdac_ext_driver *drv); -void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv); - -#define to_ehdac_driver(_drv) container_of(_drv, struct hdac_ext_driver, hdac) +int snd_hda_ext_driver_register(struct hdac_driver *drv); +void snd_hda_ext_driver_unregister(struct hdac_driver *drv); #endif /* __SOUND_HDAUDIO_EXT_H */ diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 77547ede9ae8..52f07766fff3 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -200,12 +200,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_remove); #define dev_to_hdac(dev) (container_of((dev), \ struct hdac_device, dev)) -static inline struct hdac_ext_driver *get_edrv(struct device *dev) +static inline struct hdac_driver *get_hdrv(struct device *dev) { struct hdac_driver *hdrv = drv_to_hdac_driver(dev->driver); - struct hdac_ext_driver *edrv = to_ehdac_driver(hdrv); - - return edrv; + return hdrv; } static inline struct hdac_device *get_hdev(struct device *dev) @@ -216,17 +214,17 @@ static inline struct hdac_device *get_hdev(struct device *dev) static int hda_ext_drv_probe(struct device *dev) { - return (get_edrv(dev))->probe(get_hdev(dev)); + return (get_hdrv(dev))->probe(get_hdev(dev)); } static int hdac_ext_drv_remove(struct device *dev) { - return (get_edrv(dev))->remove(get_hdev(dev)); + return (get_hdrv(dev))->remove(get_hdev(dev)); } static void hdac_ext_drv_shutdown(struct device *dev) { - return (get_edrv(dev))->shutdown(get_hdev(dev)); + return (get_hdrv(dev))->shutdown(get_hdev(dev)); } /** @@ -234,20 +232,20 @@ static void hdac_ext_drv_shutdown(struct device *dev) * * @drv: ext hda driver structure */ -int snd_hda_ext_driver_register(struct hdac_ext_driver *drv) +int snd_hda_ext_driver_register(struct hdac_driver *drv) { - drv->hdac.type = HDA_DEV_ASOC; - drv->hdac.driver.bus = &snd_hda_bus_type; + drv->type = HDA_DEV_ASOC; + drv->driver.bus = &snd_hda_bus_type; /* we use default match */ if (drv->probe) - drv->hdac.driver.probe = hda_ext_drv_probe; + drv->driver.probe = hda_ext_drv_probe; if (drv->remove) - drv->hdac.driver.remove = hdac_ext_drv_remove; + drv->driver.remove = hdac_ext_drv_remove; if (drv->shutdown) - drv->hdac.driver.shutdown = hdac_ext_drv_shutdown; + drv->driver.shutdown = hdac_ext_drv_shutdown; - return driver_register(&drv->hdac.driver); + return driver_register(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); @@ -256,8 +254,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); * * @drv: ext hda driver structure */ -void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv) +void snd_hda_ext_driver_unregister(struct hdac_driver *drv) { - driver_unregister(&drv->hdac.driver); + driver_unregister(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_unregister); diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index c3ccc8d9c91d..3e3a2a9ef310 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2186,14 +2186,12 @@ static const struct hda_device_id hdmi_list[] = { MODULE_DEVICE_TABLE(hdaudio, hdmi_list); -static struct hdac_ext_driver hdmi_driver = { - . hdac = { - .driver = { - .name = "HDMI HDA Codec", - .pm = &hdac_hdmi_pm, - }, - .id_table = hdmi_list, +static struct hdac_driver hdmi_driver = { + .driver = { + .name = "HDMI HDA Codec", + .pm = &hdac_hdmi_pm, }, + .id_table = hdmi_list, .probe = hdac_hdmi_dev_probe, .remove = hdac_hdmi_dev_remove, }; From f8a7fe1aea215e25eaf3bf04dff66fc7621ec9d7 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:54:00 -0500 Subject: [PATCH 147/529] ALSA: hdac: ext: add wait for codec to respond after link reset As per HDA spec section 4.3 - Codec Discovery, the software shall wait for atleast 521usec for codec to respond after link reset. With the multi-link capability each link is turned ON/OFF individually. Link controller drives reset signal when it is turned ON. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_controller.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 72774119dd11..5bc4a1d587d4 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -271,6 +271,15 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, } ret = snd_hdac_ext_bus_link_power_up(link); + + /* + * wait for 521usec for codec to report status + * HDA spec section 4.3 - Codec Discovery + */ + udelay(521); + bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); + snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); } mutex_unlock(&bus->lock); From 24494d3f939774c3c21d78b5e95d37f9e74d154c Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:56 -0500 Subject: [PATCH 148/529] ALSA: hda: split snd_hda_codec_new function Split snd_hda_codec_new into two separate functions. snd_hda_codec_device_init allocates memory and registers with bus. snd_hda_codec_device_new initialializes the fields and performs snd_device_new. This enables reuse of legacy HDA codec drivers as ASoC codec drivers. In addition mark some functions with EXPORT_SYMBOL_GPL so that it can be called by ASoC wrapper around the legacy HDA driver (hdac_hda). Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 68 +++++++++++++++++++++++++++++---------- sound/pci/hda/hda_codec.h | 2 ++ 2 files changed, 53 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d91c87e41756..059cfade05cc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -858,6 +858,39 @@ static void snd_hda_codec_dev_release(struct device *dev) kfree(codec); } +#define DEV_NAME_LEN 31 + +static int snd_hda_codec_device_init(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) +{ + char name[DEV_NAME_LEN]; + struct hda_codec *codec; + int err; + + dev_dbg(card->dev, "%s: entry\n", __func__); + + if (snd_BUG_ON(!bus)) + return -EINVAL; + if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) + return -EINVAL; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + sprintf(name, "hdaudioC%dD%d", card->number, codec_addr); + err = snd_hdac_device_init(&codec->core, &bus->core, name, codec_addr); + if (err < 0) { + kfree(codec); + return err; + } + + codec->core.type = HDA_DEV_LEGACY; + *codecp = codec; + + return err; +} + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -869,7 +902,19 @@ static void snd_hda_codec_dev_release(struct device *dev) int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp) { - struct hda_codec *codec; + int ret; + + ret = snd_hda_codec_device_init(bus, card, codec_addr, codecp); + if (ret < 0) + return ret; + + return snd_hda_codec_device_new(bus, card, codec_addr, *codecp); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_new); + +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec) +{ char component[31]; hda_nid_t fg; int err; @@ -879,25 +924,14 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, .dev_free = snd_hda_codec_dev_free, }; + dev_dbg(card->dev, "%s: entry\n", __func__); + if (snd_BUG_ON(!bus)) return -EINVAL; if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - sprintf(component, "hdaudioC%dD%d", card->number, codec_addr); - err = snd_hdac_device_init(&codec->core, &bus->core, component, - codec_addr); - if (err < 0) { - kfree(codec); - return err; - } - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.type = HDA_DEV_LEGACY; codec->core.exec_verb = codec_exec_verb; codec->bus = bus; @@ -957,15 +991,13 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (err < 0) goto error; - if (codecp) - *codecp = codec; return 0; error: put_device(hda_codec_dev(codec)); return err; } -EXPORT_SYMBOL_GPL(snd_hda_codec_new); +EXPORT_SYMBOL_GPL(snd_hda_codec_device_new); /** * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults @@ -2991,6 +3023,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) sync_power_up_states(codec); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_build_controls); /* * PCM stuff @@ -3196,6 +3229,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_parse_pcms); /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 681c360f29f9..8bbedf7f3f54 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -307,6 +307,8 @@ struct hda_codec { */ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); From 6298542fa33b6ba0e3effbace5b99b70b93ed9ae Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:57 -0500 Subject: [PATCH 149/529] ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init Remove memory allocation within snd_hdac_ext_bus_device_init, to make its behaviour identical to snd_hdac_bus_device_init. So that caller can allocate the parent data structure containing hdac_device. This API change helps in reusing the legacy HDA codec drivers with ASoC platform drivers. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 3 ++- sound/hda/ext/hdac_ext_bus.c | 8 ++------ sound/soc/intel/skylake/skl.c | 8 +++++++- 3 files changed, 11 insertions(+), 8 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 3c302477750b..c188b801239f 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -9,7 +9,8 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_io_ops *io_ops); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); -int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr); +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, + struct hdac_device *hdev); void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 52f07766fff3..1eb58244688e 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -135,16 +135,12 @@ static void default_release(struct device *dev) * * Returns zero for success or a negative error code. */ -int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr) +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, + struct hdac_device *hdev) { - struct hdac_device *hdev = NULL; char name[15]; int ret; - hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); - if (!hdev) - return -ENOMEM; - hdev->bus = bus; snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 9c5a701d68ac..3a7f5eb4902b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -635,6 +635,8 @@ static int probe_codec(struct hdac_bus *bus, int addr) unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; + struct skl *skl = bus_to_skl(bus); + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -644,7 +646,11 @@ static int probe_codec(struct hdac_bus *bus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK\n", addr); - return snd_hdac_ext_bus_device_init(bus, addr); + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); } /* Codec initialization */ From cb04ba33187ca571142b67c2fb60d0a8c24994c8 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Fri, 1 Jun 2018 22:53:58 -0500 Subject: [PATCH 150/529] ALSA: hdac: add extended ops in the hdac_bus Add extended ops in the hdac_bus to allow calling the ASoC HDAC library ops to reuse the legacy HDA codec drivers with ASoC framework. Extended ops are used by the legacy codec drivers to call into hdac_hda library, in the subsequent patches.. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 9 +++++++++ include/sound/hdaudio_ext.h | 3 ++- sound/hda/ext/hdac_ext_bus.c | 4 +++- sound/soc/intel/skylake/skl.c | 2 +- 4 files changed, 15 insertions(+), 3 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 59ffe63cf194..f1baaa88e766 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -213,6 +213,14 @@ struct hdac_bus_ops { int (*link_power)(struct hdac_bus *bus, bool enable); }; +/* + * ops used for ASoC HDA codec drivers + */ +struct hdac_ext_bus_ops { + int (*hdev_attach)(struct hdac_device *hdev); + int (*hdev_detach)(struct hdac_device *hdev); +}; + /* * Lowlevel I/O operators */ @@ -265,6 +273,7 @@ struct hdac_bus { struct device *dev; const struct hdac_bus_ops *ops; const struct hdac_io_ops *io_ops; + const struct hdac_ext_bus_ops *ext_ops; /* h/w resources */ unsigned long addr; diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index c188b801239f..f34aced69ca8 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -6,7 +6,8 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops); + const struct hdac_io_ops *io_ops, + const struct hdac_ext_bus_ops *ext_ops); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 1eb58244688e..9c37d9af3023 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -89,7 +89,8 @@ static const struct hdac_io_ops hdac_ext_default_io = { */ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops) + const struct hdac_io_ops *io_ops, + const struct hdac_ext_bus_ops *ext_ops) { int ret; static int idx; @@ -102,6 +103,7 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, if (ret < 0) return ret; + bus->ext_ops = ext_ops; INIT_LIST_HEAD(&bus->hlink_list); bus->idx = idx++; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3a7f5eb4902b..00e051467a40 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -803,7 +803,7 @@ static int skl_create(struct pci_dev *pci, } bus = skl_to_bus(skl); - snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops); + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); From 1adca4b0cd65c14cb8b8c9c257720385869c3d5f Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Thu, 28 Jun 2018 15:28:24 +0800 Subject: [PATCH 151/529] ALSA: hda: Add AZX_DCAPS_PM_RUNTIME for AMD Raven Ridge This patch can make audio controller in AMD Raven Ridge gets runtime suspended to D3, to save ~1W power when it's not in use. Cc: Vijendar Mukunda Signed-off-by: Kai-Heng Feng Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..a9b55d65f2bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2535,7 +2535,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, From de15d7ff5bef98746fcb76a0db7ac46de48d3560 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 12:07:25 +0200 Subject: [PATCH 152/529] ASoC: dpcm: improve runtime update predictability As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 181 +++++++++++++++++++++++--------------------- 1 file changed, 94 insertions(+), 87 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 19ebfc958b9d..63f96cde046a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2597,105 +2597,112 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) +{ + struct snd_soc_dapm_widget_list *list; + int count, paths; + + if (!fe->dai_link->dynamic) + return 0; + + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; + + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); + + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; + + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } + + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + +capture: + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; + + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } + + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); + else + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + } + + dpcm_path_put(&list); + + return 0; +} + /* Called by DAPM mixer/mux changes to update audio routing between PCMs and * any DAI links. */ int soc_dpcm_runtime_update(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *fe; - int old, new, paths; + int ret = 0; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; - - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; - - /* only check active links */ - if (!fe->cpu_dai->active) - continue; - - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); - - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } - - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - - dpcm_path_put(&list); -capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; - - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } - - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - dpcm_path_put(&list); + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; } + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: mutex_unlock(&card->mutex); - return 0; + return ret; } int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { From c54c1c5ee8e73b7cb752834e52e2129b1dab00bd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 27 Jun 2018 11:56:53 +0300 Subject: [PATCH 153/529] ASoC: qdsp6: qdafe: fix some off by one bugs The > should be >= or we could read one element beyond the end of the port_maps[] array. Fixes: 7fa2d70f9766 ("ASoC: qdsp6: q6afe: Add q6afe driver") Signed-off-by: Dan Carpenter Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 621b67b34db9..671743453fbb 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -776,7 +776,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1013,7 +1013,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1354,7 +1354,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); } From 4f2bd18b191a10660782f2f1ccc989b000b2be63 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 11:48:18 +0200 Subject: [PATCH 154/529] ASoC: dpcm: extend channel merging to the backend cpu dai Extend dpcm_merge_chan to also check backend cpu dai channels capabilities. Apply the same policy as soc_pcm_init_runtime_hw() for multicodec links and only check cpu dai in this case. Cc: Jiada Wang Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f96cde046a..6ee4131941df 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1736,12 +1736,26 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; - int i; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; - for (i = 0; i < be->num_codecs; i++) { - codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else From 4febced15ac8ddb9cf3e603edb111842e4863d9a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 17:36:38 +0200 Subject: [PATCH 155/529] ASoC: dpcm: don't merge format from invalid codec dai When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..5feae9666822 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; From fdd49c510021d389bf5979282193e084f328e031 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jun 2018 09:54:46 +0200 Subject: [PATCH 156/529] ALSA: hda/hdmi - Don't fall back to generic when i915 binding fails When i915 component binding fails, it means that HDMI isn't applicable anyway. Although the probe with the generic HDMI parser would still work, it's essentially useless, hence better to be left unbound. This patch mimics the probe_id field at failing the i915 component binding so that the generic HDMI won't be bound after that. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8a095c16ee65..993294c8fd0a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -84,6 +84,7 @@ struct hda_bus { */ typedef int (*hda_codec_patch_t)(struct hda_codec *); +#define HDA_CODEC_ID_SKIP_PROBE 0x00000001 #define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 #define HDA_CODEC_ID_GENERIC 0x00000201 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ed2318f79e3c..cbf5f4e30027 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2542,6 +2542,8 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* requires i915 binding */ if (!codec->bus->core.audio_component) { codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + /* set probe_id here to prevent generic fallback binding */ + codec->probe_id = HDA_CODEC_ID_SKIP_PROBE; return -ENODEV; } From 8f54061d001ad2da24dba89fc48adbbf4c85222b Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Thu, 28 Jun 2018 22:08:37 +0200 Subject: [PATCH 157/529] ASoC: pxa: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 5738a0abcd6a..c52b33802bf2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, From 95acb005fef2aeaeb63c20de98aca0ed5bd0efa2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:53 +0200 Subject: [PATCH 158/529] ASoC: fold pxa2xx-pcm into its only user, pxa2xx-ac97 Now that the PXA SSP bits are ported over to generic DMA, the pxa2xx-pcm code only has a single user left. This patch folds the remaining bits into its only user and removes the unnecessary glue layer along with its header file. The include dependency to linux/dma/pxa-dma.h is also gone now. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/arm/Kconfig | 5 -- sound/arm/Makefile | 3 - sound/arm/pxa2xx-ac97.c | 114 +++++++++++++++++++++----------- sound/arm/pxa2xx-pcm-lib.c | 2 - sound/arm/pxa2xx-pcm.c | 129 ------------------------------------- sound/arm/pxa2xx-pcm.h | 27 -------- sound/soc/pxa/pxa-ssp.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 2 - 8 files changed, 76 insertions(+), 207 deletions(-) delete mode 100644 sound/arm/pxa2xx-pcm.c delete mode 100644 sound/arm/pxa2xx-pcm.h diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..5fbd47a9177e 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -17,14 +17,9 @@ config SND_ARMAACI select SND_PCM select SND_AC97_CODEC -config SND_PXA2XX_PCM - tristate - select SND_PCM - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_PXA2XX_PCM select SND_AC97_CODEC select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 diff --git a/sound/arm/Makefile b/sound/arm/Makefile index e10d5b169565..34c769489877 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -6,9 +6,6 @@ obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o -obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o -snd-pxa2xx-pcm-objs := pxa2xx-pcm.o - obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 236a63cdaf9f..7d8d7b7199dc 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,7 +15,7 @@ #include #include #include -#include +#include #include #include @@ -27,8 +27,6 @@ #include #include -#include "pxa2xx-pcm.h" - static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97) { if (!pxa2xx_ac97_try_cold_reset()) @@ -63,51 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .chan_name = "pcm_pcm_stereo_out", - .maxburst = 32, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .chan_name = "pcm_pcm_stereo_in", - .maxburst = 32, -}; - static struct snd_pcm *pxa2xx_ac97_pcm; static struct snd_ac97 *pxa2xx_ac97_ac97; -static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; pxa2xx_audio_ops_t *platform_ops; - int r; + int ret, i; + + ret = __pxa2xx_pcm_open(substream); + if (ret) + return ret; runtime->hw.channels_min = 2; runtime->hw.channels_max = 2; - r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - AC97_RATES_FRONT_DAC : AC97_RATES_ADC; - runtime->hw.rates = pxa2xx_ac97_ac97->rates[r]; + i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_RATES_FRONT_DAC : AC97_RATES_ADC; + runtime->hw.rates = pxa2xx_ac97_ac97->rates[i]; snd_pcm_limit_hw_rates(runtime); - platform_ops = substream->pcm->card->dev->platform_data; - if (platform_ops && platform_ops->startup) - return platform_ops->startup(substream, platform_ops->priv); - else - return 0; + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->startup) { + ret = platform_ops->startup(substream, platform_ops->priv); + if (ret < 0) + __pxa2xx_pcm_close(substream); + } + + return ret; } -static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream) { pxa2xx_audio_ops_t *platform_ops; - platform_ops = substream->pcm->card->dev->platform_data; + platform_ops = substream->pcm->card->dev->platform_data; if (platform_ops && platform_ops->shutdown) platform_ops->shutdown(substream, platform_ops->priv); + + return 0; } static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) @@ -115,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; + int ret; + + ret = __pxa2xx_pcm_prepare(substream); + if (ret < 0) + return ret; + return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); } -static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { - .playback_params = &pxa2xx_ac97_pcm_out, - .capture_params = &pxa2xx_ac97_pcm_in, - .startup = pxa2xx_ac97_pcm_startup, - .shutdown = pxa2xx_ac97_pcm_shutdown, - .prepare = pxa2xx_ac97_pcm_prepare, -}; - #ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) @@ -183,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif +static const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_ac97_pcm_open, + .close = pxa2xx_ac97_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = __pxa2xx_pcm_hw_params, + .hw_free = __pxa2xx_pcm_hw_free, + .prepare = pxa2xx_ac97_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; + + +static int pxa2xx_ac97_pcm_new(struct snd_card *card) +{ + struct snd_pcm *pcm; + int stream, ret; + + ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm); + if (ret) + goto out; + + pcm->private_free = pxa2xx_pcm_free_dma_buffers; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_CAPTURE; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + pxa2xx_ac97_pcm = pcm; + ret = 0; + + out: + return ret; +} + static int pxa2xx_ac97_probe(struct platform_device *dev) { struct snd_card *card; @@ -204,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); - ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); + ret = pxa2xx_ac97_pcm_new(card); if (ret) goto err; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index dcbe7ecc1835..b927fa5ddbc0 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -16,8 +16,6 @@ #include #include -#include "pxa2xx-pcm.h" - static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c deleted file mode 100644 index 1c6f4b436de3..000000000000 --- a/sound/arm/pxa2xx-pcm.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include - -#include - -#include -#include -#include - -#include "pxa2xx-pcm.h" - -static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - __pxa2xx_pcm_prepare(substream); - - return client->prepare(substream); -} - -static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; - int ret; - - ret = __pxa2xx_pcm_open(substream); - if (ret) - goto out; - - rtd = runtime->private_data; - - rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - client->playback_params : client->capture_params; - - ret = client->startup(substream); - if (!ret) - goto err2; - - return 0; - - err2: - __pxa2xx_pcm_close(substream); - out: - return ret; -} - -static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - client->shutdown(substream); - - return __pxa2xx_pcm_close(substream); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, - struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int play = client->playback_params ? 1 : 0; - int capt = client->capture_params ? 1 : 0; - int ret; - - ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm); - if (ret) - goto out; - - pcm->private_data = client; - pcm->private_free = pxa2xx_pcm_free_dma_buffers; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - goto out; - - if (play) { - int stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - if (capt) { - int stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - - if (rpcm) - *rpcm = pcm; - ret = 0; - - out: - return ret; -} - -EXPORT_SYMBOL(pxa2xx_pcm_new); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h deleted file mode 100644 index 8fa2b7c9e6b8..000000000000 --- a/sound/arm/pxa2xx-pcm.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -struct pxa2xx_runtime_data { - int dma_ch; - struct snd_dmaengine_dai_dma_data *params; -}; - -struct pxa2xx_pcm_client { - struct snd_dmaengine_dai_dma_data *playback_params; - struct snd_dmaengine_dai_dma_data *capture_params; - int (*startup)(struct snd_pcm_substream *); - void (*shutdown)(struct snd_pcm_substream *); - int (*prepare)(struct snd_pcm_substream *); -}; - -extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **); - diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0b441338bdd4..c1f4af869289 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,7 +34,6 @@ #include #include -#include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" /* diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 8b6a70e94c01..445e691126e5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,8 +20,6 @@ #include #include -#include "../../arm/pxa2xx-pcm.h" - static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { From a7160670b5e2d6b59e0f7a5b7e5bcef3b532c24c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:54 +0200 Subject: [PATCH 159/529] ASoC: pxa: clean up function names in pxa2xx-lib Clean up the namespace a bit and drop the __ prefix of all functions exported by pxa2xx-lib. This improves the readability of the code. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 10 +++++----- sound/arm/pxa2xx-ac97.c | 10 +++++----- sound/arm/pxa2xx-pcm-lib.c | 22 +++++++++++----------- sound/soc/pxa/pxa2xx-pcm.c | 21 +++++++-------------- 4 files changed, 28 insertions(+), 35 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 63f75450d3db..b43de38de8b2 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -10,14 +10,14 @@ struct snd_pcm_substream; struct snd_pcm_hw_params; struct snd_pcm; -extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, +extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); -extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_open(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 7d8d7b7199dc..0d624337857b 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -70,7 +70,7 @@ static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) pxa2xx_audio_ops_t *platform_ops; int ret, i; - ret = __pxa2xx_pcm_open(substream); + ret = pxa2xx_pcm_open(substream); if (ret) return ret; @@ -86,7 +86,7 @@ static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) if (platform_ops && platform_ops->startup) { ret = platform_ops->startup(substream, platform_ops->priv); if (ret < 0) - __pxa2xx_pcm_close(substream); + pxa2xx_pcm_close(substream); } return ret; @@ -110,7 +110,7 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; int ret; - ret = __pxa2xx_pcm_prepare(substream); + ret = pxa2xx_pcm_prepare(substream); if (ret < 0) return ret; @@ -178,8 +178,8 @@ static const struct snd_pcm_ops pxa2xx_pcm_ops = { .open = pxa2xx_ac97_pcm_open, .close = pxa2xx_ac97_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, .prepare = pxa2xx_ac97_pcm_prepare, .trigger = pxa2xx_pcm_trigger, .pointer = pxa2xx_pcm_pointer, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index b927fa5ddbc0..dc56dbebf441 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -33,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .fifo_size = 32, }; -int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -62,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); +EXPORT_SYMBOL(pxa2xx_pcm_hw_params); -int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) +int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); +EXPORT_SYMBOL(pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -84,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(pxa2xx_pcm_pointer); -int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) +int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_prepare); +EXPORT_SYMBOL(pxa2xx_pcm_prepare); -int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) +int pxa2xx_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -127,13 +127,13 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) substream, dma_request_slave_channel(rtd->cpu_dai->dev, dma_params->chan_name)); } -EXPORT_SYMBOL(__pxa2xx_pcm_open); +EXPORT_SYMBOL(pxa2xx_pcm_open); -int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) +int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { return snd_dmaengine_pcm_close_release_chan(substream); } -EXPORT_SYMBOL(__pxa2xx_pcm_close); +EXPORT_SYMBOL(pxa2xx_pcm_close); int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 445e691126e5..da252d1f732e 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,8 +20,8 @@ #include #include -static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; @@ -33,23 +33,16 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - return __pxa2xx_pcm_hw_params(substream, params); -} - -static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - __pxa2xx_pcm_hw_free(substream); - - return 0; + return pxa2xx_pcm_hw_params(substream, params); } static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = __pxa2xx_pcm_open, - .close = __pxa2xx_pcm_close, + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = pxa2xx_pcm_hw_params, + .hw_params = __pxa2xx_pcm_hw_params, .hw_free = pxa2xx_pcm_hw_free, - .prepare = __pxa2xx_pcm_prepare, + .prepare = pxa2xx_pcm_prepare, .trigger = pxa2xx_pcm_trigger, .pointer = pxa2xx_pcm_pointer, .mmap = pxa2xx_pcm_mmap, From 7afd1b0b2ef9a8120951188a955010ef92bdf885 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:55 +0200 Subject: [PATCH 160/529] ASoC: pxa: move some functions to pxa2xx-lib To get rid of some intermediate platform layers, move pxa2xx_soc_pcm_new() and pxa2xx_pcm_ops in pxa2xx-lib. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 3 +++ sound/arm/pxa2xx-ac97.c | 6 ++--- sound/arm/pxa2xx-pcm-lib.c | 41 ++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-pcm.c | 55 -------------------------------------- 4 files changed, 47 insertions(+), 58 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index b43de38de8b2..6758fc12fa84 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -8,6 +8,7 @@ /* PCM */ struct snd_pcm_substream; struct snd_pcm_hw_params; +struct snd_soc_pcm_runtime; struct snd_pcm; extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, @@ -22,6 +23,8 @@ extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream); extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm); +extern int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd); +extern const struct snd_pcm_ops pxa2xx_pcm_ops; /* AC97 */ diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d624337857b..1f72672262d0 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -174,7 +174,7 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif -static const struct snd_pcm_ops pxa2xx_pcm_ops = { +static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = { .open = pxa2xx_ac97_pcm_open, .close = pxa2xx_ac97_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -203,13 +203,13 @@ static int pxa2xx_ac97_pcm_new(struct snd_card *card) goto out; stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); if (ret) goto out; stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); if (ret) goto out; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index dc56dbebf441..add23d9b4ef6 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -179,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers); +int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} +EXPORT_SYMBOL(pxa2xx_soc_pcm_new); + +const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; +EXPORT_SYMBOL(pxa2xx_pcm_ops); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx sound library"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index da252d1f732e..a1df4ec76cbd 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,61 +20,6 @@ #include #include -static int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma; - - dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - return pxa2xx_pcm_hw_params(substream, params); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - static const struct snd_soc_component_driver pxa2xx_soc_platform = { .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, From 456ec80876564edb74a0fff78499beb7ca286302 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:56 +0200 Subject: [PATCH 161/529] ASoC: pxa2xx-pcm-lib: fix indenting While at it, also fix some indenting. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index add23d9b4ef6..7931789d4a9f 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -23,8 +23,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, From d767d3ce5c48b3378e20e8cfd5d5379c4ca6001b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:57 +0200 Subject: [PATCH 162/529] ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components Now that the functions are now available through pxa2xx-lib, hook them up to pxa-sspi, pxa-ac97 and pxa-i2s. This allows DT platforms to use the DAIs without a platform driver. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 3 +++ sound/soc/pxa/pxa2xx-ac97.c | 3 +++ sound/soc/pxa/pxa2xx-i2s.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c1f4af869289..01d54697ede4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -841,6 +841,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index c52b33802bf2..9f779657bc86 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -214,6 +214,9 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e7184de0de04..42820121e5b9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -364,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) From c7b4f15ddb4f28451679019374027f5223c616ce Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:58 +0200 Subject: [PATCH 163/529] ASoC: pxa: remove bindings from pxa2xx-pcm This platform is no longer needed on DT boards, so let's remove them to avoid confusion. DT bindings should use the CPU DAIs (I2S/SSP/AC97) directly. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 --------------- sound/soc/pxa/pxa2xx-pcm.c | 9 --------- 2 files changed, 24 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt deleted file mode 100644 index 551fbb8348c2..000000000000 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt +++ /dev/null @@ -1,15 +0,0 @@ -DT bindings for ARM PXA2xx PCM platform driver - -This is just a dummy driver that registers the PXA ASoC platform driver. -It does not have any resources assigned. - -Required properties: - - - compatible 'mrvl,pxa-pcm-audio' - -Example: - - pxa_pcm_audio: snd_soc_pxa_audio { - compatible = "mrvl,pxa-pcm-audio"; - }; - diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index a1df4ec76cbd..72eaaef1b426 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -32,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev) NULL, 0); } -#ifdef CONFIG_OF -static const struct of_device_id snd_soc_pxa_audio_match[] = { - { .compatible = "mrvl,pxa-pcm-audio" }, - { } -}; -MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); -#endif - static struct platform_driver pxa_pcm_driver = { .driver = { .name = "pxa-pcm-audio", - .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, From 0a94cf3457408058f894cc4d95e58d8e18eb7f75 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:59 +0200 Subject: [PATCH 164/529] ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..95dcf97a6271 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -31,7 +31,7 @@ config SND_PXA2XX_SOC_I2S tristate config SND_PXA_SOC_SSP - tristate + tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP config SND_MMP_SOC_SSPA From f11c5db770ab675d270cb4d5a2bb90923066ef49 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Fri, 29 Jun 2018 20:29:44 +0800 Subject: [PATCH 165/529] ASoC: mediatek: sub dai use list_head use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- .../mediatek/common/mtk-afe-platform-driver.c | 64 +++++++------------ sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif From c1d9b4196ba6b311bd48a9320cd46aa125e0b034 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Fri, 29 Jun 2018 20:29:45 +0800 Subject: [PATCH 166/529] ASoC: mt6797: sub dai use list_head Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- sound/soc/mediatek/mt6797/mt6797-afe-common.h | 1 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 63 +++++++++++++------ sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 20 +++--- .../soc/mediatek/mt6797/mt6797-dai-hostless.c | 16 +++-- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 19 +++--- 5 files changed, 80 insertions(+), 39 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include +#include #include #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; + INIT_LIST_HEAD(&afe->sub_dais); - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } From 222bce5eb88d1af656419db04bcd84b2419fb900 Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Fri, 29 Jun 2018 19:07:42 +0200 Subject: [PATCH 167/529] ALSA: snd-aoa: add of_node_put() in error path Both calls to of_find_node_by_name() and of_get_next_child() return a node pointer with refcount incremented thus it must be explicidly decremented here after the last usage. As we are assured to have a refcounted np either from the initial of_find_node_by_name(NULL, name); or from the of_get_next_child(gpio, np) in the while loop if we reached the error code path below, an x of_node_put(np) is needed. Signed-off-by: Nicholas Mc Guire Fixes: commit f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Takashi Iwai --- sound/aoa/core/gpio-feature.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 71960089e207..65557421fe0b 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name, } reg = of_get_property(np, "reg", NULL); - if (!reg) + if (!reg) { + of_node_put(np); return NULL; + } *gpioptr = *reg; From d573454d9b4f00061da314460908e19476d2ff6d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:28 +0000 Subject: [PATCH 168/529] ASoC: simple-card: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card.h | 7 ++----- sound/soc/generic/simple-card.c | 17 +++++++---------- 2 files changed, 9 insertions(+), 15 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index a6a2e1547092..d264e5463f22 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -1,12 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * ASoC simple sound card support * * Copyright (C) 2012 Renesas Solutions Corp. * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SIMPLE_CARD_H diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c5b6e04cd926..64bf3560c1d1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -1,13 +1,10 @@ -/* - * ASoC simple sound card support - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple sound card support +// +// Copyright (C) 2012 Renesas Solutions Corp. +// Kuninori Morimoto + #include #include #include From d613a7f45ebb2f113444630fcbbb8a074c741998 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:44 +0000 Subject: [PATCH 169/529] ASoC: simple-card-utils: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 8 +++----- sound/soc/generic/simple-card-utils.c | 15 ++++++--------- 2 files changed, 9 insertions(+), 14 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index f82acef3b992..8bc5e2d8b13c 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,12 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ + #ifndef __SIMPLE_CARD_UTILS_H #define __SIMPLE_CARD_UTILS_H diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4398c9580929..d3f3f0fec74c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,12 +1,9 @@ -/* - * simple-card-utils.c - * - * Copyright (c) 2016 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// simple-card-utils.c +// +// Copyright (c) 2016 Kuninori Morimoto + #include #include #include From 9afe58f1cbd1fb5e9426483656ae6f65de4130e4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:58 +0000 Subject: [PATCH 170/529] ASoC: simple-scu-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 487716559deb..16a83bc51e0e 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -1,15 +1,12 @@ -/* - * ASoC simple SCU sound card support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple SCU sound card support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include #include #include From decd896121f965db2fdee0f0475c3e404746b333 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:31:16 +0000 Subject: [PATCH 171/529] ASoC: audio-graph-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index a2a3e630f11c..2094d2c8919f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -1,15 +1,12 @@ -/* - * ASoC audio graph sound card support - * - * Copyright (C) 2016 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph sound card support +// +// Copyright (C) 2016 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include #include #include From ac204c9b030fe2d27cf3a63386811254d99b3ce4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:31:33 +0000 Subject: [PATCH 172/529] ASoC: audio-graph-scu-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 25 +++++++++++------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 095ef6426d42..92882e392d6c 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -1,17 +1,14 @@ -/* - * ASoC audio graph SCU sound card support - * - * Copyright (C) 2017 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on - * ${LINUX}/sound/soc/generic/simple-scu-card.c - * ${LINUX}/sound/soc/generic/audio-graph-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph SCU sound card support +// +// Copyright (C) 2017 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on +// ${LINUX}/sound/soc/generic/simple-scu-card.c +// ${LINUX}/sound/soc/generic/audio-graph-card.c + #include #include #include From d1aaa2e68619ca4c1cc05ce7bc029cda5a8cbe87 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:21:54 +0000 Subject: [PATCH 173/529] ASoC: soc-io.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 026cd5347e53..1ff9175e9d5e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -1,15 +1,10 @@ -/* - * soc-io.c -- ASoC register I/O helpers - * - * Copyright 2009-2011 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-io.c -- ASoC register I/O helpers +// +// Copyright 2009-2011 Wolfson Microelectronics PLC. +// +// Author: Mark Brown #include #include From e2cfd2c9673f4e9d08971fed53fe1f41d8270ef0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:15 +0000 Subject: [PATCH 174/529] ASoC: soc-dai.h: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e6f8c40ed43c..a14bc0608ae9 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -1,12 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dai.h -- ALSA SoC Layer * * Copyright: 2005-2008 Wolfson Microelectronics. PLC. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * Digital Audio Interface (DAI) API. */ From 4eef5a90ca8b2aab61c98853141f9242af4a8339 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:30 +0000 Subject: [PATCH 175/529] ASoC: soc-ops.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7144a51ddfa9..592efb370c44 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -1,20 +1,15 @@ -/* - * soc-ops.c -- Generic ASoC operations - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ops.c -- Generic ASoC operations +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie #include #include From ed51758247c51dfdb526d279164cfd925496f187 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:44 +0000 Subject: [PATCH 176/529] ASoC: soc-pcm.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 28 +++++++++++----------------- 1 file changed, 11 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6ee4131941df..c2a31b51da4f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1,20 +1,14 @@ -/* - * soc-pcm.c -- ALSA SoC PCM - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Authors: Liam Girdwood - * Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-pcm.c -- ALSA SoC PCM +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Authors: Liam Girdwood +// Mark Brown #include #include From b53c34b4b73eb65e6480ada6ec8ddbcc2d4e9817 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:58 +0000 Subject: [PATCH 177/529] ASoC: soc-dpcm.h: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 806059052bfc..9bb92f187af8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -1,11 +1,8 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support * * Author: Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_DPCM_H From 8ab0215c11817fc0a89f1f82cdf10fd4c0eb9e86 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:16 +0000 Subject: [PATCH 178/529] ASoC: soc-jack.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b2b16044ae80..c7b990abdbaa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -1,15 +1,10 @@ -/* - * soc-jack.c -- ALSA SoC jack handling - * - * Copyright 2008 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-jack.c -- ALSA SoC jack handling +// +// Copyright 2008 Wolfson Microelectronics PLC. +// +// Author: Mark Brown #include #include From 632628df453ccda727fbcc1e346600162ac995e9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:30 +0000 Subject: [PATCH 179/529] ASoC: soc-utils.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..ea024236c643 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -1,17 +1,11 @@ -/* - * soc-util.c -- ALSA SoC Audio Layer utility functions - * - * Copyright 2009 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * Liam Girdwood - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-util.c -- ALSA SoC Audio Layer utility functions +// +// Copyright 2009 Wolfson Microelectronics PLC. +// +// Author: Mark Brown +// Liam Girdwood #include #include From 9e14035c7fac144f31a822f0034fe5ed79c9ef8a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:45 +0000 Subject: [PATCH 180/529] ASoC: soc-devres.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-devres.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745df1412..a9ea172a66a7 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -1,13 +1,8 @@ -/* - * soc-devres.c -- ALSA SoC Audio Layer devres functions - * - * Copyright (C) 2013 Linaro Ltd - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-devres.c -- ALSA SoC Audio Layer devres functions +// +// Copyright (C) 2013 Linaro Ltd #include #include From 7730bb13c7472620b585783f248b2dccd09d1819 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:04 +0000 Subject: [PATCH 181/529] ASoC: soc-acpi: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 14 ++------------ include/sound/soc-acpi.h | 13 ++----------- sound/soc/soc-acpi.c | 20 +++++--------------- 3 files changed, 9 insertions(+), 38 deletions(-) diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 917ddd0f2762..bb1d24b703fb 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -1,16 +1,6 @@ - -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Copyright (C) 2017, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * */ #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 082224275f52..e45b2330d16a 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -1,15 +1,6 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Copyright (C) 2013-15, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * */ #ifndef __LINUX_SND_SOC_ACPI_H diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 3d7e1ff79139..b8e72b52db30 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,18 +1,8 @@ -/* - * soc-apci.c - support for ACPI enumeration. - * - * Copyright (c) 2013-15, Intel Corporation. - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// soc-apci.c - support for ACPI enumeration. +// +// Copyright (c) 2013-15, Intel Corporation. #include From 873486ed4af3e11bfc20832dff7b124ba652bf77 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:18 +0000 Subject: [PATCH 182/529] ASoC: soc-core: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 11 ++++------- sound/soc/soc-core.c | 41 ++++++++++++++++++----------------------- 2 files changed, 22 insertions(+), 30 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..16f0bf10cc42 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc.h -- ALSA SoC Layer * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_H diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..68b08781c832 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1,26 +1,21 @@ -/* - * soc-core.c -- ALSA SoC Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * o Add hw rules to enforce rates, etc. - * o More testing with other codecs/machines. - * o Add more codecs and platforms to ensure good API coverage. - * o Support TDM on PCM and I2S - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-core.c -- ALSA SoC Audio Layer +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie +// +// TODO: +// o Add hw rules to enforce rates, etc. +// o More testing with other codecs/machines. +// o Add more codecs and platforms to ensure good API coverage. +// o Support TDM on PCM and I2S #include #include From c01f3af4d32071915119ffcb933e75d7c165378e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:31 +0000 Subject: [PATCH 183/529] ASoC: soc-dapm: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 11 ++++------- sound/soc/soc-dapm.c | 42 +++++++++++++++++----------------------- 2 files changed, 22 insertions(+), 31 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a6ce2de4e20a..af9ef16cc34d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_DAPM_H diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a099c3e45504..0602b2888d52 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1,27 +1,21 @@ -/* - * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Features: - * o Changes power status of internal codec blocks depending on the - * dynamic configuration of codec internal audio paths and active - * DACs/ADCs. - * o Platform power domain - can support external components i.e. amps and - * mic/headphone insertion events. - * o Automatic Mic Bias support - * o Jack insertion power event initiation - e.g. hp insertion will enable - * sinks, dacs, etc - * o Delayed power down of audio subsystem to reduce pops between a quick - * device reopen. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-dapm.c -- ALSA SoC Dynamic Audio Power Management +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Author: Liam Girdwood +// +// Features: +// o Changes power status of internal codec blocks depending on the +// dynamic configuration of codec internal audio paths and active +// DACs/ADCs. +// o Platform power domain - can support external components i.e. amps and +// mic/headphone insertion events. +// o Automatic Mic Bias support +// o Jack insertion power event initiation - e.g. hp insertion will enable +// sinks, dacs, etc +// o Delayed power down of audio subsystem to reduce pops between a quick +// device reopen. #include #include From f2b6a1b25fecc48a46c8a41636101af8a41c88a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:45 +0000 Subject: [PATCH 184/529] ASoC: soc-topology: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 7 ++---- sound/soc/soc-topology.c | 47 ++++++++++++++++-------------------- 2 files changed, 23 insertions(+), 31 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 401ef2c45d6c..fa4b8413d2e2 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM * * Copyright (C) 2012 Texas Instruments Inc. * Copyright (C) 2015 Intel Corporation. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc. */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 05d177d689e2..66e77e020745 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1,29 +1,24 @@ -/* - * soc-topology.c -- ALSA SoC Topology - * - * Copyright (C) 2012 Texas Instruments Inc. - * Copyright (C) 2015 Intel Corporation. - * - * Authors: Liam Girdwood - * K, Mythri P - * Prusty, Subhransu S - * B, Jayachandran - * Abdullah, Omair M - * Jin, Yao - * Lin, Mengdong - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Add support to read audio firmware topology alongside firmware text. The - * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, - * equalizers, firmware, coefficients etc. - * - * This file only manages the core ALSA and ASoC components, all other bespoke - * firmware topology data is passed to component drivers for bespoke handling. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-topology.c -- ALSA SoC Topology +// +// Copyright (C) 2012 Texas Instruments Inc. +// Copyright (C) 2015 Intel Corporation. +// +// Authors: Liam Girdwood +// K, Mythri P +// Prusty, Subhransu S +// B, Jayachandran +// Abdullah, Omair M +// Jin, Yao +// Lin, Mengdong +// +// Add support to read audio firmware topology alongside firmware text. The +// topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, +// equalizers, firmware, coefficients etc. +// +// This file only manages the core ALSA and ASoC components, all other bespoke +// firmware topology data is passed to component drivers for bespoke handling. #include #include From b3ed4c86a700b494fc5058a52531eeb14d6fe00f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:57 +0000 Subject: [PATCH 185/529] ASoC: soc-compress: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/compress_driver.h | 21 +++------------------ sound/soc/soc-compress.c | 24 +++++++++--------------- 2 files changed, 12 insertions(+), 33 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9924bc9cbc7c..ea8c93bbb0e0 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -1,27 +1,12 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * compress_driver.h - compress offload driver definations * * Copyright (C) 2011 Intel Corporation * Authors: Vinod Koul * Pierre-Louis Bossart - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * */ + #ifndef __COMPRESS_DRIVER_H #define __COMPRESS_DRIVER_H diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index e095115fa9f9..b9e1673fea51 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -1,18 +1,12 @@ -/* - * soc-compress.c -- ALSA SoC Compress - * - * Copyright (C) 2012 Intel Corp. - * - * Authors: Namarta Kohli - * Ramesh Babu K V - * Vinod Koul - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-compress.c -- ALSA SoC Compress +// +// Copyright (C) 2012 Intel Corp. +// +// Authors: Namarta Kohli +// Ramesh Babu K V +// Vinod Koul #include #include From 1356a6071cf4d7187652cd2b18dfab4763e0dba6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:25:11 +0000 Subject: [PATCH 186/529] ASoC: soc-generic-dmaengine-pcm: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 14 +++----------- sound/soc/soc-generic-dmaengine-pcm.c | 19 +++++-------------- 2 files changed, 8 insertions(+), 25 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index e3481eebdd98..2c4cfaa135a6 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -1,17 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (C) 2012, Analog Devices Inc. * Author: Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * */ + #ifndef __SOUND_DMAENGINE_PCM_H__ #define __SOUND_DMAENGINE_PCM_H__ diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..13bdca6e41c5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -1,17 +1,8 @@ -/* - * Copyright (C) 2013, Analog Devices Inc. - * Author: Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2013, Analog Devices Inc. +// Author: Lars-Peter Clausen + #include #include #include From 1a8f0a3c13c136951de7ea24ccb148e745db98a2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:26:27 +0000 Subject: [PATCH 187/529] ASoC: ac97: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/ac97/codec.h | 8 +++----- include/sound/ac97/compat.h | 9 +++------ include/sound/ac97/controller.h | 8 +++----- include/sound/ac97/regs.h | 20 ++------------------ include/sound/ac97_codec.h | 25 +++++-------------------- sound/soc/soc-ac97.c | 29 ++++++++++++----------------- 6 files changed, 28 insertions(+), 71 deletions(-) diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h index ec04be9ab119..9792d25fa369 100644 --- a/include/sound/ac97/codec.h +++ b/include/sound/ac97/codec.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik */ + #ifndef __SOUND_AC97_CODEC2_H #define __SOUND_AC97_CODEC2_H diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index 1351cba40048..57e19afa31ab 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -1,14 +1,11 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik * * This file is for backward compatibility with snd_ac97 structure and its * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops. - * */ + #ifndef AC97_COMPAT_H #define AC97_COMPAT_H diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index b36ecdd64f14..06b5afb7fa6b 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik */ + #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h index 9a4fa0c3264a..843f73f3705a 100644 --- a/include/sound/ac97/regs.h +++ b/include/sound/ac97/regs.h @@ -1,27 +1,11 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ - /* * AC'97 codec registers */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 89d311a503d3..cc383991c0fe 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -1,30 +1,15 @@ -#ifndef __SOUND_AC97_CODEC_H -#define __SOUND_AC97_CODEC_H - -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ +#ifndef __SOUND_AC97_CODEC_H +#define __SOUND_AC97_CODEC_H + #include #include #include diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f214bca..c086786e4471 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie #include #include From 1581250119daa9426c359d059e2dc14ec04bcc0c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 30 Jun 2018 22:24:33 +0200 Subject: [PATCH 188/529] ASoC: pxa: select SND_PXA2XX_LIB for drivers that depend on it Commit d767d3ce5c48b ("ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components") created a build-time dependency to SND_PXA2XX_LIB but missed to reflect that in Kconfig. Reported-by: kbuild test robot Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 95dcf97a6271..2fc02c227f69 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,18 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate From 05739375f1c0a1048fea8b9c4cb54d9e4a891938 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 29 Jun 2018 14:59:40 +0200 Subject: [PATCH 189/529] ASoC: pxa-ssp: remove .set_pll() and .set_clkdiv() callbacks The .set_pll() and .set_clkdiv() callbacks are considered legacy and should not be used anymore. In order to support PXA boards on DT platforms, remove them and let the code figure out the correct dividers and PLL base frequencies itself. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/linux/pxa2xx_ssp.h | 8 ++ sound/soc/pxa/pxa-ssp.c | 146 ++++++++++++++++++------------------- 2 files changed, 81 insertions(+), 73 deletions(-) diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 03a7ca46735b..13b4244d44c1 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -171,6 +171,14 @@ #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_ACDS_1 (0) +#define SSACD_ACDS_2 (1) +#define SSACD_ACDS_4 (2) +#define SSACD_ACDS_8 (3) +#define SSACD_ACDS_16 (4) +#define SSACD_ACDS_32 (5) +#define SSACD_SCDB_4X (0) +#define SSACD_SCDB_1X (1) #define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ /* LPSS SSP */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 01d54697ede4..f8339bb01251 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; unsigned int configured_dai_fmt; @@ -192,21 +193,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) pxa_ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -262,67 +248,18 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; - if (ssp->type == PXA3xx_SSP) - val &= ~SSACD_SCDX8; - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; - case PXA_SSP_CLK_SCDB_8: - if (ssp->type == PXA3xx_SSP) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; - default: - return -EINVAL; - } - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); - break; - default: - return -ENODEV; - } - - return 0; -} - /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); - switch (freq_out) { + switch (freq) { case 5622000: break; case 11345000: @@ -353,7 +290,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, u64 tmp = 19968; tmp *= 1000000; - do_div(tmp, freq_out); + do_div(tmp, freq); val = tmp; val = (val << 16) | 64; @@ -363,7 +300,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, dev_dbg(&ssp->pdev->dev, "Using SSACDD %x to supply %uHz\n", - val, freq_out); + val, freq); break; } @@ -568,6 +505,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) return 0; } +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + /* * Set the SSP audio DMA parameters and sample size. * Can be called multiple times by oss emulation. @@ -579,11 +534,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -623,11 +579,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, } pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd, acds; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + acds = m->acds; + + /* The values in the table are for 16 bits */ + if (width == 32) + acds--; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { + if (((priv->sysclk / bclk) == 64) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -812,8 +814,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, From 5650729f9a1bbf65b57139d855dabe0a7e6cb494 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:20 +0200 Subject: [PATCH 190/529] ASoC: es7134: remove 64kHz rate from the supported rates 64Khz is actually not supported by the es7134 according to the datasheet Fixes: 9000b59d7a12 ("ASoC: es7134: add es7134 DAC driver") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 58515bb1a303..2fbb49f5b278 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -48,7 +48,11 @@ static struct snd_soc_dai_driver es7134_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | From a016b11cc41df06b79c0c226e719d0d88373919c Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:21 +0200 Subject: [PATCH 191/529] ASoC: es7134: check if mclk rate is valid For each supported sample rate, the es7134 can work with several mclk / sample rate ratio. Check if ratio we get is actually OK. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 119 +++++++++++++++++++++++++++++++++++++- 1 file changed, 117 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 2fbb49f5b278..698289dc3e22 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -17,6 +17,7 @@ * in the file called COPYING. */ +#include #include #include @@ -24,6 +25,77 @@ * The everest 7134 is a very simple DA converter with no register */ +struct es7134_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *mclk_fs; + unsigned int mclk_fs_num; +}; + +struct es7134_chip { + const struct es7134_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7134_data { + unsigned int mclk; + const struct es7134_chip *chip; +}; + +static int es7134_check_mclk(struct snd_soc_dai *dai, + struct es7134_data *priv, + unsigned int rate) +{ + unsigned int mfs = priv->mclk / rate; + int i, j; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7134_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate > mode->rate_max) + continue; + + for (j = 0; j < mode->mclk_fs_num; j++) { + if (mode->mclk_fs[j] == mfs) + return 0; + } + + dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n", + mfs, rate); + return -EINVAL; + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7134_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + /* mclk has not been provided, assume it is OK */ + if (!priv->mclk) + return 0; + + return es7134_check_mclk(dai, priv, params_rate(params)); +} + +static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | @@ -40,6 +112,8 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, + .hw_params = es7134_hw_params, + .set_sysclk = es7134_set_sysclk, }; static struct snd_soc_dai_driver es7134_dai = { @@ -62,6 +136,33 @@ static struct snd_soc_dai_driver es7134_dai = { .ops = &es7134_dai_ops, }; +static const struct es7134_clock_mode es7134_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .mclk_fs_num = 5, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 }, + .mclk_fs_num = 5, + }, { + /* Quad speed mode */ + .rate_min = 167000, + .rate_max = 192000, + .mclk_fs = (unsigned int[]) { 128, 192, 256 }, + .mclk_fs_num = 3, + }, +}; + +static const struct es7134_chip es7134_chip = { + .modes = es7134_modes, + .mode_num = ARRAY_SIZE(es7134_modes), +}; + static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), @@ -86,6 +187,20 @@ static const struct snd_soc_component_driver es7134_component_driver = { static int es7134_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct es7134_data *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, &es7134_dai, 1); @@ -93,8 +208,8 @@ static int es7134_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { - { .compatible = "everest,es7134", }, - { .compatible = "everest,es7144", }, + { .compatible = "everest,es7134", .data = &es7134_chip }, + { .compatible = "everest,es7144", .data = &es7134_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); From 424e2b4b3521334812d833eef27df77671428698 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:23 +0200 Subject: [PATCH 192/529] ASoC: es7134: Add VDD and AVDD power supplies Add the VDD and AVDD power supplies to the DAPM graph as some board may need to enable a regulator to turn them on. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 698289dc3e22..5ad59c38fed1 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -167,11 +167,15 @@ static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, + { "Playback", NULL, "VDD" }, + { "DAC", NULL, "AVDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { From 9b11233d8e2ca80afc6e16200b680c5daf051333 Mon Sep 17 00:00:00 2001 From: John Ogness Date: Sun, 1 Jul 2018 17:28:07 +0200 Subject: [PATCH 193/529] ALSA: usb: caiaq: audio: use irqsave() in USB's complete callback The USB completion callback does not disable interrupts while acquiring the lock. We want to remove the local_irq_disable() invocation from __usb_hcd_giveback_urb() and therefore it is required for the callback handler to disable the interrupts while acquiring the lock. The callback may be invoked either in IRQ or BH context depending on the USB host controller. Use the _irqsave() variant of the locking primitives. Signed-off-by: John Ogness Signed-off-by: Sebastian Andrzej Siewior Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index f35d29f49ffe..15344d39a6cd 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -636,6 +636,7 @@ static void read_completed(struct urb *urb) struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; + unsigned long flags; size_t offset = 0; if (urb->status || !info) @@ -672,10 +673,10 @@ static void read_completed(struct urb *urb) offset += len; if (len > 0) { - spin_lock(&cdev->spinlock); + spin_lock_irqsave(&cdev->spinlock, flags); fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); - spin_unlock(&cdev->spinlock); + spin_unlock_irqrestore(&cdev->spinlock, flags); check_for_elapsed_periods(cdev, cdev->sub_playback); check_for_elapsed_periods(cdev, cdev->sub_capture); send_it = 1; From 1259d239799bf9c898091d92adc4317f2c3d74ad Mon Sep 17 00:00:00 2001 From: John Ogness Date: Sun, 1 Jul 2018 17:28:08 +0200 Subject: [PATCH 194/529] ALSA: usb-midi: use irqsave() in USB's complete callback The USB completion callback does not disable interrupts while acquiring the lock. We want to remove the local_irq_disable() invocation from __usb_hcd_giveback_urb() and therefore it is required for the callback handler to disable the interrupts while acquiring the lock. The callback may be invoked either in IRQ or BH context depending on the USB host controller. Use the _irqsave() variant of the locking primitives. Signed-off-by: John Ogness Signed-off-by: Sebastian Andrzej Siewior Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1aaa3292bf..dcfc546d81b9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -281,15 +281,16 @@ static void snd_usbmidi_out_urb_complete(struct urb *urb) struct out_urb_context *context = urb->context; struct snd_usb_midi_out_endpoint *ep = context->ep; unsigned int urb_index; + unsigned long flags; - spin_lock(&ep->buffer_lock); + spin_lock_irqsave(&ep->buffer_lock, flags); urb_index = context - ep->urbs; ep->active_urbs &= ~(1 << urb_index); if (unlikely(ep->drain_urbs)) { ep->drain_urbs &= ~(1 << urb_index); wake_up(&ep->drain_wait); } - spin_unlock(&ep->buffer_lock); + spin_unlock_irqrestore(&ep->buffer_lock, flags); if (urb->status < 0) { int err = snd_usbmidi_urb_error(urb); if (err < 0) { From 2daf3d9962c5a11fb79fe17bae03125df5d60236 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 3 Jul 2018 13:07:25 +0800 Subject: [PATCH 195/529] ASoC: rt5682: add button detection mode control We are currently using power saving mode for button detection. However, it will impact the headset recording performance. This patch will switch button detection to normal mode in capture and switch to power saving mode in the end of capture. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index baad177908ab..640d400ca013 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -857,6 +857,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) btn_type = val & 0xfff0; snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); return btn_type; } @@ -1645,6 +1647,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1807,6 +1811,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, From 1b8fc56ed514f815c4779273348c3308fb263c87 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 15:28:44 +0200 Subject: [PATCH 196/529] ASoC: es7241: add dt-bindings documentation for the es7241 adc Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/everest,es7241.txt | 28 +++++++++++++++++++ 1 file changed, 28 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.txt b/Documentation/devicetree/bindings/sound/everest,es7241.txt new file mode 100644 index 000000000000..28f82cf4959f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es7241.txt @@ -0,0 +1,28 @@ +ES7241 i2s AD converter + +Required properties: +- compatible : "everest,es7241" +- VDDP-supply: regulator phandle for the VDDA supply +- VDDA-supply: regulator phandle for the VDDP supply +- VDDD-supply: regulator phandle for the VDDD supply + +Optional properties: +- reset-gpios: gpio connected to the reset pin +- m0-gpios : gpio connected to the m0 pin +- m1-gpios : gpio connected to the m1 pin +- everest,sdout-pull-down: + Format used by the serial interface is controlled by pulling + the sdout. If the sdout is pulled down, leftj format is used. + If this property is not provided, sdout is assumed to pulled + up and i2s format is used + +Example: + +linein: audio-codec@2 { + #sound-dai-cells = <0>; + compatible = "everest,es7241"; + VDDA-supply = <&vcc_3v3>; + VDDP-supply = <&vcc_3v3>; + VDDD-supply = <&vcc_3v3>; + reset-gpios = <&gpio GPIOH_42>; +}; From 5f7bdc466c772b3af3145a71724965ecdc03e6bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 15:28:45 +0200 Subject: [PATCH 197/529] ASoC: es7241: add es7241 codec support Add support for the everest es7241 which is a simple 2 channels analog to digital converter. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 322 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 328 insertions(+) create mode 100644 sound/soc/codecs/es7241.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6d1674699385..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 + select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI select SND_SOC_ICS43432 @@ -585,6 +586,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES7241 + tristate "Everest Semi ES7241 CODEC" + config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f26ded89a1e5..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -330,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c new file mode 100644 index 000000000000..87991bd4acef --- /dev/null +++ b/sound/soc/codecs/es7241.c @@ -0,0 +1,322 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +struct es7241_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *slv_mfs; + unsigned int slv_mfs_num; + unsigned int mst_mfs; + unsigned int mst_m0:1; + unsigned int mst_m1:1; +}; + +struct es7241_chip { + const struct es7241_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7241_data { + struct gpio_desc *reset; + struct gpio_desc *m0; + struct gpio_desc *m1; + unsigned int fmt; + unsigned int mclk; + bool is_slave; + const struct es7241_chip *chip; +}; + +static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) +{ + /* put the device in reset */ + gpiod_set_value_cansleep(priv->reset, 0); + + /* set the mode */ + gpiod_set_value_cansleep(priv->m0, m0); + gpiod_set_value_cansleep(priv->m1, m1); + + /* take the device out of reset - datasheet does not specify a delay */ + gpiod_set_value_cansleep(priv->reset, 1); +} + +static int es7241_set_slave_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + int j; + + if (!mfs) + goto out_ok; + + for (j = 0; j < mode->slv_mfs_num; j++) { + if (mode->slv_mfs[j] == mfs) + goto out_ok; + } + + return -EINVAL; + +out_ok: + es7241_set_mode(priv, 1, 1); + return 0; +} + +static int es7241_set_master_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + /* + * We can't really set clock ratio, if the mclk/lrclk is different + * from what we provide, then error out + */ + if (mfs && mfs != mode->mst_mfs) + return -EINVAL; + + es7241_set_mode(priv, mode->mst_m0, mode->mst_m1); + + return 0; +} + +static int es7241_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int mfs = priv->mclk / rate; + int i; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7241_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate >= mode->rate_max) + continue; + + if (priv->is_slave) + return es7241_set_slave_mode(priv, mode, mfs); + else + return es7241_set_master_mode(priv, mode, mfs); + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + +static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(dai->dev, "Unsupported dai clock inversion\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) { + dev_err(dai->dev, "Invalid dai format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + priv->is_slave = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + priv->is_slave = false; + break; + + default: + dev_err(dai->dev, "Unsupported clock configuration\n"); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops es7241_dai_ops = { + .set_fmt = es7241_set_fmt, + .hw_params = es7241_hw_params, + .set_sysclk = es7241_set_sysclk, +}; + +static struct snd_soc_dai_driver es7241_dai = { + .name = "es7241-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7241_dai_ops, +}; + +static const struct es7241_clock_mode es7241_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .slv_mfs_num = 5, + .mst_mfs = 256, + .mst_m0 = 0, + .mst_m1 = 0, + }, { + /* Double speed mode */ + .rate_min = 50000, + .rate_max = 100000, + .slv_mfs = (unsigned int[]) { 128, 192 }, + .slv_mfs_num = 2, + .mst_mfs = 128, + .mst_m0 = 1, + .mst_m1 = 0, + }, { + /* Quad speed mode */ + .rate_min = 100000, + .rate_max = 200000, + .slv_mfs = (unsigned int[]) { 64 }, + .slv_mfs_num = 1, + .mst_mfs = 64, + .mst_m0 = 0, + .mst_m1 = 1, + }, +}; + +static const struct es7241_chip es7241_chip = { + .modes = es7241_modes, + .mode_num = ARRAY_SIZE(es7241_modes), +}; + +static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AINL"), + SND_SOC_DAPM_INPUT("AINR"), + SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0), +}; + +static const struct snd_soc_dapm_route es7241_dapm_routes[] = { + { "ADC", NULL, "AINL", }, + { "ADC", NULL, "AINR", }, + { "ADC", NULL, "VDDA", }, + { "Capture", NULL, "VDDP", }, + { "Capture", NULL, "VDDD", }, +}; + +static const struct snd_soc_component_driver es7241_component_driver = { + .dapm_widgets = es7241_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es7241_dapm_widgets), + .dapm_routes = es7241_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv) +{ + bool is_leftj; + + /* + * The format is given by a pull resistor on the SDOUT pin: + * pull-up for i2s, pull-down for left justified. + */ + is_leftj = of_property_read_bool(dev->of_node, + "everest,sdout-pull-down"); + if (is_leftj) + priv->fmt = SND_SOC_DAIFMT_LEFT_J; + else + priv->fmt = SND_SOC_DAIFMT_I2S; +} + +static int es7241_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct es7241_data *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + es7241_parse_fmt(dev, priv); + + priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset)) { + err = PTR_ERR(priv->reset); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'reset' gpio: %d", err); + return err; + } + + priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW); + if (IS_ERR(priv->m0)) { + err = PTR_ERR(priv->m0); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm0' gpio: %d", err); + return err; + } + + priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW); + if (IS_ERR(priv->m1)) { + err = PTR_ERR(priv->m1); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm1' gpio: %d", err); + return err; + } + + return devm_snd_soc_register_component(&pdev->dev, + &es7241_component_driver, + &es7241_dai, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id es7241_ids[] = { + { .compatible = "everest,es7241", .data = &es7241_chip }, + { } +}; +MODULE_DEVICE_TABLE(of, es7241_ids); +#endif + +static struct platform_driver es7241_driver = { + .driver = { + .name = "es7241", + .of_match_table = of_match_ptr(es7241_ids), + }, + .probe = es7241_probe, +}; + +module_platform_driver(es7241_driver); + +MODULE_DESCRIPTION("ASoC ES7241 audio codec driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL"); From e0431de301cbb8e3915261dfff4d0b072738de69 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Mon, 2 Jul 2018 07:17:07 -0500 Subject: [PATCH 198/529] ASoC: pxa-ssp: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..ff1e0bd8d407 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -470,6 +470,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; From 30896d3619bd80486a3f8a75d62ea3b58fc61ad5 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:50 -0600 Subject: [PATCH 199/529] ASoC: AMD: Always stop ch2 first Commit 6b116dfb4633a ("ASoC: AMD: make channel 1 dma as circular") made both channels circular, so this comment and logic no longer applies. Always stop ch2 (the channel closest to the output) before ch1. This ensures that the downstream circular DMA channel does not continue to play/capture repeated samples after the upstream circular DMA channel has already stopped. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3c3d398d0d0b..4665ae12e74e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1067,21 +1067,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->bytescount = 0; break; default: From 715cdce04487fb23d5c10693b3bc01309fea955a Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:52 -0600 Subject: [PATCH 200/529] ASoC: AMD: Always subtract bytescount It is always correct to subtract out the starting bytescount value. Even in the case of 2^64 byte rollover (292 Million Years in the future @ 48000 Hz) the math still works out. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 4665ae12e74e..034fac3de037 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -995,8 +995,7 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; + bytescount -= rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } From 55af49ac1b8627dfbfa2689af118d994d7a0ba1b Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:53 -0600 Subject: [PATCH 201/529] ASoC: AMD: Fix Capture DMA channel names On capture, audio data is first copied from I2S to ACP memory, and then to SYSRAM. For each step the channel number increases, so the names in the driver were wrong. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 30 +++++++++++++++--------------- sound/soc/amd/acp.h | 8 ++++---- 2 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 034fac3de037..df53412967e1 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,34 +697,34 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { - valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); - acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { valid_irq = true; + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); - acp_reg_write((intr_flag & - BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + acp_reg_write((intr_flag & + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if (valid_irq) return IRQ_HANDLED; else @@ -899,8 +899,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS; rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; @@ -914,8 +914,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, case I2S_SP_INSTANCE: default: rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_CH_NUM; switch (adata->asic_type) { case CHIP_STONEY: rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 3190fdce6307..0a2240bff62e 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -74,16 +74,16 @@ #define ACP_TO_I2S_DMA_CH_NUM 13 /* Capture DMA channels */ -#define ACP_TO_SYSRAM_CH_NUM 14 -#define I2S_TO_ACP_DMA_CH_NUM 15 +#define I2S_TO_ACP_DMA_CH_NUM 14 +#define ACP_TO_SYSRAM_CH_NUM 15 /* Playback DMA Channels for I2S BT instance */ #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 /* Capture DMA Channels for I2S BT Instance */ -#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 -#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10 +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11 #define NUM_DSCRS_PER_CHANNEL 2 From 8c6b964eddd2c39a9796899b2be099ece1b6c6ca Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:54 -0600 Subject: [PATCH 202/529] ASoC: AMD: Do not generate interrups for every captured sample On capture, audio data is first copied from I2S to ACP memory, and then from ACP to SYSRAM. The I2S_TO_ACP_DMA interrupt fires on every sample transferred from I2S to ACP memory. That is it fires ~48000 times per second when capturing @ 48 kHz. Since we don't do anything on this interrupt anyway, disable it to save quite a few unnecessary interrupts. The real "work" (calling snd_pcm_period_elapsed()) is done when transfer from ACP to SYSRAM is complete. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index df53412967e1..cd4d2520ac14 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -412,10 +412,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: case ACP_TO_SYSRAM_CH_NUM: - case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: - case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -704,12 +702,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); @@ -718,13 +710,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & - BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if (valid_irq) return IRQ_HANDLED; else From 1a337a1e7885085d224583c766614e5945bde671 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:51 -0600 Subject: [PATCH 203/529] ASoC: AMD: Reset bytescount when starting transaction The pointer() callback gets its value by reading the I2S BYTE_COUNT register. This is a 64-bit runnning transaction counter. If a transaction was aborted in the middle of a sample buffer, the counter will stop counting on a number divisible by the buffer size. Since we actually use it as a pointer into an aligned buffer, however, we do want to ensure that it always starts at a number divisible by the buffer size when starting a transaction, hence we reset it whenever starting a transaction. To accomplish this, it wasn't necessary to zero bytescount at the termination of each transaction, so remove this unnecessary code. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cd4d2520ac14..ab60129f4f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1013,7 +1013,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1024,9 +1023,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; + rtd->bytescount = acp_get_byte_count(rtd); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1); acp_dma_start(rtd->acp_mmio, rtd->ch2); @@ -1053,7 +1050,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - rtd->bytescount = 0; break; default: ret = -EINVAL; From df61f9f76609456efbc60d495b3235baf7d07691 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:55 -0600 Subject: [PATCH 204/529] ASoC: AMD: Simplify trigger handler Now that the I2S channel names are fixed, and DMA data flow order is consistent (ch1 then ch2), we can simplify channel start order: start the upstream channel and then the downstream channel for both playback and capture cases. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ab60129f4f26..65c1033bd51c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1024,10 +1024,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: rtd->bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); - } else { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { if (rtd->capture_channel == CAP_CHANNEL0) { acp_dma_cap_channel_disable(rtd->acp_mmio, CAP_CHANNEL1); @@ -1040,9 +1037,9 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } - acp_dma_start(rtd->acp_mmio, rtd->ch2); - acp_dma_start(rtd->acp_mmio, rtd->ch1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: From c9445d94d7fc60e7b0290df9e109daef85d26bd4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 17:05:58 +0200 Subject: [PATCH 205/529] ASoC: es7134: update DT binding with new compatible and supplies Update the documentation to add support for the es7154 and optional power supplies phandles. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/everest,es7134.txt | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/everest,es7134.txt b/Documentation/devicetree/bindings/sound/everest,es7134.txt index 5495a3cb8b7b..091666069bde 100644 --- a/Documentation/devicetree/bindings/sound/everest,es7134.txt +++ b/Documentation/devicetree/bindings/sound/everest,es7134.txt @@ -1,10 +1,15 @@ ES7134 i2s DA converter Required properties: -- compatible : "everest,es7134" or "everest,es7144" +- compatible : "everest,es7134" or + "everest,es7144" or + "everest,es7154" +- VDD-supply : regulator phandle for the VDD supply +- PVDD-supply: regulator phandle for the PVDD supply for the es7154 Example: i2s_codec: external-codec { compatible = "everest,es7134"; + VDD-supply = <&vcc_5v>; }; From 30ddfffd10b73d5d960650ea70b33a8ee0562679 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 17:05:59 +0200 Subject: [PATCH 206/529] ASoC: es7134: correct required power supplies Drop AVDD in favor of PVDD to match the names used in the datasheet and only claim PVDD on the es7154. The es7134 and es7144 don't have a separate supply for the digital I/O. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 44 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 41 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 5ad59c38fed1..80f2936cefed 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -35,6 +35,10 @@ struct es7134_clock_mode { struct es7134_chip { const struct es7134_clock_mode *modes; unsigned int mode_num; + const struct snd_soc_dapm_widget *extra_widgets; + unsigned int extra_widget_num; + const struct snd_soc_dapm_route *extra_routes; + unsigned int extra_route_num; }; struct es7134_data { @@ -110,6 +114,34 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int es7134_component_probe(struct snd_soc_component *c) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c); + struct es7134_data *priv = snd_soc_component_get_drvdata(c); + const struct es7134_chip *chip = priv->chip; + int ret; + + if (chip->extra_widget_num) { + ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets, + chip->extra_widget_num); + if (ret) { + dev_err(c->dev, "failed to add extra widgets\n"); + return ret; + } + } + + if (chip->extra_route_num) { + ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes, + chip->extra_route_num); + if (ret) { + dev_err(c->dev, "failed to add extra routes\n"); + return ret; + } + } + + return 0; +} + static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, .hw_params = es7134_hw_params, @@ -158,9 +190,16 @@ static const struct es7134_clock_mode es7134_modes[] = { }, }; +/* Digital I/O are also supplied by VDD on the es7134 */ +static const struct snd_soc_dapm_route es7134_extra_routes[] = { + { "Playback", NULL, "VDD", } +}; + static const struct es7134_chip es7134_chip = { .modes = es7134_modes, .mode_num = ARRAY_SIZE(es7134_modes), + .extra_routes = es7134_extra_routes, + .extra_route_num = ARRAY_SIZE(es7134_extra_routes), }; static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { @@ -168,17 +207,16 @@ static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, - { "Playback", NULL, "VDD" }, - { "DAC", NULL, "AVDD" }, + { "DAC", NULL, "VDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { + .probe = es7134_component_probe, .dapm_widgets = es7134_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets), .dapm_routes = es7134_dapm_routes, From 563c263248ff37dcd743549a0c0932fe2bf83980 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 17:06:00 +0200 Subject: [PATCH 207/529] ASoC: es7134: add support for the es7154 Add support for the es7154 which is basically an es7134 with an embedded power amplifier and lower maximum sample rate Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 60 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 59 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 80f2936cefed..6d7bca7b78ca 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -33,6 +33,7 @@ struct es7134_clock_mode { }; struct es7134_chip { + struct snd_soc_dai_driver *dai_drv; const struct es7134_clock_mode *modes; unsigned int mode_num; const struct snd_soc_dapm_widget *extra_widgets; @@ -196,6 +197,7 @@ static const struct snd_soc_dapm_route es7134_extra_routes[] = { }; static const struct es7134_chip es7134_chip = { + .dai_drv = &es7134_dai, .modes = es7134_modes, .mode_num = ARRAY_SIZE(es7134_modes), .extra_routes = es7134_extra_routes, @@ -227,6 +229,61 @@ static const struct snd_soc_component_driver es7134_component_driver = { .non_legacy_dai_naming = 1, }; +static struct snd_soc_dai_driver es7154_dai = { + .name = "es7154-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S18_3LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7134_dai_ops, +}; + +static const struct es7134_clock_mode es7154_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256, + 384, 512, 768, 1024 }, + .mclk_fs_num = 9, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512, + 768, 1024}, + .mclk_fs_num = 7, + } +}; + +/* Es7154 has a separate supply for digital I/O */ +static const struct snd_soc_dapm_widget es7154_extra_widgets[] = { + SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0), +}; + +static const struct snd_soc_dapm_route es7154_extra_routes[] = { + { "Playback", NULL, "PVDD", } +}; + +static const struct es7134_chip es7154_chip = { + .dai_drv = &es7154_dai, + .modes = es7154_modes, + .mode_num = ARRAY_SIZE(es7154_modes), + .extra_routes = es7154_extra_routes, + .extra_route_num = ARRAY_SIZE(es7154_extra_routes), + .extra_widgets = es7154_extra_widgets, + .extra_widget_num = ARRAY_SIZE(es7154_extra_widgets), +}; + static int es7134_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -245,13 +302,14 @@ static int es7134_probe(struct platform_device *pdev) return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, - &es7134_dai, 1); + priv->chip->dai_drv, 1); } #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { { .compatible = "everest,es7134", .data = &es7134_chip }, { .compatible = "everest,es7144", .data = &es7134_chip }, + { .compatible = "everest,es7154", .data = &es7154_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); From 73ad0df572901e03fc703b6f114e4442291f45c2 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Tue, 3 Jul 2018 17:56:30 +0300 Subject: [PATCH 208/529] ASoC: atmel-i2s: Remove unnecessary audio PLL clock (aclk) The generated clock (gclk) driver is able to set aclk as its parent and change its rate alone, if needed. This means that our driver no longer needs to configure aclk and we can let gclk select and configure its clock source. Signed-off-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-i2s.c | 46 ++++++------------------------------- 1 file changed, 7 insertions(+), 39 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 5d3b5af9fd92..d88c1d995036 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -206,7 +206,6 @@ struct atmel_i2s_dev { struct regmap *regmap; struct clk *pclk; struct clk *gclk; - struct clk *aclk; struct snd_dmaengine_dai_dma_data playback; struct snd_dmaengine_dai_dma_data capture; unsigned int fmt; @@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs) { int i, best; - if (!dev->gclk || !dev->aclk) { + if (!dev->gclk) { dev_err(dev->dev, "cannot generate the I2S Master Clock\n"); return -EINVAL; } @@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, bool enabled) { unsigned int mr, mr_mask; - unsigned long aclk_rate; + unsigned long gclk_rate; int ret; mr = 0; @@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, /* Disable/unprepare the PMC generated clock. */ clk_disable_unprepare(dev->gclk); - /* Disable/unprepare the PLL audio clock. */ - clk_disable_unprepare(dev->aclk); return 0; } if (!dev->gck_param) return -EINVAL; - aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); + gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); - /* Fist change the PLL audio clock frequency ... */ - ret = clk_set_rate(dev->aclk, aclk_rate); + ret = clk_set_rate(dev->gclk, gclk_rate); if (ret) return ret; - /* - * ... then set the PMC generated clock rate to the very same frequency - * to set the gclk parent to aclk. - */ - ret = clk_set_rate(dev->gclk, aclk_rate); - if (ret) - return ret; - - /* Prepare and enable the PLL audio clock first ... */ - ret = clk_prepare_enable(dev->aclk); - if (ret) - return ret; - - /* ... then prepare and enable the PMC generated clock. */ ret = clk_prepare_enable(dev->gclk); if (ret) return ret; @@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev) return err; } - /* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */ - dev->aclk = devm_clk_get(&pdev->dev, "aclk"); + /* Get audio clock to generate the I2S Master Clock (I2S_MCK) */ dev->gclk = devm_clk_get(&pdev->dev, "gclk"); - if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) { - if (PTR_ERR(dev->aclk) == -EPROBE_DEFER || - PTR_ERR(dev->gclk) == -EPROBE_DEFER) + if (IS_ERR(dev->gclk)) { + if (PTR_ERR(dev->gclk) == -EPROBE_DEFER) return -EPROBE_DEFER; /* Master Mode not supported */ - dev->aclk = NULL; dev->gclk = NULL; - } else if (IS_ERR(dev->gclk)) { - err = PTR_ERR(dev->gclk); - dev_err(&pdev->dev, - "failed to get the PMC generated clock: %d\n", err); - return err; - } else if (IS_ERR(dev->aclk)) { - err = PTR_ERR(dev->aclk); - dev_err(&pdev->dev, - "failed to get the PLL audio clock: %d\n", err); - return err; } - dev->dev = &pdev->dev; dev->regmap = regmap; platform_set_drvdata(pdev, dev); From 2bd368d7bf21028f37a123041a138922254d4840 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Tue, 3 Jul 2018 17:56:31 +0300 Subject: [PATCH 209/529] ASoC: atmel-i2s: dt-bindings: Remove unnecessary phandle to aclk The optional clock phandle to aclk (Audio PLL clock) is no longer needed by the driver. Signed-off-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/atmel-i2s.txt | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt index 735368b8a73f..40549f496a81 100644 --- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt +++ b/Documentation/devicetree/bindings/sound/atmel-i2s.txt @@ -15,7 +15,6 @@ Required properties: - clock-names: Should be one of each entry matching the clocks phandles list: - "pclk" (peripheral clock) Required. - "gclk" (generated clock) Optional (1). - - "aclk" (Audio PLL clock) Optional (1). - "muxclk" (I2S mux clock) Optional (1). Optional properties: @@ -23,9 +22,9 @@ Optional properties: - princtrl-names: Should contain only one value - "default". -(1) : Only the peripheral clock is required. The generated clock, the Audio - PLL clock adn the I2S mux clock are optional and should only be set - together, when Master Mode is required. +(1) : Only the peripheral clock is required. The generated clock and the I2S + mux clock are optional and should only be set together, when Master Mode + is required. Example: @@ -40,8 +39,8 @@ Example: (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | AT91_XDMAC_DT_PERID(32))>; dma-names = "tx", "rx"; - clocks = <&i2s0_clk>, <&i2s0_gclk>, <&audio_pll_pmc>, <&i2s0muxck>; - clock-names = "pclk", "gclk", "aclk", "muxclk"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2s0_default>; }; From a655de808cbde6c58b3298e704d786b53f59fb5d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 2 Jul 2018 16:59:54 +0100 Subject: [PATCH 210/529] ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 13 ++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ 3 files changed, 123 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 16f0bf10cc42..870ba6b64817 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -803,6 +803,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -960,6 +968,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -999,6 +1010,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1008,6 +1020,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 68b08781c832..00bd58d167dd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -847,6 +847,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1456,7 +1459,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1502,9 +1507,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1514,7 +1538,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1841,6 +1865,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1850,6 +1942,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c2a31b51da4f..b7e67b871c0c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -859,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { From 9cd641ed31f576d08f7b784850ba93eef050f32f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 14:46:27 +0200 Subject: [PATCH 211/529] ALSA: pcm: trace XRUN event at injection, too The PCM xrun injection triggers directly snd_pcm_stop() without the standard xrun handler, hence it's not recorded on the event buffer. Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl. They are inconvenient from the debugging POV. Let's make them to trigger XRUN via the standard helper more consistently. Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 2 +- sound/core/pcm_lib.c | 7 ++++--- sound/core/pcm_local.h | 2 ++ sound/core/pcm_native.c | 8 ++++---- 4 files changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c352bfb973cc..6f037a4b8b52 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -497,7 +497,7 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, snd_pcm_stream_lock_irq(substream); runtime = substream->runtime; if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irq(substream); } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..c1d2e8e1fc6b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name); dump_stack(); \ } while (0) -static void xrun(struct snd_pcm_substream *substream) +/* call with stream lock held */ +void __snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, } } else { if (avail >= runtime->stop_threshold) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } } @@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } if (pos >= runtime->buffer_size) { diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 7a499d02df6c..c515612969a4 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} #endif +void __snd_pcm_xrun(struct snd_pcm_substream *substream); + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cecc79772c94..20174d0c0527 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { unsigned long flags; - int ret = 0; snd_pcm_stream_lock_irqsave(substream, flags); if (snd_pcm_running(substream)) - ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) result = 0; /* already there */ break; case SNDRV_PCM_STATE_RUNNING: - result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); + result = 0; break; default: result = -EBADFD; From e647f5a5c5d165c87750e8c0dcbe341b5a378ffd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 15:08:05 +0200 Subject: [PATCH 212/529] ALSA: pcm: Use snd_pcm_stop_xrun() for xrun injection Basically the xrun injection routine can simply call the standard helper snd_pcm_stop_xrun(), but with one exception: it may be called even when the stream is closed. Make snd_pcm_stop_xrun() more robust and check the NULL runtime state, and simplify xrun injection code by calling it. Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 7 +------ sound/core/pcm_native.c | 2 +- 2 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6f037a4b8b52..fdb9b92fc8d6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -492,13 +492,8 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime; - snd_pcm_stream_lock_irq(substream); - runtime = substream->runtime; - if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - __snd_pcm_xrun(substream); - snd_pcm_stream_unlock_irq(substream); + snd_pcm_stop_xrun(substream); } static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 20174d0c0527..66c90f486af9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1339,7 +1339,7 @@ int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) unsigned long flags; snd_pcm_stream_lock_irqsave(substream, flags); - if (snd_pcm_running(substream)) + if (substream->runtime && snd_pcm_running(substream)) __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; From 110743189c863e96dc08a581d56c50b965870a3f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:43 +0100 Subject: [PATCH 213/529] ASoC: qdsp6: q6afe-dai: do not close port if its not opened afe ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..a373ca5523ff 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -315,6 +315,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); int rc; + if (!dai_data->is_port_started[dai->id]) + return; + rc = q6afe_port_stop(dai_data->port[dai->id]); if (rc < 0) dev_err(dai->dev, "fail to close AFE port (%d)\n", rc); From 5dffc1752cabde6396fca28ff8343febfa524512 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:44 +0100 Subject: [PATCH 214/529] ASoC: qdsp6: q6asm-dai: do not close port if its not opened asm ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..360936703b3d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -390,7 +390,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream) struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + if (prtd->state) + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6asm_audio_client_free(prtd->audio_client); From da13ed1d80fe6a4d95043aaf2e0aff292ade5708 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 4 Jul 2018 09:28:28 -0500 Subject: [PATCH 215/529] ASoC: nau8825: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea4987b7d..b9fed99d8b5e 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; From dae35d1f4f7dab9ccef20037df91c43e680bad0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:43 +0200 Subject: [PATCH 216/529] ASoC: davinci: Use snd_pcm_stop_xrun() helper Replace open-codes with the standard snd_pcm_stop_xrun() helper. It simplifies codes a lot. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 47c0c821d325..f70db8412c7c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) handled_mask |= XUNDRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) @@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) handled_mask |= ROVRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) From 1a42e7e3aff1aa4789378020318dff7432317d25 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:44 +0200 Subject: [PATCH 217/529] ASoC: qcom: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis, too. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 31fe78aa207f..d07271ea4c45 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return IRQ_NONE; } dev_warn(soc_runtime->dev, "xrun warning\n"); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(substream); ret = IRQ_HANDLED; } From dc865fb9e7c2251c9585ff6a7bf185d499db13e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:45 +0200 Subject: [PATCH 218/529] ASoC: sti: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis, too. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e..313dab2857ef 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index ee0055e60852..7b63d35ef428 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(reader->substream); ret = IRQ_HANDLED; } From b1625fbb3b87affbedf14545b65d69ff182a0611 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:46 +0200 Subject: [PATCH 219/529] ASoC: stm32: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It simplifies the locking as well. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index c4f15ea14197..06fba9650ac4 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -300,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) status = SNDRV_PCM_STATE_XRUN; } - if (status != SNDRV_PCM_STATE_RUNNING) { - snd_pcm_stream_lock(sai->substream); - snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(sai->substream); - } + if (status != SNDRV_PCM_STATE_RUNNING) + snd_pcm_stop_xrun(sai->substream); return IRQ_HANDLED; } From 25090bc3f36cc3c171ec020dcc89c71db6bd0a67 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:39 +0100 Subject: [PATCH 220/529] ASoC: qdsp6: q6afe: Add missing slimbus capture ports Existing code already has support for SLIMbus TX and RX, only thing that was missing from TX side was mapping between virtual to actual DSP port ids. This patch adds those mappings. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 671743453fbb..000775b4bba8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -514,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { SLIMBUS_5_RX, 1, 1}, [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, SLIMBUS_6_RX, 1, 1}, + [SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX, + SLIMBUS_0_TX, 0, 1}, + [SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX, + SLIMBUS_1_TX, 0, 1}, + [SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX, + SLIMBUS_2_TX, 0, 1}, + [SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX, + SLIMBUS_3_TX, 0, 1}, + [SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX, + SLIMBUS_4_TX, 0, 1}, + [SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX, + SLIMBUS_5_TX, 0, 1}, + [SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX, + SLIMBUS_6_TX, 0, 1}, [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX, PRIMARY_MI2S_RX, 1, 1}, [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX, @@ -1372,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_MULTICHAN_HDMI_RX: cfg_type = AFE_PARAM_ID_HDMI_CONFIG; break; + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX: From f03d6b1b4d2460a749fb2826aa71e15a66104a88 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:40 +0100 Subject: [PATCH 221/529] ASoC: qdsp6: q6afe-dai: add support to slim tx dais This patch adds support to SLIMbus TX dais in AFE module. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 179 ++++++++++++++++++++++++++++--- 1 file changed, 163 insertions(+), 16 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 1d2e5013c121..8dd3683eb367 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -382,24 +382,32 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai, struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id]; int i; - if (!rx_slot) { - pr_err("%s: rx slot not found\n", __func__); - return -EINVAL; + if (dai->id & 0x1) { + /* TX */ + if (!tx_slot) { + pr_err("%s: tx slot not found\n", __func__); + return -EINVAL; + } + + for (i = 0; i < tx_num; i++) + pcfg->slim.ch_mapping[i] = tx_slot[i]; + + pcfg->slim.num_channels = tx_num; + + + } else { + if (!rx_slot) { + pr_err("%s: rx slot not found\n", __func__); + return -EINVAL; + } + + for (i = 0; i < rx_num; i++) + pcfg->slim.ch_mapping[i] = rx_slot[i]; + + pcfg->slim.num_channels = rx_num; + } - for (i = 0; i < rx_num; i++) { - pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: find number of channels[%d] ch[%d]\n", - __func__, i, rx_slot[i]); - } - - pcfg->slim.num_channels = rx_num; - - pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__, - (dai->id - SLIMBUS_0_RX) / 2, rx_num, - pcfg->slim.ch_mapping[0], - pcfg->slim.ch_mapping[1]); - return 0; } @@ -443,6 +451,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"}, {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"}, + {"SLIMBUS_0_TX", NULL, "Slimbus Capture"}, + {"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"}, + {"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"}, + {"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"}, + {"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"}, + {"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"}, + {"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"}, + {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"}, {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"}, {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, @@ -636,6 +652,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rate_min = 8000, .rate_max = 192000, }, + }, { + .name = "SLIMBUS_0_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_0_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus1 Playback", @@ -654,6 +688,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_1_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + }, { + .name = "SLIMBUS_1_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_1_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus1 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus2 Playback", @@ -672,6 +724,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_2_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_2_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_2_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus2 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus3 Playback", @@ -690,6 +761,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_3_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_3_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_3_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus3 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus4 Playback", @@ -708,6 +798,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_4_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_4_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_4_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus4 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus5 Playback", @@ -726,6 +835,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_5_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_5_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_5_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus5 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus6 Playback", @@ -744,6 +872,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_6_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_6_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_6_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus6 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Primary MI2S Playback", From 9191ffe2d212f64aa2ec311f4294ba7066d1f8a1 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:41 +0100 Subject: [PATCH 222/529] ASoC: qdsp6: q6routing: add slim rx routings This patch add routings mixer controls for slim rx ports. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index c80fdbc2442e..35269b492761 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -68,6 +68,13 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ + { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ + { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ + { mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" }, \ + { mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" }, \ + { mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" }, \ + { mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" }, \ { mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"}, \ { mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"}, \ { mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"}, \ @@ -122,6 +129,27 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ From f1478a1476d45c5d7b070c726f46a7b581d3103c Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:42 +0100 Subject: [PATCH 223/529] ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels num_channels for slim dais are aready set int set_channel_map, do not overwrite them in hw_params. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8dd3683eb367..074582afda85 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -81,7 +81,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; - slim->num_channels = params_channels(params); slim->sample_rate = params_rate(params); switch (params_format(params)) { From b999a7a9e72bd2d37b5d03772cedfc4dd45875bf Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 4 Jul 2018 09:18:33 -0500 Subject: [PATCH 224/529] ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 64 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: rate[index] * txclk_df * 64 Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87b61bf..740b90df44bb 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else From 74b37e299f038c910dc728d736e3f071ba0ead2a Mon Sep 17 00:00:00 2001 From: Andrew Gabbasov Date: Thu, 5 Jul 2018 11:20:04 +0900 Subject: [PATCH 225/529] ASoC: rsnd: cmd: Add missing newline to debug message To comply with the style of all kernel messages, add newline to the end of every message. Fixes: 70fb10529f61 ("ASoC: rsnd: add MIX (Mixer) support") Signed-off-by: Andrew Gabbasov Signed-off-by: Jiada Wang Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/cmd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index d8043ad33540..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -86,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); From 90eb6b59d311e6facd040124cb5b659a865125b8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 2 Jul 2018 17:11:00 +0200 Subject: [PATCH 226/529] ASoC: pxa-ssp: add support for an external clock in devicetree Allow setting a clock called 'extclk' in the device of the ssp-dai device. If specified, this clock will be set to the mclk rate from the DAI's .set_sysclk() callback. The DAI will also configure itself to use that external clock. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../bindings/sound/mrvl,pxa-ssp.txt | 8 ++++++ sound/soc/pxa/pxa-ssp.c | 25 +++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index 74c9ba6c2823..93b982e9419f 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ff1e0bd8d407..69033e1a84e6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; @@ -205,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -774,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { From f7ddff54d0a0f068442414b48bec7f22aa777de7 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 5 Jul 2018 08:06:17 -0500 Subject: [PATCH 227/529] ASoC: nau8824: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd14453f06e..468d5143e2c4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; From d64c5cf8e89d124355924c513a42b16f0d7d3a03 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 6 Jul 2018 13:50:36 +0100 Subject: [PATCH 228/529] ALSA: pcm: Allow drivers to set R/W wait time. Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index c1d2e8e1fc6b..5736860f325b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1833,12 +1833,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { From 8db339d66774821091f73bd0e57c8a7511c5988e Mon Sep 17 00:00:00 2001 From: "benjamin.gaignard@linaro.org" Date: Fri, 6 Jul 2018 15:07:03 +0200 Subject: [PATCH 229/529] ASoC: stm32: replace "%p" with "%pK" The format specifier "%p" can leak kernel addresses. Use "%pK" instead. Signed-off-by: Benjamin Gaignard Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..0e9373064032 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { From 9d1310daedae494874fe36a995172e64b06e29a4 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 6 Jul 2018 15:30:48 +0200 Subject: [PATCH 230/529] ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are building for ARM PXA or MMP: WARNING: unmet direct dependencies detected for PXA_SSP Depends on [n]: PLAT_PXA [=n] Selected by [y]: - SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] This adds an explicit dependency for it. Fixes: 0a94cf345740 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 2fc02c227f69..776e148b0aa2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -34,6 +34,7 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA select PXA_SSP select SND_PXA2XX_LIB From 5bea327962fa296efd16f2d3369dd339ddd7ce6f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Jul 2018 16:19:14 +0300 Subject: [PATCH 231/529] ASoC: adau171x1: Connect playback DAI to the DSP The playback DAI is connected to the DSP and the DSP might be sourcing signals from the playback stream. Add a DAPM route between the two to make sure that the playback DAI is powered up, when the DSP is active. Signed-off-by: Lars-Peter Clausen Signed-off-by: Alexandru Ardelean Signed-off-by: Mark Brown --- sound/soc/codecs/adau17x1.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index ae41edd1c406..57169b8ff14e 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { { "DSP", NULL, "Left Decimator" }, { "DSP", NULL, "Right Decimator" }, + { "DSP", NULL, "Playback" }, }; static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { From 81583afe790c5bd86300537783b23f3b12794f03 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 7 Jul 2018 12:22:10 +0200 Subject: [PATCH 232/529] ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital mic on DMIC1 and has a mono-speaker. The jack-detect uses the default settings.. Reported-and-tested-by: russianneuromancer@ya.ru Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7456566c5648..657910a08261 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -552,6 +552,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 2 8 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), From fbea16dbc0a31484811c5f335ae344b2bbc66f40 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:29 +0200 Subject: [PATCH 233/529] ASoC: Intel: bytcr_rt5651: Remove is_valleyview helper Remove is_valleyview helper, this is not necessary, we can simply call x86_match_cpu() directly instead. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index ba2753e0e12a..80f47a45cb10 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -680,17 +681,10 @@ static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ -static bool is_valleyview(void) -{ - static const struct x86_cpu_id cpu_ids[] = { - { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ - {} - }; - - if (!x86_match_cpu(cpu_ids)) - return false; - return true; -} +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + {} +}; struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ @@ -741,7 +735,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) */ - if (is_valleyview()) { + if (x86_match_cpu(baytrail_cpu_ids)) { struct sst_platform_info *p_info = mach->pdata; const struct sst_res_info *res_info = p_info->res_info; From 2c375204bfad2f481feb006a82cdb67cc570b670 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:30 +0200 Subject: [PATCH 234/529] ASoC: Intel: bytcr_rt5651: Move getting of codec_dev into probe() Move the getting of the codec_dev, to add device-props to it, out of byt_rt5651_add_codec_device_props() and into its caller, snd_byt_rt5651_mc_probe(). This is a preparation patch for adding support for an external amplifier enable GPIO, which requires further accesses to the codec_dev. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 80f47a45cb10..d920725ce603 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -403,15 +403,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) +static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) { struct property_entry props[MAX_NO_PROPS] = {}; - struct device *i2c_dev; - int ret, cnt = 0; - - i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name); - if (!i2c_dev) - return -EPROBE_DEFER; + int cnt = 0; props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -425,10 +420,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en"); - ret = device_add_properties(i2c_dev, props); - put_device(i2c_dev); - - return ret; + return device_add_properties(i2c_dev, props); } static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) @@ -696,6 +688,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) const char * const mic_name[] = { "dmic", "in1", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; + struct device *codec_dev; const char *i2c_name = NULL; const char *hp_swapped; bool is_bytcr = false; @@ -731,6 +724,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) "%s%s", "i2c-", i2c_name); byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name; + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + byt_rt5651_codec_name); + if (!codec_dev) + return -EPROBE_DEFER; + /* * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) @@ -794,7 +792,8 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) dmi_check_system(byt_rt5651_quirk_table); /* Must be called before register_card, also see declaration comment. */ - ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name); + ret_val = byt_rt5651_add_codec_device_props(codec_dev); + put_device(codec_dev); if (ret_val) return ret_val; From 5f6fb23d2e114506ddb8437a3e65f4a20d081013 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:31 +0200 Subject: [PATCH 235/529] ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO The rt5651 does not have a built-in speaker amplifier, so it is often used together with an external amplifier. On Cherry Trail boards this external amplifier's enable pin is driven through a GPIO, which is given as the first GPIO in the ACPI resources of the codec fwnode. This commit adds support to the bytcr_rt5651 for this GPIO, fixing the speaker not working on CHT devices with a rt5651 codec. Cc: Carlo Caione Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 65 +++++++++++++++++++++++++-- 1 file changed, 62 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d920725ce603..5301205496be 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include #include @@ -86,6 +88,7 @@ enum { struct byt_rt5651_private { struct clk *mclk; + struct gpio_desc *ext_amp_gpio; struct snd_soc_jack jack; }; @@ -208,6 +211,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->ext_amp_gpio, 1); + else + gpiod_set_value_cansleep(priv->ext_amp_gpio, 0); + + return 0; +} + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -217,7 +234,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - + SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0, + rt5651_ext_amp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -225,6 +244,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Speaker", NULL, "Ext Amp Power"}, {"Line In", NULL, "Platform Clock"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ @@ -678,6 +698,18 @@ static const struct x86_cpu_id baytrail_cpu_ids[] = { {} }; +static const struct x86_cpu_id cherrytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */ + {} +}; + +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping byt_rt5651_gpios[] = { + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1 }, + { }, +}; + struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -793,9 +825,36 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Must be called before register_card, also see declaration comment. */ ret_val = byt_rt5651_add_codec_device_props(codec_dev); - put_device(codec_dev); - if (ret_val) + if (ret_val) { + put_device(codec_dev); return ret_val; + } + + /* Cherry Trail devices use an external amplifier enable gpio */ + if (x86_match_cpu(cherrytrail_cpu_ids)) { + devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); + priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( + &pdev->dev, "ext-amp-enable", 0, + codec_dev->fwnode, + GPIOD_OUT_LOW, "speaker-amp"); + if (IS_ERR(priv->ext_amp_gpio)) { + ret_val = PTR_ERR(priv->ext_amp_gpio); + switch (ret_val) { + case -ENOENT: + priv->ext_amp_gpio = NULL; + break; + default: + dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", + ret_val); + /* fall through */ + case -EPROBE_DEFER: + put_device(codec_dev); + return ret_val; + } + } + } + + put_device(codec_dev); log_quirks(&pdev->dev); From 8d2d7bcdc1645dc243f7735278675b083c0e506c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:31 +0200 Subject: [PATCH 236/529] ASoC: rt5651: Fix workqueue cancel vs irq free race on remove On removal we must free the IRQ *before* cancelling the jack-detect work, so that the jack-detect work cannot be rescheduled by the IRQ. Before this commit we were cancelling the jack-detect work from the driver remove callback, while relying on devm to free the IRQ, which happens after the remove callback. This is the wrong order. This commit uses a devm-action to register a devm callback which cancels the work, before requesting the IRQ (devm tears things down in reverse order). This also allows us to remove the now empty remove driver callback. Cc: Carlo Caione Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 6b5669f3e85d..39d2c67cd064 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1696,6 +1696,13 @@ static irqreturn_t rt5651_irq(int irq, void *data) return IRQ_HANDLED; } +static void rt5651_cancel_work(void *data) +{ + struct rt5651_priv *rt5651 = data; + + cancel_work_sync(&rt5651->jack_detect_work); +} + static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { @@ -2036,6 +2043,11 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); + /* Make sure work is stopped on probe-error / remove */ + ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651); + if (ret) + return ret; + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); @@ -2043,15 +2055,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, return ret; } -static int rt5651_i2c_remove(struct i2c_client *i2c) -{ - struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c); - - cancel_work_sync(&rt5651->jack_detect_work); - - return 0; -} - static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", @@ -2059,7 +2062,6 @@ static struct i2c_driver rt5651_i2c_driver = { .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, - .remove = rt5651_i2c_remove, .id_table = rt5651_i2c_id, }; module_i2c_driver(rt5651_i2c_driver); From 34c906ddacd237511808fb2bbd941e6b91e9095a Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:32 +0200 Subject: [PATCH 237/529] ASoC: rt5651: Allow disabling jack-detect by calling set_jack(NULL) Allow the machine driver to disable jack-detect over a suspend/resume by calling snd_soc_component_set_jack(NULL). Note this renames rt5651_set_jack, where all the jack-enable work was done to rt5651_enable_jack_detect. This function can now no longer fail as it does not request the IRQ anymore. It can still be passed an invalid jack source, but that should never happen, so this is now logged and treated as no jack source. Cc: Carlo Caione Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 55 +++++++++++++++++++++++++-------------- 1 file changed, 36 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 39d2c67cd064..40bd1e70fee7 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1703,14 +1703,10 @@ static void rt5651_cancel_work(void *data) cancel_work_sync(&rt5651->jack_detect_work); } -static int rt5651_set_jack(struct snd_soc_component *component, - struct snd_soc_jack *hp_jack, void *data) +static void rt5651_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hp_jack) { struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); - int ret; - - if (!rt5651->irq) - return -EINVAL; /* IRQ output on GPIO1 */ snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, @@ -1737,10 +1733,10 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); break; case RT5651_JD_NULL: - return 0; + return; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); - return -EINVAL; + return; } /* Enable jack detect power */ @@ -1774,19 +1770,28 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; - - ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL, - rt5651_irq, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, "rt5651", rt5651); - if (ret) { - dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret); - return ret; - } - + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); +} + +static void rt5651_disable_jack_detect(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + disable_irq(rt5651->irq); + rt5651_cancel_work(rt5651); + + rt5651->hp_jack = NULL; +} + +static int rt5651_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + rt5651_enable_jack_detect(component, jack); + else + rt5651_disable_jack_detect(component); return 0; } @@ -2048,6 +2053,18 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, if (ret) return ret; + ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5651", rt5651); + if (ret == 0) { + /* Gets re-enabled by rt5651_set_jack() */ + disable_irq(rt5651->irq); + } else { + dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + rt5651->irq, ret); + rt5651->irq = -ENXIO; + } + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); From df1569f2006b157caa944367d0d431eb4ea08624 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:33 +0200 Subject: [PATCH 238/529] ASoC: rt5651: Add button press support Enable button press detection for headsets by using the ovcd IRQ to get notified of button presses. This is modelled after (almost exactly copied from) the button press code for the rt5640 which has identical ovcd hardware. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 158 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt5651.h | 8 ++ 2 files changed, 159 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 40bd1e70fee7..0462049e739c 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1581,6 +1581,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component snd_soc_dapm_mutex_unlock(dapm); } +static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR); + rt5651->ovcd_irq_enabled = true; +} + +static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP); + rt5651->ovcd_irq_enabled = false; +} + static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component) { snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, @@ -1622,10 +1640,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) return val == 0; } -/* Jack detect timings */ +/* Jack detect and button-press timings */ #define JACK_SETTLE_TIME 100 /* milli seconds */ #define JACK_DETECT_COUNT 5 #define JACK_DETECT_MAXCOUNT 20 /* Aprox. 2 seconds worth of tries */ +#define JACK_UNPLUG_TIME 80 /* milli seconds */ +#define BP_POLL_TIME 10 /* milli seconds */ +#define BP_POLL_MAXCOUNT 200 /* assume something is wrong after this */ +#define BP_THRESHOLD 3 + +static void rt5651_start_button_press_work(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + rt5651->poll_count = 0; + rt5651->press_count = 0; + rt5651->release_count = 0; + rt5651->pressed = false; + rt5651->press_reported = false; + rt5651_clear_micbias1_ovcd(component); + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} + +static void rt5651_button_press_work(struct work_struct *work) +{ + struct rt5651_priv *rt5651 = + container_of(work, struct rt5651_priv, bp_work.work); + struct snd_soc_component *component = rt5651->component; + + /* Check the jack was not removed underneath us */ + if (!rt5651_jack_inserted(component)) + return; + + if (rt5651_micbias1_ovcd(component)) { + rt5651->release_count = 0; + rt5651->press_count++; + /* Remember till after JACK_UNPLUG_TIME wait */ + if (rt5651->press_count >= BP_THRESHOLD) + rt5651->pressed = true; + rt5651_clear_micbias1_ovcd(component); + } else { + rt5651->press_count = 0; + rt5651->release_count++; + } + + /* + * The pins get temporarily shorted on jack unplug, so we poll for + * at least JACK_UNPLUG_TIME milli-seconds before reporting a press. + */ + rt5651->poll_count++; + if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) { + schedule_delayed_work(&rt5651->bp_work, + msecs_to_jiffies(BP_POLL_TIME)); + return; + } + + if (rt5651->pressed && !rt5651->press_reported) { + dev_dbg(component->dev, "headset button press\n"); + snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + rt5651->press_reported = true; + } + + if (rt5651->release_count >= BP_THRESHOLD) { + if (rt5651->press_reported) { + dev_dbg(component->dev, "headset button release\n"); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + /* Re-enable OVCD IRQ to detect next press */ + rt5651_enable_micbias1_ovcd_irq(component); + return; /* Stop polling */ + } + + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} static int rt5651_detect_headset(struct snd_soc_component *component) { @@ -1676,15 +1764,58 @@ static void rt5651_jack_detect_work(struct work_struct *work) { struct rt5651_priv *rt5651 = container_of(work, struct rt5651_priv, jack_detect_work); + struct snd_soc_component *component = rt5651->component; int report = 0; - if (rt5651_jack_inserted(rt5651->component)) { - rt5651_enable_micbias1_for_ovcd(rt5651->component); - report = rt5651_detect_headset(rt5651->component); - rt5651_disable_micbias1_for_ovcd(rt5651->component); - } + if (!rt5651_jack_inserted(component)) { + /* Jack removed, or spurious IRQ? */ + if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) { + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + cancel_delayed_work_sync(&rt5651->bp_work); + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + } + snd_soc_jack_report(rt5651->hp_jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + dev_dbg(component->dev, "jack unplugged\n"); + } + } else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) { + /* Jack inserted */ + WARN_ON(rt5651->ovcd_irq_enabled); + rt5651_enable_micbias1_for_ovcd(component); + report = rt5651_detect_headset(component); + if (report == SND_JACK_HEADSET) { + /* Enable ovcd IRQ for button press detect. */ + rt5651_enable_micbias1_ovcd_irq(component); + } else { + /* No more need for overcurrent detect. */ + rt5651_disable_micbias1_for_ovcd(component); + } + dev_dbg(component->dev, "detect report %#02x\n", report); + snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) { + dev_dbg(component->dev, "OVCD IRQ\n"); - snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + /* + * The ovcd IRQ keeps firing while the button is pressed, so + * we disable it and start polling the button until released. + * + * The disable will make the IRQ pin 0 again and since we get + * IRQs on both edges (so as to detect both jack plugin and + * unplug) this means we will immediately get another IRQ. + * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP. + */ + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_start_button_press_work(component); + + /* + * If the jack-detect IRQ flag goes high (unplug) after our + * above rt5651_jack_inserted() check and before we have + * disabled the OVCD IRQ, the IRQ pin will stay high and as + * we react to edges, we miss the unplug event -> recheck. + */ + queue_work(system_long_wq, &rt5651->jack_detect_work); + } } static irqreturn_t rt5651_irq(int irq, void *data) @@ -1701,6 +1832,7 @@ static void rt5651_cancel_work(void *data) struct rt5651_priv *rt5651 = data; cancel_work_sync(&rt5651->jack_detect_work); + cancel_delayed_work_sync(&rt5651->bp_work); } static void rt5651_enable_jack_detect(struct snd_soc_component *component, @@ -1770,6 +1902,11 @@ static void rt5651_enable_jack_detect(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_enable_micbias1_for_ovcd(component); + rt5651_enable_micbias1_ovcd_irq(component); + } + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); @@ -1782,6 +1919,12 @@ static void rt5651_disable_jack_detect(struct snd_soc_component *component) disable_irq(rt5651->irq); rt5651_cancel_work(rt5651); + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + rt5651->hp_jack = NULL; } @@ -2046,6 +2189,7 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, rt5651->irq = i2c->irq; rt5651->hp_mute = 1; + INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work); INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); /* Make sure work is stopped on probe-error / remove */ diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 3a0968c53fde..ac6de6fb5414 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2071,8 +2071,16 @@ struct rt5651_pll_code { struct rt5651_priv { struct snd_soc_component *component; struct regmap *regmap; + /* Jack and button detect data */ struct snd_soc_jack *hp_jack; struct work_struct jack_detect_work; + struct delayed_work bp_work; + bool ovcd_irq_enabled; + bool pressed; + bool press_reported; + int press_count; + int release_count; + int poll_count; unsigned int jd_src; unsigned int ovcd_th; unsigned int ovcd_sf; From b91f432cbc3326f715b8c3f02ff4066ab398833f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:34 +0200 Subject: [PATCH 239/529] ASoC: Intel: bytcr_rt5651: Disable jack-detect over suspend/resume Disable jack-detection and thus the codec IRQ over suspend/resume. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 49 ++++++++++++++++++++++++--- 1 file changed, 44 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 5301205496be..2a8f86dfe4cb 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -676,6 +676,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { }; /* SoC card */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; +static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ +static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ + +static int byt_rt5651_suspend(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "disabling jack detect before suspend\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static int byt_rt5651_resume(struct snd_soc_card *card) +{ + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "re-enabling jack detect after resume\n"); + snd_soc_component_set_jack(component, &priv->jack, NULL); + break; + } + } + + return 0; +} + static struct snd_soc_card byt_rt5651_card = { .name = "bytcr-rt5651", .owner = THIS_MODULE, @@ -686,13 +728,10 @@ static struct snd_soc_card byt_rt5651_card = { .dapm_routes = byt_rt5651_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map), .fully_routed = true, + .suspend_pre = byt_rt5651_suspend, + .resume_post = byt_rt5651_resume, }; -static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ -static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ - static const struct x86_cpu_id baytrail_cpu_ids[] = { { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ {} From caed9d636e857997e923dfe473b9310de645d916 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:35 +0200 Subject: [PATCH 240/529] ASoC: Intel: bytcr_rt5651: Reporting button presses Enable reporting of button presses now that the codec driver recently has gotten support for this. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 2a8f86dfe4cb..b687043c8425 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -531,13 +532,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET, &priv->jack, - bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); + SND_JACK_HEADSET | SND_JACK_BTN_0, + &priv->jack, bytcr_jack_pins, + ARRAY_SIZE(bytcr_jack_pins)); if (ret) { dev_err(runtime->dev, "jack creation failed %d\n", ret); return ret; } + snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, + KEY_PLAYPAUSE); + ret = snd_soc_component_set_jack(codec, &priv->jack, NULL); if (ret) return ret; From ff2d6acdf6f13d9f8fdcd890844c6d7535ac1f10 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Tue, 10 Jul 2018 17:28:45 +0200 Subject: [PATCH 241/529] ALSA: pcm: Fix snd_interval_refine first/last with open min/max Without this commit the following intervals [x y), (x y) were be replaced to (y-1 y) by snd_interval_refine_last(). This was also done if y-1 is part of the previous interval. With this changes it will be replaced with [y-1 y) in case of y-1 is part of the previous interval. A similar behavior will be used for snd_interval_refine_first(). This commit adapts the changes for alsa-lib of commit 9bb985c ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5736860f325b..4e6110d778bd 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -627,27 +627,33 @@ EXPORT_SYMBOL(snd_interval_refine); static int snd_interval_refine_first(struct snd_interval *i) { + const unsigned int last_max = i->max; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->max = i->min; - i->openmax = i->openmin; - if (i->openmax) + if (i->openmin) i->max++; + /* only exclude max value if also excluded before refine */ + i->openmax = (i->openmax && i->max >= last_max); return 1; } static int snd_interval_refine_last(struct snd_interval *i) { + const unsigned int last_min = i->min; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->min = i->max; - i->openmin = i->openmax; - if (i->openmin) + if (i->openmax) i->min--; + /* only exclude min value if also excluded before refine */ + i->openmin = (i->openmin && i->min <= last_min); return 1; } From 435ffb76f8b35be108a52bf1c43233a57b3c72b4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Jul 2018 12:13:48 +0200 Subject: [PATCH 242/529] ASoC: dpcm: rework runtime stream merge The goal of this patch is to simplify a bit dpcm runtime stream merge by removing several local variables. ATM, merge functions return the BE 'filter' values which should then be filtered against the FE stream values. This create a lot of local variable and unnecessary init of min and max. Instead of this, we can pass the FE stream values directly and let the BE filtering functions perform the merge 'in-place' Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b7e67b871c0c..114e6c060cae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1672,29 +1672,28 @@ unwind: } static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *stream, - u64 formats) + struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) - runtime->hw.formats &= formats & stream->formats; + runtime->hw.formats &= stream->formats; else - runtime->hw.formats = formats & stream->formats; + runtime->hw.formats = stream->formats; runtime->hw.rates = stream->rates; } -static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) +static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, + u64 *formats) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; - u64 formats = ULLONG_MAX; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) - return formats; + return; /* * It returns merged BE codec format @@ -1714,16 +1713,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) else codec_stream = &codec_dai_drv->capture; - formats &= codec_stream->formats; + *formats &= codec_stream->formats; } } - - return formats; } -static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, - unsigned int *channels_min, - unsigned int *channels_max) +static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; @@ -1732,9 +1729,6 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, if (!fe->dai_link->dpcm_merged_chan) return; - *channels_min = 0; - *channels_max = UINT_MAX; - /* * It returns merged BE codec channel; * if FE want to use it (= dpcm_merged_chan) @@ -1781,18 +1775,15 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - u64 format = dpcm_runtime_base_format(substream); - unsigned int channels_min = 0, channels_max = UINT_MAX; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); - dpcm_runtime_base_chan(substream, &channels_min, &channels_max); - - runtime->hw.channels_min = max(channels_min, runtime->hw.channels_min); - runtime->hw.channels_max = min(channels_max, runtime->hw.channels_max); + dpcm_runtime_merge_format(substream, &runtime->hw.formats); + dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, + &runtime->hw.channels_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); From baacd8d100d571aa713c3c60b1471b9962e6ec8a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Jul 2018 12:13:49 +0200 Subject: [PATCH 243/529] ASoC: dpcm: add rate merge to the BE stream merge As done for format and channels, add the possibility to merge the backend rates on the frontend rates. This useful if the backend does not support all rates supported by the frontend, or if several backends (cpu and codecs) with different capabilities are connected to the same frontend. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 60 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 870ba6b64817..a4915148f739 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -964,6 +964,8 @@ struct snd_soc_dai_link { unsigned int dpcm_merged_format:1; /* DPCM used FE & BE merged channel */ unsigned int dpcm_merged_chan:1; + /* DPCM used FE & BE merged rate */ + unsigned int dpcm_merged_rate:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 114e6c060cae..4019bc10897c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1769,6 +1769,64 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, } } +static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, + unsigned int *rates, + unsigned int *rate_min, + unsigned int *rate_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_rate) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *rate_min = max(*rate_min, cpu_stream->rate_min); + *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); + + for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *rate_min = max(*rate_min, codec_stream->rate_min); + *rate_max = min_not_zero(*rate_max, + codec_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, + codec_stream->rates); + } + } +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1784,6 +1842,8 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) dpcm_runtime_merge_format(substream, &runtime->hw.formats); dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, &runtime->hw.channels_max); + dpcm_runtime_merge_rate(substream, &runtime->hw.rates, + &runtime->hw.rate_min, &runtime->hw.rate_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); From d5418ae3f9443f911d4324c0cade988ced39cfbe Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 13 Jul 2018 14:50:42 +0200 Subject: [PATCH 244/529] ASoC: add DT documentation for the sound-name-prefix property Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/name-prefix.txt | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt diff --git a/Documentation/devicetree/bindings/sound/name-prefix.txt b/Documentation/devicetree/bindings/sound/name-prefix.txt new file mode 100644 index 000000000000..645775908657 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/name-prefix.txt @@ -0,0 +1,24 @@ +Name prefix: + +Card implementing the routing property define the connection between +audio components as list of string pair. Component using the same +sink/source names may use the name prefix property to prepend the +name of their sinks/sources with the provided string. + +Optional name prefix property: +- sound-name-prefix : string using as prefix for the sink/source names of + the component. + +Example: Two instances of the same component. + +amp0: analog-amplifier@0 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; + sound-name-prefix = "FRONT"; +}; + +amp1: analog-amplifier@1 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_4 0>; + sound-name-prefix = "BACK"; +}; From aefba45539bc4868c1fae336410aec907ee0882a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 13 Jul 2018 14:50:43 +0200 Subject: [PATCH 245/529] ASoC: allow soc-core to pick up name prefixes from component nodes When the component does not match the configuration table provided by the card, let soc-core check the component node for a name prefix Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 00bd58d167dd..3be0310d5c81 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1193,15 +1193,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); +static void soc_set_of_name_prefix(struct snd_soc_component *component) +{ + struct device_node *component_of_node = component->dev->of_node; + const char *str; + int ret; + + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + ret = of_property_read_string(component_of_node, "sound-name-prefix", + &str); + if (!ret) + component->name_prefix = str; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { int i; - if (card->codec_conf == NULL) - return; - - for (i = 0; i < card->num_configs; i++) { + for (i = 0; i < card->num_configs && card->codec_conf; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; struct device_node *component_of_node = component->dev->of_node; @@ -1213,8 +1225,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card, if (map->dev_name && strcmp(component->name, map->dev_name)) continue; component->name_prefix = map->name_prefix; - break; + return; } + + /* + * If there is no configuration table or no match in the table, + * check if a prefix is provided in the node + */ + soc_set_of_name_prefix(component); } static int soc_probe_component(struct snd_soc_card *card, From b8110a87b75f948d978c06e130cc68026645c4a1 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:05:57 +0300 Subject: [PATCH 246/529] ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data() The main thing is that the data->priv[] array has AFE_PORT_MAX elements so the > condition should be >=. But we may as well check for negative values as well just to be safe. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index a373ca5523ff..9ba95956ada8 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1183,7 +1183,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } From 090345ce7265dd111299e3a63cdc79c3ef924481 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:11:04 +0300 Subject: [PATCH 247/529] ASoC: qdsp6: q6routing: off by one in routing_hw_params() The data->port_data[] array has AFE_MAX_PORTS elements so the check should be >= instead of > or we write one element beyond the end of the array. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..7a19d6278406 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -899,7 +899,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id]; From eeef847de593e2de4ddbf7ae3439c9fc9ebc8e84 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 09:34:51 +0100 Subject: [PATCH 248/529] ALSA: opl3: remove redundant pointer opl3 Variable opl3 is being assigned but is never used hence it is redundant and can be removed. Cleans up several clang warnings: warning: variable 'opl3' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_midi.c | 12 ------------ sound/drivers/opl3/opl3_oss.c | 6 ------ 2 files changed, 18 deletions(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index bb3f3a5a6951..71cd5a2fbe82 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -721,9 +721,6 @@ void snd_opl3_note_off(void *p, int note, int vel, */ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -735,9 +732,6 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha */ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -861,9 +855,6 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -876,9 +867,6 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "SYSEX\n"); #endif diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 22c3e4bca220..8a0ce3f43f42 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -233,11 +233,8 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: snd_printk(KERN_ERR "OPL3: " @@ -261,11 +258,8 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, /* reset device */ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; return 0; } From a34e8aac49e6d2e3438998edba5f753bd5d5210f Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 09:43:08 +0100 Subject: [PATCH 249/529] ALSA: es1688: remove redundant pointer chip Pointer chip is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'chip' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..3dfe7e592c25 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -260,7 +260,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, struct snd_card *card; static unsigned int dev; int error; - struct snd_es1688 *chip; if (snd_es968_pnp_is_probed) return -EBUSY; @@ -276,7 +275,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, sizeof(struct snd_es1688), &card); if (error < 0) return error; - chip = card->private_data; error = snd_card_es968_pnp(card, dev, pcard, pid); if (error < 0) { From 29fba9230de237312337fb7372a6ee2d5f2f893c Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 09:49:38 +0100 Subject: [PATCH 250/529] ALSA: gus: remove redundant pointer private_data Pointer private_data is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'private_data' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_reset.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 3d1fed0c2620..59b3f683d49b 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -292,7 +292,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) { unsigned long flags; void (*private_free)(struct snd_gus_voice *voice); - void *private_data; if (voice == NULL || !voice->use) return; @@ -300,7 +299,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) snd_gf1_clear_voices(gus, voice->number, voice->number); spin_lock_irqsave(&gus->voice_alloc, flags); private_free = voice->private_free; - private_data = voice->private_data; voice->private_free = NULL; voice->private_data = NULL; if (voice->pcm) From 7527cd209eb88ed03aeab9f4f9bde0923fd6b5d5 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 09:52:56 +0100 Subject: [PATCH 251/529] ALSA: sb8: remove redundant pointer runtime Pointer runtime is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/isa/sb/sb8_main.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index d45df5c54423..80e7dcaa551f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -381,7 +381,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) { struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; snd_sb_ack_8bit(chip); switch (chip->mode) { @@ -391,7 +390,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; - runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); @@ -402,7 +400,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; - runtime = substream->runtime; if (chip->capture_format == SB_DSP_INPUT) snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); From c888443951949530b2eb9075cb24f6db00b8118d Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 09:57:38 +0100 Subject: [PATCH 252/529] ALSA: ali5451: remove redundant pointer 'codec' Pointer 'codec' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'codec' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 39547e32e584..9f569379b77e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1484,12 +1484,9 @@ static struct snd_pcm_hardware snd_ali_capture = static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime) { struct snd_ali_voice *pvoice = runtime->private_data; - struct snd_ali *codec; - if (pvoice) { - codec = pvoice->codec; + if (pvoice) snd_ali_free_voice(pvoice->codec, pvoice); - } } static int snd_ali_open(struct snd_pcm_substream *substream, int rec, From d6e08c7eabefc9b027d31d56024810eba76ce113 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 16 Jul 2018 10:03:15 +0100 Subject: [PATCH 253/529] ALSA: cs46xx: remove redundant pointer 'ins' Pointer 'ins' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'ins' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/dsp_spos_scb_lib.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index abb01ce66983..8d0a3d357345 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -73,13 +73,10 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, { struct proc_scb_info * scb_info = entry->private_data; struct dsp_scb_descriptor * scb = scb_info->scb_desc; - struct dsp_spos_instance * ins; struct snd_cs46xx *chip = scb_info->chip; int j,col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - ins = chip->dsp_spos_instance; - mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); From d30e23d69981a4b665f5ce8711335df986576389 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sat, 14 Jul 2018 16:01:06 +0100 Subject: [PATCH 254/529] ASoC: hdmi-codec: fix routing Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, From 8452112baac67c3235d15de67fb190d29bbba98f Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Wed, 11 Jul 2018 16:05:36 +0530 Subject: [PATCH 255/529] ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar Signed-off-by: Harsha Priya Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 643 ++++++++++++++++++ 3 files changed, 659 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..cccda87f4b34 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..87ef8b4058e5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 000000000000..c4b94e2617c5 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); From fa9d2f17c23fb3ea6b659b1bfe4ca10551a19e56 Mon Sep 17 00:00:00 2001 From: "Agrawal, Akshu" Date: Mon, 16 Jul 2018 15:02:40 +0800 Subject: [PATCH 256/529] ASoC: AMD: Send correct channel for configuring DMA descriptors Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for ACP-I2S transfer. With recent patches ch1 is used to define channel order number 1 and ch2 as channel order number 2. Thus, Playback: ch1:SYSMEM->ACP ch2:ACP->I2S Capture: ch1:I2S->ACP ch1:ACP->SYSMEM Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 65c1033bd51c..eeb867767252 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -322,17 +322,27 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } @@ -995,16 +1005,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; From 19e023e3befb4cb64b4a81b47a92a0c687672661 Mon Sep 17 00:00:00 2001 From: "Agrawal, Akshu" Date: Mon, 16 Jul 2018 15:02:41 +0800 Subject: [PATCH 257/529] ASoC: AMD: For capture have interrupts on I2S->ACP channel Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in for an interrupt storm. For both playback and capture interrupts should be enabled for I2S<->ACP DMA. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index eeb867767252..94bcf69008df 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) | (size / 2); } @@ -421,9 +419,9 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: - case ACP_TO_SYSRAM_CH_NUM: + case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: - case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: + case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -705,18 +703,18 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); - acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & - BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } From c77e1ef1cdf74dba09edfa706fecc9fddd7bd084 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Wed, 11 Jul 2018 13:37:51 +0100 Subject: [PATCH 258/529] ALSA: usb-audio: Add support for Selector Units in UAC3 This patch add support for Selector Units and Clock Selector Units defined in the new UAC3 spec. Selector Units play a really important role in the new UAC3 spec as Processing Units do not define an on/off switch control anymore. This forces topology designers to add bypass paths in the topology to enable/dissable the Processing Units. Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 54 ++++++++++++++++++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca963e94ec03..a51f2320a3dd 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -940,6 +940,19 @@ static int check_input_term(struct mixer_build *state, int id, return 0; } + case UAC3_SELECTOR_UNIT: + case UAC3_CLOCK_SELECTOR: { + struct uac_selector_unit_descriptor *d = p1; + /* call recursively to retrieve the channel info */ + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = 0; /* TODO: UAC3 Class-specific strings */ + + return 0; + } default: return -ENODEV; } @@ -2509,11 +2522,20 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->res = 1; cval->initialized = 1; - if (state->mixer->protocol == UAC_VERSION_1) + switch (state->mixer->protocol) { + case UAC_VERSION_1: + default: cval->control = 0; - else /* UAC_VERSION_2 */ - cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? - UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; + break; + case UAC_VERSION_2: + case UAC_VERSION_3: + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR || + desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR) + cval->control = UAC2_CX_CLOCK_SELECTOR; + else /* UAC2/3_SELECTOR_UNIT */ + cval->control = UAC2_SU_SELECTOR; + break; + } namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL); if (!namelist) { @@ -2555,12 +2577,22 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (!len) { /* no mapping ? */ + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: /* if iSelector is given, use it */ - nameid = uac_selector_unit_iSelector(desc); - if (nameid) - len = snd_usb_copy_string_desc(state->chip, nameid, - kctl->id.name, - sizeof(kctl->id.name)); + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state->chip, + nameid, kctl->id.name, + sizeof(kctl->id.name)); + break; + case UAC_VERSION_3: + /* TODO: Class-Specific strings not yet supported */ + break; + } + /* ... or pick up the terminal name at next */ if (!len) len = get_term_name(state->chip, &state->oterm, @@ -2570,7 +2602,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); /* and add the proper suffix */ - if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR || + desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) append_ctl_name(kctl, " Capture Source"); @@ -2641,6 +2674,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_mixer_unit(state, unitid, p1); case UAC3_CLOCK_SOURCE: return parse_clock_source_unit(state, unitid, p1); + case UAC3_SELECTOR_UNIT: case UAC3_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); case UAC3_FEATURE_UNIT: From 4e887af31cedec0d3b24521a5c3c14c3a190422b Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Wed, 11 Jul 2018 13:37:52 +0100 Subject: [PATCH 259/529] ALSA: usb-audio: Processing Unit controls parsing in UAC2 Current support for UAC2 Processing Units does the parsing as one control per bit in the bitmap. However, the UAC2 spec defines the controls as bit pairs where b01 means read-only and b11 means read/write control. This patch fixes that and uses the helper functions for checking controls readability/writability when the control is defined as bit pairs (UAC2 and UAC3). Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a51f2320a3dd..bfb3484096a6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2300,8 +2300,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, for (valinfo = info->values; valinfo->control; valinfo++) { __u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol); - if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) - continue; + if (state->mixer->protocol == UAC_VERSION_1) { + if (!(controls[valinfo->control / 8] & + (1 << ((valinfo->control % 8) - 1)))) + continue; + } else { /* UAC_VERSION_2/3 */ + if (!uac_v2v3_control_is_readable(controls[valinfo->control / 8], + valinfo->control)) + continue; + } + map = find_map(state->map, unitid, valinfo->control); if (check_ignored_ctl(map)) continue; @@ -2313,6 +2321,11 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, cval->val_type = valinfo->val_type; cval->channels = 1; + if (state->mixer->protocol > UAC_VERSION_1 && + !uac_v2v3_control_is_writeable(controls[valinfo->control / 8], + valinfo->control)) + cval->master_readonly = 1; + /* get min/max values */ if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) { __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); From 0f292f023ffcc67ec49d63dcb7fe388711cbb83a Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Wed, 11 Jul 2018 13:37:53 +0100 Subject: [PATCH 260/529] ALSA: usb-audio: Add support for Processing Units in UAC3 This patch adds support for the Processig Units defined in the UAC3 spec. The main difference with the previous specs is the lack of on/off switches in the controls for these units and the addiction of the new Multi Function Processing Unit. The current version of the UAC3 spec doesn't define any useful controls for the new Multi Function Processing Unit so no control will get created once this unit is parsed. Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v3.h | 15 ++++++++++ include/uapi/linux/usb/audio.h | 49 ++++++++++++++++++++++++++------ sound/usb/mixer.c | 51 ++++++++++++++++++++++++++++++++-- 3 files changed, 104 insertions(+), 11 deletions(-) diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h index a710e28b5215..334bfa6dfb47 100644 --- a/include/linux/usb/audio-v3.h +++ b/include/linux/usb/audio-v3.h @@ -387,6 +387,12 @@ struct uac3_interrupt_data_msg { #define UAC3_CONNECTORS 0x0f #define UAC3_POWER_DOMAIN 0x10 +/* A.20 PROCESSING UNIT PROCESS TYPES */ +#define UAC3_PROCESS_UNDEFINED 0x00 +#define UAC3_PROCESS_UP_DOWNMIX 0x01 +#define UAC3_PROCESS_STEREO_EXTENDER 0x02 +#define UAC3_PROCESS_MULTI_FUNCTION 0x03 + /* A.22 AUDIO CLASS-SPECIFIC REQUEST CODES */ /* see audio-v2.h for the rest, which is identical to v2 */ #define UAC3_CS_REQ_INTEN 0x04 @@ -406,6 +412,15 @@ struct uac3_interrupt_data_msg { #define UAC3_TE_OVERFLOW 0x04 #define UAC3_TE_LATENCY 0x05 +/* A.23.10 PROCESSING UNITS CONTROL SELECTROS */ + +/* Up/Down Mixer */ +#define UAC3_UD_MODE_SELECT 0x01 + +/* Stereo Extender */ +#define UAC3_EXT_WIDTH_CONTROL 0x01 + + /* BADD predefined Unit/Terminal values */ #define UAC3_BADD_IT_ID1 1 /* Input Terminal ID1: bTerminalID = 1 */ #define UAC3_BADD_FU_ID2 2 /* Feature Unit ID2: bUnitID = 2 */ diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index 74e520fb944f..ddc5396800aa 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -390,33 +390,64 @@ static inline __u8 uac_processing_unit_iChannelNames(struct uac_processing_unit_ static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_descriptor *desc, int protocol) { - return (protocol == UAC_VERSION_1) ? - desc->baSourceID[desc->bNrInPins + 4] : - 2; /* in UAC2, this value is constant */ + switch (protocol) { + case UAC_VERSION_1: + return desc->baSourceID[desc->bNrInPins + 4]; + case UAC_VERSION_2: + return 2; /* in UAC2, this value is constant */ + case UAC_VERSION_3: + return 4; /* in UAC3, this value is constant */ + default: + return 1; + } } static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc, int protocol) { - return (protocol == UAC_VERSION_1) ? - &desc->baSourceID[desc->bNrInPins + 5] : - &desc->baSourceID[desc->bNrInPins + 6]; + switch (protocol) { + case UAC_VERSION_1: + return &desc->baSourceID[desc->bNrInPins + 5]; + case UAC_VERSION_2: + return &desc->baSourceID[desc->bNrInPins + 6]; + case UAC_VERSION_3: + return &desc->baSourceID[desc->bNrInPins + 2]; + default: + return NULL; + } } static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc, int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return *(uac_processing_unit_bmControls(desc, protocol) - + control_size); + + switch (protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return *(uac_processing_unit_bmControls(desc, protocol) + + control_size); + case UAC_VERSION_3: + return 0; /* UAC3 does not have this field */ + } } static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return uac_processing_unit_bmControls(desc, protocol) + + switch (protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return uac_processing_unit_bmControls(desc, protocol) + control_size + 1; + case UAC_VERSION_3: + return uac_processing_unit_bmControls(desc, protocol) + + control_size; + } } /* 4.5.2 Class-Specific AS Interface Descriptor */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index bfb3484096a6..39fde49e8749 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -953,6 +953,23 @@ static int check_input_term(struct mixer_build *state, int id, return 0; } + case UAC3_PROCESSING_UNIT: { + struct uac_processing_unit_descriptor *d = p1; + + if (!d->bNrInPins) + return -EINVAL; + + /* call recursively to retrieve the channel info */ + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = 0; /* TODO: UAC3 Class-specific strings */ + + return 0; + } default: return -ENODEV; } @@ -2180,6 +2197,11 @@ struct procunit_info { struct procunit_value_info *values; }; +static struct procunit_value_info undefined_proc_info[] = { + { 0x00, "Control Undefined", 0 }, + { 0 } +}; + static struct procunit_value_info updown_proc_info[] = { { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, @@ -2228,6 +2250,23 @@ static struct procunit_info procunits[] = { { UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info }, { 0 }, }; + +static struct procunit_value_info uac3_updown_proc_info[] = { + { UAC3_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, + { 0 } +}; +static struct procunit_value_info uac3_stereo_ext_proc_info[] = { + { UAC3_EXT_WIDTH_CONTROL, "Width Control", USB_MIXER_U8 }, + { 0 } +}; + +static struct procunit_info uac3_procunits[] = { + { UAC3_PROCESS_UP_DOWNMIX, "Up Down", uac3_updown_proc_info }, + { UAC3_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", uac3_stereo_ext_proc_info }, + { UAC3_PROCESS_MULTI_FUNCTION, "Multi-Function", undefined_proc_info }, + { 0 }, +}; + /* * predefined data for extension units */ @@ -2388,8 +2427,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, static int parse_audio_processing_unit(struct mixer_build *state, int unitid, void *raw_desc) { - return build_audio_procunit(state, unitid, raw_desc, - procunits, "Processing Unit"); + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return build_audio_procunit(state, unitid, raw_desc, + procunits, "Processing Unit"); + case UAC_VERSION_3: + return build_audio_procunit(state, unitid, raw_desc, + uac3_procunits, "Processing Unit"); + } } static int parse_audio_extension_unit(struct mixer_build *state, int unitid, From 8b3a087f7f6580e685a4f1ac980d07f96ad250a5 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Wed, 11 Jul 2018 13:37:54 +0100 Subject: [PATCH 261/529] ALSA: usb-audio: Unify virtual type units type to UAC3 values The Audio Control interface descriptor subtypes do not match across all the UAC versions. That makes reusability of the "virtual type" (Mixer, Processors, Selectors, etc) terminals difficult. It also makes the mixer get the default names for the virtual terminals wrong due to the overlap. This patch proposes an unified approach by always using the most comprehensive spec version to define them all (in this case UAC3). Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 39fde49e8749..87f18cb74ca3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -675,16 +675,16 @@ static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iter if (term_only) return 0; switch (iterm->type >> 16) { - case UAC_SELECTOR_UNIT: + case UAC3_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case UAC1_PROCESSING_UNIT: + case UAC3_PROCESSING_UNIT: strcpy(name, "Process Unit"); return 12; - case UAC1_EXTENSION_UNIT: + case UAC3_EXTENSION_UNIT: strcpy(name, "Ext Unit"); return 8; - case UAC_MIXER_UNIT: + case UAC3_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: @@ -832,7 +832,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC_MIXER_UNIT: { struct uac_mixer_unit_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ term->channels = uac_mixer_unit_bNrChannels(d); term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); term->name = uac_mixer_unit_iMixer(d); @@ -845,15 +845,23 @@ static int check_input_term(struct mixer_build *state, int id, err = check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); return 0; } case UAC1_PROCESSING_UNIT: + /* UAC2_EFFECT_UNIT */ + if (protocol == UAC_VERSION_1) + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + else /* UAC_VERSION_2 */ + term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ + if (protocol == UAC_VERSION_1 && !term->type) + term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + else if (protocol == UAC_VERSION_2 && !term->type) + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; @@ -869,7 +877,9 @@ static int check_input_term(struct mixer_build *state, int id, id = d->baSourceID[0]; break; /* continue to parse */ } - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + if (!term->type) + term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + term->channels = uac_processing_unit_bNrChannels(d); term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); term->name = uac_processing_unit_iProcessing(d, protocol); @@ -878,7 +888,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SOURCE: { struct uac_clock_source_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ term->id = id; term->name = d->iClockSource; return 0; @@ -923,7 +933,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC3_CLOCK_SOURCE: { struct uac3_clock_source_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ term->id = id; term->name = le16_to_cpu(d->wClockSourceStr); return 0; @@ -936,7 +946,7 @@ static int check_input_term(struct mixer_build *state, int id, return err; term->channels = err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ return 0; } @@ -947,7 +957,7 @@ static int check_input_term(struct mixer_build *state, int id, err = check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ term->id = id; term->name = 0; /* TODO: UAC3 Class-specific strings */ @@ -964,7 +974,7 @@ static int check_input_term(struct mixer_build *state, int id, if (err < 0) return err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ term->id = id; term->name = 0; /* TODO: UAC3 Class-specific strings */ From 55b8cb46a71137aa19eba2ae5a4be384500db89e Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Wed, 11 Jul 2018 13:37:55 +0100 Subject: [PATCH 262/529] ALSA: usb-audio: Tidy up logic for Processing Unit min/max values This patch refactors the processing units min/max calculation logic for the mixer controls and fixes an issue where the Mode Select checking of the Up/Down mixers doesn't differentiate between the UAC1 and UAC2 Control Selector (0x02) and the UAC3 one which is different (0x01). Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 58 +++++++++++++++++++++++++++++++++-------------- 1 file changed, 41 insertions(+), 17 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 87f18cb74ca3..73e811f86a95 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2376,25 +2376,49 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, cval->master_readonly = 1; /* get min/max values */ - if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) { - __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); - /* FIXME: hard-coded */ - cval->min = 1; - cval->max = control_spec[0]; - cval->res = 1; - cval->initialized = 1; - } else { - if (type == USB_XU_CLOCK_RATE) { - /* - * E-Mu USB 0404/0202/TrackerPre/0204 - * samplerate control quirk - */ - cval->min = 0; - cval->max = 5; + switch (type) { + case UAC_PROCESS_UP_DOWNMIX: { + bool mode_sel = false; + + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + if (cval->control == UAC_UD_MODE_SELECT) + mode_sel = true; + break; + case UAC_VERSION_3: + if (cval->control == UAC3_UD_MODE_SELECT) + mode_sel = true; + break; + } + + if (mode_sel) { + __u8 *control_spec = uac_processing_unit_specific(desc, + state->mixer->protocol); + cval->min = 1; + cval->max = control_spec[0]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + break; + } + + get_min_max(cval, valinfo->min_value); + break; + } + case USB_XU_CLOCK_RATE: + /* + * E-Mu USB 0404/0202/TrackerPre/0204 + * samplerate control quirk + */ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + break; + default: + get_min_max(cval, valinfo->min_value); + break; } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); From b6d7b3622b6e7685767a616bb663aed40d04fdc6 Mon Sep 17 00:00:00 2001 From: Jim Qu Date: Mon, 16 Jul 2018 14:06:34 +0800 Subject: [PATCH 263/529] ALSA: hda: use PCI_BASE_CLASS_DISPLAY to replace PCI_CLASS_DISPLAY_VGA Except PCI_CLASS_DISPLAY_VGA, some PCI class is sometimes PCI_CLASS_DISPLAY_3D or PCI_CLASS_DISPLAY_OTHER. Signed-off-by: Jim Qu Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a9b55d65f2bd..daedf662b940 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1429,7 +1429,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } From 4aaf448fa9754e2d5ee188d32327b24ffc15ca4d Mon Sep 17 00:00:00 2001 From: Jim Qu Date: Tue, 17 Jul 2018 16:20:50 +0800 Subject: [PATCH 264/529] vga_switcheroo: set audio client id according to bound GPU id On modern laptop, there are more and more platforms have two GPUs, and each of them maybe have audio codec for HDMP/DP output. For some dGPU which is no output, audio codec usually is disabled. In currect HDA audio driver, it will set all codec as VGA_SWITCHEROO_DIS, the audio which is binded to UMA will be suspended if user use debugfs to contorl power In HDA driver side, it is difficult to know which GPU the audio has binded to. So set the bound gpu pci dev to vga_switcheroo. if the audio client is not the third registration, audio id will set in vga_switcheroo enable function. if the audio client is the last registration when vga_switcheroo _ready() get true, we should get audio client id from bound GPU directly. Signed-off-by: Jim Qu Reviewed-by: Lukas Wunner Signed-off-by: Takashi Iwai --- drivers/gpu/vga/vga_switcheroo.c | 63 ++++++++++++++++++++++++++------ include/linux/vga_switcheroo.h | 8 ++-- sound/pci/hda/hda_intel.c | 11 +++--- 3 files changed, 62 insertions(+), 20 deletions(-) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index fc4adf3d34e8..a96bf46bc483 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -103,9 +103,11 @@ * runtime pm. If true, writing ON and OFF to the vga_switcheroo debugfs * interface is a no-op so as not to interfere with runtime pm * @list: client list + * @vga_dev: pci device, indicate which GPU is bound to current audio client * * Registered client. A client can be either a GPU or an audio device on a GPU. - * For audio clients, the @fb_info and @active members are bogus. + * For audio clients, the @fb_info and @active members are bogus. For GPU + * clients, the @vga_dev is bogus. */ struct vga_switcheroo_client { struct pci_dev *pdev; @@ -116,6 +118,7 @@ struct vga_switcheroo_client { bool active; bool driver_power_control; struct list_head list; + struct pci_dev *vga_dev; }; /* @@ -161,9 +164,8 @@ struct vgasr_priv { }; #define ID_BIT_AUDIO 0x100 -#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) -#define client_is_vga(c) ((c)->id == VGA_SWITCHEROO_UNKNOWN_ID || \ - !client_is_audio(c)) +#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) +#define client_is_vga(c) (!client_is_audio(c)) #define client_id(c) ((c)->id & ~ID_BIT_AUDIO) static int vga_switcheroo_debugfs_init(struct vgasr_priv *priv); @@ -192,14 +194,29 @@ static void vga_switcheroo_enable(void) vgasr_priv.handler->init(); list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->id != VGA_SWITCHEROO_UNKNOWN_ID) + if (!client_is_vga(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) continue; + ret = vgasr_priv.handler->get_client_id(client->pdev); if (ret < 0) return; client->id = ret; } + + list_for_each_entry(client, &vgasr_priv.clients, list) { + if (!client_is_audio(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) + continue; + + ret = vgasr_priv.handler->get_client_id(client->vga_dev); + if (ret < 0) + return; + + client->id = ret | ID_BIT_AUDIO; + } + vga_switcheroo_debugfs_init(&vgasr_priv); vgasr_priv.active = true; } @@ -272,7 +289,9 @@ EXPORT_SYMBOL(vga_switcheroo_handler_flags); static int register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id, bool active, + enum vga_switcheroo_client_id id, + struct pci_dev *vga_dev, + bool active, bool driver_power_control) { struct vga_switcheroo_client *client; @@ -287,6 +306,7 @@ static int register_client(struct pci_dev *pdev, client->id = id; client->active = active; client->driver_power_control = driver_power_control; + client->vga_dev = vga_dev; mutex_lock(&vgasr_mutex); list_add_tail(&client->list, &vgasr_priv.clients); @@ -319,7 +339,7 @@ int vga_switcheroo_register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, bool driver_power_control) { - return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, + return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, NULL, pdev == vga_default_device(), driver_power_control); } @@ -329,19 +349,40 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); * vga_switcheroo_register_audio_client - register audio client * @pdev: client pci device * @ops: client callbacks - * @id: client identifier + * @vga_dev: pci device which is bound to current audio client * * Register audio client (audio device on a GPU). The client is assumed * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer() * shall be called to ensure that all prerequisites are met. * - * Return: 0 on success, -ENOMEM on memory allocation error. + * Return: 0 on success, -ENOMEM on memory allocation error, -EINVAL on getting + * client id error. */ int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) + struct pci_dev *vga_dev) { - return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true); + enum vga_switcheroo_client_id id = VGA_SWITCHEROO_UNKNOWN_ID; + + /* + * if vga_switcheroo has enabled, that mean two GPU clients and also + * handler are registered. Get audio client id from bound GPU client + * id directly, otherwise, set it as VGA_SWITCHEROO_UNKNOWN_ID, + * it will set to correct id in later when vga_switcheroo_enable() + * is called. + */ + mutex_lock(&vgasr_mutex); + if (vgasr_priv.active) { + id = vgasr_priv.handler->get_client_id(vga_dev); + if (id < 0) { + mutex_unlock(&vgasr_mutex); + return -EINVAL; + } + } + mutex_unlock(&vgasr_mutex); + + return register_client(pdev, ops, id | ID_BIT_AUDIO, vga_dev, + false, true); } EXPORT_SYMBOL(vga_switcheroo_register_audio_client); diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index 77f0f0af3a71..a34539b7f750 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -84,8 +84,8 @@ enum vga_switcheroo_state { * Client identifier. Audio clients use the same identifier & 0x100. */ enum vga_switcheroo_client_id { - VGA_SWITCHEROO_UNKNOWN_ID = -1, - VGA_SWITCHEROO_IGD, + VGA_SWITCHEROO_UNKNOWN_ID = 0x1000, + VGA_SWITCHEROO_IGD = 0, VGA_SWITCHEROO_DIS, VGA_SWITCHEROO_MAX_CLIENTS, }; @@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev, bool driver_power_control); int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id); + struct pci_dev *vga_dev); void vga_switcheroo_client_fb_set(struct pci_dev *dev, struct fb_info *info); @@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha enum vga_switcheroo_handler_flags_t handler_flags) { return 0; } static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) { return 0; } + struct pci_dev *vga_dev) { return 0; } static inline void vga_switcheroo_unregister_handler(void) {} static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; } static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..1967a5537d68 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1319,15 +1319,16 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int register_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct pci_dev *p; int err; if (!hda->use_vga_switcheroo) return 0; - /* FIXME: currently only handling DIS controller - * is there any machine with two switchable HDMI audio controllers? - */ - err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, - VGA_SWITCHEROO_DIS); + + p = get_bound_vga(chip->pci); + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p); + pci_dev_put(p); + if (err < 0) return err; hda->vga_switcheroo_registered = 1; From e2d2f240497c63a157f897c87567b93ed43a2213 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 17 Jul 2018 10:00:43 -0500 Subject: [PATCH 265/529] ALSA: emu10k1_patch: Use swap macro in snd_emu10k1_sample_new Make use of the swap macro and remove unnecessary variable *tmp*. This makes the code easier to read and maintain. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 0e069aeab86d..c32eb7053715 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -70,11 +70,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, loopend = sampleend; /* be sure loop points start < end */ - if (sp->v.loopstart >= sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart >= sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; From 7373c2a99abf2b11b5e8a226071331c0176253ff Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 17 Jul 2018 10:06:10 -0500 Subject: [PATCH 266/529] ALSA: emu8000: Use swap macro in snd_emu8000_sample_new Make use of the swap macro and remove unnecessary variable *tmp*. This makes the code easier to read and maintain. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_patch.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index c2e41d2762f7..d45a6b9d6437 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -165,11 +165,8 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return 0; /* be sure loop points start < end */ - if (sp->v.loopstart > sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart > sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size; From ae891abe7c2ccf75b69ca8330225e37ecc06924e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Jul 2018 15:17:22 +0200 Subject: [PATCH 267/529] drm/i915: Split audio component to a generic type For allowing other drivers to use the DRM audio component, rename the i915_audio_component_* with drm_audio_component_*, and split the generic part into drm_audio_component.h. The i915 specific stuff remains in struct i915_audio_component, which contains drm_audio_component as the base. The license of drm_audio_component.h is kept to MIT as same as the the original i915_component.h. This is a preliminary change for further development, and no functional changes by this patch itself, merely code-split and renames. v1->v2: Use SPDX for drm_audio_component.h, fix remaining i915 argument in drm_audio_component.h Reviewed-by: Rodrigo Vivi Signed-off-by: Takashi Iwai --- drivers/gpu/drm/i915/intel_audio.c | 22 +++---- include/drm/drm_audio_component.h | 95 ++++++++++++++++++++++++++++++ include/drm/i915_component.h | 85 ++------------------------ include/sound/hda_i915.h | 6 +- include/sound/hdaudio.h | 6 +- sound/hda/hdac_i915.c | 40 +++++++------ sound/pci/hda/patch_hdmi.c | 8 +-- sound/soc/codecs/hdac_hdmi.c | 2 +- 8 files changed, 144 insertions(+), 120 deletions(-) create mode 100644 include/drm/drm_audio_component.h diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 3ea566f99450..7dd5605d94ae 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -639,11 +639,12 @@ void intel_audio_codec_enable(struct intel_encoder *encoder, dev_priv->av_enc_map[pipe] = encoder; mutex_unlock(&dev_priv->av_mutex); - if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) { + if (acomp && acomp->base.audio_ops && + acomp->base.audio_ops->pin_eld_notify) { /* audio drivers expect pipe = -1 to indicate Non-MST cases */ if (!intel_crtc_has_type(crtc_state, INTEL_OUTPUT_DP_MST)) pipe = -1; - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr, (int) port, (int) pipe); } @@ -681,11 +682,12 @@ void intel_audio_codec_disable(struct intel_encoder *encoder, dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); - if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) { + if (acomp && acomp->base.audio_ops && + acomp->base.audio_ops->pin_eld_notify) { /* audio drivers expect pipe = -1 to indicate Non-MST cases */ if (!intel_crtc_has_type(old_crtc_state, INTEL_OUTPUT_DP_MST)) pipe = -1; - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr, (int) port, (int) pipe); } @@ -880,7 +882,7 @@ static int i915_audio_component_get_eld(struct device *kdev, int port, return ret; } -static const struct i915_audio_component_ops i915_audio_component_ops = { +static const struct drm_audio_component_ops i915_audio_component_ops = { .owner = THIS_MODULE, .get_power = i915_audio_component_get_power, .put_power = i915_audio_component_put_power, @@ -897,12 +899,12 @@ static int i915_audio_component_bind(struct device *i915_kdev, struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev); int i; - if (WARN_ON(acomp->ops || acomp->dev)) + if (WARN_ON(acomp->base.ops || acomp->base.dev)) return -EEXIST; drm_modeset_lock_all(&dev_priv->drm); - acomp->ops = &i915_audio_component_ops; - acomp->dev = i915_kdev; + acomp->base.ops = &i915_audio_component_ops; + acomp->base.dev = i915_kdev; BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++) acomp->aud_sample_rate[i] = 0; @@ -919,8 +921,8 @@ static void i915_audio_component_unbind(struct device *i915_kdev, struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev); drm_modeset_lock_all(&dev_priv->drm); - acomp->ops = NULL; - acomp->dev = NULL; + acomp->base.ops = NULL; + acomp->base.dev = NULL; dev_priv->audio_component = NULL; drm_modeset_unlock_all(&dev_priv->drm); } diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h new file mode 100644 index 000000000000..e85689f212c2 --- /dev/null +++ b/include/drm/drm_audio_component.h @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: MIT +// Copyright © 2014 Intel Corporation + +#ifndef _DRM_AUDIO_COMPONENT_H_ +#define _DRM_AUDIO_COMPONENT_H_ + +/** + * struct drm_audio_component_ops - Ops implemented by DRM driver, called by hda driver + */ +struct drm_audio_component_ops { + /** + * @owner: drm module to pin down + */ + struct module *owner; + /** + * @get_power: get the POWER_DOMAIN_AUDIO power well + * + * Request the power well to be turned on. + */ + void (*get_power)(struct device *); + /** + * @put_power: put the POWER_DOMAIN_AUDIO power well + * + * Allow the power well to be turned off. + */ + void (*put_power)(struct device *); + /** + * @codec_wake_override: Enable/disable codec wake signal + */ + void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Get the Core Display Clock in kHz + */ + int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); + /** + * @get_eld: fill the audio state and ELD bytes for the given port + * + * Called from audio driver to get the HDMI/DP audio state of the given + * digital port, and also fetch ELD bytes to the given pointer. + * + * It returns the byte size of the original ELD (not the actually + * copied size), zero for an invalid ELD, or a negative error code. + * + * Note that the returned size may be over @max_bytes. Then it + * implies that only a part of ELD has been copied to the buffer. + */ + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, + unsigned char *buf, int max_bytes); +}; + +/** + * struct drm_audio_component_audio_ops - Ops implemented by hda driver, called by DRM driver + */ +struct drm_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ + void *audio_ptr; + /** + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the DRM driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). + */ + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); +}; + +/** + * struct drm_audio_component - Used for direct communication between DRM and hda drivers + */ +struct drm_audio_component { + /** + * @dev: DRM device, used as parameter for ops + */ + struct device *dev; + /** + * @ops: Ops implemented by DRM driver, called by hda driver + */ + const struct drm_audio_component_ops *ops; + /** + * @audio_ops: Ops implemented by hda driver, called by DRM driver + */ + const struct drm_audio_component_audio_ops *audio_ops; +}; + +#endif /* _DRM_AUDIO_COMPONENT_H_ */ diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 346b1f5cb180..fca22d463e1b 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,101 +24,26 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +#include "drm_audio_component.h" + /* MAX_PORT is the number of port * It must be sync with I915_MAX_PORTS defined i915_drv.h */ #define MAX_PORTS 6 -/** - * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver - */ -struct i915_audio_component_ops { - /** - * @owner: i915 module - */ - struct module *owner; - /** - * @get_power: get the POWER_DOMAIN_AUDIO power well - * - * Request the power well to be turned on. - */ - void (*get_power)(struct device *); - /** - * @put_power: put the POWER_DOMAIN_AUDIO power well - * - * Allow the power well to be turned off. - */ - void (*put_power)(struct device *); - /** - * @codec_wake_override: Enable/disable codec wake signal - */ - void (*codec_wake_override)(struct device *, bool enable); - /** - * @get_cdclk_freq: Get the Core Display Clock in kHz - */ - int (*get_cdclk_freq)(struct device *); - /** - * @sync_audio_rate: set n/cts based on the sample rate - * - * Called from audio driver. After audio driver sets the - * sample rate, it will call this function to set n/cts - */ - int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); - /** - * @get_eld: fill the audio state and ELD bytes for the given port - * - * Called from audio driver to get the HDMI/DP audio state of the given - * digital port, and also fetch ELD bytes to the given pointer. - * - * It returns the byte size of the original ELD (not the actually - * copied size), zero for an invalid ELD, or a negative error code. - * - * Note that the returned size may be over @max_bytes. Then it - * implies that only a part of ELD has been copied to the buffer. - */ - int (*get_eld)(struct device *, int port, int pipe, bool *enabled, - unsigned char *buf, int max_bytes); -}; - -/** - * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver - */ -struct i915_audio_component_audio_ops { - /** - * @audio_ptr: Pointer to be used in call to pin_eld_notify - */ - void *audio_ptr; - /** - * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed - * - * Called when the i915 driver has set up audio pipeline or has just - * begun to tear it down. This allows the HDA driver to update its - * status accordingly (even when the HDA controller is in power save - * mode). - */ - void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); -}; - /** * struct i915_audio_component - Used for direct communication between i915 and hda drivers */ struct i915_audio_component { /** - * @dev: i915 device, used as parameter for ops + * @base: the drm_audio_component base class */ - struct device *dev; + struct drm_audio_component base; + /** * @aud_sample_rate: the array of audio sample rate per port */ int aud_sample_rate[MAX_PORTS]; - /** - * @ops: Ops implemented by i915 driver, called by hda driver - */ - const struct i915_audio_component_ops *ops; - /** - * @audio_ops: Ops implemented by hda driver, called by i915 driver - */ - const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index a94f5b6f92ac..f69ea84e7b65 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -5,7 +5,7 @@ #ifndef __SOUND_HDA_I915_H #define __SOUND_HDA_I915_H -#include +#include #ifdef CONFIG_SND_HDA_I915 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); @@ -17,7 +17,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); -int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *); +int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *); #else static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { @@ -49,7 +49,7 @@ static inline int snd_hdac_i915_exit(struct hdac_bus *bus) { return 0; } -static inline int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *ops) +static inline int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *ops) { return -ENODEV; } diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index f1baaa88e766..ab5ee3ef2198 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -333,9 +333,9 @@ struct hdac_bus { spinlock_t reg_lock; struct mutex cmd_mutex; - /* i915 component interface */ - struct i915_audio_component *audio_component; - int i915_power_refcount; + /* DRM component interface */ + struct drm_audio_component *audio_component; + int drm_power_refcount; /* parameters required for enhanced capabilities */ int num_streams; diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index cbe818eda336..1a88c1aaf9bb 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -16,13 +16,13 @@ #include #include #include -#include +#include #include #include #include #include -static struct i915_audio_component *hdac_acomp; +static struct drm_audio_component *hdac_acomp; /** * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup @@ -39,7 +39,7 @@ static struct i915_audio_component *hdac_acomp; */ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; if (!acomp || !acomp->ops) return -ENODEV; @@ -74,7 +74,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); */ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; if (!acomp || !acomp->ops) return -ENODEV; @@ -83,14 +83,14 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) enable ? "enable" : "disable"); if (enable) { - if (!bus->i915_power_refcount++) { + if (!bus->drm_power_refcount++) { acomp->ops->get_power(acomp->dev); snd_hdac_set_codec_wakeup(bus, true); snd_hdac_set_codec_wakeup(bus, false); } } else { - WARN_ON(!bus->i915_power_refcount); - if (!--bus->i915_power_refcount) + WARN_ON(!bus->drm_power_refcount); + if (!--bus->drm_power_refcount) acomp->ops->put_power(acomp->dev); } @@ -119,7 +119,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_display_power); */ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; struct pci_dev *pci = to_pci_dev(bus->dev); int cdclk_freq; unsigned int bclk_m, bclk_n; @@ -206,7 +206,7 @@ int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int dev_id, int rate) { struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) @@ -244,7 +244,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes) { struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->get_eld) @@ -262,7 +262,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); static int hdac_component_master_bind(struct device *dev) { - struct i915_audio_component *acomp = hdac_acomp; + struct drm_audio_component *acomp = hdac_acomp; int ret; ret = component_bind_all(dev, acomp); @@ -294,7 +294,7 @@ out_unbind: static void hdac_component_master_unbind(struct device *dev) { - struct i915_audio_component *acomp = hdac_acomp; + struct drm_audio_component *acomp = hdac_acomp; module_put(acomp->ops->owner); component_unbind_all(dev, acomp); @@ -323,7 +323,7 @@ static int hdac_component_master_match(struct device *dev, void *data) * * Returns zero for success or a negative error code. */ -int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) +int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *aops) { if (!hdac_acomp) return -ENODEV; @@ -361,7 +361,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus) { struct component_match *match = NULL; struct device *dev = bus->dev; - struct i915_audio_component *acomp; + struct i915_audio_component *i915_acomp; + struct drm_audio_component *acomp; int ret; if (WARN_ON(hdac_acomp)) @@ -370,9 +371,10 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); - if (!acomp) + i915_acomp = kzalloc(sizeof(*i915_acomp), GFP_KERNEL); + if (!i915_acomp) return -ENOMEM; + acomp = &i915_acomp->base; bus->audio_component = acomp; hdac_acomp = acomp; @@ -421,13 +423,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_i915_init); int snd_hdac_i915_exit(struct hdac_bus *bus) { struct device *dev = bus->dev; - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; if (!acomp) return 0; - WARN_ON(bus->i915_power_refcount); - if (bus->i915_power_refcount > 0 && acomp->ops) + WARN_ON(bus->drm_power_refcount); + if (bus->drm_power_refcount > 0 && acomp->ops) acomp->ops->put_power(acomp->dev); component_master_del(dev, &hdac_component_master_ops); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8a49415aebac..c0847017114c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -177,7 +177,7 @@ struct hdmi_spec { /* i915/powerwell (Haswell+/Valleyview+) specific */ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ - struct i915_audio_component_audio_ops i915_audio_ops; + struct drm_audio_component_audio_ops drm_audio_ops; struct hdac_chmap chmap; hda_nid_t vendor_nid; @@ -2511,14 +2511,14 @@ static void register_i915_notifier(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; spec->use_acomp_notifier = true; - spec->i915_audio_ops.audio_ptr = codec; + spec->drm_audio_ops.audio_ptr = codec; /* intel_audio_codec_enable() or intel_audio_codec_disable() * will call pin_eld_notify with using audio_ptr pointer * We need make sure audio_ptr is really setup */ wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->drm_audio_ops); } /* setup_stream ops override for HSW+ */ diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3e3a2a9ef310..460075475f20 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1583,7 +1583,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) } -static struct i915_audio_component_audio_ops aops = { +static struct drm_audio_component_audio_ops aops = { .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; From 82887c0beb1ee6b33eed8318d8e8d41c5b3eddae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Jul 2018 15:48:18 +0200 Subject: [PATCH 268/529] ALSA: hda/i915: Associate audio component with devres The HD-audio i915 binding code contains a single pointer, hdac_acomp, for allowing the access to audio component from the master bind/unbind callbacks. This was needed because the callbacks pass only the device pointer and we can't guarantee the object type assigned to the drvdata (which is free for each controller driver implementation). And this implementation will be a problem if we support multiple components for different DRM drivers, not only i915. As a solution, allocate the audio component object via devres and associate it with the given device, so that the component callbacks can refer to it via devres_find(). The removal of the object is still done half-manually via devres_destroy() to make the code consistent (although it may work without the explicit call). Also, the snd_hda_i915_register_notifier() had the reference to hdac_acomp as well. In this patch, the corresponding code is removed by passing hdac_bus object to the function, too. Signed-off-by: Takashi Iwai --- include/sound/hda_i915.h | 6 ++++-- sound/hda/hdac_i915.c | 34 +++++++++++++++++++++------------- sound/pci/hda/patch_hdmi.c | 5 +++-- sound/soc/codecs/hdac_hdmi.c | 2 +- 4 files changed, 29 insertions(+), 18 deletions(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index f69ea84e7b65..648263791559 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -17,7 +17,8 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); -int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *); +int snd_hdac_i915_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops); #else static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { @@ -49,7 +50,8 @@ static inline int snd_hdac_i915_exit(struct hdac_bus *bus) { return 0; } -static inline int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *ops) +static inline int snd_hdac_i915_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops) { return -ENODEV; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 1a88c1aaf9bb..861b77bbc7df 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -22,7 +22,14 @@ #include #include -static struct drm_audio_component *hdac_acomp; +static void hdac_acomp_release(struct device *dev, void *res) +{ +} + +static struct drm_audio_component *hdac_get_acomp(struct device *dev) +{ + return devres_find(dev, hdac_acomp_release, NULL, NULL); +} /** * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup @@ -262,7 +269,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); static int hdac_component_master_bind(struct device *dev) { - struct drm_audio_component *acomp = hdac_acomp; + struct drm_audio_component *acomp = hdac_get_acomp(dev); int ret; ret = component_bind_all(dev, acomp); @@ -294,7 +301,7 @@ out_unbind: static void hdac_component_master_unbind(struct device *dev) { - struct drm_audio_component *acomp = hdac_acomp; + struct drm_audio_component *acomp = hdac_get_acomp(dev); module_put(acomp->ops->owner); component_unbind_all(dev, acomp); @@ -314,6 +321,7 @@ static int hdac_component_master_match(struct device *dev, void *data) /** * snd_hdac_i915_register_notifier - Register i915 audio component ops + * @bus: HDA core bus * @aops: i915 audio component ops * * This function is supposed to be used only by a HD-audio controller @@ -323,12 +331,13 @@ static int hdac_component_master_match(struct device *dev, void *data) * * Returns zero for success or a negative error code. */ -int snd_hdac_i915_register_notifier(const struct drm_audio_component_audio_ops *aops) +int snd_hdac_i915_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops) { - if (!hdac_acomp) + if (!bus->audio_component) return -ENODEV; - hdac_acomp->audio_ops = aops; + bus->audio_component->audio_ops = aops; return 0; } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); @@ -365,18 +374,19 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct drm_audio_component *acomp; int ret; - if (WARN_ON(hdac_acomp)) + if (WARN_ON(hdac_get_acomp(dev))) return -EBUSY; if (!i915_gfx_present()) return -ENODEV; - i915_acomp = kzalloc(sizeof(*i915_acomp), GFP_KERNEL); + i915_acomp = devres_alloc(hdac_acomp_release, sizeof(*i915_acomp), + GFP_KERNEL); if (!i915_acomp) return -ENOMEM; acomp = &i915_acomp->base; bus->audio_component = acomp; - hdac_acomp = acomp; + devres_add(dev, acomp); component_match_add(dev, &match, hdac_component_master_match, bus); ret = component_master_add_with_match(dev, &hdac_component_master_ops, @@ -400,9 +410,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus) out_master_del: component_master_del(dev, &hdac_component_master_ops); out_err: - kfree(acomp); bus->audio_component = NULL; - hdac_acomp = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; @@ -434,9 +443,8 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) component_master_del(dev, &hdac_component_master_ops); - kfree(acomp); bus->audio_component = NULL; - hdac_acomp = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c0847017114c..bf174a013f2d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2288,7 +2288,7 @@ static void generic_hdmi_free(struct hda_codec *codec) int pin_idx, pcm_idx; if (codec_has_acomp(codec)) - snd_hdac_i915_register_notifier(NULL); + snd_hdac_i915_register_notifier(&codec->bus->core, NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2518,7 +2518,8 @@ static void register_i915_notifier(struct hda_codec *codec) */ wmb(); spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->drm_audio_ops); + snd_hdac_i915_register_notifier(&codec->bus->core, + &spec->drm_audio_ops); } /* setup_stream ops override for HSW+ */ diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 460075475f20..2b7c33db4ded 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1812,7 +1812,7 @@ static int hdmi_codec_probe(struct snd_soc_component *component) return ret; aops.audio_ptr = hdev; - ret = snd_hdac_i915_register_notifier(&aops); + ret = snd_hdac_i915_register_notifier(hdev->bus, &aops); if (ret < 0) { dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret); return ret; From a57942bfdd61b46df94021c9c33b8faaae5b65e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Jul 2018 16:23:16 +0200 Subject: [PATCH 269/529] ALSA: hda: Make audio component support more generic This is the final step for more generic support of DRM audio component. The generic audio component code is now moved to its own file, and the symbols are renamed from snd_hac_i915_* to snd_hdac_acomp_*, respectively. The generic code is enabled via the new kconfig, CONFIG_SND_HDA_COMPONENT, while CONFIG_SND_HDA_I915 is kept as the super-class. Along with the split, three new callbacks are added to audio_ops: pin2port is for providing the conversion between the pin number and the widget id, and master_bind/master_unbin are called at binding / unbinding the master component, respectively. All these are optional, but used in i915 implementation and also other later implementations. A note about the new snd_hdac_acomp_init() function: there is a slight difference between this and the old snd_hdac_i915_init(). The latter (still) synchronizes with the master component binding, i.e. it assures that the relevant DRM component gets bound when it returns, or gives a negative error. Meanwhile the new function doesn't synchronize but just leaves as is. It's the responsibility by the caller's side to synchronize, or the caller may accept the asynchronous binding on the fly. v1->v2: Fix missing NULL check in master_bind/unbind Signed-off-by: Takashi Iwai --- drivers/gpu/drm/i915/Kconfig | 1 + include/drm/drm_audio_component.h | 23 ++ include/sound/hda_component.h | 61 ++++++ include/sound/hda_i915.h | 39 +--- sound/hda/Kconfig | 7 +- sound/hda/Makefile | 1 + sound/hda/hdac_component.c | 335 +++++++++++++++++++++++++++++ sound/hda/hdac_i915.c | 343 ++---------------------------- sound/pci/hda/patch_hdmi.c | 50 +++-- sound/soc/codecs/hdac_hdmi.c | 8 +- 10 files changed, 486 insertions(+), 382 deletions(-) create mode 100644 include/sound/hda_component.h create mode 100644 sound/hda/hdac_component.c diff --git a/drivers/gpu/drm/i915/Kconfig b/drivers/gpu/drm/i915/Kconfig index dfd95889f4b7..5c607f2c707b 100644 --- a/drivers/gpu/drm/i915/Kconfig +++ b/drivers/gpu/drm/i915/Kconfig @@ -23,6 +23,7 @@ config DRM_I915 select SYNC_FILE select IOSF_MBI select CRC32 + select SND_HDA_I915 if SND_HDA_CORE help Choose this option if you have a system that has "Intel Graphics Media Accelerator" or "HD Graphics" integrated graphics, diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h index e85689f212c2..4923b00328c1 100644 --- a/include/drm/drm_audio_component.h +++ b/include/drm/drm_audio_component.h @@ -4,6 +4,8 @@ #ifndef _DRM_AUDIO_COMPONENT_H_ #define _DRM_AUDIO_COMPONENT_H_ +struct drm_audio_component; + /** * struct drm_audio_component_ops - Ops implemented by DRM driver, called by hda driver */ @@ -72,6 +74,27 @@ struct drm_audio_component_audio_ops { * mode). */ void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); + /** + * @pin2port: Check and convert from pin node to port number + * + * Called by HDA driver to check and convert from the pin widget node + * number to a port number in the graphics side. + */ + int (*pin2port)(void *audio_ptr, int pin); + /** + * @master_bind: (Optional) component master bind callback + * + * Called at binding master component, for HDA codec-specific + * handling of dynamic binding. + */ + int (*master_bind)(struct device *dev, struct drm_audio_component *); + /** + * @master_unbind: (Optional) component master unbind callback + * + * Called at unbinding master component, for HDA codec-specific + * handling of dynamic unbinding. + */ + void (*master_unbind)(struct device *dev, struct drm_audio_component *); }; /** diff --git a/include/sound/hda_component.h b/include/sound/hda_component.h new file mode 100644 index 000000000000..78626cde7081 --- /dev/null +++ b/include/sound/hda_component.h @@ -0,0 +1,61 @@ +// SPDX-License-Identifier: GPL-2.0 +// HD-Audio helpers to sync with DRM driver + +#ifndef __SOUND_HDA_COMPONENT_H +#define __SOUND_HDA_COMPONENT_H + +#include + +#ifdef CONFIG_SND_HDA_COMPONENT +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int snd_hdac_display_power(struct hdac_bus *bus, bool enable); +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, + bool *audio_enabled, char *buffer, int max_bytes); +int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size); +int snd_hdac_acomp_exit(struct hdac_bus *bus); +int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops); +#else +static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, + hda_nid_t nid, int dev_id, int rate) +{ + return 0; +} +static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, + int dev_id, bool *audio_enabled, + char *buffer, int max_bytes) +{ + return -ENODEV; +} +static inline int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size) +{ + return -ENODEV; +} +static inline int snd_hdac_acomp_exit(struct hdac_bus *bus) +{ + return 0; +} +static inline int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops) +{ + return -ENODEV; +} +#endif + +#endif /* __SOUND_HDA_COMPONENT_H */ diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index 648263791559..6b79614a893b 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -5,56 +5,23 @@ #ifndef __SOUND_HDA_I915_H #define __SOUND_HDA_I915_H -#include +#include "hda_component.h" #ifdef CONFIG_SND_HDA_I915 -int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); -int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, - int dev_id, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, - bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); -int snd_hdac_i915_exit(struct hdac_bus *bus); -int snd_hdac_i915_register_notifier(struct hdac_bus *bus, - const struct drm_audio_component_audio_ops *ops); #else -static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) -{ - return 0; -} -static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) -{ - return 0; -} static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } -static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int dev_id, int rate) -{ - return 0; -} -static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - int dev_id, bool *audio_enabled, - char *buffer, int max_bytes) -{ - return -ENODEV; -} static inline int snd_hdac_i915_init(struct hdac_bus *bus) { return -ENODEV; } +#endif static inline int snd_hdac_i915_exit(struct hdac_bus *bus) { - return 0; + return snd_hdac_acomp_exit(bus); } -static inline int snd_hdac_i915_register_notifier(struct hdac_bus *bus, - const struct drm_audio_component_audio_ops *ops) -{ - return -ENODEV; -} -#endif #endif /* __SOUND_HDA_I915_H */ diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3129546398d0..2d90e11b3eaa 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -5,11 +5,12 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool +config SND_HDA_COMPONENT + bool + config SND_HDA_I915 bool - default y - depends on DRM_I915 - depends on SND_HDA_CORE + select SND_HDA_COMPONENT config SND_HDA_EXT_CORE tristate diff --git a/sound/hda/Makefile b/sound/hda/Makefile index e4e726f2ce98..2160202e2dc1 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -6,6 +6,7 @@ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) # for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_COMPONENT) += hdac_component.o snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c new file mode 100644 index 000000000000..6e46a9c73aed --- /dev/null +++ b/sound/hda/hdac_component.c @@ -0,0 +1,335 @@ +// SPDX-License-Identifier: GPL-2.0 +// hdac_component.c - routines for sync between HD-A core and DRM driver + +#include +#include +#include +#include +#include +#include +#include +#include + +static void hdac_acomp_release(struct device *dev, void *res) +{ +} + +static struct drm_audio_component *hdac_get_acomp(struct device *dev) +{ + return devres_find(dev, hdac_acomp_release, NULL, NULL); +} + +/** + * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup + * @bus: HDA core bus + * @enable: enable or disable the wakeup + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function should be called during the chip reset, also called at + * resume for updating STATESTS register read. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) + return 0; + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); + +/** + * snd_hdac_display_power - Power up / down the power refcount + * @bus: HDA core bus + * @enable: power up or down + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function manages a refcount and calls the get_power() and + * put_power() ops accordingly, toggling the codec wakeup, too. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->drm_power_refcount++) { + if (acomp->ops->get_power) + acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } + } else { + WARN_ON(!bus->drm_power_refcount); + if (!--bus->drm_power_refcount) + if (acomp->ops->put_power) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_display_power); + +/** + * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @rate: the sample rate to set + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets N/CTS value based on the given sample rate. + * Returns zero for success, or a negative error code. + */ +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) + return -ENODEV; + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); +} +EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); + +/** + * snd_hdac_acomp_get_eld - Get the audio state and ELD via component + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @audio_enabled: the pointer to store the current audio state + * @buffer: the buffer pointer to store ELD bytes + * @max_bytes: the max bytes to be stored on @buffer + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function queries the current state of the audio on the given + * digital port and fetches the ELD bytes onto the given buffer. + * It returns the number of bytes for the total ELD data, zero for + * invalid ELD, or a negative error code. + * + * The return size is the total bytes required for the whole ELD bytes, + * thus it may be over @max_bytes. If it's over @max_bytes, it implies + * that only a part of ELD bytes have been fetched. + */ +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, + bool *audio_enabled, char *buffer, int max_bytes) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->get_eld) + return -ENODEV; + + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, + buffer, max_bytes); +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); + +static int hdac_component_master_bind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + int ret; + + if (WARN_ON(!acomp)) + return -EINVAL; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops))) { + ret = -EINVAL; + goto out_unbind; + } + + /* pin the module to avoid dynamic unbinding, but only if given */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + if (acomp->audio_ops && acomp->audio_ops->master_bind) { + ret = acomp->audio_ops->master_bind(dev, acomp); + if (ret < 0) + goto module_put; + } + + return 0; + + module_put: + module_put(acomp->ops->owner); +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hdac_component_master_unbind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + + if (acomp->audio_ops && acomp->audio_ops->master_unbind) + acomp->audio_ops->master_unbind(dev, acomp); + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hdac_component_master_ops = { + .bind = hdac_component_master_bind, + .unbind = hdac_component_master_unbind, +}; + +/** + * snd_hdac_acomp_register_notifier - Register audio component ops + * @bus: HDA core bus + * @aops: audio component ops + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets the given ops to be called by the graphics driver. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops) +{ + if (!bus->audio_component) + return -ENODEV; + + bus->audio_component->audio_ops = aops; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_register_notifier); + +/** + * snd_hdac_acomp_init - Initialize audio component + * @bus: HDA core bus + * @match_master: match function for finding components + * @extra_size: Extra bytes to allocate + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function initializes and sets up the audio component to communicate + * with graphics driver. + * + * Unlike snd_hdac_i915_init(), this function doesn't synchronize with the + * binding with the DRM component. Each caller needs to sync via master_bind + * audio_ops. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct drm_audio_component *acomp; + int ret; + + if (WARN_ON(hdac_get_acomp(dev))) + return -EBUSY; + + acomp = devres_alloc(hdac_acomp_release, sizeof(*acomp) + extra_size, + GFP_KERNEL); + if (!acomp) + return -ENOMEM; + acomp->audio_ops = aops; + bus->audio_component = acomp; + devres_add(dev, acomp); + + component_match_add(dev, &match, match_master, bus); + ret = component_master_add_with_match(dev, &hdac_component_master_ops, + match); + if (ret < 0) + goto out_err; + + return 0; + +out_err: + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + dev_info(dev, "failed to add audio component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_init); + +/** + * snd_hdac_acomp_exit - Finalize audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function releases the audio component that has been used. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp) + return 0; + + WARN_ON(bus->drm_power_refcount); + if (bus->drm_power_refcount > 0 && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hdac_component_master_ops); + + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_exit); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 861b77bbc7df..8f2aa8bc1185 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -15,96 +15,11 @@ #include #include #include -#include -#include #include #include #include #include -static void hdac_acomp_release(struct device *dev, void *res) -{ -} - -static struct drm_audio_component *hdac_get_acomp(struct device *dev) -{ - return devres_find(dev, hdac_acomp_release, NULL, NULL); -} - -/** - * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup - * @bus: HDA core bus - * @enable: enable or disable the wakeup - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function should be called during the chip reset, also called at - * resume for updating STATESTS register read. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) -{ - struct drm_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(bus->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(bus->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); - -/** - * snd_hdac_display_power - Power up / down the power refcount - * @bus: HDA core bus - * @enable: power up or down - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function manages a refcount and calls the i915 get_power() and - * put_power() ops accordingly, toggling the codec wakeup, too. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_display_power(struct hdac_bus *bus, bool enable) -{ - struct drm_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - dev_dbg(bus->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!bus->drm_power_refcount++) { - acomp->ops->get_power(acomp->dev); - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); - } - } else { - WARN_ON(!bus->drm_power_refcount); - if (!--bus->drm_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_display_power); - #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -165,183 +80,11 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/* There is a fixed mapping between audio pin node and display port. - * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: - * Pin Widget 5 - PORT B (port = 1 in i915 driver) - * Pin Widget 6 - PORT C (port = 2 in i915 driver) - * Pin Widget 7 - PORT D (port = 3 in i915 driver) - * - * on VLV, ILK: - * Pin Widget 4 - PORT B (port = 1 in i915 driver) - * Pin Widget 5 - PORT C (port = 2 in i915 driver) - * Pin Widget 6 - PORT D (port = 3 in i915 driver) - */ -static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) +static int i915_component_master_match(struct device *dev, void *data) { - int base_nid; - - switch (codec->vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - base_nid = 3; - break; - default: - base_nid = 4; - break; - } - - if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) - return -1; - return pin_nid - base_nid; -} - -/** - * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @rate: the sample rate to set - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets N/CTS value based on the given sample rate. - * Returns zero for success, or a negative error code. - */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, - int dev_id, int rate) -{ - struct hdac_bus *bus = codec->bus; - struct drm_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) - return -ENODEV; - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - pipe = dev_id; - return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); -} -EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); - -/** - * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @audio_enabled: the pointer to store the current audio state - * @buffer: the buffer pointer to store ELD bytes - * @max_bytes: the max bytes to be stored on @buffer - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function queries the current state of the audio on the given - * digital port and fetches the ELD bytes onto the given buffer. - * It returns the number of bytes for the total ELD data, zero for - * invalid ELD, or a negative error code. - * - * The return size is the total bytes required for the whole ELD bytes, - * thus it may be over @max_bytes. If it's over @max_bytes, it implies - * that only a part of ELD bytes have been fetched. - */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, - bool *audio_enabled, char *buffer, int max_bytes) -{ - struct hdac_bus *bus = codec->bus; - struct drm_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->get_eld) - return -ENODEV; - - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - - pipe = dev_id; - return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, - buffer, max_bytes); -} -EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); - -static int hdac_component_master_bind(struct device *dev) -{ - struct drm_audio_component *acomp = hdac_get_acomp(dev); - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hdac_component_master_unbind(struct device *dev) -{ - struct drm_audio_component *acomp = hdac_get_acomp(dev); - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hdac_component_master_ops = { - .bind = hdac_component_master_bind, - .unbind = hdac_component_master_unbind, -}; - -static int hdac_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ return !strcmp(dev->driver->name, "i915"); } -/** - * snd_hdac_i915_register_notifier - Register i915 audio component ops - * @bus: HDA core bus - * @aops: i915 audio component ops - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets the given ops to be called by the i915 graphics driver. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_register_notifier(struct hdac_bus *bus, - const struct drm_audio_component_audio_ops *aops) -{ - if (!bus->audio_component) - return -ENODEV; - - bus->audio_component->audio_ops = aops; - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); - /* check whether intel graphics is present */ static bool i915_gfx_present(void) { @@ -368,84 +111,26 @@ static bool i915_gfx_present(void) */ int snd_hdac_i915_init(struct hdac_bus *bus) { - struct component_match *match = NULL; - struct device *dev = bus->dev; - struct i915_audio_component *i915_acomp; struct drm_audio_component *acomp; - int ret; - - if (WARN_ON(hdac_get_acomp(dev))) - return -EBUSY; + int err; if (!i915_gfx_present()) return -ENODEV; - i915_acomp = devres_alloc(hdac_acomp_release, sizeof(*i915_acomp), - GFP_KERNEL); - if (!i915_acomp) - return -ENOMEM; - acomp = &i915_acomp->base; - bus->audio_component = acomp; - devres_add(dev, acomp); - - component_match_add(dev, &match, hdac_component_master_match, bus); - ret = component_master_add_with_match(dev, &hdac_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - + err = snd_hdac_acomp_init(bus, NULL, + i915_component_master_match, + sizeof(struct i915_audio_component) - sizeof(*acomp)); + if (err < 0) + return err; + acomp = bus->audio_component; + if (!acomp) + return -ENODEV; + if (!acomp->ops) + request_module("i915"); if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; + snd_hdac_acomp_exit(bus); + return -ENODEV; } - dev_dbg(dev, "bound to i915 component master\n"); - return 0; -out_master_del: - component_master_del(dev, &hdac_component_master_ops); -out_err: - bus->audio_component = NULL; - devres_destroy(dev, hdac_acomp_release, NULL, NULL); - dev_info(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; } EXPORT_SYMBOL_GPL(snd_hdac_i915_init); - -/** - * snd_hdac_i915_exit - Finalize i915 audio component - * @bus: HDA core bus - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function releases the i915 audio component that has been used. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_exit(struct hdac_bus *bus) -{ - struct device *dev = bus->dev; - struct drm_audio_component *acomp = bus->audio_component; - - if (!acomp) - return 0; - - WARN_ON(bus->drm_power_refcount); - if (bus->drm_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hdac_component_master_ops); - - bus->audio_component = NULL; - devres_destroy(dev, hdac_acomp_release, NULL, NULL); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_exit); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bf174a013f2d..1de5491fb9bf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -183,7 +183,7 @@ struct hdmi_spec { hda_nid_t vendor_nid; }; -#ifdef CONFIG_SND_HDA_I915 +#ifdef CONFIG_SND_HDA_COMPONENT static inline bool codec_has_acomp(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2288,7 +2288,7 @@ static void generic_hdmi_free(struct hda_codec *codec) int pin_idx, pcm_idx; if (codec_has_acomp(codec)) - snd_hdac_i915_register_notifier(&codec->bus->core, NULL); + snd_hdac_acomp_register_notifier(&codec->bus->core, NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2471,6 +2471,38 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: + * Pin Widget 5 - PORT B (port = 1 in i915 driver) + * Pin Widget 6 - PORT C (port = 2 in i915 driver) + * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) + */ +static int intel_base_nid(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + return 4; + default: + return 5; + } +} + +static int intel_pin2port(void *audio_ptr, int pin_nid) +{ + int base_nid = intel_base_nid(audio_ptr); + + if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) + return -1; + return pin_nid - base_nid + 1; /* intel port is 1-based */ +} + static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; @@ -2481,16 +2513,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (port < 1 || port > 3) return; - switch (codec->core.vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - pin_nid = port + 0x03; - break; - default: - pin_nid = port + 0x04; - break; - } + pin_nid = port + intel_base_nid(codec) - 1; /* intel port is 1-based */ /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume @@ -2517,8 +2540,9 @@ static void register_i915_notifier(struct hda_codec *codec) * We need make sure audio_ptr is really setup */ wmb(); + spec->drm_audio_ops.pin2port = intel_pin2port; spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&codec->bus->core, + snd_hdac_acomp_register_notifier(&codec->bus->core, &spec->drm_audio_ops); } diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2b7c33db4ded..4748a9d5de3b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1530,6 +1530,11 @@ free_widgets: return ret; } +static int hdac_hdmi_pin2port(void *aptr, int pin) +{ + return pin - 4; /* map NID 0x05 -> port #1 */ +} + static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { struct hdac_device *hdev = aptr; @@ -1584,6 +1589,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) } static struct drm_audio_component_audio_ops aops = { + .pin2port = hdac_hdmi_pin2port, .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; @@ -1812,7 +1818,7 @@ static int hdmi_codec_probe(struct snd_soc_component *component) return ret; aops.audio_ptr = hdev; - ret = snd_hdac_i915_register_notifier(hdev->bus, &aops); + ret = snd_hdac_acomp_register_notifier(hdev->bus, &aops); if (ret < 0) { dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret); return ret; From 5bed9139727f3bad06c9444bd092336a59397e9d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jul 2018 22:32:52 +0200 Subject: [PATCH 270/529] ALSA: rawmidi: Tidy up coding styles Just minor coding style fixes like removal of superfluous white space, adding missing blank lines, etc. No actual code changes at all. Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 101 +++++++++++++++++++++++++------------------ 1 file changed, 59 insertions(+), 42 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b53026a72e73..14dec13ecde9 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -88,6 +88,7 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + return runtime->avail >= runtime->avail_min; } @@ -95,6 +96,7 @@ static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substre size_t count) { struct snd_rawmidi_runtime *runtime = substream->runtime; + return runtime->avail >= runtime->avail_min && (!substream->append || runtime->avail >= count); } @@ -103,6 +105,7 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) { struct snd_rawmidi_runtime *runtime = container_of(work, struct snd_rawmidi_runtime, event_work); + if (runtime->event) runtime->event(runtime->substream); } @@ -111,7 +114,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; - if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL) + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) return -ENOMEM; runtime->substream = substream; spin_lock_init(&runtime->lock); @@ -124,7 +128,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) runtime->avail = 0; else runtime->avail = runtime->buffer_size; - if ((runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) { + runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL); + if (!runtime->buffer) { kfree(runtime); return -ENOMEM; } @@ -143,7 +148,7 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) return 0; } -static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up) +static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) { if (!substream->opened) return; @@ -330,7 +335,7 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, /* called from sound/core/seq/seq_midi.c */ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, - int mode, struct snd_rawmidi_file * rfile) + int mode, struct snd_rawmidi_file *rfile) { struct snd_rawmidi *rmidi; int err; @@ -370,7 +375,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_entry_t wait; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) return -EINVAL; /* invalid combination */ err = nonseekable_open(inode, file); @@ -520,7 +525,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) if (snd_BUG_ON(!rfile)) return -ENXIO; - + rmidi = rfile->rmidi; rawmidi_release_priv(rfile); module_put(rmidi->card->module); @@ -548,7 +553,7 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; - + if (substream == NULL) return -ENODEV; rmidi = substream->rmidi; @@ -568,11 +573,13 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, } static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_info __user * _info) + struct snd_rawmidi_info __user *_info) { struct snd_rawmidi_info info; int err; - if ((err = snd_rawmidi_info(substream, &info)) < 0) + + err = snd_rawmidi_info(substream, &info); + if (err < 0) return err; if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info))) return -EFAULT; @@ -619,13 +626,15 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, { int err; struct snd_rawmidi_info info; + if (get_user(info.device, &_info->device)) return -EFAULT; if (get_user(info.stream, &_info->stream)) return -EFAULT; if (get_user(info.subdevice, &_info->subdevice)) return -EFAULT; - if ((err = snd_rawmidi_info_select(card, &info)) < 0) + err = snd_rawmidi_info_select(card, &info); + if (err < 0) return err; if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info))) return -EFAULT; @@ -633,20 +642,18 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, } int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_params * params) + struct snd_rawmidi_params *params) { char *newbuf, *oldbuf; struct snd_rawmidi_runtime *runtime = substream->runtime; - + if (substream->append && substream->use_count > 1) return -EBUSY; snd_rawmidi_drain_output(substream); - if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) { + if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; - } - if (params->avail_min < 1 || params->avail_min > params->buffer_size) { + if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; - } if (params->buffer_size != runtime->buffer_size) { newbuf = kmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) @@ -667,18 +674,16 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_params * params) + struct snd_rawmidi_params *params) { char *newbuf, *oldbuf; struct snd_rawmidi_runtime *runtime = substream->runtime; snd_rawmidi_drain_input(substream); - if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) { + if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; - } - if (params->avail_min < 1 || params->avail_min > params->buffer_size) { + if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; - } if (params->buffer_size != runtime->buffer_size) { newbuf = kmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) @@ -697,7 +702,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, EXPORT_SYMBOL(snd_rawmidi_input_params); static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_status * status) + struct snd_rawmidi_status *status) { struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -710,7 +715,7 @@ static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, } static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_status * status) + struct snd_rawmidi_status *status) { struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -739,6 +744,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long { int stream; struct snd_rawmidi_info __user *info = argp; + if (get_user(stream, &info->stream)) return -EFAULT; switch (stream) { @@ -753,6 +759,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; + if (copy_from_user(¶ms, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; switch (params.stream) { @@ -772,6 +779,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long { int err = 0; struct snd_rawmidi_status status; + if (copy_from_user(&status, argp, sizeof(struct snd_rawmidi_status))) return -EFAULT; switch (status.stream) { @@ -797,6 +805,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_DROP: { int val; + if (get_user(val, (int __user *) argp)) return -EFAULT; switch (val) { @@ -811,6 +820,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_DRAIN: { int val; + if (get_user(val, (int __user *) argp)) return -EFAULT; switch (val) { @@ -844,7 +854,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE: { int device; - + if (get_user(device, (int __user *)argp)) return -EFAULT; if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ @@ -866,7 +876,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE: { int val; - + if (get_user(val, (int __user *)argp)) return -EFAULT; control->preferred_subdevice[SND_CTL_SUBDEV_RAWMIDI] = val; @@ -1020,6 +1030,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun spin_lock_irq(&runtime->lock); while (!snd_rawmidi_ready(substream)) { wait_queue_entry_t wait; + if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { spin_unlock_irq(&runtime->lock); return result > 0 ? result : -EAGAIN; @@ -1072,7 +1083,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) spin_lock_irqsave(&runtime->lock, flags); result = runtime->avail >= runtime->buffer_size; spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); @@ -1210,7 +1221,7 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); * @substream: the rawmidi substream * @buffer: the buffer pointer * @count: the data size to transfer - * + * * Copies data from the buffer to the device and advances the pointer. * * Return: The copied size if successful, or a negative error code on failure. @@ -1324,6 +1335,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, spin_lock_irq(&runtime->lock); while (!snd_rawmidi_ready_append(substream, count)) { wait_queue_entry_t wait; + if (file->f_flags & O_NONBLOCK) { spin_unlock_irq(&runtime->lock); return result > 0 ? result : -EAGAIN; @@ -1357,6 +1369,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, while (runtime->avail != runtime->buffer_size) { wait_queue_entry_t wait; unsigned int last_avail = runtime->avail; + init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); @@ -1374,7 +1387,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return result; } -static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait) +static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait) { struct snd_rawmidi_file *rfile; struct snd_rawmidi_runtime *runtime; @@ -1411,7 +1424,6 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait) #endif /* - */ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, @@ -1479,8 +1491,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, * Register functions */ -static const struct file_operations snd_rawmidi_f_ops = -{ +static const struct file_operations snd_rawmidi_f_ops = { .owner = THIS_MODULE, .read = snd_rawmidi_read, .write = snd_rawmidi_write, @@ -1535,7 +1546,7 @@ static void release_rawmidi_device(struct device *dev) */ int snd_rawmidi_new(struct snd_card *card, char *id, int device, int output_count, int input_count, - struct snd_rawmidi ** rrawmidi) + struct snd_rawmidi **rrawmidi) { struct snd_rawmidi *rmidi; int err; @@ -1566,21 +1577,24 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, rmidi->dev.release = release_rawmidi_device; dev_set_name(&rmidi->dev, "midiC%iD%i", card->number, device); - if ((err = snd_rawmidi_alloc_substreams(rmidi, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT], - SNDRV_RAWMIDI_STREAM_INPUT, - input_count)) < 0) { + err = snd_rawmidi_alloc_substreams(rmidi, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT], + SNDRV_RAWMIDI_STREAM_INPUT, + input_count); + if (err < 0) { snd_rawmidi_free(rmidi); return err; } - if ((err = snd_rawmidi_alloc_substreams(rmidi, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT], - SNDRV_RAWMIDI_STREAM_OUTPUT, - output_count)) < 0) { + err = snd_rawmidi_alloc_substreams(rmidi, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT], + SNDRV_RAWMIDI_STREAM_OUTPUT, + output_count); + if (err < 0) { snd_rawmidi_free(rmidi); return err; } - if ((err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops); + if (err < 0) { snd_rawmidi_free(rmidi); return err; } @@ -1624,6 +1638,7 @@ static int snd_rawmidi_free(struct snd_rawmidi *rmidi) static int snd_rawmidi_dev_free(struct snd_device *device) { struct snd_rawmidi *rmidi = device->device_data; + return snd_rawmidi_free(rmidi); } @@ -1631,6 +1646,7 @@ static int snd_rawmidi_dev_free(struct snd_device *device) static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device) { struct snd_rawmidi *rmidi = device->private_data; + rmidi->seq_dev = NULL; } #endif @@ -1732,6 +1748,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) list_del_init(&rmidi->list); for (dir = 0; dir < 2; dir++) { struct snd_rawmidi_substream *s; + list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { if (s->runtime) wake_up(&s->runtime->sleep); @@ -1769,7 +1786,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, const struct snd_rawmidi_ops *ops) { struct snd_rawmidi_substream *substream; - + list_for_each_entry(substream, &rmidi->streams[stream].substreams, list) substream->ops = ops; } From 7fdc9b08071b6a3fc85bf90b79e13f6e973a7e5e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jul 2018 22:45:50 +0200 Subject: [PATCH 271/529] ALSA: rawmidi: Simplify error paths Apply the standard idiom: rewrite the multiple unlocks in error paths in the goto-error-and-single-unlock way. Just a code refactoring, and no functional changes. Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 76 +++++++++++++++++++++++--------------------- 1 file changed, 39 insertions(+), 37 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 14dec13ecde9..6b24c2d2dae6 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -338,22 +338,20 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, int mode, struct snd_rawmidi_file *rfile) { struct snd_rawmidi *rmidi; - int err; + int err = 0; if (snd_BUG_ON(!rfile)) return -EINVAL; mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, device); - if (rmidi == NULL) { - mutex_unlock(®ister_mutex); - return -ENODEV; - } - if (!try_module_get(rmidi->card->module)) { - mutex_unlock(®ister_mutex); - return -ENXIO; - } + if (!rmidi) + err = -ENODEV; + else if (!try_module_get(rmidi->card->module)) + err = -ENXIO; mutex_unlock(®ister_mutex); + if (err < 0) + return err; mutex_lock(&rmidi->open_mutex); err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); @@ -1581,26 +1579,25 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT], SNDRV_RAWMIDI_STREAM_INPUT, input_count); - if (err < 0) { - snd_rawmidi_free(rmidi); - return err; - } + if (err < 0) + goto error; err = snd_rawmidi_alloc_substreams(rmidi, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT], SNDRV_RAWMIDI_STREAM_OUTPUT, output_count); - if (err < 0) { - snd_rawmidi_free(rmidi); - return err; - } + if (err < 0) + goto error; err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops); - if (err < 0) { - snd_rawmidi_free(rmidi); - return err; - } + if (err < 0) + goto error; + if (rrawmidi) *rrawmidi = rmidi; return 0; + + error: + snd_rawmidi_free(rmidi); + return err; } EXPORT_SYMBOL(snd_rawmidi_new); @@ -1660,30 +1657,27 @@ static int snd_rawmidi_dev_register(struct snd_device *device) if (rmidi->device >= SNDRV_RAWMIDI_DEVICES) return -ENOMEM; + err = 0; mutex_lock(®ister_mutex); - if (snd_rawmidi_search(rmidi->card, rmidi->device)) { - mutex_unlock(®ister_mutex); - return -EBUSY; - } - list_add_tail(&rmidi->list, &snd_rawmidi_devices); + if (snd_rawmidi_search(rmidi->card, rmidi->device)) + err = -EBUSY; + else + list_add_tail(&rmidi->list, &snd_rawmidi_devices); mutex_unlock(®ister_mutex); + if (err < 0) + return err; + err = snd_register_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device, &snd_rawmidi_f_ops, rmidi, &rmidi->dev); if (err < 0) { rmidi_err(rmidi, "unable to register\n"); - mutex_lock(®ister_mutex); - list_del(&rmidi->list); - mutex_unlock(®ister_mutex); - return err; + goto error; } - if (rmidi->ops && rmidi->ops->dev_register && - (err = rmidi->ops->dev_register(rmidi)) < 0) { - snd_unregister_device(&rmidi->dev); - mutex_lock(®ister_mutex); - list_del(&rmidi->list); - mutex_unlock(®ister_mutex); - return err; + if (rmidi->ops && rmidi->ops->dev_register) { + err = rmidi->ops->dev_register(rmidi); + if (err < 0) + goto error_unregister; } #ifdef CONFIG_SND_OSSEMUL rmidi->ossreg = 0; @@ -1735,6 +1729,14 @@ static int snd_rawmidi_dev_register(struct snd_device *device) } #endif return 0; + + error_unregister: + snd_unregister_device(&rmidi->dev); + error: + mutex_lock(®ister_mutex); + list_del(&rmidi->list); + mutex_unlock(®ister_mutex); + return err; } static int snd_rawmidi_dev_disconnect(struct snd_device *device) From f5beb598b0c4dd023833ae1a7c188ecd987b7125 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jul 2018 23:07:29 +0200 Subject: [PATCH 272/529] ALSA: rawmidi: Minor code refactoring Unify a few open codes with helper functions to improve the readability. Minor behavior changes (rather fixes) are: - runtime->drain clearance is done within lock - active_sensing is updated before resizing buffer in SNDRV_RAWMIDI_IOCTL_PARAMS ioctl. Other than that, simply code cleanups. Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 77 +++++++++++++++++++------------------------- 1 file changed, 33 insertions(+), 44 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6b24c2d2dae6..cc944a3637a2 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -164,17 +164,28 @@ static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, i cancel_work_sync(&substream->runtime->event_work); } -int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) +static void __reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, + bool is_input) +{ + runtime->drain = 0; + runtime->appl_ptr = runtime->hw_ptr = 0; + runtime->avail = is_input ? 0 : runtime->buffer_size; +} + +static void reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, + bool is_input) { unsigned long flags; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_output_trigger(substream, 0); - runtime->drain = 0; spin_lock_irqsave(&runtime->lock, flags); - runtime->appl_ptr = runtime->hw_ptr = 0; - runtime->avail = runtime->buffer_size; + __reset_runtime_ptrs(runtime, is_input); spin_unlock_irqrestore(&runtime->lock, flags); +} + +int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) +{ + snd_rawmidi_output_trigger(substream, 0); + reset_runtime_ptrs(substream->runtime, false); return 0; } EXPORT_SYMBOL(snd_rawmidi_drop_output); @@ -213,15 +224,8 @@ EXPORT_SYMBOL(snd_rawmidi_drain_output); int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) { - unsigned long flags; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_input_trigger(substream, 0); - runtime->drain = 0; - spin_lock_irqsave(&runtime->lock, flags); - runtime->appl_ptr = runtime->hw_ptr = 0; - runtime->avail = 0; - spin_unlock_irqrestore(&runtime->lock, flags); + reset_runtime_ptrs(substream->runtime, true); return 0; } EXPORT_SYMBOL(snd_rawmidi_drain_input); @@ -639,15 +643,12 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, return 0; } -int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_params *params) +static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, + struct snd_rawmidi_params *params, + bool is_input) { char *newbuf, *oldbuf; - struct snd_rawmidi_runtime *runtime = substream->runtime; - if (substream->append && substream->use_count > 1) - return -EBUSY; - snd_rawmidi_drain_output(substream); if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) @@ -660,42 +661,30 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; - runtime->avail = runtime->buffer_size; - runtime->appl_ptr = runtime->hw_ptr = 0; + __reset_runtime_ptrs(runtime, is_input); spin_unlock_irq(&runtime->lock); kfree(oldbuf); } runtime->avail_min = params->avail_min; - substream->active_sensing = !params->no_active_sensing; return 0; } + +int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, + struct snd_rawmidi_params *params) +{ + if (substream->append && substream->use_count > 1) + return -EBUSY; + snd_rawmidi_drain_output(substream); + substream->active_sensing = !params->no_active_sensing; + return resize_runtime_buffer(substream->runtime, params, false); +} EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { - char *newbuf, *oldbuf; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_drain_input(substream); - if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) - return -EINVAL; - if (params->avail_min < 1 || params->avail_min > params->buffer_size) - return -EINVAL; - if (params->buffer_size != runtime->buffer_size) { - newbuf = kmalloc(params->buffer_size, GFP_KERNEL); - if (!newbuf) - return -ENOMEM; - spin_lock_irq(&runtime->lock); - oldbuf = runtime->buffer; - runtime->buffer = newbuf; - runtime->buffer_size = params->buffer_size; - runtime->appl_ptr = runtime->hw_ptr = 0; - spin_unlock_irq(&runtime->lock); - kfree(oldbuf); - } - runtime->avail_min = params->avail_min; - return 0; + return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); From ef4db239cda2d74f53120e223643b0f5bbf947c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jul 2018 23:12:33 +0200 Subject: [PATCH 273/529] ALSA: rawmidi: Use kvmalloc() for buffers The size of in-kernel rawmidi buffers may be big up to 1MB, and it can be specified freely by user-space; which implies that user-space may trigger kmalloc() errors frequently. This patch replaces the buffer allocation via kvmalloc() for dealing with bigger buffers gracefully. Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index cc944a3637a2..69517e18ef07 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -128,7 +129,7 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) runtime->avail = 0; else runtime->avail = runtime->buffer_size; - runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL); + runtime->buffer = kvmalloc(runtime->buffer_size, GFP_KERNEL); if (!runtime->buffer) { kfree(runtime); return -ENOMEM; @@ -142,7 +143,7 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; - kfree(runtime->buffer); + kvfree(runtime->buffer); kfree(runtime); substream->runtime = NULL; return 0; @@ -654,7 +655,7 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { - newbuf = kmalloc(params->buffer_size, GFP_KERNEL); + newbuf = kvmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; spin_lock_irq(&runtime->lock); @@ -663,7 +664,7 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, runtime->buffer_size = params->buffer_size; __reset_runtime_ptrs(runtime, is_input); spin_unlock_irq(&runtime->lock); - kfree(oldbuf); + kvfree(oldbuf); } runtime->avail_min = params->avail_min; return 0; From fa84cf094ef9667e2b91c104b0a788fd1896f482 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Jul 2018 07:40:53 +0200 Subject: [PATCH 274/529] ALSA: pcm: Nuke snd_pcm_lib_mmap_vmalloc() snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with somewhat special for vmalloc handling, but in the end, this turned to just the default handler, i.e. NULL. As the situation has never changed over decades, let's rip it off. Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- drivers/staging/most/sound/sound.c | 1 - include/sound/pcm.h | 2 -- sound/drivers/aloop.c | 1 - sound/drivers/vx/vx_pcm.c | 2 -- sound/firewire/bebob/bebob_pcm.c | 1 - sound/firewire/dice/dice-pcm.c | 2 -- sound/firewire/digi00x/digi00x-pcm.c | 1 - sound/firewire/fireface/ff-pcm.c | 1 - sound/firewire/fireworks/fireworks_pcm.c | 1 - sound/firewire/isight.c | 1 - sound/firewire/motu/motu-pcm.c | 2 -- sound/firewire/oxfw/oxfw-pcm.c | 2 -- sound/firewire/tascam/tascam-pcm.c | 1 - sound/mips/sgio2audio.c | 3 --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/soc/codecs/rt5514-spi.c | 1 - sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 1 - sound/usb/hiface/pcm.c | 1 - sound/usb/misc/ua101.c | 2 -- sound/usb/pcm.c | 2 -- 21 files changed, 30 deletions(-) diff --git a/drivers/staging/most/sound/sound.c b/drivers/staging/most/sound/sound.c index 04c18323c2ea..89b02fc305b8 100644 --- a/drivers/staging/most/sound/sound.c +++ b/drivers/staging/most/sound/sound.c @@ -457,7 +457,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = pcm_trigger, .pointer = pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int split_arg_list(char *buf, char **card_name, u16 *ch_num, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fcdf358a25f0..1206045ccf03 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1342,8 +1342,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -#define snd_pcm_lib_mmap_vmalloc NULL - /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer * @dma: DMA number diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 78a2fdc38531..1e34e6381baa 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -778,7 +778,6 @@ static const struct snd_pcm_ops loopback_pcm_ops = { .trigger = loopback_trigger, .pointer = loopback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int loopback_pcm_new(struct loopback *loopback, diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 380a028469c4..ba80f459bdc5 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -883,7 +883,6 @@ static const struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1105,7 +1104,6 @@ static const struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index e6adab3ef42e..ea9b86450580 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -373,7 +373,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 80351b29fe0d..bb3ef5ff3488 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -412,7 +412,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -425,7 +424,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int capture, playback; diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 796f4b4645f5..fdcff0460c53 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -352,7 +352,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index e3c16308363d..bf47f9ec8703 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -383,7 +383,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 40faed5e6968..aed566d82726 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -397,7 +397,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 3919e186a30b..30957477e005 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -454,7 +454,6 @@ static int isight_create_pcm(struct isight *isight) .trigger = isight_trigger, .pointer = isight_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 4330220890e8..ab69d7e6ac05 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -363,7 +363,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -376,7 +375,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 3dd46285c0e2..b3f6503dd34d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -389,7 +389,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -402,7 +401,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int cap = 0; diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 6ec8ec634d4d..e4cc8990e195 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -279,7 +279,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9fb68b35de5a..3ec9391a4736 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -685,7 +685,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -698,7 +697,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -711,7 +709,6 @@ static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 4a2354b5ae00..98a6863e933c 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -276,7 +276,6 @@ static const struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 18686ffb0cd5..6478d10c4f4a 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -268,7 +268,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, .pointer = rt5514_spi_pcm_pointer, - .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, }; diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2dd2518a71d3..f8ef3e2a8ca0 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -565,7 +565,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 15344d39a6cd..c6108a3d7f8f 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -348,7 +348,6 @@ static const struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 396c317115b1..e1fbb9cc9ea7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -523,7 +523,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = hiface_pcm_trigger, .pointer = hiface_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int hiface_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 386fbfd5c617..a0b6d039017f 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -900,7 +900,6 @@ static const struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_pcm_ops = { @@ -913,7 +912,6 @@ static const struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 160f52c4871b..4b930fa47277 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,7 +1694,6 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_substream_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_capture_ops = { @@ -1707,7 +1706,6 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_substream_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_playback_dev_ops = { From bb4b894addb09a069c072a0a032f644cc470d17f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:28 +0100 Subject: [PATCH 275/529] ASoC: core: add support to card re-bind using component framework This patch aims at achieving dynamic behaviour of audio card when the dependent components disappear and reappear. With this patch the card is removed if any of the dependent component is removed and card is added back if the dependent component comes back. All this is done using component framework and matching based on component name. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc.h | 7 +++++ sound/soc/soc-core.c | 62 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..a23ecdf3eff1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -1090,6 +1091,12 @@ struct snd_soc_card { struct work_struct deferred_resume_work; + /* component framework related */ + bool components_added; + /* set in machine driver to enable/disable auto re-binding */ + bool auto_bind; + struct component_match *match; + /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3be0310d5c81..08e189485009 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -279,11 +279,28 @@ static inline void snd_soc_debugfs_exit(void) #endif +static int snd_soc_card_comp_compare(struct device *dev, void *data) +{ + struct snd_soc_component *component; + + lockdep_assert_held(&client_mutex); + list_for_each_entry(component, &component_list, list) { + if (dev == component->dev) { + if (!strcmp(component->name, data)) + return 1; + break; + } + } + + return 0; +} + static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, struct snd_soc_component *component) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_rtdcom_list *new_rtdcom; + char *cname; for_each_rtdcom(rtd, rtdcom) { /* already connected */ @@ -300,6 +317,13 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, list_add_tail(&new_rtdcom->list, &rtd->component_list); + if (rtd->card->auto_bind && !rtd->card->components_added) { + cname = devm_kasprintf(rtd->card->dev, GFP_KERNEL, + "%s", component->name); + component_match_add(rtd->card->dev, &rtd->card->match, + snd_soc_card_comp_compare, cname); + } + return 0; } @@ -835,6 +859,25 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, return false; } +static int snd_soc_card_comp_bind(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + if (card->instantiated) + return 0; + + return snd_soc_register_card(card); +} + +static void snd_soc_card_comp_unbind(struct device *dev) +{ +} + +static const struct component_master_ops snd_soc_card_comp_ops = { + .bind = snd_soc_card_comp_bind, + .unbind = snd_soc_card_comp_unbind, +}; + static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2126,6 +2169,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); + if (card->auto_bind && !card->components_added) { + component_master_add_with_match(card->dev, + &snd_soc_card_comp_ops, + card->match); + card->components_added = true; + } mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2749,6 +2798,9 @@ int snd_soc_unregister_card(struct snd_soc_card *card) dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } + if (!card->auto_bind && card->components_added) + component_master_del(card->dev, &snd_soc_card_comp_ops); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); @@ -3161,8 +3213,17 @@ int snd_soc_add_component(struct device *dev, snd_soc_component_add(component); + ret = component_add(dev, NULL); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to add Component: %d\n", ret); + goto err_comp; + } + return 0; +err_comp: + soc_remove_component(component); + snd_soc_unregister_dais(component); err_cleanup: snd_soc_component_cleanup(component); err_free: @@ -3210,6 +3271,7 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_unlock(&client_mutex); if (found) { + component_del(dev, NULL); snd_soc_component_cleanup(component); } From 605fcb69918528e1a448cba4d358cbd8ed532146 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:29 +0100 Subject: [PATCH 276/529] ASoC: qdsp6: q6afe-dai: remove component fw related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 34 ++++---------------------------- 1 file changed, 4 insertions(+), 30 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 074582afda85..e988692a3ced 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -4,7 +4,6 @@ #include #include -#include #include #include #include @@ -1395,11 +1394,12 @@ static void of_q6afe_parse_dai_data(struct device *dev, } } -static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) +static int q6afe_dai_dev_probe(struct platform_device *pdev) { struct q6afe_dai_data *dai_data; + struct device *dev = &pdev->dev; - dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL); + dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL); if (!dai_data) return -ENOMEM; @@ -1407,35 +1407,10 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) of_q6afe_parse_dai_data(dev, dai_data); - return snd_soc_register_component(dev, &q6afe_dai_component, + return devm_snd_soc_register_component(dev, &q6afe_dai_component, q6afe_dais, ARRAY_SIZE(q6afe_dais)); } -static void q6afe_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6afe_dai_data *dai_data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - kfree(dai_data); -} - -static const struct component_ops q6afe_dai_comp_ops = { - .bind = q6afe_dai_bind, - .unbind = q6afe_dai_unbind, -}; - -static int q6afe_dai_dev_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6afe_dai_comp_ops); -} - -static int q6afe_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6afe_dai_comp_ops); - return 0; -} - static const struct of_device_id q6afe_dai_device_id[] = { { .compatible = "qcom,q6afe-dais" }, {}, @@ -1448,7 +1423,6 @@ static struct platform_driver q6afe_dai_platform_driver = { .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, - .remove = q6afe_dai_dev_remove, }; module_platform_driver(q6afe_dai_platform_driver); From f924e4fd89659908107a5529f02c21edb4770dba Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:30 +0100 Subject: [PATCH 277/529] ASoC: qdsp6: q6asm-dai: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 ++++---------------------------- 1 file changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 1196dc7483d2..acf96c6549fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -7,7 +7,6 @@ #include #include #include -#include #include #include #include @@ -561,14 +560,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; -static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) +static int q6asm_dai_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct device_node *node = dev->of_node; struct of_phandle_args args; struct q6asm_dai_data *pdata; int rc; - pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL); + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) return -ENOMEM; @@ -580,36 +580,10 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(dev, pdata); - return snd_soc_register_component(dev, &q6asm_fe_dai_component, + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, q6asm_fe_dais, ARRAY_SIZE(q6asm_fe_dais)); } -static void q6asm_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6asm_dai_data *pdata = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(pdata); - -} - -static const struct component_ops q6asm_dai_comp_ops = { - .bind = q6asm_dai_bind, - .unbind = q6asm_dai_unbind, -}; - -static int q6asm_dai_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6asm_dai_comp_ops); -} - -static int q6asm_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6asm_dai_comp_ops); - return 0; -} static const struct of_device_id q6asm_dai_device_id[] = { { .compatible = "qcom,q6asm-dais" }, @@ -623,7 +597,6 @@ static struct platform_driver q6asm_dai_platform_driver = { .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, - .remove = q6asm_dai_dev_remove, }; module_platform_driver(q6asm_dai_platform_driver); From 791940779d651c2219e97702d2245b5420b0c8ae Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:31 +0100 Subject: [PATCH 278/529] ASoC: qdsp6: q6routing: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 34 +++++++------------------------- 1 file changed, 7 insertions(+), 27 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 35269b492761..1d33b00e5b44 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -8,7 +8,6 @@ #include #include #include -#include #include #include #include @@ -977,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = { .num_dapm_routes = ARRAY_SIZE(intercon), }; -static int q6routing_dai_bind(struct device *dev, struct device *master, - void *data) +static int q6pcm_routing_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL); if (!routing_data) return -ENOMEM; @@ -989,35 +989,15 @@ static int q6routing_dai_bind(struct device *dev, struct device *master, mutex_init(&routing_data->lock); dev_set_drvdata(dev, routing_data); - return snd_soc_register_component(dev, &msm_soc_routing_component, + return devm_snd_soc_register_component(dev, &msm_soc_routing_component, NULL, 0); } -static void q6routing_dai_unbind(struct device *dev, struct device *master, - void *d) -{ - struct msm_routing_data *data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(data); - - routing_data = NULL; -} - -static const struct component_ops q6routing_dai_comp_ops = { - .bind = q6routing_dai_bind, - .unbind = q6routing_dai_unbind, -}; - -static int q6pcm_routing_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6routing_dai_comp_ops); -} - static int q6pcm_routing_remove(struct platform_device *pdev) { - component_del(&pdev->dev, &q6routing_dai_comp_ops); + kfree(routing_data); + routing_data = NULL; + return 0; } From 90ae7105eaf19342bb11e554059d62b84e01da12 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:32 +0100 Subject: [PATCH 279/529] ASoC: qcom: apq8096: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 81 +++++----------------------------------- 1 file changed, 9 insertions(+), 72 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index cab8c4ff7c00..a56156281c8d 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -129,17 +129,18 @@ err: return ret; } -static int apq8096_bind(struct device *dev) +static int apq8096_platform_probe(struct platform_device *pdev) { struct snd_soc_card *card; + struct device *dev = &pdev->dev; int ret; card = kzalloc(sizeof(*card), GFP_KERNEL); if (!card) return -ENOMEM; - component_bind_all(dev, card); card->dev = dev; + card->auto_bind = true; dev_set_drvdata(dev, card); ret = apq8096_sbc_parse_of(card); if (ret) { @@ -154,82 +155,18 @@ static int apq8096_bind(struct device *dev) return 0; err: - component_unbind_all(dev, card); kfree(card); return ret; } -static void apq8096_unbind(struct device *dev) -{ - struct snd_soc_card *card = dev_get_drvdata(dev); - - snd_soc_unregister_card(card); - component_unbind_all(dev, card); - kfree(card->dai_link); - kfree(card); -} - -static const struct component_master_ops apq8096_ops = { - .bind = apq8096_bind, - .unbind = apq8096_unbind, -}; - -static int apq8016_compare_of(struct device *dev, void *data) -{ - return dev->of_node == data; -} - -static void apq8016_release_of(struct device *dev, void *data) -{ - of_node_put(data); -} - -static int add_audio_components(struct device *dev, - struct component_match **matchptr) -{ - struct device_node *np, *platform, *cpu, *node, *dai_node; - - node = dev->of_node; - - for_each_child_of_node(node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (cpu) { - dai_node = of_parse_phandle(cpu, "sound-dai", 0); - of_node_get(dai_node); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - - platform = of_get_child_by_name(np, "platform"); - if (platform) { - dai_node = of_parse_phandle(platform, "sound-dai", 0); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - } - - return 0; -} - -static int apq8096_platform_probe(struct platform_device *pdev) -{ - struct component_match *match = NULL; - int ret; - - ret = add_audio_components(&pdev->dev, &match); - if (ret) - return ret; - - return component_master_add_with_match(&pdev->dev, &apq8096_ops, match); -} - static int apq8096_platform_remove(struct platform_device *pdev) { - component_master_del(&pdev->dev, &apq8096_ops); + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + + card->auto_bind = false; + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(card); return 0; } From bf270262b7b8bb7b48a846c613f74e800abba392 Mon Sep 17 00:00:00 2001 From: Sriram Periyasamy Date: Mon, 16 Jul 2018 15:32:34 +0530 Subject: [PATCH 280/529] ASoC: hdac_hdmi: Add documentation for power management Add documentation for power management of HDAC HDMI codec device for various scenarios such as S0/S3, probe and playback use case. Signed-off-by: Sriram Periyasamy Signed-off-by: Sanyog Kale Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 69 ++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..30ccc902e5cf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2103,6 +2103,75 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_ext_device *edev = to_hda_ext_device(dev); From e32d99af6830c9a8f37b4f2637ef0cdc60fa79fb Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:50 +0200 Subject: [PATCH 281/529] ASoC: meson: add axg fifos DT binding documentation Add the DT bindings documentation for axg's FIFOs: TODDR and FRDDR. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,axg-fifo.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt new file mode 100644 index 000000000000..3dfc2515e5c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt @@ -0,0 +1,23 @@ +* Amlogic Audio FIFO controllers + +Required properties: +- compatible: 'amlogic,axg-toddr' or + 'amlogic,axg-frddr' +- reg: physical base address of the controller and length of memory + mapped region. +- interrupts: interrupt specifier for the fifo. +- clocks: phandle to the fifo peripheral clock provided by the audio + clock controller. +- resets: phandle to memory ARB line provided by the arb reset controller. +- #sound-dai-cells: must be 0. + +Example of FRDDR A on the A113 SoC: + +frddr_a: audio-controller@1c0 { + compatible = "amlogic,axg-frddr"; + reg = <0x0 0x1c0 0x0 0x1c>; + #sound-dai-cells = <0>; + interrupts = ; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + resets = <&arb AXG_ARB_FRDDR_A>; +}; From 6dc4fa179fb86d2c986b2bc8a8377fe4d8c0428d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:51 +0200 Subject: [PATCH 282/529] ASoC: meson: add axg fifo base driver Amlogic's axg SoCs have two types of fifos which are the memory interfaces of the audio subsystem. FRDDR provides the playback interface while TODDR provides the capture interface. The way these fifos operate is very similar. Only a few settings are specific to each. They implement the same pcm driver here and the specifics of each will be dealt with the related DAI driver. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 5 + sound/soc/meson/axg-fifo.c | 341 +++++++++++++++++++++++++++++++++++++ sound/soc/meson/axg-fifo.h | 80 +++++++++ 6 files changed, 436 insertions(+) create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 41af6b9cc350..1cf11cf51e1d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" +source "sound/soc/meson/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/qcom/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 06389a5385d7..62a5f87c3cfc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ +obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig new file mode 100644 index 000000000000..c3eb5e050308 --- /dev/null +++ b/sound/soc/meson/Kconfig @@ -0,0 +1,8 @@ +menu "ASoC support for Amlogic platforms" + depends on ARCH_MESON || COMPILE_TEST + +config SND_MESON_AXG_FIFO + tristate + select REGMAP_MMIO + +endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile new file mode 100644 index 000000000000..75289b6b3ade --- /dev/null +++ b/sound/soc/meson/Makefile @@ -0,0 +1,5 @@ +# SPDX-License-Identifier: (GPL-2.0 OR MIT) + +snd-soc-meson-axg-fifo-objs := axg-fifo.o + +obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c new file mode 100644 index 000000000000..db367d85290f --- /dev/null +++ b/sound/soc/meson/axg-fifo.c @@ -0,0 +1,341 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +/* + * This file implements the platform operations common to the playback and + * capture frontend DAI. The logic behind this two types of fifo is very + * similar but some difference exist. + * These differences the respective DAI drivers + */ + +static struct snd_pcm_hardware axg_fifo_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = AXG_FIFO_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .period_bytes_min = AXG_FIFO_MIN_DEPTH, + .period_bytes_max = UINT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return rtd->cpu_dai; +} + +static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return snd_soc_dai_get_drvdata(dai); +} + +static struct device *axg_fifo_dev(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return dai->dev; +} + +static void __dma_enable(struct axg_fifo *fifo, bool enable) +{ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN, + enable ? CTRL0_DMA_EN : 0); +} + +static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + __dma_enable(fifo, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + __dma_enable(fifo, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct snd_pcm_runtime *runtime = ss->runtime; + unsigned int addr; + + regmap_read(fifo->map, FIFO_STATUS2, &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = ss->runtime; + struct axg_fifo *fifo = axg_fifo_data(ss); + dma_addr_t end_ptr; + unsigned int burst_num; + int ret; + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup dma memory pointers */ + end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr); + regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr); + + /* Setup interrupt periodicity */ + burst_num = params_period_bytes(params) / AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_INT_ADDR, burst_num); + + /* Enable block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + + return 0; +} + +static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + /* Disable the block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + + return snd_pcm_lib_free_pages(ss); +} + +static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) +{ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + CTRL1_INT_CLR(mask)); + + /* Clear must also be cleared */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + 0); +} + +static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) +{ + struct snd_pcm_substream *ss = dev_id; + struct axg_fifo *fifo = axg_fifo_data(ss); + unsigned int status; + + regmap_read(fifo->map, FIFO_STATUS1, &status); + + status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + if (status & FIFO_INT_COUNT_REPEAT) + snd_pcm_period_elapsed(ss); + else + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); + + /* Ack irqs */ + axg_fifo_ack_irq(fifo, status); + + return !status ? IRQ_NONE : IRQ_HANDLED; +} + +static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct device *dev = axg_fifo_dev(ss); + int ret; + + snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw); + + /* + * Make sure the buffer and period size are multiple of the FIFO + * minimum depth size + */ + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, + dev_name(dev), ss); + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Setup status2 so it reports the memory pointer */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_STATUS2_SEL_MASK, + CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + + /* Make sure the dma is initially disabled */ + __dma_enable(fifo, false); + + /* Disable irqs until params are ready */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_MASK), 0); + + /* Clear any pending interrupt */ + axg_fifo_ack_irq(fifo, FIFO_INT_MASK); + + /* Take memory arbitror out of reset */ + ret = reset_control_deassert(fifo->arb); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +static int axg_fifo_pcm_close(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + int ret; + + /* Put the memory arbitror back in reset */ + ret = reset_control_assert(fifo->arb); + + /* Disable fifo ip and register access */ + clk_disable_unprepare(fifo->pclk); + + /* remove IRQ */ + free_irq(fifo->irq, ss); + + return ret; +} + +const struct snd_pcm_ops axg_fifo_pcm_ops = { + .open = axg_fifo_pcm_open, + .close = axg_fifo_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = axg_fifo_pcm_hw_params, + .hw_free = axg_fifo_pcm_hw_free, + .pointer = axg_fifo_pcm_pointer, + .trigger = axg_fifo_pcm_trigger, +}; +EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops); + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = axg_fifo_hw.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream, + SNDRV_DMA_TYPE_DEV, card->dev, + size, size); +} +EXPORT_SYMBOL_GPL(axg_fifo_pcm_new); + +static const struct regmap_config axg_fifo_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = FIFO_STATUS2, +}; + +int axg_fifo_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_fifo_match_data *data; + struct axg_fifo *fifo; + struct resource *res; + void __iomem *regs; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + platform_set_drvdata(pdev, fifo); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg); + if (IS_ERR(fifo->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(fifo->map)); + return PTR_ERR(fifo->map); + } + + fifo->pclk = devm_clk_get(dev, NULL); + if (IS_ERR(fifo->pclk)) { + if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %ld\n", + PTR_ERR(fifo->pclk)); + return PTR_ERR(fifo->pclk); + } + + fifo->arb = devm_reset_control_get_exclusive(dev, NULL); + if (IS_ERR(fifo->arb)) { + if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) + dev_err(dev, "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); + return PTR_ERR(fifo->arb); + } + + fifo->irq = of_irq_get(dev->of_node, 0); + if (fifo->irq <= 0) { + dev_err(dev, "failed to get irq: %d\n", fifo->irq); + return fifo->irq; + } + + return devm_snd_soc_register_component(dev, data->component_drv, + data->dai_drv, 1); +} +EXPORT_SYMBOL_GPL(axg_fifo_probe); + +MODULE_DESCRIPTION("Amlogic AXG fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h new file mode 100644 index 000000000000..cb6c4013ca33 --- /dev/null +++ b/sound/soc/meson/axg-fifo.h @@ -0,0 +1,80 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_FIFO_H +#define _MESON_AXG_FIFO_H + +struct clk; +struct platform_device; +struct regmap; +struct reset_control; + +struct snd_soc_component_driver; +struct snd_soc_dai; +struct snd_soc_dai_driver; +struct snd_pcm_ops; +struct snd_soc_pcm_runtime; + +#define AXG_FIFO_CH_MAX 128 +#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AXG_FIFO_BURST 8 +#define AXG_FIFO_MIN_CNT 64 +#define AXG_FIFO_MIN_DEPTH (AXG_FIFO_BURST * AXG_FIFO_MIN_CNT) + +#define FIFO_INT_ADDR_FINISH BIT(0) +#define FIFO_INT_ADDR_INT BIT(1) +#define FIFO_INT_COUNT_REPEAT BIT(2) +#define FIFO_INT_COUNT_ONCE BIT(3) +#define FIFO_INT_FIFO_ZERO BIT(4) +#define FIFO_INT_FIFO_DEPTH BIT(5) +#define FIFO_INT_MASK GENMASK(7, 0) + +#define FIFO_CTRL0 0x00 +#define CTRL0_DMA_EN BIT(31) +#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_SEL_MASK GENMASK(2, 0) +#define CTRL0_SEL_SHIFT 0 +#define FIFO_CTRL1 0x04 +#define CTRL1_INT_CLR(x) ((x) << 0) +#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) +#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define STATUS2_SEL_DDR_READ 0 +#define CTRL1_THRESHOLD_MASK GENMASK(23, 16) +#define CTRL1_THRESHOLD(x) ((x) << 16) +#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) +#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define FIFO_START_ADDR 0x08 +#define FIFO_FINISH_ADDR 0x0c +#define FIFO_INT_ADDR 0x10 +#define FIFO_STATUS1 0x14 +#define STATUS1_INT_STS(x) ((x) << 0) +#define FIFO_STATUS2 0x18 + +struct axg_fifo { + struct regmap *map; + struct clk *pclk; + struct reset_control *arb; + int irq; +}; + +struct axg_fifo_match_data { + const struct snd_soc_component_driver *component_drv; + struct snd_soc_dai_driver *dai_drv; +}; + +extern const struct snd_pcm_ops axg_fifo_pcm_ops; + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type); +int axg_fifo_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_FIFO_H */ From 57d552e3ea76003643b2e771042659ce71bac7c2 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:52 +0200 Subject: [PATCH 283/529] ASoC: meson: add axg frddr driver Add the playback memory interface of Amlogic's axg SoCs. This device pulls data from DDR to an internal FIFO. This FIFO is then used to feed TDM and SPDIF Output devices. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-frddr.c | 141 ++++++++++++++++++++++++++++++++++++ 3 files changed, 150 insertions(+) create mode 100644 sound/soc/meson/axg-frddr.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index c3eb5e050308..cdd78f62e8d7 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -5,4 +5,11 @@ config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO +config SND_MESON_AXG_FRDDR + tristate "Amlogic AXG Playback FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend playback interfaces + embedded in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 75289b6b3ade..9c5d7d4a8e33 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,5 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) snd-soc-meson-axg-fifo-objs := axg-fifo.o +snd-soc-meson-axg-frddr-objs := axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o +obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c new file mode 100644 index 000000000000..a6f6f6a2eca8 --- /dev/null +++ b/sound/soc/meson/axg-frddr.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +/* This driver implements the frontend playback DAI of AXG based SoCs */ + +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +#define CTRL0_FRDDR_PP_MODE BIT(30) + +static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_depth, fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); + + /* + * TODO: We could adapt the fifo depth and the fifo threshold + * depending on the expected memory throughput and lantencies + * For now, we'll just use the same values as the vendor kernel + * Depth and threshold are zero based. + */ + fifo_depth = AXG_FIFO_MIN_CNT - 1; + fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK, + CTRL1_FRDDR_DEPTH(fifo_depth) | + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK); +} + +static const struct snd_soc_dai_ops axg_frddr_ops = { + .startup = axg_frddr_dai_startup, + .shutdown = axg_frddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_frddr_dai_drv = { + .name = "FRDDR", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_frddr_ops, + .pcm_new = axg_frddr_pcm_new, +}; + +static const char * const axg_frddr_sel_texts[] = { + "OUT 0", "OUT 1", "OUT 2", "OUT 3" +}; + +static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT, + axg_frddr_sel_texts); + +static const struct snd_kcontrol_new axg_frddr_out_demux = + SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = { + SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0, + &axg_frddr_out_demux), + SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = { + { "SINK SEL", NULL, "Playback" }, + { "OUT 0", "OUT 0", "SINK SEL" }, + { "OUT 1", "OUT 1", "SINK SEL" }, + { "OUT 2", "OUT 2", "SINK SEL" }, + { "OUT 3", "OUT 3", "SINK SEL" }, +}; + +static const struct snd_soc_component_driver axg_frddr_component_drv = { + .dapm_widgets = axg_frddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_frddr_dapm_widgets), + .dapm_routes = axg_frddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_frddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_frddr_match_data = { + .component_drv = &axg_frddr_component_drv, + .dai_drv = &axg_frddr_dai_drv +}; + +static const struct of_device_id axg_frddr_of_match[] = { + { + .compatible = "amlogic,axg-frddr", + .data = &axg_frddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_frddr_of_match); + +static struct platform_driver axg_frddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-frddr", + .of_match_table = axg_frddr_of_match, + }, +}; +module_platform_driver(axg_frddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG playback fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From 7ed4877b403c9343a8e2c7581d9bcfceef0f40cf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:53 +0200 Subject: [PATCH 284/529] ASoC: meson: add axg toddr driver Add the capture memory interface of Amlogic's axg SoCs. TDM, SPDIF or PDM input devices place audio samples inside this FIFO. The FIFO content is then pushed to DDR Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-toddr.c | 199 ++++++++++++++++++++++++++++++++++++ 3 files changed, 208 insertions(+) create mode 100644 sound/soc/meson/axg-toddr.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index cdd78f62e8d7..3916060edcae 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -12,4 +12,11 @@ config SND_MESON_AXG_FRDDR Select Y or M to add support for the frontend playback interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_TODDR + tristate "Amlogic AXG Capture FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend capture interfaces + embedded in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 9c5d7d4a8e33..12edf2db2a24 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -2,6 +2,8 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o +snd-soc-meson-axg-toddr-objs := axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o +obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c new file mode 100644 index 000000000000..c2c9bb312586 --- /dev/null +++ b/sound/soc/meson/axg-toddr.c @@ -0,0 +1,199 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +/* This driver implements the frontend capture DAI of AXG based SoCs */ + +#include +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +#define CTRL0_TODDR_SEL_RESAMPLE BIT(30) +#define CTRL0_TODDR_EXT_SIGNED BIT(29) +#define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) +#define CTRL0_TODDR_TYPE(x) ((x) << 13) +#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) +#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) +#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) +#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) + +static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE); +} + +static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int type, width, msb = 31; + + /* + * NOTE: + * Almost all backend will place the MSB at bit 31, except SPDIF Input + * which will put it at index 28. When adding support for the SPDIF + * Input, we'll need to find which type of backend we are connected to. + */ + + switch (params_physical_width(params)) { + case 8: + type = 0; /* 8 samples of 8 bits */ + break; + case 16: + type = 2; /* 4 samples of 16 bits - right justified */ + break; + case 32: + type = 4; /* 2 samples of 32 bits - right justified */ + break; + default: + return -EINVAL; + } + + width = params_width(params); + + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_TODDR_TYPE_MASK | + CTRL0_TODDR_MSB_POS_MASK | + CTRL0_TODDR_LSB_POS_MASK, + CTRL0_TODDR_TYPE(type) | + CTRL0_TODDR_MSB_POS(msb) | + CTRL0_TODDR_LSB_POS(msb - (width - 1))); + + return 0; +} + +static int axg_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Select orginal data - resampling not supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0); + + /* Only signed format are supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED, + CTRL0_TODDR_EXT_SIGNED); + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0); + + /* TODDR does not have a configurable fifo depth */ + fifo_threshold = AXG_FIFO_MIN_CNT - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK, + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static const struct snd_soc_dai_ops axg_toddr_ops = { + .hw_params = axg_toddr_dai_hw_params, + .startup = axg_toddr_dai_startup, + .shutdown = axg_toddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_toddr_dai_drv = { + .name = "TODDR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_toddr_ops, + .pcm_new = axg_toddr_pcm_new, +}; + +static const char * const axg_toddr_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6" +}; + +static const unsigned int axg_toddr_sel_values[] = { + 0, 1, 2, 3, 4, 6 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0, + CTRL0_SEL_SHIFT, CTRL0_SEL_MASK, + axg_toddr_sel_texts, axg_toddr_sel_values); + +static const struct snd_kcontrol_new axg_toddr_in_mux = + SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux), + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = { + { "Capture", NULL, "SRC SEL" }, + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 6", "IN 6" }, +}; + +static const struct snd_soc_component_driver axg_toddr_component_drv = { + .dapm_widgets = axg_toddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets), + .dapm_routes = axg_toddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_toddr_match_data = { + .component_drv = &axg_toddr_component_drv, + .dai_drv = &axg_toddr_dai_drv +}; + +static const struct of_device_id axg_toddr_of_match[] = { + { + .compatible = "amlogic,axg-toddr", + .data = &axg_toddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_toddr_of_match); + +static struct platform_driver axg_toddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-toddr", + .of_match_table = axg_toddr_of_match, + }, +}; +module_platform_driver(axg_toddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG capture fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From eb257e6607f96fd70a443750e9eaddbb49ba87ff Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:54 +0200 Subject: [PATCH 285/529] ASoC: meson: add axg spdif output DT bindings documentation Add the DT bindings documentation for axg's SPDIF output. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,axg-spdifout.txt | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt new file mode 100644 index 000000000000..521c38ad89e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt @@ -0,0 +1,20 @@ +* Amlogic Audio SPDIF Output + +Required properties: +- compatible: 'amlogic,axg-spdifout' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "mclk" : master clock +- #sound-dai-cells: must be 0. + +Example on the A113 SoC: + +spdifout: audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x0 0x480 0x0 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; +}; From 53eb4b7aaa045e23b6e8edb0ae0d047a4a3612ef Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:55 +0200 Subject: [PATCH 286/529] ASoC: meson: add axg spdif output Add support for the spdif output serializer of the axg SoC family Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-spdifout.c | 456 +++++++++++++++++++++++++++++++++ 3 files changed, 465 insertions(+) create mode 100644 sound/soc/meson/axg-spdifout.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 3916060edcae..9408214b5854 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -19,4 +19,11 @@ config SND_MESON_AXG_TODDR Select Y or M to add support for the frontend capture interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_SPDIFOUT + tristate "Amlogic AXG SPDIF Output Support" + imply SND_SOC_SPDIF + help + Select Y or M to add support for SPDIF output serializer embedded + in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 12edf2db2a24..d51ae045ef08 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -3,7 +3,9 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c new file mode 100644 index 000000000000..9dea528053ad --- /dev/null +++ b/sound/soc/meson/axg-spdifout.c @@ -0,0 +1,456 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include + +/* + * NOTE: + * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite + * of what the documentation says. Manual control on V, U and C bits is + * applied when the related sel bits are cleared + */ + +#define SPDIFOUT_STAT 0x00 +#define SPDIFOUT_GAIN0 0x04 +#define SPDIFOUT_GAIN1 0x08 +#define SPDIFOUT_CTRL0 0x0c +#define SPDIFOUT_CTRL0_EN BIT(31) +#define SPDIFOUT_CTRL0_RST_OUT BIT(29) +#define SPDIFOUT_CTRL0_RST_IN BIT(28) +#define SPDIFOUT_CTRL0_USEL BIT(26) +#define SPDIFOUT_CTRL0_USET BIT(25) +#define SPDIFOUT_CTRL0_CHSTS_SEL BIT(24) +#define SPDIFOUT_CTRL0_DATA_SEL BIT(20) +#define SPDIFOUT_CTRL0_MSB_FIRST BIT(19) +#define SPDIFOUT_CTRL0_VSEL BIT(18) +#define SPDIFOUT_CTRL0_VSET BIT(17) +#define SPDIFOUT_CTRL0_MASK_MASK GENMASK(11, 4) +#define SPDIFOUT_CTRL0_MASK(x) ((x) << 4) +#define SPDIFOUT_CTRL1 0x10 +#define SPDIFOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define SPDIFOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define SPDIFOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define SPDIFOUT_CTRL1_TYPE(x) ((x) << 4) +#define SPDIFOUT_PREAMB 0x14 +#define SPDIFOUT_SWAP 0x18 +#define SPDIFOUT_CHSTS0 0x1c +#define SPDIFOUT_CHSTS1 0x20 +#define SPDIFOUT_CHSTS2 0x24 +#define SPDIFOUT_CHSTS3 0x28 +#define SPDIFOUT_CHSTS4 0x2c +#define SPDIFOUT_CHSTS5 0x30 +#define SPDIFOUT_CHSTS6 0x34 +#define SPDIFOUT_CHSTS7 0x38 +#define SPDIFOUT_CHSTS8 0x3c +#define SPDIFOUT_CHSTS9 0x40 +#define SPDIFOUT_CHSTSA 0x44 +#define SPDIFOUT_CHSTSB 0x48 +#define SPDIFOUT_MUTE_VAL 0x4c + +struct axg_spdifout { + struct regmap *map; + struct clk *mclk; + struct clk *pclk; +}; + +static void axg_spdifout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN, + 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT); + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_IN, SPDIFOUT_CTRL0_RST_IN); + + /* Enable spdifout */ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, + SPDIFOUT_CTRL0_EN); +} + +static void axg_spdifout_disable(struct regmap *map) +{ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0); +} + +static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_spdifout_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_spdifout_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + /* Use spdif valid bit to perform digital mute */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET, + mute ? SPDIFOUT_CTRL0_VSET : 0); + + return 0; +} + +static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + /* Set the samples spdifout will pull from the FIFO */ + switch (params_channels(params)) { + case 1: + val = SPDIFOUT_CTRL0_MASK(0x1); + break; + case 2: + val = SPDIFOUT_CTRL0_MASK(0x3); + break; + default: + dev_err(dai->dev, "too many channels for spdif dai: %u\n", + params_channels(params)); + return -EINVAL; + } + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MASK_MASK, val); + + /* FIFO data are arranged in chunks of 64bits */ + switch (params_physical_width(params)) { + case 8: + /* 8 samples of 8 bits */ + val = SPDIFOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(4); + break; + default: + dev_err(dai->dev, "Unsupported physical width: %u\n", + params_physical_width(params)); + return -EINVAL; + } + + /* Position of the MSB in FIFO samples */ + val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL1, + SPDIFOUT_CTRL1_MSB_POS_MASK | + SPDIFOUT_CTRL1_TYPE_MASK, val); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + return 0; +} + +static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int offset; + int ret; + u8 cs[4]; + u32 val; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24; + + /* Setup channel status A bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS0, val); + + /* Clear channel status A bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + /* Setup channel status B bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS6, val); + + /* Clear channel status B bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + return 0; +} + +static int axg_spdifout_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + int ret; + + /* 2 * 32bits per subframe * 2 channels = 128 */ + ret = clk_set_rate(priv->mclk, rate * 128); + if (ret) { + dev_err(dai->dev, "failed to set spdif clock\n"); + return ret; + } + + ret = axg_spdifout_sample_fmt(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup sample format\n"); + return ret; + } + + ret = axg_spdifout_set_chsts(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup channel status words\n"); + return ret; + } + + return 0; +} + +static int axg_spdifout_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + /* Clock the spdif output block */ + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "failed to enable pclk\n"); + return ret; + } + + /* Make sure the block is initially stopped */ + axg_spdifout_disable(priv->map); + + /* Insert data from bit 27 lsb first */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + /* Manual control of V, C and U, U = 0 */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL | + SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET, + 0); + + /* Static SWAP configuration ATM */ + regmap_write(priv->map, SPDIFOUT_SWAP, 0x10); + + return 0; +} + +static void axg_spdifout_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->pclk); +} + +static const struct snd_soc_dai_ops axg_spdifout_ops = { + .trigger = axg_spdifout_trigger, + .digital_mute = axg_spdifout_digital_mute, + .hw_params = axg_spdifout_hw_params, + .startup = axg_spdifout_startup, + .shutdown = axg_spdifout_shutdown, +}; + +static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = { + { + .name = "SPDIF Output", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &axg_spdifout_ops, + }, +}; + +static const char * const spdifout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24, + spdifout_sel_texts); + +static const struct snd_kcontrol_new axg_spdifout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum); + +static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux), +}; + +static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "Playback", NULL, "SRC SEL" }, +}; + +static const struct snd_kcontrol_new axg_spdifout_controls[] = { + SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1), + SOC_SINGLE("Playback Gain Enable Switch", + SPDIFOUT_CTRL1, 26, 1, 0), + SOC_SINGLE("Playback Channels Mix Switch", + SPDIFOUT_CTRL0, 23, 1, 0), +}; + +static int axg_spdifout_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_spdifout *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(priv->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(priv->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_spdifout_component_drv = { + .controls = axg_spdifout_controls, + .num_controls = ARRAY_SIZE(axg_spdifout_controls), + .dapm_widgets = axg_spdifout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_spdifout_dapm_widgets), + .dapm_routes = axg_spdifout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes), + .set_bias_level = axg_spdifout_set_bias_level, +}; + +static const struct regmap_config axg_spdifout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = SPDIFOUT_MUTE_VAL, +}; + +static const struct of_device_id axg_spdifout_of_match[] = { + { .compatible = "amlogic,axg-spdifout", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_spdifout_of_match); + +static int axg_spdifout_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_spdifout *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv, + axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv)); +} + +static struct platform_driver axg_spdifout_pdrv = { + .probe = axg_spdifout_probe, + .driver = { + .name = "axg-spdifout", + .of_match_table = axg_spdifout_of_match, + }, +}; +module_platform_driver(axg_spdifout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From 7713a70034f2cb54168d134ac523fdfcdda92a13 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:56 +0200 Subject: [PATCH 287/529] ASoC: meson: add axg tdm formatters DT bindings documentation Add the DT bindings documentation for axg's TDM formatters: TDMIN and TDMOUT. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../sound/amlogic,axg-tdm-formatters.txt | 28 +++++++++++++++++++ 1 file changed, 28 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt new file mode 100644 index 000000000000..1c1b7490554e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -0,0 +1,28 @@ +* Amlogic Audio TDM formatters + +Required properties: +- compatible: 'amlogic,axg-tdmin' or + 'amlogic,axg-tdmout' +- reg: physical base address of the controller and length of memory + mapped region. +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "sclk" : bit clock. + * "sclk_sel" : bit clock input multiplexer. + * "lrclk" : sample clock + * "lrclk_sel": sample clock input multiplexer + +Example of TDMOUT_A on the A113 SoC: + +tdmout_a: audio-controller@500 { + compatible = "amlogic,axg-tdmout"; + reg = <0x0 0x500 0x0 0x40>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; +}; From 58cabe8715f20b7fb33431bb1f2c5bd7a438b11b Mon Sep 17 00:00:00 2001 From: Adam Goode Date: Wed, 18 Jul 2018 16:41:05 -0400 Subject: [PATCH 288/529] ALSA: usb-audio: Allow changing from a bad sample rate If the audio device is externally clocked and set to a rate that does not match the external clock, the clock will never be valid and we cannot set the rate successfully. To fix this, allow a rate change even if the clock is initially invalid, and validate again after the rate is changed. This fixes problems with MOTU UltraLite AVB hardware over USB. Signed-off-by: Adam Goode Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index c79749613fa6..db5e39d67a90 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -513,14 +513,28 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bool writeable; u32 bmControls; + /* First, try to find a valid clock. This may trigger + * automatic clock selection if the current clock is not + * valid. + */ clock = snd_usb_clock_find_source(chip, fmt->protocol, fmt->clock, true); - if (clock < 0) - return clock; + if (clock < 0) { + /* We did not find a valid clock, but that might be + * because the current sample rate does not match an + * external clock source. Try again without validation + * and we will do another validation after setting the + * rate. + */ + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, false); + if (clock < 0) + return clock; + } prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) - return 0; + goto validation; if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc; @@ -577,6 +591,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, snd_usb_set_interface_quirk(dev); } +validation: + /* validate clock after rate change */ + if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + return -ENXIO; return 0; } From 868e49a4a00afaca07d2c450a02e49581eaece6c Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Thu, 19 Jul 2018 11:50:37 +0100 Subject: [PATCH 289/529] ASoC: wm_adsp: Ensure DSP boot work complete before preloader_put return All controls derived from the loaded firmware should be created prior to returning from the preloader's put function, such that they are immediately available to user-space. Signed-off-by: Stuart Henderson Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b7b914963c62..4e7f8525449e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2672,6 +2672,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); + flush_work(&dsp->boot_work); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); From 517ee74e1b3124b696f293aa4e220418f8125b4c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Jul 2018 11:50:35 +0100 Subject: [PATCH 290/529] ASoC: wm_adsp: Correct algorithm list allocation size Commit 6396bb221514 ("treewide: kzalloc() -> kcalloc()") was overlooked when doing some refactoring to the algorithm list handling, which lead to twice as much buffer being allocated as required for reading the algorithm list. A kcalloc is no longer appropriate since the allocation size is now in bytes not registers, as such change back to kzalloc. Fixes: 7f7cca08abf4 ("ASoC: wm_adsp: Simplify handling of alg offset and length") Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 07c17acc8a4f..99108d18de8d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,7 +1906,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, /* Convert length from DSP words to bytes */ len *= sizeof(u32); - alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); From b7ede5af62ab6bfad0980fad58e82d3fb56866df Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Jul 2018 11:50:36 +0100 Subject: [PATCH 291/529] ASoC: wm_adsp: Take prefix into account in control name length Currently when creating ALSA control names for the DSP the length of any prefix applied to the CODEC is not taken into account. Whilst this is mostly harmless it does result in ALSA doing the truncation of the control names and printing a warning. It is better to have the driver do the truncation so it can truncate from the start of parameter name itself to give a greater chance of the result maintain a unique name. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 99108d18de8d..eec73c98a141 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1343,6 +1343,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; int skip = 0; + if (dsp->component->name_prefix) + avail -= strlen(dsp->component->name_prefix) + 1; + if (subname_len > avail) skip = subname_len - avail; From 3bbc2705a3d132b9a86a0e4083f82a2b3c9bfdfd Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 19 Jul 2018 11:50:38 +0100 Subject: [PATCH 292/529] ASoC: wm_adsp: Allow up to 8 channels for voice control Newer voice control firmwares can capture multiple audio channels. Allow up to 8 channels for future-proofing. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index eec73c98a141..aeb1b8c27670 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -418,7 +418,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = { { .id = SND_AUDIOCODEC_BESPOKE, .desc = { - .max_ch = 1, + .max_ch = 8, .sample_rates = { 16000 }, .num_sample_rates = 1, .formats = SNDRV_PCM_FMTBIT_S16_LE, From d52ed4b0bc73c1c7816f5b7a36229a95acfc76c8 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 19 Jul 2018 11:50:39 +0100 Subject: [PATCH 293/529] ASoC: wm_adsp: Parse HOST_BUFFER controls Currently the compressed streams in DSP firmwares are identified essentially by looking at a fixed location inside the firmware. This is fragile and also limits things to a single compressed stream. Here a new form of firmware parameter is added, the HOST_BUFFER which identifies a compressed stream from meta-data in the firmware file. This is more robust and allows for the possiblity of using multiple streams per core in the future. Currently the implementation is still limited to a single stream and will use the first HOST_BUFFER parameter encountered. If there aren't any HOST_BUFFER parameters it will fall back to the legacy way of finding the host buffer. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 66 +++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wmfw.h | 1 + 2 files changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index aeb1b8c27670..e39b0e0b04df 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1607,6 +1607,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp, if (ret) return -EINVAL; break; + case WMFW_CTL_TYPE_HOST_BUFFER: + ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk, + WMFW_CTL_FLAG_SYS | + WMFW_CTL_FLAG_VOLATILE | + WMFW_CTL_FLAG_READABLE, + 0); + if (ret) + return -EINVAL; + break; default: adsp_err(dsp, "Unknown control type: %d\n", coeff_blk.ctl_type); @@ -3200,7 +3209,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; struct wm_adsp *dsp = buf->dsp; @@ -3239,6 +3248,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) return 0; } +static struct wm_coeff_ctl * +wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + + list_for_each_entry(ctl, &dsp->ctl_list, list) { + if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER) + continue; + + if (!ctl->enabled) + continue; + + return ctl; + } + + return NULL; +} + +static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + unsigned int reg; + u32 val; + int i, ret; + + ctl = wm_adsp_find_host_buffer_ctrl(buf); + if (!ctl) + return wm_adsp_legacy_host_buf_addr(buf); + + ret = wm_coeff_base_reg(ctl, ®); + if (ret) + return ret; + + for (i = 0; i < 5; ++i) { + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); + if (ret < 0) + return ret; + + if (val) + break; + + usleep_range(1000, 2000); + } + + if (!val) + return -EIO; + + buf->host_buf_ptr = be32_to_cpu(val); + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + + return 0; +} + static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) { const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index ec78b9da020f..0c3f50acb8b1 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -29,6 +29,7 @@ /* Non-ALSA coefficient types start at 0x1000 */ #define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */ #define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */ +#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */ struct wmfw_header { char magic[4]; From eea1662525bd4a158a67ac836b2a1fd9cf77cc81 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:37 +0200 Subject: [PATCH 294/529] ASoC: rt5651: Add IN3 Boost volume control Add a mixer control for the IN3 Boost volume, IN3 is used for the headset mic on most devices, so this is necessary to control the headset mic volume. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 0462049e739c..985852fd9723 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL, RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT, From 0a3badd141f78535315cca9ff5062a7ebf414281 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:38 +0200 Subject: [PATCH 295/529] ASoC: Intel: bytcr_rt5651: Fix using the wrong GPIO for the ext-amp on some boards Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the GpioIo one for the ext-amp-enable-gpio. So far we've been assuming that the GpioIo one always comes first, this commit adds code to detect which one comes first and to add the right gpio-mapping. This fixes sound not working on the Vios LTH17 laptop. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 69 +++++++++++++++++++++++++-- 1 file changed, 65 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b687043c8425..601e47c33ba8 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -747,13 +747,74 @@ static const struct x86_cpu_id cherrytrail_cpu_ids[] = { {} }; -static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; +static const struct acpi_gpio_params first_gpio = { 0, 0, false }; +static const struct acpi_gpio_params second_gpio = { 1, 0, false }; -static const struct acpi_gpio_mapping byt_rt5651_gpios[] = { - { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1 }, +static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { + { "ext-amp-enable-gpios", &first_gpio, 1 }, { }, }; +static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { + { "ext-amp-enable-gpios", &second_gpio, 1 }, + { }, +}; + +/* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other + * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the + * GpioIo one for the ext-amp-enable-gpio and both count for the index in + * acpi_gpio_params index. So we have 2 different mappings and the code + * below figures out which one to use. + */ +struct byt_rt5651_acpi_resource_data { + int gpio_count; + int gpio_int_idx; +}; + +static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) +{ + struct byt_rt5651_acpi_resource_data *data = arg; + + if (ares->type != ACPI_RESOURCE_TYPE_GPIO) + return 0; + + if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) + data->gpio_int_idx = data->gpio_count; + + data->gpio_count++; + return 0; +} + +static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) +{ + struct byt_rt5651_acpi_resource_data data = { 0, -1 }; + LIST_HEAD(resources); + int ret; + + ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, + snd_byt_rt5651_acpi_resource, &data); + if (ret < 0) { + dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); + return; + } + + /* All info we need is gathered during the walk */ + acpi_dev_free_resource_list(&resources); + + switch (data.gpio_int_idx) { + case 0: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second); + break; + case 1: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first); + break; + default: + dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", + data.gpio_int_idx); + } +} + struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -876,7 +937,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Cherry Trail devices use an external amplifier enable gpio */ if (x86_match_cpu(cherrytrail_cpu_ids)) { - devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); + snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev); priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( &pdev->dev, "ext-amp-enable", 0, codec_dev->fwnode, From 8627fb257e1673d2c2277494545642921097da86 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:39 +0200 Subject: [PATCH 296/529] ASoC: Intel: bytcr_rt5651: Set OVCD limit for VIOS LTH17 to 2000uA With the default over current detect limit of 1500uA headsets on often get detected as headphones on the VIOS LTH17 and even when detected as headset the OVCD current triggers often while plugged in, resulting in false-positive button press detection. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 601e47c33ba8..53ac97c15fc6 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -414,8 +414,11 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_IN2_MAP), + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | + BYT_RT5651_JD1_1 | + BYT_RT5651_OVCD_TH_2000UA | + BYT_RT5651_OVCD_SF_1P0 | + BYT_RT5651_MCLK_EN), }, {} }; From ac275ee5aa67abe9b65d66071ee333c6b0905b93 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:40 +0200 Subject: [PATCH 297/529] ASoC: Intel: bytcr_rt5651: Add IN2 input mapping During the recent cleanup series 3 of the 6 input mappings where removed from the bytcr_rt5651 machine driver because testing showed that none of them were used. However some devices do actually have their internal mic on IN2 (and only IN2, not IN1 and IN2), this did not show during previous tests due to a bug in the userspace UCM input device switching code. This commit re-adds the IN2 mapping for devices with the internal mic. on IN2 and the headser mic on IN3 and enables this mapping on devices with their internal mic on IN2. This commit also changes the default internal mic input to IN2, because all my 7 test devices have their mic there. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 53ac97c15fc6..d85530b1cc8e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -44,6 +44,7 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, + BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, }; @@ -93,9 +94,9 @@ struct byt_rt5651_private { struct snd_soc_jack jack; }; -/* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ +/* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP; + BYT_RT5651_IN2_MAP; static void log_quirks(struct device *dev) { @@ -103,6 +104,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) + dev_info(dev, "quirk IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { @@ -270,6 +273,12 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"IN3P", NULL, "Headset Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN2P", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, +}; + static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, @@ -364,7 +373,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | + BYT_RT5651_IN2_MAP | BYT_RT5651_HP_LR_SWAPPED), }, { @@ -375,7 +384,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | + BYT_RT5651_IN2_MAP | BYT_RT5651_HP_LR_SWAPPED), }, { @@ -468,6 +477,10 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; + case BYT_RT5651_IN2_MAP: + custom_map = byt_rt5651_intmic_in2_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); + break; case BYT_RT5651_IN1_IN2_MAP: custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); @@ -825,7 +838,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "dmic", "in1", "in12" }; + const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; struct device *codec_dev; From a0d1d867c262f4ad5d8e4925e2212711ebdbf2b7 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:41 +0200 Subject: [PATCH 298/529] ASoC: Intel: bytcr_rt5651: Add mono speaker quirk During my initial round of bytcr_rt5651 long-name patches I did not include a difference for mono vs stereo speaker setups in the longname because it seems that all 5651 devices with only a single speaker do some mixing of left + right on the PCB. However further testing has shown that while this works great when only playing audio on the left or right channel, the output becomes garbled when using both channels at once. Something which does not happen when using the Stereo DAC MIXL / MIXR switches to mix the channels together inside the codec and then only outputting on a single channel. So we need to have separate UCM profiles and thus separate long-names for devices with a mono speaker vs stereo speakers. Just as we already have for the bytcr_rt5640 case. This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk" or "mono-spk" to the long-name based on this and enables this mapping on devices with a mono speaker. Changing the long-name like this is ok for now, since I'm still working on the UCM profiles, so they are not in upstream alsa-lib yet. This brings the long-name naming scheme fully in sync with the bytcr_rt5640 case, which is good from a consistency pov. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d85530b1cc8e..8374e633796d 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -79,6 +79,7 @@ enum { #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) #define BYT_RT5651_HP_LR_SWAPPED BIT(22) +#define BYT_RT5651_MONO_SPEAKER BIT(23) #define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ BYT_RT5651_JD1_1 | \ @@ -128,6 +129,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled\n"); if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); } #define BYT_CODEC_DAI1 "rt5651-aif1" @@ -374,7 +377,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP | - BYT_RT5651_HP_LR_SWAPPED), + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { /* Chuwi Vi8 Plus (CWI519) */ @@ -385,7 +389,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP | - BYT_RT5651_HP_LR_SWAPPED), + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { /* KIANO SlimNote 14.2 */ @@ -700,7 +705,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ +static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ static int byt_rt5651_suspend(struct snd_soc_card *card) { @@ -1025,7 +1030,9 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) hp_swapped = ""; snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-mic%s", + "bytcr-rt5651-%s-spk-%s-mic%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? + "mono" : "stereo", mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; From 06aa6e51273c9ac458af0bb9be95603cfbad14ec Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:42 +0200 Subject: [PATCH 299/529] ASoC: Intel: bytcr_rt5651: Add quirk table entries for various devices Add quirk table entries for the following tablets: ITWorks TW701 Ployer Momo7w Trekstor win7 Yours 8" These all use the default settings, except that they only have a single speaker and thus need the mono-speaker quirk. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 28 +++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 8374e633796d..f8a68bdb3885 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -392,6 +392,21 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_HP_LR_SWAPPED | BYT_RT5651_MONO_SPEAKER), }, + { + /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6 + * (these all use the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all of itWORKS.G.WI71C.JGBMRBA, + * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */ + DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), + }, { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, @@ -434,6 +449,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_OVCD_SF_1P0 | BYT_RT5651_MCLK_EN), }, + { + /* Yours Y8W81 (and others using the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all devs with a W86C mainboard */ + DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), + }, {} }; From 9ee6f8a8cbbd203fb0a844f937007a9525422697 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 11:30:23 +0200 Subject: [PATCH 300/529] ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet Add a quirk for the "Connect Tablet 9" tablet, this tablet has a mono-speaker. Otherwise it works fine with the defaults. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 657910a08261..d32844f94d74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -486,6 +486,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), From 486c16f2b5d1eb12339fb7a020961ea1c40e619f Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler Date: Fri, 20 Jul 2018 09:53:02 +0200 Subject: [PATCH 301/529] ASoC: sgtl5000: fix spelling in devicetree binding document This fixes a spelling mistake. Signed-off-by: Marcel Ziswiler Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0f214457476f..9c58f724396a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -17,7 +17,7 @@ Optional properties: - VDDD-supply : the regulator provider of VDDD -- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then From e1548b1ba1643d4b5015f7168c64f240ebe114c8 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Fri, 20 Jul 2018 10:02:50 +0200 Subject: [PATCH 302/529] MAINTAINERS: add entry for STI audio drivers Add sound/soc/sti drivers entry for STI audio drivers from ST Microelectronics. Signed-off-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 9d5eeff51b5f..b44d7167a9e9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13557,6 +13557,13 @@ L: linux-block@vger.kernel.org S: Maintained F: drivers/block/skd*[ch] +STI AUDIO (ASoC) DRIVERS +M: Arnaud Pouliquen +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +F: sound/soc/sti/ + STI CEC DRIVER M: Benjamin Gaignard S: Maintained From faa80b66c767ef4995e33835e9844a5df142c6cc Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Fri, 20 Jul 2018 10:02:51 +0200 Subject: [PATCH 303/529] MAINTAINERS: add entry for STM32 audio drivers Add sound/soc/stm drivers entry for STM32 audio drivers from ST Microelectronics. Signed-off-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- MAINTAINERS | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index b44d7167a9e9..fc711d23dc83 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13577,6 +13577,14 @@ T: git git://linuxtv.org/media_tree.git S: Maintained F: drivers/media/usb/stk1160/ +STM32 AUDIO (ASoC) DRIVERS +M: Olivier Moysan +M: Arnaud Pouliquen +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,stm32-*.txt +F: sound/soc/stm/ + STM32 TIMER/LPTIMER DRIVERS M: Fabrice Gasnier S: Maintained From 2ec42486358f63c4a426514c395d13f4b9de5da8 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler Date: Fri, 20 Jul 2018 10:04:23 +0200 Subject: [PATCH 304/529] ASoC: tegra: improve goto error label While the two error labels "err" and "err_clk_put" goto the same place it is rather confusing that the earlier one is certainly used later again. Signed-off-by: Marcel Ziswiler Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); From 9e960c0298b5811e5a2c1ebceea6aa3b7bbc61c6 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:57 +0200 Subject: [PATCH 305/529] ASoC: meson: add axg tdm interface DT bindings documentation Add the DT bindings documentation for axg's TDM interfaces Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 +++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt new file mode 100644 index 000000000000..cabfb26a5f22 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt @@ -0,0 +1,22 @@ +* Amlogic Audio TDM Interfaces + +Required properties: +- compatible: 'amlogic,axg-tdm-iface' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "sclk" : bit clock. + * "lrclk": sample clock + * "mclk" : master clock + -> optional if the interface is in clock slave mode. +- #sound-dai-cells: must be 0. + +Example of TDM_A on the A113 SoC: + +tdmif_a: audio-controller@0 { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>, + <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>; + clock-names = "mclk", "sclk", "lrclk"; +}; From 1a11d88f499ceb69e9b4098ddc36866820335a54 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:58 +0200 Subject: [PATCH 306/529] ASoC: meson: add tdm formatter base driver Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more complex than usual, mainly because the different TDM input decoders can be attached to any of TDM pad interface, including the output pads. For the this, TDM on this SoC is modeled like this: - TDM interface provides the DAIs the codecs will be attached to. The main responsibility of this driver is to manage the pad format and the TDM clock rates. - TDM Formatters: These are the entities which are actually dealing with the TDM signal. TDMOUT produce a TDM signal from the audio sample provided by FRDDR using the clocks provided the TDM interface. TDMIN feeds TODDR with audio sample using the clocks and TDM signal provided by the TDM Interface. - TDM Streams: This provides the link between 1 DAI stream of the TDM interface and one (or more) TDM formatters. This driver provides the TDM formatter and TDM stream operations. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdm-formatter.c | 381 ++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-formatter.h | 39 +++ sound/soc/meson/axg-tdm.h | 74 ++++++ 5 files changed, 500 insertions(+) create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm.h diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 9408214b5854..80a88689491f 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -19,6 +19,10 @@ config SND_MESON_AXG_TODDR Select Y or M to add support for the frontend capture interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_TDM_FORMATTER + tristate + select REGMAP_MMIO + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index d51ae045ef08..a06b56a1c995 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -3,9 +3,11 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c new file mode 100644 index 000000000000..43e390f9358a --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +struct axg_tdm_formatter { + struct list_head list; + struct axg_tdm_stream *stream; + const struct axg_tdm_formatter_driver *drv; + struct clk *pclk; + struct clk *sclk; + struct clk *lrclk; + struct clk *sclk_sel; + struct clk *lrclk_sel; + bool enabled; + struct regmap *map; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset) +{ + unsigned int val, ch = ts->channels; + unsigned long mask; + int i, j; + + /* + * Distribute the channels of the stream over the available slots + * of each TDM lane + */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + val = 0; + mask = ts->mask[i]; + + for (j = find_first_bit(&mask, 32); + (j < 32) && ch; + j = find_next_bit(&mask, 32, j + 1)) { + val |= 1 << j; + ch -= 1; + } + + regmap_write(map, offset, val); + offset += regmap_get_reg_stride(map); + } + + /* + * If we still have channel left at the end of the process, it means + * the stream has more channels than we can accommodate and we should + * have caught this earlier. + */ + if (WARN_ON(ch != 0)) { + pr_err("channel mask error\n"); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); + +static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + bool invert = formatter->drv->invert_sclk; + int ret; + + /* Do nothing if the formatter is already enabled */ + if (formatter->enabled) + return 0; + + /* + * If sclk is inverted, invert it back and provide the inversion + * required by the formatter + */ + invert ^= axg_tdm_sclk_invert(ts->iface->fmt); + ret = clk_set_phase(formatter->sclk, invert ? 180 : 0); + if (ret) + return ret; + + /* Setup the stream parameter in the formatter */ + ret = formatter->drv->ops->prepare(formatter->map, formatter->stream); + if (ret) + return ret; + + /* Enable the signal clocks feeding the formatter */ + ret = clk_prepare_enable(formatter->sclk); + if (ret) + return ret; + + ret = clk_prepare_enable(formatter->lrclk); + if (ret) { + clk_disable_unprepare(formatter->sclk); + return ret; + } + + /* Finally, actually enable the formatter */ + formatter->drv->ops->enable(formatter->map); + formatter->enabled = true; + + return 0; +} + +static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter) +{ + /* Do nothing if the formatter is already disabled */ + if (!formatter->enabled) + return; + + formatter->drv->ops->disable(formatter->map); + clk_disable_unprepare(formatter->lrclk); + clk_disable_unprepare(formatter->sclk); + formatter->enabled = false; +} + +static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + int ret = 0; + + mutex_lock(&ts->lock); + + /* Catch up if the stream is already running when we attach */ + if (ts->ready) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to enable formatter\n"); + goto out; + } + } + + list_add_tail(&formatter->list, &ts->formatter_list); +out: + mutex_unlock(&ts->lock); + return ret; +} + +static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + + mutex_lock(&ts->lock); + list_del(&formatter->list); + mutex_unlock(&ts->lock); + + axg_tdm_formatter_disable(formatter); +} + +static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter, + struct snd_soc_dapm_widget *w) +{ + struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w); + int ret; + + /* + * If we don't get a stream at this stage, it would mean that the + * widget is powering up but is not attached to any backend DAI. + * It should not happen, ever ! + */ + if (WARN_ON(!ts)) + return -ENODEV; + + /* Clock our device */ + ret = clk_prepare_enable(formatter->pclk); + if (ret) + return ret; + + /* Reparent the bit clock to the TDM interface */ + ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk); + if (ret) + goto disable_pclk; + + /* Reparent the sample clock to the TDM interface */ + ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk); + if (ret) + goto disable_pclk; + + formatter->stream = ts; + ret = axg_tdm_formatter_attach(formatter); + if (ret) + goto disable_pclk; + + return 0; + +disable_pclk: + clk_disable_unprepare(formatter->pclk); + return ret; +} + +static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter) +{ + axg_tdm_formatter_dettach(formatter); + clk_disable_unprepare(formatter->pclk); + formatter->stream = NULL; +} + +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = axg_tdm_formatter_power_up(formatter, w); + break; + + case SND_SOC_DAPM_PRE_PMD: + axg_tdm_formatter_power_down(formatter); + break; + + default: + dev_err(c->dev, "Unexpected event %d\n", event); + return -EINVAL; + } + + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_event); + +int axg_tdm_formatter_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_tdm_formatter_driver *drv; + struct axg_tdm_formatter *formatter; + struct resource *res; + void __iomem *regs; + int ret; + + drv = of_device_get_match_data(dev); + if (!drv) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL); + if (!formatter) + return -ENOMEM; + platform_set_drvdata(pdev, formatter); + formatter->drv = drv; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg); + if (IS_ERR(formatter->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(formatter->map)); + return PTR_ERR(formatter->map); + } + + /* Peripharal clock */ + formatter->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(formatter->pclk)) { + ret = PTR_ERR(formatter->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock */ + formatter->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(formatter->sclk)) { + ret = PTR_ERR(formatter->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Formatter sample clock */ + formatter->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(formatter->lrclk)) { + ret = PTR_ERR(formatter->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock input multiplexer */ + formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); + if (IS_ERR(formatter->sclk_sel)) { + ret = PTR_ERR(formatter->sclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk_sel: %d\n", ret); + return ret; + } + + /* Formatter sample clock input multiplexer */ + formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); + if (IS_ERR(formatter->lrclk_sel)) { + ret = PTR_ERR(formatter->lrclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, drv->component_drv, + NULL, 0); +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe); + +int axg_tdm_stream_start(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + int ret = 0; + + mutex_lock(&ts->lock); + ts->ready = true; + + /* Start all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to start tdm stream\n"); + goto out; + } + } + +out: + mutex_unlock(&ts->lock); + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_start); + +void axg_tdm_stream_stop(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + + mutex_lock(&ts->lock); + ts->ready = false; + + /* Stop all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + axg_tdm_formatter_disable(formatter); + } + + mutex_unlock(&ts->lock); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_stop); + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface) +{ + struct axg_tdm_stream *ts; + + ts = kzalloc(sizeof(*ts), GFP_KERNEL); + if (ts) { + INIT_LIST_HEAD(&ts->formatter_list); + mutex_init(&ts->lock); + ts->iface = iface; + } + + return ts; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc); + +void axg_tdm_stream_free(struct axg_tdm_stream *ts) +{ + /* + * If the list is not empty, it would mean that one of the formatter + * widget is still powered and attached to the interface while we + * we are removing the TDM DAI. It should not be possible + */ + WARN_ON(!list_empty(&ts->formatter_list)); + mutex_destroy(&ts->lock); + kfree(ts); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_free); + +MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h new file mode 100644 index 000000000000..cf947caf3cb1 --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -0,0 +1,39 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_TDM_FORMATTER_H +#define _MESON_AXG_TDM_FORMATTER_H + +#include "axg-tdm.h" + +struct platform_device; +struct regmap; +struct snd_soc_dapm_widget; +struct snd_kcontrol; + +struct axg_tdm_formatter_ops { + struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w); + void (*enable)(struct regmap *map); + void (*disable)(struct regmap *map); + int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts); +}; + +struct axg_tdm_formatter_driver { + const struct snd_soc_component_driver *component_drv; + const struct regmap_config *regmap_cfg; + const struct axg_tdm_formatter_ops *ops; + bool invert_sclk; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset); +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event); +int axg_tdm_formatter_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_TDM_FORMATTER_H */ diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h new file mode 100644 index 000000000000..435d95b86457 --- /dev/null +++ b/sound/soc/meson/axg-tdm.h @@ -0,0 +1,74 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_TDM_H +#define _MESON_AXG_TDM_H + +#include +#include +#include +#include +#include + +#define AXG_TDM_NUM_LANES 4 +#define AXG_TDM_CHANNEL_MAX 128 +#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct axg_tdm_iface { + struct clk *sclk; + struct clk *lrclk; + struct clk *mclk; + unsigned long mclk_rate; + + /* format is common to all the DAIs of the iface */ + unsigned int fmt; + unsigned int slots; + unsigned int slot_width; + + /* For component wide symmetry */ + int rate; +}; + +static inline bool axg_tdm_lrclk_invert(unsigned int fmt) +{ + return (fmt & SND_SOC_DAIFMT_I2S) ^ + !!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF)); +} + +static inline bool axg_tdm_sclk_invert(unsigned int fmt) +{ + return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF); +} + +struct axg_tdm_stream { + struct axg_tdm_iface *iface; + struct list_head formatter_list; + struct mutex lock; + unsigned int channels; + unsigned int width; + unsigned int physical_width; + u32 *mask; + bool ready; +}; + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface); +void axg_tdm_stream_free(struct axg_tdm_stream *ts); +int axg_tdm_stream_start(struct axg_tdm_stream *ts); +void axg_tdm_stream_stop(struct axg_tdm_stream *ts); + +static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) +{ + axg_tdm_stream_stop(ts); + return axg_tdm_stream_start(ts); +} + +#endif /* _MESON_AXG_TDM_H */ From d60e4f1e4be5e2dfb55fb084b119aed094227a35 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:59 +0200 Subject: [PATCH 307/529] ASoC: meson: add tdm interface driver Add Amlogic's axg TDM interface driver. This driver manages the format and clocks provided on the pads. On this SoC, each stream direction provides 4 serial lanes. This makes a maximum of 8 channels in i2s modes and 128 channels in DSP modes. While each lanes operate on the same slot number (same bit clock), they may have different TDM masks. This requires to provide a function to let the card set the 4 masks, in lieu of the usual set_tdm_slots() callback of the dai driver. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdm-interface.c | 542 ++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm.h | 4 + 4 files changed, 552 insertions(+) create mode 100644 sound/soc/meson/axg-tdm-interface.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 80a88689491f..869b359c2ce7 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -23,6 +23,10 @@ config SND_MESON_AXG_TDM_FORMATTER tristate select REGMAP_MMIO +config SND_MESON_AXG_TDM_INTERFACE + tristate + select SND_MESON_AXG_TDM_FORMATTER + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index a06b56a1c995..1a8eb77402e3 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -4,10 +4,12 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o +snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o +obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c new file mode 100644 index 000000000000..7b8baf46d968 --- /dev/null +++ b/sound/soc/meson/axg-tdm-interface.c @@ -0,0 +1,542 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include + +#include "axg-tdm.h" + +enum { + TDM_IFACE_PAD, + TDM_IFACE_LOOPBACK, +}; + +static unsigned int axg_tdm_slots_total(u32 *mask) +{ + unsigned int slots = 0; + int i; + + if (!mask) + return 0; + + /* Count the total number of slots provided by all 4 lanes */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) + slots += hweight32(mask[i]); + + return slots; +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *tx = (struct axg_tdm_stream *) + dai->playback_dma_data; + struct axg_tdm_stream *rx = (struct axg_tdm_stream *) + dai->capture_dma_data; + unsigned int tx_slots, rx_slots; + + tx_slots = axg_tdm_slots_total(tx_mask); + rx_slots = axg_tdm_slots_total(rx_mask); + + /* We should at least have a slot for a valid interface */ + if (!tx_slots && !rx_slots) { + dev_err(dai->dev, "interface has no slot\n"); + return -EINVAL; + } + + /* + * Amend the dai driver channel number and let dpcm channel merge do + * its job + */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + } + + iface->slots = slots; + + switch (slot_width) { + case 0: + /* defaults width to 32 if not provided */ + iface->slot_width = 32; + break; + case 8: + case 16: + case 24: + case 32: + iface->slot_width = slot_width; + break; + default: + dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); + +static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret = -ENOTSUPP; + + if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) { + if (!iface->mclk) { + dev_warn(dai->dev, "master clock not provided\n"); + } else { + ret = clk_set_rate(iface->mclk, freq); + if (!ret) + iface->mclk_rate = freq; + } + } + + return ret; +} + +static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + /* These modes are not supported */ + if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) { + dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + return -EINVAL; + } + + /* If the TDM interface is the clock master, it requires mclk */ + if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(dai->dev, "cpu clock master: mclk missing\n"); + return -ENODEV; + } + + iface->fmt = fmt; + return 0; +} + +static int axg_tdm_iface_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = + snd_soc_dai_get_dma_data(dai, substream); + int ret; + + if (!axg_tdm_slots_total(ts->mask)) { + dev_err(dai->dev, "interface has not slots\n"); + return -EINVAL; + } + + /* Apply component wide rate symmetry */ + if (dai->component->active) { + ret = snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + iface->rate); + if (ret < 0) { + dev_err(dai->dev, + "can't set iface rate constraint\n"); + return ret; + } + } + + return 0; +} + +static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + unsigned int channels = params_channels(params); + unsigned int width = params_width(params); + + /* Save rate and sample_bits for component symmetry */ + iface->rate = params_rate(params); + + /* Make sure this interface can cope with the stream */ + if (axg_tdm_slots_total(ts->mask) < channels) { + dev_err(dai->dev, "not enough slots for channels\n"); + return -EINVAL; + } + + if (iface->slot_width < width) { + dev_err(dai->dev, "incompatible slots width for stream\n"); + return -EINVAL; + } + + /* Save the parameter for tdmout/tdmin widgets */ + ts->physical_width = params_physical_width(params); + ts->width = params_width(params); + ts->channels = params_channels(params); + + return 0; +} + +static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned int ratio_num; + int ret; + + ret = clk_set_rate(iface->lrclk, params_rate(params)); + if (ret) { + dev_err(dai->dev, "setting sample clock failed: %d\n", ret); + return ret; + } + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + /* 50% duty cycle ratio */ + ratio_num = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* + * A zero duty cycle ratio will result in setting the mininum + * ratio possible which, for this clock, is 1 cycle of the + * parent bclk clock high and the rest low, This is exactly + * what we want here. + */ + ratio_num = 0; + break; + + default: + return -EINVAL; + } + + ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); + if (ret) { + dev_err(dai->dev, + "setting sample clock duty cycle failed: %d\n", ret); + return ret; + } + + /* Set sample clock inversion */ + ret = clk_set_phase(iface->lrclk, + axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); + if (ret) { + dev_err(dai->dev, + "setting sample clock phase failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned long srate; + int ret; + + srate = iface->slots * iface->slot_width * params_rate(params); + + if (!iface->mclk_rate) { + /* If no specific mclk is requested, default to bit clock * 4 */ + clk_set_rate(iface->mclk, 4 * srate); + } else { + /* Check if we can actually get the bit clock from mclk */ + if (iface->mclk_rate % srate) { + dev_err(dai->dev, + "can't derive sclk %lu from mclk %lu\n", + srate, iface->mclk_rate); + return -EINVAL; + } + } + + ret = clk_set_rate(iface->sclk, srate); + if (ret) { + dev_err(dai->dev, "setting bit clock failed: %d\n", ret); + return ret; + } + + /* Set the bit clock inversion */ + ret = clk_set_phase(iface->sclk, + axg_tdm_sclk_invert(iface->fmt) ? 0 : 180); + if (ret) { + dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + if (iface->slots > 2) { + dev_err(dai->dev, "bad slot number for format: %d\n", + iface->slots); + return -EINVAL; + } + break; + + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + break; + + default: + dev_err(dai->dev, "unsupported dai format\n"); + return -EINVAL; + } + + ret = axg_tdm_iface_set_stream(substream, params, dai); + if (ret) + return ret; + + if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) { + ret = axg_tdm_iface_set_sclk(dai, params); + if (ret) + return ret; + + ret = axg_tdm_iface_set_lrclk(dai, params); + if (ret) + return ret; + } + + return 0; +} + +static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Stop all attached formatters */ + axg_tdm_stream_stop(ts); + + return 0; +} + +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); +} + +static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) +{ + if (dai->capture_dma_data) + axg_tdm_stream_free(dai->capture_dma_data); + + if (dai->playback_dma_data) + axg_tdm_stream_free(dai->playback_dma_data); + + return 0; +} + +static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_widget) { + dai->capture_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->capture_dma_data) + return -ENOMEM; + } + + if (dai->playback_widget) { + dai->playback_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->playback_dma_data) { + axg_tdm_iface_remove_dai(dai); + return -ENOMEM; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops axg_tdm_iface_ops = { + .set_sysclk = axg_tdm_iface_set_sysclk, + .set_fmt = axg_tdm_iface_set_fmt, + .startup = axg_tdm_iface_startup, + .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, + .hw_free = axg_tdm_iface_hw_free, +}; + +/* TDM Backend DAIs */ +static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { + [TDM_IFACE_PAD] = { + .name = "TDM Pad", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_PAD, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, + [TDM_IFACE_LOOPBACK] = { + .name = "TDM Loopback", + .capture = { + .stream_name = "Loopback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_LOOPBACK, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, +}; + +static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(iface->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(iface->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { + .set_bias_level = axg_tdm_iface_set_bias_level, +}; + +static const struct of_device_id axg_tdm_iface_of_match[] = { + { .compatible = "amlogic,axg-tdm-iface", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match); + +static int axg_tdm_iface_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_soc_dai_driver *dai_drv; + struct axg_tdm_iface *iface; + int ret, i; + + iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL); + if (!iface) + return -ENOMEM; + platform_set_drvdata(pdev, iface); + + /* + * Duplicate dai driver: depending on the slot masks configuration + * We'll change the number of channel provided by DAI stream, so dpcm + * channel merge can be done properly + */ + dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv), + sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++) + memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i], + sizeof(*dai_drv)); + + /* Bit clock provided on the pad */ + iface->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(iface->sclk)) { + ret = PTR_ERR(iface->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Sample clock provided on the pad */ + iface->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(iface->lrclk)) { + ret = PTR_ERR(iface->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* + * mclk maybe be missing when the cpu dai is in slave mode and + * the codec does not require it to provide a master clock. + * At this point, ignore the error if mclk is missing. We'll + * throw an error if the cpu dai is master and mclk is missing + */ + iface->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(iface->mclk)) { + ret = PTR_ERR(iface->mclk); + if (ret == -ENOENT) { + iface->mclk = NULL; + } else { + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + } + + return devm_snd_soc_register_component(dev, + &axg_tdm_iface_component_drv, dai_drv, + ARRAY_SIZE(axg_tdm_iface_dai_drv)); +} + +static struct platform_driver axg_tdm_iface_pdrv = { + .probe = axg_tdm_iface_probe, + .driver = { + .name = "axg-tdm-iface", + .of_match_table = axg_tdm_iface_of_match, + }, +}; +module_platform_driver(axg_tdm_iface_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM interface driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 435d95b86457..e578b6f40a07 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -71,4 +71,8 @@ static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) return axg_tdm_stream_start(ts); } +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width); + #endif /* _MESON_AXG_TDM_H */ From c41c2a355b86368608377eaf3df442ec0f342f1e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:00 +0200 Subject: [PATCH 308/529] ASoC: meson: add tdm output driver Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo and produce the TDM signals for 4 output lanes. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 8 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdmout.c | 259 +++++++++++++++++++++++++++++++++++ 3 files changed, 269 insertions(+) create mode 100644 sound/soc/meson/axg-tdmout.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 869b359c2ce7..08a522b77749 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -27,6 +27,14 @@ config SND_MESON_AXG_TDM_INTERFACE tristate select SND_MESON_AXG_TDM_FORMATTER +config SND_MESON_AXG_TDMOUT + tristate "Amlogic AXG TDM Output Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM output formatter embedded + in the Amlogic AXG SoC family + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 1a8eb77402e3..a665c6b76a95 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -5,6 +5,7 @@ snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -12,4 +13,5 @@ obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c new file mode 100644 index 000000000000..f73368ee1088 --- /dev/null +++ b/sound/soc/meson/axg-tdmout.c @@ -0,0 +1,259 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +#define TDMOUT_CTRL0 0x00 +#define TDMOUT_CTRL0_BITNUM_MASK GENMASK(4, 0) +#define TDMOUT_CTRL0_BITNUM(x) ((x) << 0) +#define TDMOUT_CTRL0_SLOTNUM_MASK GENMASK(9, 5) +#define TDMOUT_CTRL0_SLOTNUM(x) ((x) << 5) +#define TDMOUT_CTRL0_INIT_BITNUM_MASK GENMASK(19, 15) +#define TDMOUT_CTRL0_INIT_BITNUM(x) ((x) << 15) +#define TDMOUT_CTRL0_ENABLE BIT(31) +#define TDMOUT_CTRL0_RST_OUT BIT(29) +#define TDMOUT_CTRL0_RST_IN BIT(28) +#define TDMOUT_CTRL1 0x04 +#define TDMOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define TDMOUT_CTRL1_TYPE(x) ((x) << 4) +#define TDMOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define TDMOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define TDMOUT_CTRL1_SEL_SHIFT 24 +#define TDMOUT_CTRL1_GAIN_EN 26 +#define TDMOUT_CTRL1_WS_INV BIT(28) +#define TDMOUT_SWAP 0x08 +#define TDMOUT_MASK0 0x0c +#define TDMOUT_MASK1 0x10 +#define TDMOUT_MASK2 0x14 +#define TDMOUT_MASK3 0x18 +#define TDMOUT_STAT 0x1c +#define TDMOUT_GAIN0 0x20 +#define TDMOUT_GAIN1 0x24 +#define TDMOUT_MUTE_VAL 0x28 +#define TDMOUT_MUTE0 0x2c +#define TDMOUT_MUTE1 0x30 +#define TDMOUT_MUTE2 0x34 +#define TDMOUT_MUTE3 0x38 +#define TDMOUT_MASK_VAL 0x3c + +static const struct regmap_config axg_tdmout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMOUT_MASK_VAL, +}; + +static const struct snd_kcontrol_new axg_tdmout_controls[] = { + SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0), + SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1, 0, 8, 255, 0), + SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0), + SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1, + TDMOUT_CTRL1_GAIN_EN, 1, 0), +}; + +static const char * const tdmout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1, + TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts); + +static const struct snd_kcontrol_new axg_tdmout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum); + +static struct snd_soc_dai * +axg_tdmout_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (!p->connect) + continue; + + if (p->sink->id == snd_soc_dapm_dai_in) + return (struct snd_soc_dai *)p->sink->priv; + + be = axg_tdmout_get_be(p->sink); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmout_get_be(w); + + if (!be) + return NULL; + + return be->playback_dma_data; +} + +static void axg_tdmout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT); + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_IN, TDMOUT_CTRL0_RST_IN); + + /* Actually enable tdmout */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE); +} + +static void axg_tdmout_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0); +} + +static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set the stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMOUT_CTRL0_INIT_BITNUM(1); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val |= TDMOUT_CTRL0_INIT_BITNUM(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set the slot width */ + val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1); + + /* Set the slot number */ + val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1); + + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_INIT_BITNUM_MASK | + TDMOUT_CTRL0_BITNUM_MASK | + TDMOUT_CTRL0_SLOTNUM_MASK, val); + + /* Set the sample width */ + val = TDMOUT_CTRL1_MSB_POS(ts->width - 1); + + /* FIFO data are arranged in chunks of 64bits */ + switch (ts->physical_width) { + case 8: + /* 8 samples of 8 bits */ + val |= TDMOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(4); + break; + default: + pr_err("Unsupported physical width: %u\n", + ts->physical_width); + return -EINVAL; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMOUT_CTRL1_WS_INV; + + regmap_update_bits(map, TDMOUT_CTRL1, + (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK | + TDMOUT_CTRL1_WS_INV), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMOUT_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux), + SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "ENC", NULL, "SRC SEL" }, + { "OUT", NULL, "ENC" }, +}; + +static const struct snd_soc_component_driver axg_tdmout_component_drv = { + .controls = axg_tdmout_controls, + .num_controls = ARRAY_SIZE(axg_tdmout_controls), + .dapm_widgets = axg_tdmout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmout_dapm_widgets), + .dapm_routes = axg_tdmout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmout_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmout_ops = { + .get_stream = axg_tdmout_get_tdm_stream, + .prepare = axg_tdmout_prepare, + .enable = axg_tdmout_enable, + .disable = axg_tdmout_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmout_drv = { + .component_drv = &axg_tdmout_component_drv, + .regmap_cfg = &axg_tdmout_regmap_cfg, + .ops = &axg_tdmout_ops, + .invert_sclk = true, +}; + +static const struct of_device_id axg_tdmout_of_match[] = { + { + .compatible = "amlogic,axg-tdmout", + .data = &axg_tdmout_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmout_of_match); + +static struct platform_driver axg_tdmout_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmout", + .of_match_table = axg_tdmout_of_match, + }, +}; +module_platform_driver(axg_tdmout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From 13a22e6a98f8b47d61948fcd095d862377b3b143 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:01 +0200 Subject: [PATCH 309/529] ASoC: meson: add tdm input driver Add Amlogic's axg TDM input driver which take the TDM signal of 4 input lanes and push the decoded audio samples to TODDR fifo Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 8 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdmin.c | 229 ++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+) create mode 100644 sound/soc/meson/axg-tdmin.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 08a522b77749..00d05df67b52 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -27,6 +27,14 @@ config SND_MESON_AXG_TDM_INTERFACE tristate select SND_MESON_AXG_TDM_FORMATTER +config SND_MESON_AXG_TDMIN + tristate "Amlogic AXG TDM Input Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM input formatter embedded + in the Amlogic AXG SoC family + config SND_MESON_AXG_TDMOUT tristate "Amlogic AXG TDM Output Support" select SND_MESON_AXG_TDM_FORMATTER diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index a665c6b76a95..f62833fb44d8 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -5,6 +5,7 @@ snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o @@ -13,5 +14,6 @@ obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c new file mode 100644 index 000000000000..bbac44c81688 --- /dev/null +++ b/sound/soc/meson/axg-tdmin.c @@ -0,0 +1,229 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +#define TDMIN_CTRL 0x00 +#define TDMIN_CTRL_ENABLE BIT(31) +#define TDMIN_CTRL_I2S_MODE BIT(30) +#define TDMIN_CTRL_RST_OUT BIT(29) +#define TDMIN_CTRL_RST_IN BIT(28) +#define TDMIN_CTRL_WS_INV BIT(25) +#define TDMIN_CTRL_SEL_SHIFT 20 +#define TDMIN_CTRL_IN_BIT_SKEW_MASK GENMASK(18, 16) +#define TDMIN_CTRL_IN_BIT_SKEW(x) ((x) << 16) +#define TDMIN_CTRL_LSB_FIRST BIT(5) +#define TDMIN_CTRL_BITNUM_MASK GENMASK(4, 0) +#define TDMIN_CTRL_BITNUM(x) ((x) << 0) +#define TDMIN_SWAP 0x04 +#define TDMIN_MASK0 0x08 +#define TDMIN_MASK1 0x0c +#define TDMIN_MASK2 0x10 +#define TDMIN_MASK3 0x14 +#define TDMIN_STAT 0x18 +#define TDMIN_MUTE_VAL 0x1c +#define TDMIN_MUTE0 0x20 +#define TDMIN_MUTE1 0x24 +#define TDMIN_MUTE2 0x28 +#define TDMIN_MUTE3 0x2c + +static const struct regmap_config axg_tdmin_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMIN_MUTE3, +}; + +static const char * const axg_tdmin_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5", +}; + +/* Change to special mux control to reset dapm */ +static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL, + TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts); + +static const struct snd_kcontrol_new axg_tdmin_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum); + +static struct snd_soc_dai * +axg_tdmin_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + if (p->source->id == snd_soc_dapm_dai_out) + return (struct snd_soc_dai *)p->source->priv; + + be = axg_tdmin_get_be(p->source); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmin_get_be(w); + + if (!be) + return NULL; + + return be->capture_dma_data; +} + +static void axg_tdmin_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT); + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_IN, TDMIN_CTRL_RST_IN); + + /* Actually enable tdmin */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE); +} + +static void axg_tdmin_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0); +} + +static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMIN_CTRL_IN_BIT_SKEW(3); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val = TDMIN_CTRL_IN_BIT_SKEW(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set stream format mode */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + val |= TDMIN_CTRL_I2S_MODE; + break; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMIN_CTRL_WS_INV; + + /* Set the slot width */ + val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1); + + /* + * The following also reset LSB_FIRST which result in the formatter + * placing the first bit received at bit 31 + */ + regmap_update_bits(map, TDMIN_CTRL, + (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV | + TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST | + TDMIN_CTRL_BITNUM_MASK), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMIN_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux), + SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 5", "IN 5" }, + { "DEC", NULL, "SRC SEL" }, + { "OUT", NULL, "DEC" }, +}; + +static const struct snd_soc_component_driver axg_tdmin_component_drv = { + .dapm_widgets = axg_tdmin_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmin_dapm_widgets), + .dapm_routes = axg_tdmin_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmin_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmin_ops = { + .get_stream = axg_tdmin_get_tdm_stream, + .prepare = axg_tdmin_prepare, + .enable = axg_tdmin_enable, + .disable = axg_tdmin_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmin_drv = { + .component_drv = &axg_tdmin_component_drv, + .regmap_cfg = &axg_tdmin_regmap_cfg, + .ops = &axg_tdmin_ops, + .invert_sclk = false, +}; + +static const struct of_device_id axg_tdmin_of_match[] = { + { + .compatible = "amlogic,axg-tdmin", + .data = &axg_tdmin_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); + +static struct platform_driver axg_tdmin_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmin", + .of_match_table = axg_tdmin_of_match, + }, +}; +module_platform_driver(axg_tdmin_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From cbdfab3b675f4c34258b0ec9e4707de44e1f6989 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:02 +0200 Subject: [PATCH 310/529] ASoC: export snd_soc_of_get_slot_mask Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot() directly because it needs to handle 4 mask for each direction. Yet the parsing of each mask is the same, so export snd_soc_of_get_slot_mask() to reuse the the existing code. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 7 ++++--- 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a23ecdf3eff1..ace474e8649e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1433,6 +1433,9 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, unsigned int *rx_mask, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08e189485009..ad5b0ef16d82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3428,9 +3428,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); -static int snd_soc_of_get_slot_mask(struct device_node *np, - const char *prop_name, - unsigned int *mask) +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) { u32 val; const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); @@ -3445,6 +3445,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return val; } +EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, From 2a05c71ea17b09c88a212e8fa6be1ccddd4613ab Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:03 +0200 Subject: [PATCH 311/529] ASoC: meson: add axg sound card DT bindings documentation Add the DT bindings documentation for axg sound card Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,axg-sound-card.txt | 124 ++++++++++++++++++ 1 file changed, 124 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt new file mode 100644 index 000000000000..39e005da0407 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt @@ -0,0 +1,124 @@ +Amlogic AXG sound card: + +Required properties: + +- compatible: "amlogic,axg-sound-card" +- amlogic,name : User specified audio sound card name, one string + +Optional properties: + +- amlogic,aux-devs : List of phandles pointing to auxiliary devices +- amlogic,widgets : Please refer to widgets.txt. +- amlogic,routing : A list of the connections between audio components. + +Subnodes: + +- amlogic,dai-link: Container for dai-link level properties and the + CODEC sub-nodes. There should be at least one (and + probably) subnode of this type. + +Required dai-link properties: + +- sound-dai: phandle and port of the CPU DAI. + +Required TDM Backend dai-link properties: +- dai-format : CPU/CODEC common audio format + +Optional TDM Backend dai-link properties: +- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks +- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks + When omitted, mask is assumed to have to no + slots. A valid must have at one slot, so at + least one these mask should be provided with + an enabled slot. +- dai-tdm-slot-num : Please refer to tdm-slot.txt. + If omitted, slot number is set to accommodate the largest + mask provided. +- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted. +- mclk-fs : Multiplication factor between stream rate and mclk + +Backend dai-link subnodes: + +- codec: dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. + dai-link representing frontend links have no codec, therefore have no + subnodes + +Required codec subnodes properties: + +- sound-dai: phandle and port of the CODEC DAI. + +Optional codec subnodes properties: + +- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt. +- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt. + +Example: + +sound { + compatible = "amlogic,axg-sound-card"; + amlogic,name = "AXG-S420"; + amlogic,aux-devs = <&tdmin_a>, <&tdmout_c>; + amlogic,widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right"; + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + amlogic,routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + amlogic,dai-link@0 { + sound-dai = <&frddr_a>; + }; + + amlogic,dai-link@1 { + sound-dai = <&toddr_a>; + }; + + amlogic,dai-link@2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec@0 { + sound-dai = <&lineout>; + }; + + codec@1 { + sound-dai = <&speaker_amp1>; + }; + + codec@2 { + sound-dai = <&speaker_amp2>; + }; + + codec@3 { + sound-dai = <&linein>; + }; + + }; + + amlogic,dai-link@4 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; +}; From 7864a79f37b55769b817d5e6c5ae0ca4bfdba93b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:04 +0200 Subject: [PATCH 312/529] ASoC: meson: add axg sound card support Add the axg sound card to handle the specifities of the axg audio sub system. This card is required to: * setup the dpcm links specific to the AXG (with a cpu sound dai) * handle the 4 lanes masks of the tdm interfaces * add the loopback link when a tdm pad interface has a playback stream * handle multi-codec links Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 11 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 671 +++++++++++++++++++++++++++++++++++++ 3 files changed, 684 insertions(+) create mode 100644 sound/soc/meson/axg-card.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 00d05df67b52..4cf93c05a982 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -43,6 +43,17 @@ config SND_MESON_AXG_TDMOUT Select Y or M to add support for TDM output formatter embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_SOUND_CARD + tristate "Amlogic AXG Sound Card Support" + select SND_MESON_AXG_TDM_INTERFACE + imply SND_MESON_AXG_FRDDR + imply SND_MESON_AXG_TODDR + imply SND_MESON_AXG_TDMIN + imply SND_MESON_AXG_TDMOUT + imply SND_MESON_AXG_SPDIFOUT + help + Select Y or M to add support for the AXG SoC sound card + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index f62833fb44d8..c5e003b093db 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -7,6 +7,7 @@ snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o +snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -16,4 +17,5 @@ obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o +obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c new file mode 100644 index 000000000000..d6d1081d94ad --- /dev/null +++ b/sound/soc/meson/axg-card.c @@ -0,0 +1,671 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include "axg-tdm.h" + +struct axg_card { + struct snd_soc_card card; + void **link_data; +}; + +struct axg_dai_link_tdm_mask { + u32 tx; + u32 rx; +}; + +struct axg_dai_link_tdm_data { + unsigned int mclk_fs; + unsigned int slots; + unsigned int slot_width; + u32 *tx_mask; + u32 *rx_mask; + struct axg_dai_link_tdm_mask *codec_masks; +}; + +#define PREFIX "amlogic," + +static int axg_card_reallocate_links(struct axg_card *priv, + unsigned int num_links) +{ + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} + +static int axg_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} + +static int axg_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, link->cpu_of_node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +static void axg_card_clean_references(struct axg_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + int i, j; + + if (card->dai_link) { + for (i = 0; i < card->num_links; i++) { + link = &card->dai_link[i]; + of_node_put(link->cpu_of_node); + for (j = 0; j < link->num_codecs; j++) + of_node_put(link->codecs[j].of_node); + } + } + + if (card->aux_dev) { + for (i = 0; i < card->num_aux_devs; i++) + of_node_put(card->aux_dev[i].codec_of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +static int axg_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, PREFIX "aux-devs", NULL); + if (num == -ENOENT) { + /* + * It is ok to have no auxiliary devices but for this card it + * is a strange situtation. Let's warn the about it. + */ + dev_warn(card->dev, "card has no auxiliary devices\n"); + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for (i = 0; i < card->num_aux_devs; i++, aux++) { + aux->codec_of_node = of_parse_phandle(node, + PREFIX "aux-devs", i); + if (!aux->codec_of_node) + return -EINVAL; + } + + return 0; +} + +static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (be->mclk_fs) { + mclk = params_rate(params) * be->mclk_fs; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + } + + return 0; +} + +static const struct snd_soc_ops axg_card_tdm_be_ops = { + .hw_params = axg_card_tdm_be_hw_params, +}; + +static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + int ret, i; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_tdm_slot(codec_dai, + be->codec_masks[i].tx, + be->codec_masks[i].rx, + be->slots, be->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(codec_dai->dev, + "setting tdm link slots failed\n"); + return ret; + } + } + + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + int ret; + + /* The loopback rx_mask is the pad tx_mask */ + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_add_tdm_loopback(struct snd_soc_card *card, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *lb; + int ret; + + /* extend links */ + ret = axg_card_reallocate_links(priv, card->num_links + 1); + if (ret) + return ret; + + lb = &card->dai_link[*index + 1]; + + lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + if (!lb->name) + return -ENOMEM; + + lb->stream_name = lb->name; + lb->cpu_of_node = pad->cpu_of_node; + lb->cpu_dai_name = "TDM Loopback"; + lb->codec_name = "snd-soc-dummy"; + lb->codec_dai_name = "snd-soc-dummy-dai"; + lb->dpcm_capture = 1; + lb->no_pcm = 1; + lb->ops = &axg_card_tdm_be_ops; + lb->init = axg_card_tdm_dai_lb_init; + + /* Provide the same link data to the loopback */ + priv->link_data[*index + 1] = priv->link_data[*index]; + + /* + * axg_card_clean_references() will iterate over this link, + * make sure the node count is balanced + */ + of_node_get(lb->cpu_of_node); + + /* Let add_links continue where it should */ + *index += 1; + + return 0; +} + +static unsigned int axg_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} + +static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + char propname[32]; + u32 tx, rx; + int i; + + be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->tx_mask), GFP_KERNEL); + be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->rx_mask), GFP_KERNEL); + if (!be->tx_mask || !be->rx_mask) + return -ENOMEM; + + for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]); + tx = max(tx, be->tx_mask[i]); + } + + /* Disable playback is the interface has no tx slots */ + if (!tx) + link->dpcm_playback = 0; + + for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]); + rx = max(rx, be->rx_mask[i]); + } + + /* Disable capture is the interface has no rx slots */ + if (!rx) + link->dpcm_capture = 0; + + /* ... but the interface should at least have one of them */ + if (!tx && !rx) { + dev_err(card->dev, "tdm link has no cpu slots\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-num", &be->slots); + if (!be->slots) { + /* + * If the slot number is not provided, set it such as it + * accommodates the largest mask + */ + be->slots = fls(max(tx, rx)); + } else if (be->slots < fls(max(tx, rx)) || be->slots > 32) { + /* + * Error if the slots can't accommodate the largest mask or + * if it is just too big + */ + dev_err(card->dev, "bad slot number\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width); + + return 0; +} + +static int axg_card_parse_codecs_masks(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + struct axg_dai_link_tdm_mask *codec_mask; + struct device_node *np; + + codec_mask = devm_kcalloc(card->dev, link->num_codecs, + sizeof(*codec_mask), GFP_KERNEL); + if (!codec_mask) + return -ENOMEM; + + be->codec_masks = codec_mask; + + for_each_child_of_node(node, np) { + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", + &codec_mask->rx); + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", + &codec_mask->tx); + + codec_mask++; + } + + return 0; +} + +static int axg_card_parse_tdm(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct axg_dai_link_tdm_data *be; + int ret; + + /* Allocate tdm link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup tdm link */ + link->ops = &axg_card_tdm_be_ops; + link->init = axg_card_tdm_dai_init; + link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + ret = axg_card_parse_cpu_tdm_slots(card, link, node, be); + if (ret) { + dev_err(card->dev, "error parsing tdm link slots\n"); + return ret; + } + + ret = axg_card_parse_codecs_masks(card, link, node, be); + if (ret) + return ret; + + /* Add loopback if the pad dai has playback */ + if (link->dpcm_playback) { + ret = axg_card_add_tdm_loopback(card, index); + if (ret) + return ret; + } + + return 0; +} + +static int axg_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = axg_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = axg_card_set_link_name(card, link, "be"); + if (ret) + dev_err(card->dev, "error setting %s link name\n", np->name); + + return ret; +} + +static int axg_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + bool is_playback) +{ + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return axg_card_set_link_name(card, link, "fe"); +} + +static int axg_card_cpu_is_capture_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-toddr"); +} + +static int axg_card_cpu_is_playback_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-frddr"); +} + +static int axg_card_cpu_is_tdm_iface(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); +} + +static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + int ret; + + ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); + if (ret) + return ret; + + if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, true); + else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, false); + else + ret = axg_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node)) + ret = axg_card_parse_tdm(card, np, index); + + return ret; +} + +static int axg_card_add_links(struct snd_soc_card *card) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = axg_card_reallocate_links(priv, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = axg_card_add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int axg_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static const struct of_device_id axg_card_of_match[] = { + { .compatible = "amlogic,axg-sound-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_card_of_match); + +static int axg_card_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_card *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + + ret = snd_soc_of_parse_card_name(&priv->card, PREFIX "name"); + if (ret < 0) + return ret; + + ret = axg_card_parse_of_optional(&priv->card, PREFIX "routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = axg_card_parse_of_optional(&priv->card, PREFIX "widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = axg_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = axg_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + axg_card_clean_references(priv); + return ret; +} + +static int axg_card_remove(struct platform_device *pdev) +{ + struct axg_card *priv = platform_get_drvdata(pdev); + + axg_card_clean_references(priv); + + return 0; +} + +static struct platform_driver axg_card_pdrv = { + .probe = axg_card_probe, + .remove = axg_card_remove, + .driver = { + .name = "axg-sound-card", + .of_match_table = axg_card_of_match, + }, +}; +module_platform_driver(axg_card_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From a8e43c21a8a32a3af4abc605b6ebcab039f28e00 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 16 Jul 2018 08:24:45 +0200 Subject: [PATCH 313/529] ASoC: pxa: remove clock divider and pll setup from zylonite and magician The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 106 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 2 insertions(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); From d10ee9c542365bdc0a7497306e21ff6c7f2172b0 Mon Sep 17 00:00:00 2001 From: Srikanth K H Date: Fri, 20 Jul 2018 11:13:51 +0530 Subject: [PATCH 314/529] ALSA: timer: catch invalid timer object creation A timer object for the classes SNDRV_TIMER_CLASS_CARD and SNDRV_TIMER_CLASS_PCM has to be associated with a card object, but we have no check at creation time. Such a timer object with NULL card causes various unexpected problems, e.g. NULL dereference at reading the sound timer proc file. So as preventive measure while the creating the sound timer object is created the card information availability is checked for the mentioned entries and returned error if its NULL. Signed-off-by: Srikanth K H Signed-off-by: Takashi Iwai --- sound/core/timer.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index b6f076bbc72d..61a0cec6e1f6 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -883,6 +883,11 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, if (snd_BUG_ON(!tid)) return -EINVAL; + if (tid->dev_class == SNDRV_TIMER_CLASS_CARD || + tid->dev_class == SNDRV_TIMER_CLASS_PCM) { + if (WARN_ON(!card)) + return -EINVAL; + } if (rtimer) *rtimer = NULL; timer = kzalloc(sizeof(*timer), GFP_KERNEL); From dfef01e150824b0e6da750cacda8958188d29aea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jul 2018 11:01:04 +0200 Subject: [PATCH 315/529] ALSA: memalloc: Don't exceed over the requested size snd_dma_alloc_pages_fallback() tries to allocate pages again when the allocation fails with reduced size. But the first try actually *increases* the size to power-of-two, which may give back a larger chunk than the requested size. This confuses the callers, e.g. sgbuf assumes that the size is equal or less, and it may result in a bad loop due to the underflow and eventually lead to Oops. The code of this function seems incorrectly assuming the usage of get_order(). We need to decrease at first, then align to power-of-two. Reported-and-tested-by: he, bo Reported-by: zhang jun Cc: Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 7f89d3c79a4b..753d5fc4b284 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,16 +242,12 @@ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size, int err; while ((err = snd_dma_alloc_pages(type, device, size, dmab)) < 0) { - size_t aligned_size; if (err != -ENOMEM) return err; if (size <= PAGE_SIZE) return -ENOMEM; - aligned_size = PAGE_SIZE << get_order(size); - if (size != aligned_size) - size = aligned_size; - else - size >>= 1; + size >>= 1; + size = PAGE_SIZE << get_order(size); } if (! dmab->area) return -ENOMEM; From 1ea0358ecb848058b35b6da13d7f4c08610a73a8 Mon Sep 17 00:00:00 2001 From: Yue Wang Date: Mon, 23 Jul 2018 01:56:46 -0700 Subject: [PATCH 316/529] ALSA: usb-audio: Generic DSD detection for Thesycon-based implementations Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..06ae3f5adf4b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1443,6 +1443,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; From 2f562a4739602d96928de0b040ce46b00d8d3adc Mon Sep 17 00:00:00 2001 From: Liang Chen Date: Mon, 23 Jul 2018 17:25:21 +0800 Subject: [PATCH 317/529] ASoC: rockchip-i2s: add description for px30 Add "rockchip,px30-i2s", "rockchip,rk3066-i2s" for i2s on px30 platform. Acked-by: Rob Herring Signed-off-by: Liang Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b208a752576c..54aefab71f2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,6 +7,7 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 From f9b54e1961c7052e7d7817d707826eb2b9a1ca09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Jul 2018 15:05:06 +0200 Subject: [PATCH 318/529] ALSA: hda/i915: Allow delayed i915 audio component binding Currently HD-audio i915 audio binding doesn't support any delayed binding, and supposes that the i915 driver registers the component immediately. This has been OK, so far, but the work-in-progress change in i915 may introduce the asynchronous binding, which effectively delays the component registration. For addressing it, implement a completion to be synced with the master binding. The timeout is set to 10 seconds which should be long enough and hopefully be not too annoying if anyone boots up a debugging session with i915 KMS turned off. Signed-off-by: Takashi Iwai --- sound/hda/hdac_i915.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8f2aa8bc1185..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,8 @@ #include #include +static struct completion bind_complete; + #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -97,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -117,7 +132,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - err = snd_hdac_acomp_init(bus, NULL, + init_completion(&bind_complete); + + err = snd_hdac_acomp_init(bus, &i915_init_ops, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -125,8 +142,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) acomp = bus->audio_component; if (!acomp) return -ENODEV; - if (!acomp->ops) + if (!acomp->ops) { request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } if (!acomp->ops) { snd_hdac_acomp_exit(bus); return -ENODEV; From a241c3d95b8b985834d218eedde3923ed683e862 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 24 Jul 2018 11:36:45 +0200 Subject: [PATCH 319/529] ASoC: meson: axg-spdifout: select SND_PCM_IEC958 When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the new driver: sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params': axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params' The other users use 'select', so we should do the same here. Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann Acked-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 4cf93c05a982..8af8bc358a90 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -56,6 +56,7 @@ config SND_MESON_AXG_SOUND_CARD config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 imply SND_SOC_SPDIF help Select Y or M to add support for SPDIF output serializer embedded From d96f8bd28cd0bae3e6702ae90df593628ef6906f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 24 Jul 2018 15:49:23 +0800 Subject: [PATCH 320/529] ASoC: rt5514: Fix the issue of the delay volume applied The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91bf018..dca82dd6e3bf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, From 467b061f1ac80f323bedb56d20a24f7bc0d2cec5 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 23 Jul 2018 16:54:03 +0100 Subject: [PATCH 321/529] ASoC: core: add support to snd_soc_dai_get_channel_map() On Qualcomm platforms, specifically with SLIMbus interfaced codecs, the codec slim channel numbers are passed to DSP while configuring the slim audio path. Having get_channel_map() would allow dais to share such information across multiple dais. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 8 ++++++++ sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 2 files changed, 30 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index a14bc0608ae9..f5d70041108f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -138,6 +138,11 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); + +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); + int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); struct snd_soc_dai_ops { @@ -165,6 +170,9 @@ struct snd_soc_dai_ops { int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); + int (*get_channel_map)(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); int (*set_sdw_stream)(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad5b0ef16d82..81b27923303d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2679,6 +2679,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); +/** + * snd_soc_dai_get_channel_map - Get DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + */ +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->get_channel_map) + return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map); + /** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI From aa624a0a9243460b9037889a05663d3b9a9e8f99 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:30 -0300 Subject: [PATCH 322/529] ASoC: fsl-asoc-card: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750aa3637..07808c6d5461 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include From 2ba28053686334075c8c8bff6c4f6e3fc9ac51ae Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:31 -0300 Subject: [PATCH 323/529] ASoC: fsl_asrc: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 18 +++++++----------- sound/soc/fsl/fsl_asrc.h | 5 +---- sound/soc/fsl/fsl_asrc_dma.c | 18 +++++++----------- 3 files changed, 15 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index adfb8135d739..528e8b108422 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index d558dd5499a5..c60075112570 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -1,13 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_asrc.h - Freescale ASRC ALSA SoC header file * * Copyright (C) 2014 Freescale Semiconductor, Inc. * * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_ASRC_H diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 565e16d8fe85..1033ac6631b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Platform (DMA) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include From ad47191a72ca0a95327e8bb16155e5e216f1d97d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:32 -0300 Subject: [PATCH 324/529] ASoC: fsl_utils: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 18 +++++++----------- sound/soc/fsl/fsl_utils.h | 7 ++----- 2 files changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7592b0406370..7f0fa4b52223 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -1,14 +1,10 @@ -/** - * Freescale ALSA SoC Machine driver utility - * - * Author: Timur Tabi - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ALSA SoC Machine driver utility +// +// Author: Timur Tabi +// +// Copyright 2010 Freescale Semiconductor, Inc. #include #include diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index 1687b66ef18e..c5dc2a14b492 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -1,13 +1,10 @@ -/** +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Freescale ALSA SoC Machine driver utility * * Author: Timur Tabi * * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_UTILS_H From 6b24e03ecf1ce188071d142115233b10ed8767b2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:33 -0300 Subject: [PATCH 325/529] ASoC: imx-sgtl5000: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..c29200cf755a 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -1,14 +1,7 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. #include #include From d77a760842751eb620a7e74f74e5a5ad6b3d6cef Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Jul 2018 09:48:20 +0800 Subject: [PATCH 326/529] ASoC: rt5631: add Volume to the name of volume control add Volume to the name of volume control. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index cf6dce69eb2a..e52e4670cf65 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), - SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), - SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), /* MONO IN */ SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), From 92beb0a26975c6459794d885d27f48357c1aefd0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:44 -0500 Subject: [PATCH 327/529] ASoC: Intel: Haswell: fix endianness handling Make all Sparse warnings go away by using le16/32_to_cpu. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-dsp.c | 53 +++++++++++++---------- 1 file changed, 29 insertions(+), 24 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index b2bec36d074c..a28220e67cdf 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module_template template; int count, ret; void __iomem *ram; + int type = le16_to_cpu(module->type); + int entry_point = le32_to_cpu(module->entry_point); /* TODO: allowed module types need to be configurable */ - if (module->type != SST_HSW_MODULE_BASE_FW - && module->type != SST_HSW_MODULE_PCM_SYSTEM - && module->type != SST_HSW_MODULE_PCM - && module->type != SST_HSW_MODULE_PCM_REFERENCE - && module->type != SST_HSW_MODULE_PCM_CAPTURE - && module->type != SST_HSW_MODULE_WAVES - && module->type != SST_HSW_MODULE_LPAL) + if (type != SST_HSW_MODULE_BASE_FW && + type != SST_HSW_MODULE_PCM_SYSTEM && + type != SST_HSW_MODULE_PCM && + type != SST_HSW_MODULE_PCM_REFERENCE && + type != SST_HSW_MODULE_PCM_CAPTURE && + type != SST_HSW_MODULE_WAVES && + type != SST_HSW_MODULE_LPAL) return 0; dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", module->signature, module->mod_size, - module->blocks, module->type); - dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + module->blocks, type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point); dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", module->info.persistent_size, module->info.scratch_size); memset(&template, 0, sizeof(template)); - template.id = module->type; - template.entry = module->entry_point - 4; - template.persistent_size = module->info.persistent_size; - template.scratch_size = module->info.scratch_size; + template.id = type; + template.entry = entry_point - 4; + template.persistent_size = le32_to_cpu(module->info.persistent_size); + template.scratch_size = le32_to_cpu(module->info.scratch_size); mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)module + sizeof(*module); - for (count = 0; count < module->blocks; count++) { + for (count = 0; count < le32_to_cpu(module->blocks); count++) { - if (block->size <= 0) { + if (le32_to_cpu(block->size) <= 0) { dev_err(dsp->dev, "error: block %d size invalid\n", count); sst_module_free(mod); return -EINVAL; } - switch (block->type) { + switch (le32_to_cpu(block->type)) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - mod->offset = - block->ram_offset + dsp->addr.iram_offset; + mod->offset = le32_to_cpu(block->ram_offset) + + dsp->addr.iram_offset; mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: case SST_HSW_REGS: ram = dsp->addr.lpe; - mod->offset = block->ram_offset; + mod->offset = le32_to_cpu(block->ram_offset); mod->type = SST_MEM_DRAM; break; default: @@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - mod->size = block->size; + mod->size = le32_to_cpu(block->size); mod->data = (void *)block + sizeof(*block); mod->data_offset = mod->data - fw->dma_buf; @@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return ret; } - block = (void *)block + sizeof(*block) + block->size; + block = (void *)block + sizeof(*block) + + le32_to_cpu(block->size); } mod->state = SST_MODULE_STATE_LOADED; @@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* verify FW */ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || - (sst_fw->size != header->file_size + sizeof(*header))) { + (sst_fw->size != + le32_to_cpu(header->file_size) + sizeof(*header))) { dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); return -EINVAL; } @@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* parse each module */ module = (void *)sst_fw->dma_buf + sizeof(*header); - for (count = 0; count < header->modules; count++) { + for (count = 0; count < le32_to_cpu(header->modules); count++) { /* module */ ret = hsw_parse_module(dsp, sst_fw, module); @@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) dev_err(dsp->dev, "error: invalid module %d\n", count); return ret; } - module = (void *)module + sizeof(*module) + module->mod_size; + module = (void *)module + sizeof(*module) + + le32_to_cpu(module->mod_size); } return 0; From 86efd35ec1e34094914494dacb83077e5df2d1b8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:45 -0500 Subject: [PATCH 328/529] ASoC: Intel: Skylake: BDL definitions should be __le32 Make sure definitions are consistent with usage. Detected with Sparse. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-sst-cldma.c | 8 ++++---- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index d2b1d60fec02..5bc0d38da7e3 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp *ctx) /* Code loader helper APIs */ static void skl_cldma_setup_bdle(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc) + __le32 **bdlp, int size, int with_ioc) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; ctx->cl_dev.frags = 0; while (size > 0) { @@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx) int skl_cldma_prepare(struct sst_dsp *ctx) { int ret; - u32 *bdl; + __le32 *bdl; ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; @@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx) ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); return ret; } - bdl = (u32 *)ctx->cl_dev.dmab_bdl.area; + bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area; /* Allocate BDLs */ ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h index 5b730a1a0ae4..ec736921a083 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.h +++ b/sound/soc/intel/skylake/skl-sst-cldma.h @@ -203,7 +203,7 @@ struct sst_dsp; struct skl_cl_dev_ops { void (*cl_setup_bdle)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc); + __le32 **bdlp, int size, int with_ioc); void (*cl_setup_controller)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_bdl, unsigned int max_size, u32 page_count); From ef3cb7423358ff0cadacb76f26e3e1f4da8be4ce Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:46 -0500 Subject: [PATCH 329/529] ASoC: Intel: common: make sst_dma functions static sst_dma_new and sst_dma_free are not used in any other file and don't have a prototype. Move to static functions and remove EXPORT_SYMBOL_GPL statement. Reported by sparse warnings. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/sst-firmware.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 657afc02f1c4..11041aedea31 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp) } EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel); -int sst_dma_new(struct sst_dsp *sst) +static int sst_dma_new(struct sst_dsp *sst) { struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; @@ -320,9 +320,8 @@ err_dma_dev: devm_kfree(sst->dev, dma); return ret; } -EXPORT_SYMBOL(sst_dma_new); -void sst_dma_free(struct sst_dma *dma) +static void sst_dma_free(struct sst_dma *dma) { if (dma == NULL) @@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma) dw_remove(dma->chip); } -EXPORT_SYMBOL(sst_dma_free); /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, From ce1cfe295abaa7436f9049bcb2562c843089abfc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:47 -0500 Subject: [PATCH 330/529] ASoC: Intel: Atom: simplify iomem address and casts Simplify code and add relevant casts to make Sparse warnings go away Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_drv_interface.c | 29 ++++++++++---------- 1 file changed, 15 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 6a8b253c58d2..5455d6e0ab53 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id, stream->cumm_bytes += bytes; dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = ((void __iomem *)(ctx->mailbox + ctx->tstamp)) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); fw_tstamp.bytes_copied = stream->cumm_bytes; dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", fw_tstamp.bytes_copied, bytes); - addr = ((void *)(ctx->mailbox + ctx->tstamp)) + - (str_id * sizeof(fw_tstamp)); offset = offsetof(struct snd_sst_tstamp, bytes_copied); sst_shim_write(addr, offset, fw_tstamp.bytes_copied); return 0; @@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, struct snd_sst_tstamp fw_tstamp = {0,}; struct stream_info *stream; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); stream = get_stream_info(ctx, str_id); if (!stream) @@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) struct snd_sst_tstamp fw_tstamp; unsigned int str_id; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; str_id = info->str_id; stream = get_stream_info(ctx, str_id); @@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) return -EINVAL; substream = stream->pcm_substream; - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - + (str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); + return sst_calc_tstamp(ctx, info, substream, &fw_tstamp); } From 9a0daaab31e9e39047ced79409313c34dae4635a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:48 -0500 Subject: [PATCH 331/529] ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32 This looks like a copy/paste issue, but clearly there is an inversion that is obvious when checking the arguments. Detected with Sparse - now that we have fewer warnings this one was easy to find. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_loader.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index a686eef2cf7f..27413ebae956 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count) /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __iowrite32_copy(dst, src, count / 4); } void memcpy32_fromio(void *dst, const void __iomem *src, int count) { - /* __iowrite32_copy uses 32-bit count values so divide by 4 for + /* __ioread32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __ioread32_copy(dst, src, count / 4); } /** From fe65324e3f5205072a2d55ac9c63ec77155fa528 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Tue, 24 Jul 2018 19:50:48 -0500 Subject: [PATCH 332/529] ASoC: Intel: Skylake: fix widget handling include DAPM Mux and output widgets into the list. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 647e52aecdc3..9845d853644c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; From a49a71f6e25da2acc637fcd31e73debd96ca18f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 16:34:12 +0200 Subject: [PATCH 333/529] ALSA: seq: Fix poll() error return The sanity checks in ALSA sequencer and OSS sequencer emulation codes return falsely -ENXIO from poll callback. They should be EPOLLERR instead. This was caught thanks to the recent change to the return value. Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 5f64d0d88320..e1f44fc86885 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -203,7 +203,7 @@ odev_poll(struct file *file, poll_table * wait) struct seq_oss_devinfo *dp; dp = file->private_data; if (snd_BUG_ON(!dp)) - return -ENXIO; + return EPOLLERR; return snd_seq_oss_poll(dp, file, wait); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 56ca78423040..6fd4b074b206 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1101,7 +1101,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait) /* check client structures are in place */ if (snd_BUG_ON(!client)) - return -ENXIO; + return EPOLLERR; if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) && client->data.user.fifo) { From fff71a4c050ba46e305d910c837b99ba1728135e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 17:10:11 +0200 Subject: [PATCH 334/529] ALSA: vx222: Fix invalid endian conversions The endian conversions used in vx2_dma_read() and vx2_dma_write() are superfluous and even wrong on big-endian machines, as inl() and outl() already do conversions. Kill them. Spotted by sparse, a warning like: sound/pci/vx222/vx222_ops.c:278:30: warning: incorrect type in argument 1 (different base types) Cc: Signed-off-by: Takashi Iwai --- sound/pci/vx222/vx222_ops.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index d4298af6d3ee..c0d0bf44f365 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -275,7 +275,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } addr = (u32 *)runtime->dma_area; @@ -285,7 +285,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } @@ -313,7 +313,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -321,7 +321,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); vx2_release_pseudo_dma(chip); } From 3acd3e3bab95ec3622ff98da313290ee823a0f68 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 17:11:38 +0200 Subject: [PATCH 335/529] ALSA: vxpocket: Fix invalid endian conversions The endian conversions used in vxp_dma_read() and vxp_dma_write() are superfluous and even wrong on big-endian machines, as inw() and outw() already do conversions. Kill them. Cc: Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxp_ops.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 8cde40226355..4c4ef1fec69f 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -375,7 +375,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outw(cpu_to_le16(*addr), port); + outw(*addr, port); addr++; } addr = (unsigned short *)runtime->dma_area; @@ -385,7 +385,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outw(cpu_to_le16(*addr), port); + outw(*addr, port); addr++; } vx_release_pseudo_dma(chip); @@ -417,7 +417,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le16_to_cpu(inw(port)); + *addr++ = inw(port); addr = (unsigned short *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -425,12 +425,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; count > 1; count--) - *addr++ = le16_to_cpu(inw(port)); + *addr++ = inw(port); /* Disable DMA */ pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK; vx_outb(chip, DIALOG, pchip->regDIALOG); /* Read the last word (16 bits) */ - *addr = le16_to_cpu(inw(port)); + *addr = inw(port); /* Disable 16-bit accesses */ pchip->regDIALOG &= ~VXP_DLG_DMA16_SEL_MASK; vx_outb(chip, DIALOG, pchip->regDIALOG); From 69756930f2de0457d51db7d505a1e4f40e9fd116 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 17:59:26 +0200 Subject: [PATCH 336/529] ALSA: cs5535audio: Fix invalid endian conversion One place in cs5535audio_build_dma_packets() does an extra conversion via cpu_to_le32(); namely jmpprd_addr is passed to setup_prd() ops, which writes the value via cs_writel(). That is, the callback does the conversion by itself, and we don't need to convert beforehand. This patch fixes that bogus conversion. Cc: Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.h | 6 +++--- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index f4fcdf93f3c8..d84620a0c26c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -67,9 +67,9 @@ struct cs5535audio_dma_ops { }; struct cs5535audio_dma_desc { - u32 addr; - u16 size; - u16 ctlreserved; + __le32 addr; + __le16 size; + __le16 ctlreserved; }; struct cs5535audio_dma { diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index ee7065f6e162..326caec854e1 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -158,8 +158,8 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr); lastdesc->size = 0; lastdesc->ctlreserved = cpu_to_le16(PRD_JMP); - jmpprd_addr = cpu_to_le32(lastdesc->addr + - (sizeof(struct cs5535audio_dma_desc)*periods)); + jmpprd_addr = (u32)dma->desc_buf.addr + + sizeof(struct cs5535audio_dma_desc) * periods; dma->substream = substream; dma->period_bytes = period_bytes; From b080dc5bd0dfc0b33c6cfc31f909c93d5e63c186 Mon Sep 17 00:00:00 2001 From: Jeff Crukley Date: Wed, 25 Jul 2018 15:05:01 -0400 Subject: [PATCH 337/529] ALSA: usb-audio: Add support for Encore mDSD USB DAC This patch adds native DSD playback support for the Encore mDSD USB DAC by specifying the vendor and product ID's Signed-off-by: Jeff Crukley Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 06ae3f5adf4b..dde87d64bc32 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ From bd1cd0eb2ce9141100628d476ead4de485501b29 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:46 +0200 Subject: [PATCH 338/529] ALSA: usb-audio: Fix multiple definitions in AU0828_DEVICE() macro AU0828_DEVICE() macro in quirks-table.h uses USB_DEVICE_VENDOR_SPEC() for expanding idVendor and idProduct fields. However, the latter macro adds also match_flags and bInterfaceClass, which are different from the values AU0828_DEVICE() macro sets after that. For fixing them, just expand idVendor and idProduct fields manually in AU0828_DEVICE(). This fixes sparse warnings like: sound/usb/quirks-table.h:2892:1: warning: Initializer entry defined twice Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8aac48f9c322..08aa78007020 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2875,7 +2875,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ #define AU0828_DEVICE(vid, pid, vname, pname) { \ - USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .idVendor = vid, \ + .idProduct = pid, \ .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ USB_DEVICE_ID_MATCH_INT_CLASS | \ USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ From ab647a2d62f70cf45d7c64bf9f1574b8c52406e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:47 +0200 Subject: [PATCH 339/529] ALSA: msnd: Add missing __iomem annotations The io-mapped buffers used in msnd drivers need __iomem annotations. This fixes sparse warnings like: sound/isa/msnd/msnd_pinnacle.c:172:45: warning: incorrect type in initializer (different address spaces) Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd.c | 18 +++++++++--------- sound/isa/msnd/msnd.h | 2 +- sound/isa/msnd/msnd_midi.c | 2 +- sound/isa/msnd/msnd_pinnacle.c | 2 +- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 569897f64fda..7c3203fe4869 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -54,7 +54,7 @@ #define LOGNAME "msnd" -void snd_msnd_init_queue(void *base, int start, int size) +void snd_msnd_init_queue(void __iomem *base, int start, int size) { writew(PCTODSP_BASED(start), base + JQS_wStart); writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); @@ -270,7 +270,7 @@ int snd_msnd_DARQ(struct snd_msnd *chip, int bank) udelay(1); if (chip->capturePeriods == 2) { - void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + void __iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + bank * DAQDS__size + DAQDS_wStart; unsigned short offset = 0x3000 + chip->capturePeriodBytes; @@ -309,7 +309,7 @@ int snd_msnd_DAPQ(struct snd_msnd *chip, int start) { u16 DAPQ_tail; int protect = start, nbanks = 0; - void *DAQD; + void __iomem *DAQD; static int play_banks_submitted; /* unsigned long flags; spin_lock_irqsave(&chip->lock, flags); not necessary */ @@ -370,7 +370,7 @@ static void snd_msnd_play_reset_queue(struct snd_msnd *chip, unsigned int pcm_count) { int n; - void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; chip->last_playbank = -1; chip->playLimit = pcm_count * (pcm_periods - 1); @@ -398,7 +398,7 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, unsigned int pcm_count) { int n; - void *pDAQ; + void __iomem *pDAQ; /* unsigned long flags; */ /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ @@ -485,7 +485,7 @@ static int snd_msnd_playback_open(struct snd_pcm_substream *substream) clear_bit(F_WRITING, &chip->flags); snd_msnd_enable_irq(chip); - runtime->dma_area = chip->mappedbase; + runtime->dma_area = (__force void *)chip->mappedbase; runtime->dma_bytes = 0x3000; chip->playback_substream = substream; @@ -508,7 +508,7 @@ static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, { int i; struct snd_msnd *chip = snd_pcm_substream_chip(substream); - void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; chip->play_sample_size = snd_pcm_format_width(params_format(params)); chip->play_channels = params_channels(params); @@ -589,7 +589,7 @@ static int snd_msnd_capture_open(struct snd_pcm_substream *substream) set_bit(F_AUDIO_READ_INUSE, &chip->flags); snd_msnd_enable_irq(chip); - runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_area = (__force void *)chip->mappedbase + 0x3000; runtime->dma_bytes = 0x3000; memset(runtime->dma_area, 0, runtime->dma_bytes); chip->capture_substream = substream; @@ -654,7 +654,7 @@ static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, { int i; struct snd_msnd *chip = snd_pcm_substream_chip(substream); - void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; chip->capture_sample_size = snd_pcm_format_width(params_format(params)); chip->capture_channels = params_channels(params); diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h index 5f3c7dcd9f9d..80c718757eef 100644 --- a/sound/isa/msnd/msnd.h +++ b/sound/isa/msnd/msnd.h @@ -283,7 +283,7 @@ struct snd_msnd { }; -void snd_msnd_init_queue(void *base, int start, int size); +void snd_msnd_init_queue(void __iomem *base, int start, int size); int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); int snd_msnd_send_word(struct snd_msnd *chip, diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 013d8d1170fe..42876b0cb68b 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -119,7 +119,7 @@ void snd_msndmidi_input_read(void *mpuv) { unsigned long flags; struct snd_msndmidi *mpu = mpuv; - void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + void __iomem *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; u16 head, tail, size; spin_lock_irqsave(&mpu->input_lock, flags); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 6c584d9b6c42..642609f7eda9 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -169,7 +169,7 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) { struct snd_msnd *chip = dev_id; - void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + void __iomem *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; u16 head, tail, size; /* Send ack to DSP */ From 7c500f9ea139d0c9b80fdea5a9c911db3166ea54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:48 +0200 Subject: [PATCH 340/529] ALSA: msnd: Fix the default sample sizes The default sample sizes set by msnd driver are bogus; it sets ALSA PCM format, not the actual bit width. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 642609f7eda9..7e3f52c8a6c6 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -82,10 +82,10 @@ static void set_default_audio_parameters(struct snd_msnd *chip) { - chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_size = snd_pcm_format_width(DEFSAMPLESIZE); chip->play_sample_rate = DEFSAMPLERATE; chip->play_channels = DEFCHANNELS; - chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_size = snd_pcm_format_width(DEFSAMPLESIZE); chip->capture_sample_rate = DEFSAMPLERATE; chip->capture_channels = DEFCHANNELS; } From bb86124c80780d9d2a2d9bef7f0840e61007ebb7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:49 +0200 Subject: [PATCH 341/529] ALSA: hda/ca0132 - Use NULL instead of 0 Use NULL for initializing the snd_kcontrol_new.tlv field, instead of 0, as warned by sparse: sound/pci/hda/patch_ca0132.c:5519:22: warning: Using plain integer as NULL pointer Also, the driver does the same initialization twice, once for knew.tlv.c and another for knew.tlv.p while both point to the same address (these are union). Drop the latter superfluous one. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 321e95c409c1..27d3388cd2a2 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5516,8 +5516,7 @@ static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid, sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]); - knew.tlv.c = 0; - knew.tlv.p = 0; + knew.tlv.c = NULL; switch (nid) { case XBASS_XOVER: From dcda6f7853c52c42d242b474bfbd1d17d02c8870 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:50 +0200 Subject: [PATCH 342/529] ALSA: msnd: Use NULL instead of 0 Fix a sparse warning: sound/isa/msnd/msnd_pinnacle.c:813:1: warning: Using plain integer as NULL pointer Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 7e3f52c8a6c6..11af9c40bc05 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -810,7 +810,7 @@ module_param(calibrate_signal, int, 0444); #ifndef MSND_CLASSIC module_param_array(digital, int, NULL, 0444); module_param_hw_array(cfg, long, ioport, NULL, 0444); -module_param_array(reset, int, 0, 0444); +module_param_array(reset, int, NULL, 0444); module_param_hw_array(mpu_io, long, ioport, NULL, 0444); module_param_hw_array(mpu_irq, int, irq, NULL, 0444); module_param_hw_array(ide_io0, long, ioport, NULL, 0444); From ebd836edfc4324da016ce0c09f809f882a133f50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:51 +0200 Subject: [PATCH 343/529] ALSA: hda - Fix a sparse warning about snd_ctl_elem_iface_t The knew->iface field is in snd_ctl_elem_iface_t, which is with __bitwise, hence it can't be converted implicitly from integer. Give an explicit cast for the invalid type. Spotted by sparse: sound/pci/hda/hda_codec.c:3280:25: warning: restricted snd_ctl_elem_iface_t degrades to integer Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6d0c0b143270..0a5085537034 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3277,8 +3277,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, for (; knew->name; knew++) { struct snd_kcontrol *kctl; int addr = 0, idx = 0; - if (knew->iface == -1) /* skip this codec private value */ - continue; + if (knew->iface == (__force snd_ctl_elem_iface_t)-1) + continue; /* skip this codec private value */ for (;;) { kctl = snd_ctl_new1(knew, codec); if (!kctl) From 7e9c20f40304a16d5f69dbdc44a551cc48252266 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:52 +0200 Subject: [PATCH 344/529] ALSA: opl3: Declare common variables properly Move the declarations of common variables into opl3_voice.h instead of declaring at each file multiple times, which was error-prone. This fixes sparse warnings like: sound/drivers/opl3/opl3_synth.c:51:6: warning: symbol 'snd_opl3_regmap' was not declared. Should it be static? Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_drums.c | 2 -- sound/drivers/opl3/opl3_lib.c | 3 +-- sound/drivers/opl3/opl3_midi.c | 4 ---- sound/drivers/opl3/opl3_oss.c | 2 -- sound/drivers/opl3/opl3_synth.c | 1 + sound/drivers/opl3/opl3_voice.h | 4 ++++ 6 files changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/drivers/opl3/opl3_drums.c b/sound/drivers/opl3/opl3_drums.c index 73694380734a..14929822956c 100644 --- a/sound/drivers/opl3/opl3_drums.c +++ b/sound/drivers/opl3/opl3_drums.c @@ -21,8 +21,6 @@ #include "opl3_voice.h" -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - static char snd_opl3_drum_table[47] = { OPL3_BASSDRUM_ON, OPL3_BASSDRUM_ON, OPL3_HIHAT_ON, /* 35 - 37 */ diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 588963d6be28..1a5355b747ec 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -31,13 +31,12 @@ #include #include #include +#include "opl3_voice.h" MODULE_AUTHOR("Jaroslav Kysela , Hannu Savolainen 1993-1996, Rob Hooft"); MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)"); MODULE_LICENSE("GPL"); -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - static void snd_opl2_command(struct snd_opl3 * opl3, unsigned short cmd, unsigned char val) { unsigned long flags; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 71cd5a2fbe82..471916ca0b6b 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -25,10 +25,6 @@ #include "opl3_voice.h" #include -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - -extern bool use_internal_drums; - static void snd_opl3_note_off_unsafe(void *p, int note, int vel, struct snd_midi_channel *chan); /* diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 8a0ce3f43f42..869220ced4ed 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -29,8 +29,6 @@ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg); /* operators */ -extern struct snd_midi_op opl3_ops; - static struct snd_seq_oss_callback oss_callback = { .owner = THIS_MODULE, .open = snd_opl3_open_seq_oss, diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 42920a243328..d522925fc5c0 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,6 +24,7 @@ #include #include #include +#include "opl3_voice.h" #if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index a2445163008e..5b02bd49fde4 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -52,4 +52,8 @@ void snd_opl3_free_seq_oss(struct snd_opl3 *opl3); #define snd_opl3_free_seq_oss(opl3) /* NOP */ #endif +extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; +extern bool use_internal_drums; +extern struct snd_midi_op opl3_ops; + #endif From 00966dcdf0e324e10ab79d505732c339f72417d9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:53 +0200 Subject: [PATCH 345/529] ALSA: usb-audio: Declare the common variable in header file Declare snd_usb_feature_unit_ctl properly in mixer.h. Otherwise it's error-prone. This fixes the sparse warning: sound/usb/mixer.c:1464:25: warning: symbol 'snd_usb_feature_unit_ctl' was not declared. Should it be static? Signed-off-by: Takashi Iwai --- sound/usb/mixer.h | 2 ++ sound/usb/mixer_quirks.c | 2 -- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index e02653465e29..3d12af8bf191 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -109,4 +109,6 @@ int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, extern void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl); +extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; + #endif /* __USBMIXER_H */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e82a72fea9a1..cbfb48bdea51 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -47,8 +47,6 @@ #include "mixer_us16x08.h" #include "helper.h" -extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; - struct std_mono_table { unsigned int unitid, control, cmask; int val_type; From 95a48b7d4459948b6bacf809809cf01a7dc06d1d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:54 +0200 Subject: [PATCH 346/529] ALSA: pcm: Add __force to cast in snd_pcm_lib_read/write() The snd_pcm_lib_read() and snd_pcm_lib_write() inline functions have the explicit cast from a user pointer to a kernel pointer, but they lacks of __force prefix. This fixes sparse warnings like: ./include/sound/pcm.h:1093:47: warning: cast removes address space of expression Fixes: 68541213720d ("ALSA: pcm: Direct in-kernel read/write support") Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1206045ccf03..d6bd3caf6878 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1090,14 +1090,14 @@ static inline snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const void __user *buf, snd_pcm_uframes_t frames) { - return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false); + return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false); } static inline snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream, void __user *buf, snd_pcm_uframes_t frames) { - return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false); + return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false); } static inline snd_pcm_sframes_t From 191bb51e72c3b2b1ed1b2bd7ac30dd9f7ccea306 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:55 +0200 Subject: [PATCH 347/529] ALSA: pcm: Use standard lower_32_bits() and upper_32_bits() Instead of open codes, use the standard macros for obtaining the lower and upper 32bit values. Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 85a56af104bd..85bab922ce69 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -281,10 +281,10 @@ static int snd_pcm_plug_formats(const struct snd_mask *mask, SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE); snd_mask_set(&formats, (__force int)SNDRV_PCM_FORMAT_MU_LAW); - if (formats.bits[0] & (u32)linfmts) - formats.bits[0] |= (u32)linfmts; - if (formats.bits[1] & (u32)(linfmts >> 32)) - formats.bits[1] |= (u32)(linfmts >> 32); + if (formats.bits[0] & lower_32_bits(linfmts)) + formats.bits[0] |= lower_32_bits(linfmts); + if (formats.bits[1] & upper_32_bits(linfmts)) + formats.bits[1] |= upper_32_bits(linfmts); return snd_mask_test(&formats, (__force int)format); } From 0701492c86ac0446a834da35ef4991bcf8adb6c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:56 +0200 Subject: [PATCH 348/529] ALSA: korg1212: Add __force annotation to cast in user-copy callbacks The user-copy callbacks in korg1212 driver contain the explicit cast from a user pointer to a kernel pointer, but they missed __force prefix. It's mandatory for converting between them. Spotted by sparse, a warning like: sound/pci/korg1212/korg1212.c:1329:33: warning: cast removes address space of expression Signed-off-by: Takashi Iwai --- sound/pci/korg1212/korg1212.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 4206ba44d8bb..4e189a93f475 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1326,7 +1326,7 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, } #endif if (in_kernel) - memcpy((void *)dst, src, size); + memcpy((__force void *)dst, src, size); else if (copy_to_user(dst, src, size)) return -EFAULT; src++; @@ -1365,7 +1365,7 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, } #endif if (in_kernel) - memcpy((void *)dst, src, size); + memcpy(dst, (__force void *)src, size); else if (copy_from_user(dst, src, size)) return -EFAULT; dst++; From 63623646a06358833766016d8c1fe14d6576b3d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:00:57 +0200 Subject: [PATCH 349/529] ALSA: emu10k1: Fix missing __force annotation for user/kernel pointer cast The cast between user-space and kernel-space needs an explicit __force prefix, but it's missing in many places in emu10k1 driver code. Spotted by sparse as a warning like: sound/pci/emu10k1/emufx.c:529:33: warning: cast removes address space of expression Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emufx.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index de2ecbe95d6c..90713741c2dc 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -526,7 +526,7 @@ static int snd_emu10k1_gpr_poke(struct snd_emu10k1 *emu, if (!test_bit(gpr, icode->gpr_valid)) continue; if (in_kernel) - val = *(u32 *)&icode->gpr_map[gpr]; + val = *(__force u32 *)&icode->gpr_map[gpr]; else if (get_user(val, &icode->gpr_map[gpr])) return -EFAULT; snd_emu10k1_ptr_write(emu, emu->gpr_base + gpr, 0, val); @@ -560,8 +560,8 @@ static int snd_emu10k1_tram_poke(struct snd_emu10k1 *emu, if (!test_bit(tram, icode->tram_valid)) continue; if (in_kernel) { - val = *(u32 *)&icode->tram_data_map[tram]; - addr = *(u32 *)&icode->tram_addr_map[tram]; + val = *(__force u32 *)&icode->tram_data_map[tram]; + addr = *(__force u32 *)&icode->tram_addr_map[tram]; } else { if (get_user(val, &icode->tram_data_map[tram]) || get_user(addr, &icode->tram_addr_map[tram])) @@ -611,8 +611,8 @@ static int snd_emu10k1_code_poke(struct snd_emu10k1 *emu, if (!test_bit(pc / 2, icode->code_valid)) continue; if (in_kernel) { - lo = *(u32 *)&icode->code[pc + 0]; - hi = *(u32 *)&icode->code[pc + 1]; + lo = *(__force u32 *)&icode->code[pc + 0]; + hi = *(__force u32 *)&icode->code[pc + 1]; } else { if (get_user(lo, &icode->code[pc + 0]) || get_user(hi, &icode->code[pc + 1])) @@ -666,7 +666,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel) if (!_tlv) return NULL; if (in_kernel) - memcpy(data, (void *)_tlv, sizeof(data)); + memcpy(data, (__force void *)_tlv, sizeof(data)); else if (copy_from_user(data, _tlv, sizeof(data))) return NULL; if (data[1] >= MAX_TLV_SIZE) @@ -676,7 +676,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel) return NULL; memcpy(tlv, data, sizeof(data)); if (in_kernel) { - memcpy(tlv + 2, (void *)(_tlv + 2), data[1]); + memcpy(tlv + 2, (__force void *)(_tlv + 2), data[1]); } else if (copy_from_user(tlv + 2, _tlv + 2, data[1])) { kfree(tlv); return NULL; @@ -693,7 +693,7 @@ static int copy_gctl(struct snd_emu10k1 *emu, if (emu->support_tlv) { if (in_kernel) - memcpy(gctl, (void *)&_gctl[idx], sizeof(*gctl)); + memcpy(gctl, (__force void *)&_gctl[idx], sizeof(*gctl)); else if (copy_from_user(gctl, &_gctl[idx], sizeof(*gctl))) return -EFAULT; return 0; @@ -701,7 +701,7 @@ static int copy_gctl(struct snd_emu10k1 *emu, octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl; if (in_kernel) - memcpy(gctl, (void *)&octl[idx], sizeof(*octl)); + memcpy(gctl, (__force void *)&octl[idx], sizeof(*octl)); else if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) return -EFAULT; gctl->tlv = NULL; @@ -735,7 +735,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (in_kernel) - id = *(struct snd_ctl_elem_id *)_id; + id = *(__force struct snd_ctl_elem_id *)_id; else if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; if (snd_emu10k1_look_for_ctl(emu, &id) == NULL) @@ -833,7 +833,7 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, knew.device = gctl->id.device; knew.subdevice = gctl->id.subdevice; knew.info = snd_emu10k1_gpr_ctl_info; - knew.tlv.p = copy_tlv(gctl->tlv, in_kernel); + knew.tlv.p = copy_tlv((__force const unsigned int __user *)gctl->tlv, in_kernel); if (knew.tlv.p) knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ; @@ -897,7 +897,7 @@ static int snd_emu10k1_del_controls(struct snd_emu10k1 *emu, for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (in_kernel) - id = *(struct snd_ctl_elem_id *)_id; + id = *(__force struct snd_ctl_elem_id *)_id; else if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); From d6b340d7cb33c816ef4abe8143764ec5ab14a5cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jul 2018 14:58:03 +0200 Subject: [PATCH 350/529] ALSA: trident: Suppress gcc string warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The meddlesome gcc warns about the possible shortname string in trident driver code: sound/pci/trident/trident.c: In function ‘snd_trident_probe’: sound/pci/trident/trident.c:126:2: warning: ‘strcat’ accessing 17 or more bytes at offsets 36 and 20 may overlap 1 byte at offset 36 [-Wrestrict] strcat(card->shortname, card->driver); It happens since gcc calculates the possible string size from card->driver, but this can't be true since we did set the string just before that, and they are much shorter. For shutting it up, use the exactly same string set to card->driver for strcat() to card->shortname, too. Signed-off-by: Takashi Iwai --- sound/pci/trident/trident.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index cedf13b64803..2f18b1cdc2cd 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -123,7 +123,7 @@ static int snd_trident_probe(struct pci_dev *pci, } else { strcpy(card->shortname, "Trident "); } - strcat(card->shortname, card->driver); + strcat(card->shortname, str); sprintf(card->longname, "%s PCI Audio at 0x%lx, irq %d", card->shortname, trident->port, trident->irq); From c183fec10ae6b3d1dfb77471d4158e576f6b43ae Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 25 Jul 2018 17:00:59 +0800 Subject: [PATCH 351/529] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 DA7219 for our platform need to be configured for 1.8V. Hence, we add a volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Adam Thomson Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4..58c1dcb4d255 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e5879e..cd3cf6e691a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include #include #include +#include +#include #include #include #include @@ -320,11 +322,52 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); From 7699676081ded2e149e798f9205f1f089aa3a6bc Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Thu, 26 Jul 2018 14:04:08 +0800 Subject: [PATCH 352/529] ASoC: AMD: Fix build warning Fixes sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ret); Reported-by: Stephen Rothwell Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cd3cf6e691a9..8e3275a96a82 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -365,7 +365,7 @@ static int cz_probe(struct platform_device *pdev) &acp_da7219_cfg); if (IS_ERR(rdev)) { dev_err(&pdev->dev, "Failed to register regulator: %d\n", - ret); + (int)PTR_ERR(rdev)); return -EINVAL; } From 036e4963bfb2d4513c30efd80e4bd50ff6c79e3e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 26 Jul 2018 14:45:42 +0200 Subject: [PATCH 353/529] ASoC: meson: use IRQ_RETVAL in the fifo irq handler Use IRQ_RETVAL instead of the open coded ternary operation. Suggested-by: Takashi Iwai Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index db367d85290f..30262550e37b 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -174,7 +174,7 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) /* Ack irqs */ axg_fifo_ack_irq(fifo, status); - return !status ? IRQ_NONE : IRQ_HANDLED; + return IRQ_RETVAL(status); } static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) From d9e81048127604fe6373fb76181fefcf218965f4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 26 Jul 2018 14:45:43 +0200 Subject: [PATCH 354/529] ASoC: meson: update axg sound card bindings Remove the amlogic prefix in front of the generic properties and change the card 'name' property to 'model' Suggested-by: Rob Herring Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,axg-sound-card.txt | 66 +++++++++---------- 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt index 39e005da0407..80b411296480 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt @@ -3,19 +3,19 @@ Amlogic AXG sound card: Required properties: - compatible: "amlogic,axg-sound-card" -- amlogic,name : User specified audio sound card name, one string +- model : User specified audio sound card name, one string Optional properties: -- amlogic,aux-devs : List of phandles pointing to auxiliary devices -- amlogic,widgets : Please refer to widgets.txt. -- amlogic,routing : A list of the connections between audio components. +- audio-aux-devs : List of phandles pointing to auxiliary devices +- audio-widgets : Please refer to widgets.txt. +- audio-routing : A list of the connections between audio components. Subnodes: -- amlogic,dai-link: Container for dai-link level properties and the - CODEC sub-nodes. There should be at least one (and - probably) subnode of this type. +- dai-link: Container for dai-link level properties and the CODEC + sub-nodes. There should be at least one (and probably more) + subnode of this type. Required dai-link properties: @@ -57,38 +57,38 @@ Example: sound { compatible = "amlogic,axg-sound-card"; - amlogic,name = "AXG-S420"; - amlogic,aux-devs = <&tdmin_a>, <&tdmout_c>; - amlogic,widgets = "Line", "Lineout", - "Line", "Linein", - "Speaker", "Speaker1 Left", - "Speaker", "Speaker1 Right"; - "Speaker", "Speaker2 Left", - "Speaker", "Speaker2 Right"; - amlogic,routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", - "SPDIFOUT IN 0", "FRDDR_A OUT 3", - "TDM_C Playback", "TDMOUT_C OUT", - "TDMIN_A IN 2", "TDM_C Capture", - "TDMIN_A IN 5", "TDM_C Loopback", - "TODDR_A IN 0", "TDMIN_A OUT", - "Lineout", "Lineout AOUTL", - "Lineout", "Lineout AOUTR", - "Speaker1 Left", "SPK1 OUT_A", - "Speaker2 Left", "SPK2 OUT_A", - "Speaker1 Right", "SPK1 OUT_B", - "Speaker2 Right", "SPK2 OUT_B", - "Linein AINL", "Linein", - "Linein AINR", "Linein"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right"; + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; - amlogic,dai-link@0 { + dai-link@0 { sound-dai = <&frddr_a>; }; - amlogic,dai-link@1 { + dai-link@1 { sound-dai = <&toddr_a>; }; - amlogic,dai-link@2 { + dai-link@2 { sound-dai = <&tdmif_c>; dai-format = "i2s"; dai-tdm-slot-tx-mask-2 = <1 1>; @@ -114,7 +114,7 @@ sound { }; - amlogic,dai-link@4 { + dai-link@3 { sound-dai = <&spdifout>; codec { From 435857e015dc7b337c5f21d195ac2e4ffd694283 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 26 Jul 2018 14:45:44 +0200 Subject: [PATCH 355/529] ASoC: meson: align axg card driver with DT bindings documentation Drop amlogic prefix in front of the generic DT properties and change property "name" to "model". Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index d6d1081d94ad..2914ba0d965b 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -123,7 +123,7 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card) struct snd_soc_aux_dev *aux; int num, i; - num = of_count_phandle_with_args(node, PREFIX "aux-devs", NULL); + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); if (num == -ENOENT) { /* * It is ok to have no auxiliary devices but for this card it @@ -144,8 +144,8 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card) card->num_aux_devs = num; for (i = 0; i < card->num_aux_devs; i++, aux++) { - aux->codec_of_node = of_parse_phandle(node, - PREFIX "aux-devs", i); + aux->codec_of_node = + of_parse_phandle(node, "audio-aux-devs", i); if (!aux->codec_of_node) return -EINVAL; } @@ -610,18 +610,18 @@ static int axg_card_probe(struct platform_device *pdev) priv->card.owner = THIS_MODULE; priv->card.dev = dev; - ret = snd_soc_of_parse_card_name(&priv->card, PREFIX "name"); + ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret < 0) return ret; - ret = axg_card_parse_of_optional(&priv->card, PREFIX "routing", + ret = axg_card_parse_of_optional(&priv->card, "audio-routing", snd_soc_of_parse_audio_routing); if (ret) { dev_err(dev, "error while parsing routing\n"); return ret; } - ret = axg_card_parse_of_optional(&priv->card, PREFIX "widgets", + ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", snd_soc_of_parse_audio_simple_widgets); if (ret) { dev_err(dev, "error while parsing widgets\n"); From 150a6dc8fc764c1912fa2a46e18d1505dfd3536f Mon Sep 17 00:00:00 2001 From: Hiroyuki Yokoyama Date: Thu, 26 Jul 2018 05:40:15 +0900 Subject: [PATCH 356/529] ASoC: rsnd: Document R-Car M3-N support Document support for the sound modules in the Renesas M3-N (r8a77965) SoC. No driver update is needed. Signed-off-by: Hiroyuki Yokoyama Signed-off-by: Yoshihiro Kaneko Acked-by: Kuninori Morimoto Reviewed-by: Simon Horman Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790f630f..9e764270c36b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 From c889a45d229938a94b50aadb819def8bb11a6a54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:40:49 +0200 Subject: [PATCH 357/529] ASoC: zte: Fix incorrect PCM format bit usages zx-tdm driver sets the DAI driver definitions with the format bits wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*. This patch corrects the definitions. Spotted by a sparse warning: sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer Fixes: 870e0ddc4345 ("ASoC: zx-tdm: add zte's tdm controller driver") Cc: Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/zte/zx-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { From 40d1299f87bf915931970c8e6ea3852acacd1889 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:42:08 +0200 Subject: [PATCH 358/529] ASoC: dmaengine: Fix missing __user prefix in copy_user callback It seems that __user prefix was forgotten to be added to dmaengine_copy_user callback while we refactored the user-copy PCM core. This patch adds the missing prefix, remove the superfluous cast, and add the needed cast (__force is needed for downgrading from user pointer to kernel pointer), too. Spotted by a sparse warning like: sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces)) Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 13bdca6e41c5..120f7b39e256 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -334,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -350,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; From 3ba66feb59810e2ce616da0c4f1a5230c74768a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:43:26 +0200 Subject: [PATCH 359/529] ASoC: dapm: Use int for format bit position fmt in snd_soc_dai_link_event() contains the format bit position, not the format bit itself. Hence it can be a simple integer instead of the explicit u64. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0602b2888d52..7e96793050c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3656,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || From 4cae99d9b5305ab8cccc839fccceb81ec9e5abda Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Jul 2018 15:15:56 -0500 Subject: [PATCH 360/529] ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally Make this helper inline function available for all platforms. This helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not defined. Reported-by: kbuild test robot Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/memalloc.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..c669900e6cbe 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -67,6 +67,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -90,14 +98,6 @@ struct snd_sg_buf { struct device *dev; }; -/* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - /* * return the physical address at the corresponding offset */ From 0b62834e73e332fea76a340d62aaf50c732b17e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:17 +0200 Subject: [PATCH 361/529] ALSA: pcm: Add snd_mask_set_format() helper for standard usages Many drivers calling snd_mask_set() need to do ugly cast with __force for shutting up the sparse warnings. Actually almost all of them are about setting the format, so it's far better to provide a common helper snd_mask_set_format() to pass SNDRV_PCM_FORMAT_* directly without the cast. There are a few other calls of snd_mask_set(), but they are in the PCM core code, so we leave them for now. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm_params.h | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index c704357775fc..2dd37cada7c0 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -87,6 +87,13 @@ static inline void snd_mask_set(struct snd_mask *mask, unsigned int val) mask->bits[MASK_OFS(val)] |= MASK_BIT(val); } +/* Most of drivers need only this one */ +static inline void snd_mask_set_format(struct snd_mask *mask, + snd_pcm_format_t format) +{ + snd_mask_set(mask, (__force unsigned int)format); +} + static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val) { mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val); @@ -369,8 +376,7 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) static inline void params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) { - snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), - (__force int)fmt); + snd_mask_set_format(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), fmt); } #endif /* __SOUND_PCM_PARAMS_H */ From 533a9274850b041b32fbe6d1df58a5c5b0b9e652 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:18 +0200 Subject: [PATCH 362/529] ASoC: doc: Replace open code with params_set_format() The example code in dpcm.rst contains an open code calling snd_mask_set(), and this can be better represented with params_set_format() instead. This automatically fixes the sparse warning about snd_pcm_format_t handling, too. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- Documentation/sound/soc/dpcm.rst | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index 395e5a516282..fe61e02277f8 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -254,9 +254,7 @@ configuration. channels->min = channels->max = 2; /* set DAI0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } From b5453e8ca311fdb6003c6583ad101d2b9131b994 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:19 +0200 Subject: [PATCH 363/529] ASoC: intel: Fix snd_pcm_format_t handling As sparse warns, the PCM format type can't be dealt as integer as found in Intel SST driver codes. Fix them in the following two ways: - The open code with snd_mask_set() and params->masks reference is replaced with params_set_format() - The rest codes with snd_mask_set(fmt, SNDRV_PCM_FORMAT_XXX) are replaced with the new helper, snd_mask_set_format(). Reported-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- 9 files changed, 11 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 6ea360f33575..efcfd906c856 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3aba5bcf806a..be6e4b40bf03 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index b68c289558a8..27308337ab12 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP5 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 7961f1fd18bd..38f6ab74709d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3a61252fe450..21a6490746a6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } /* * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 92f5fb2ae0a3..a892b37eab7c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) @@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ff6646cfa21..d31482b8c9bb 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index b0610bba3cfa..e877bb60beb1 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 38a1495c29cf..0e1818dd4cc6 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } From ebc22af0c9268dc4032326c20b02f2f227203330 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:20 +0200 Subject: [PATCH 364/529] ASoC: fsl: Use snd_mask_set_format() Use the new helper function snd_mask_set_format() for avoiding the ugly cast with __force prefix. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 07808c6d5461..44433b20435c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -195,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); - snd_mask_set(mask, (__force int)priv->asrc_format); + snd_mask_set_format(mask, priv->asrc_format); return 0; } From 79b8a50813a80ac15fdcdc96674098dee15eaaf5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:21 +0200 Subject: [PATCH 365/529] ASoC: pcm186x: Declare PCM format with snd_pcm_format_t The PCM format type is with __bitwise, so we should use the dedicated snd_pcm_format_t instead of int. This fixes the sparse warning like: sound/soc/codecs/pcm186x.c:268:44: warning: incorrect type in initializer (different base types) Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/pcm186x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 88fde70b1e9e..690c26e7389e 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component); unsigned int rate = params_rate(params); - unsigned int format = params_format(params); + snd_pcm_format_t format = params_format(params); unsigned int width = params_width(params); unsigned int channels = params_channels(params); unsigned int div_lrck; From 8adf3df4156345f1edcdfa8c7f7beeb0de351ce2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:22 +0200 Subject: [PATCH 366/529] ASoC: dmaengine: Use standard pcm_format_to_bits() macro The conversion from PCM format type to bits needs an explicit cast, and it'll be uglier. Since we have a standard macro for that, let's use it instead. This patch fixes the sparse warning: sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 120f7b39e256..52fd7af952a5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -188,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea case 32: case 64: if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); + hw.formats |= pcm_format_to_bits(i); break; default: /* Unsupported types */ From 50e9ffb1996a5d11ff5040a266585bad4ceeca0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jul 2018 14:27:59 +0200 Subject: [PATCH 367/529] ALSA: virmidi: Fix too long output trigger loop The virmidi output trigger tries to parse the all available bytes and process sequencer events as much as possible. In a normal situation, this is supposed to be relatively short, but a program may give a huge buffer and it'll take a long time in a single spin lock, which may eventually lead to a soft lockup. This patch simply adds a workaround, a cond_resched() call in the loop if applicable. A better solution would be to move the event processor into a work, but let's put a duct-tape quickly at first. Reported-and-tested-by: Dae R. Jeong Reported-by: syzbot+619d9f40141d826b097e@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_virmidi.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 289ae6bb81d9..8ebbca554e99 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -163,6 +163,7 @@ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int count, res; unsigned char buf[32], *pbuf; unsigned long flags; + bool check_resched = !in_atomic(); if (up) { vmidi->trigger = 1; @@ -200,6 +201,15 @@ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, vmidi->event.type = SNDRV_SEQ_EVENT_NONE; } } + if (!check_resched) + continue; + /* do temporary unlock & cond_resched() for avoiding + * CPU soft lockup, which may happen via a write from + * a huge rawmidi buffer + */ + spin_unlock_irqrestore(&substream->runtime->lock, flags); + cond_resched(); + spin_lock_irqsave(&substream->runtime->lock, flags); } out: spin_unlock_irqrestore(&substream->runtime->lock, flags); From a6ea5fe95ab4a1a7af6d57429fe3ecde9acf5b5a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:36 +0200 Subject: [PATCH 368/529] ALSA: hda: Fix implicit PCM format type conversion The PCM format type is defined with __bitwise, hence it can't be passed as integer but needs an explicit cast. In this patch, instead of the messy cast flood, define the format argument of snd_hdac_calc_stream_format() to be the proper snd_pcm_format_t type. This fixes sparse warnings like: sound/hda/hdac_device.c:760:38: warning: incorrect type in argument 1 (different base types) Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 3 ++- sound/hda/hdac_device.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 8305e7971035..6f1e1f3b3063 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -11,6 +11,7 @@ #include #include #include +#include #include #include #include @@ -133,7 +134,7 @@ int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *start_id); unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int channels, - unsigned int format, + snd_pcm_format_t format, unsigned int maxbps, unsigned short spdif_ctls); int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 7ba100bb1c3f..dbf02a3a8d2f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -738,7 +738,7 @@ static struct hda_rate_tbl rate_bits[] = { */ unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int channels, - unsigned int format, + snd_pcm_format_t format, unsigned int maxbps, unsigned short spdif_ctls) { From 94dfee0c1a33baa974cba0bd8b83021b1d801297 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:37 +0200 Subject: [PATCH 369/529] ALSA: riptide: Fix PCM format type conversion The PCM format type is with __bitwise, hence it needs to be explicitly declared as snd_pcm_format_t, as warned by sparse: sound/pci/riptide/riptide.c:1028:34: warning: incorrect type in argument 1 (different base types) sound/pci/riptide/riptide.c:1028:34: expected restricted snd_pcm_format_t [usertype] format sound/pci/riptide/riptide.c:1028:34: got unsigned char [unsigned] format Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 44f3b48d47b1..9d2b2ef15c6b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1017,7 +1017,7 @@ getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate) static int setsampleformat(struct cmdif *cif, unsigned char mixer, unsigned char id, - unsigned char channels, unsigned char format) + unsigned char channels, snd_pcm_format_t format) { unsigned char w, ch, sig, order; From f8b6c0cfbdd7359db9bb4da38dd54217296f9264 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:38 +0200 Subject: [PATCH 370/529] ALSA: pcm: Fix sparse warning wrt PCM format type The PCM format type is with __bitwise, hence it needs the explicit cast with __force. It's ugly, but there is a reason for that cost... This fixes the sparse warning: sound/core/oss/pcm_oss.c:1854:55: warning: incorrect type in argument 1 (different base types) Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 905a53c1cde5..f8d4a419f3af 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1851,7 +1851,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); for (fmt = 0; fmt < 32; ++fmt) { if (snd_mask_test(format_mask, fmt)) { - int f = snd_pcm_oss_format_to(fmt); + int f = snd_pcm_oss_format_to((__force snd_pcm_format_t)fmt); if (f >= 0) formats |= f; } From d63f33d3f083bdb3a7c2dfd623f4d811b2a8d772 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:39 +0200 Subject: [PATCH 371/529] ALSA: ad1816a: Fix sparse warning wrt PCM format type The PCM format type is with __bitwise, and it can't be converted from integer implicitly. Instead of an ugly cast, declare the function argument of snd_ad1816a_get_format() with the proper snd_pcm_format_t type. This fixes the sparse warning like: sound/isa/ad1816a/ad1816a_lib.c:93:14: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 923201414469..fba6d22f7f4b 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -85,7 +85,8 @@ static void snd_ad1816a_write_mask(struct snd_ad1816a *chip, unsigned char reg, static unsigned char snd_ad1816a_get_format(struct snd_ad1816a *chip, - unsigned int format, int channels) + snd_pcm_format_t format, + int channels) { unsigned char retval = AD1816A_FMT_LINEAR_8; From 10d3d91e3bc4e152a580bf523e4fd6bf279ae532 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:40 +0200 Subject: [PATCH 372/529] ALSA: au88x0: Fix sparse warning wrt PCM format type The PCM format type is with __bitwise, and it can't be converted from integer implicitly. Instead of an ugly cast, declare the function argument of vortex_alsafmt_aspfmt() with the proper snd_pcm_format_t type. This fixes the sparse warning like: sound/pci/au88x0/au88x0_core.c:2778:14: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.h | 2 +- sound/pci/au88x0/au88x0_core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index bcc648bf6478..e3e31f07d766 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -241,7 +241,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); +static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4083c8b01619..2e5b460a847c 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2770,7 +2770,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) +static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v) { int fmt; From a91a0e774984aa57090c39dc3269a812417737ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:41 +0200 Subject: [PATCH 373/529] ALSA: asihpi: Fix PCM format notations asihpi driver treats -1 as an own invalid PCM format, but this needs a proper cast with __force prefix since PCM format type is __bitwise. Define a constant with the proper type and use it allover. This fixes sparse warnings like: sound/pci/asihpi/asihpi.c:315:9: warning: incorrect type in initializer (different base types) Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 64e0961f93ba..a31fe1550903 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -311,27 +311,29 @@ static void print_hwparams(struct snd_pcm_substream *substream, snd_pcm_format_width(params_format(p)) / 8); } +#define INVALID_FORMAT (__force snd_pcm_format_t)(-1) + static snd_pcm_format_t hpi_to_alsa_formats[] = { - -1, /* INVALID */ + INVALID_FORMAT, /* INVALID */ SNDRV_PCM_FORMAT_U8, /* HPI_FORMAT_PCM8_UNSIGNED 1 */ SNDRV_PCM_FORMAT_S16, /* HPI_FORMAT_PCM16_SIGNED 2 */ - -1, /* HPI_FORMAT_MPEG_L1 3 */ + INVALID_FORMAT, /* HPI_FORMAT_MPEG_L1 3 */ SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L2 4 */ SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L3 5 */ - -1, /* HPI_FORMAT_DOLBY_AC2 6 */ - -1, /* HPI_FORMAT_DOLBY_AC3 7 */ + INVALID_FORMAT, /* HPI_FORMAT_DOLBY_AC2 6 */ + INVALID_FORMAT, /* HPI_FORMAT_DOLBY_AC3 7 */ SNDRV_PCM_FORMAT_S16_BE,/* HPI_FORMAT_PCM16_BIGENDIAN 8 */ - -1, /* HPI_FORMAT_AA_TAGIT1_HITS 9 */ - -1, /* HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_HITS 9 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ SNDRV_PCM_FORMAT_S32, /* HPI_FORMAT_PCM32_SIGNED 11 */ - -1, /* HPI_FORMAT_RAW_BITSTREAM 12 */ - -1, /* HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ + INVALID_FORMAT, /* HPI_FORMAT_RAW_BITSTREAM 12 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ SNDRV_PCM_FORMAT_FLOAT, /* HPI_FORMAT_PCM32_FLOAT 14 */ #if 1 /* ALSA can't handle 3 byte sample size together with power-of-2 * constraint on buffer_bytes, so disable this format */ - -1 + INVALID_FORMAT #else /* SNDRV_PCM_FORMAT_S24_3LE */ /* HPI_FORMAT_PCM24_SIGNED 15 */ #endif @@ -1023,7 +1025,7 @@ static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi, format, sample_rate, 128000, 0); if (!err) err = hpi_outstream_query_format(h_stream, &hpi_format); - if (!err && (hpi_to_alsa_formats[format] != -1)) + if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT)) formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; @@ -1205,7 +1207,7 @@ static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, format, sample_rate, 128000, 0); if (!err) err = hpi_instream_query_format(h_stream, &hpi_format); - if (!err && (hpi_to_alsa_formats[format] != -1)) + if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT)) formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; From 6be9a60efb401487a4d658ef23d652a9e6860b34 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:42 +0200 Subject: [PATCH 374/529] ALSA: wss: Fix sparse warning wrt PCM format type The PCM format type is with __bitwise, and it can't be converted from integer implicitly. Instead of an ugly cast, declare the function argument of snd_wss_get_format() with the proper snd_pcm_format_t type. This fixes the sparse warnings like: sound/isa/wss/wss_lib.c:551:14: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 8a852042a066..d23cc8abe1ef 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -541,7 +541,7 @@ static unsigned char snd_wss_get_rate(unsigned int rate) } static unsigned char snd_wss_get_format(struct snd_wss *chip, - int format, + snd_pcm_format_t format, int channels) { unsigned char rformat; From 55ff2d1ea5487fe131cce399baf4503dcf5cc8e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:43 +0200 Subject: [PATCH 375/529] ALSA: sb: Fix PCM format bit calculation The PCM format type in snd_pcm_format_t can't be treated as integer implicitly since it's with __bitwise. We have already a helper function to get the bit index of the given type, and use it in each place instead. This fixes sparse warnings like: sound/isa/sb/sb16_main.c:61:44: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- sound/isa/sb/sb16_main.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 3e39ba220c39..11ed4a6e5bf1 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -49,6 +49,9 @@ MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); +#define runtime_format_bits(runtime) \ + ((unsigned int)pcm_format_to_bits((runtime)->format)) + #ifdef CONFIG_SND_SB16_CSP static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_runtime *runtime) { @@ -58,7 +61,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru if (csp->running & SNDRV_SB_CSP_ST_LOADED) { /* manually loaded codec */ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) && - ((1U << runtime->format) == csp->acc_format)) { + (runtime_format_bits(runtime) == csp->acc_format)) { /* Supported runtime PCM format for playback */ if (csp->ops.csp_use(csp) == 0) { /* If CSP was successfully acquired */ @@ -66,7 +69,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru } } else if ((csp->mode & SNDRV_SB_CSP_MODE_QSOUND) && (csp->q_enabled)) { /* QSound decoder is loaded and enabled */ - if ((1 << runtime->format) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + if (runtime_format_bits(runtime) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE)) { /* Only for simple PCM formats */ if (csp->ops.csp_use(csp) == 0) { @@ -106,7 +109,7 @@ static void snd_sb16_csp_capture_prepare(struct snd_sb *chip, struct snd_pcm_run if (csp->running & SNDRV_SB_CSP_ST_LOADED) { /* manually loaded codec */ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_READ) && - ((1U << runtime->format) == csp->acc_format)) { + (runtime_format_bits(runtime) == csp->acc_format)) { /* Supported runtime PCM format for capture */ if (csp->ops.csp_use(csp) == 0) { /* If CSP was successfully acquired */ From e5d3765b6c4cb3ba64295a4205a2f68a4e8fe083 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:44 +0200 Subject: [PATCH 376/529] ALSA: sb: Fix sparse warning wrt PCM format type The PCM format type is with __bitwise, and it can't be converted from integer implicitly. Instead of an ugly cast, declare the function argument of snd_sb_csp_autoload() with the proper snd_pcm_format_t type. This fixes the sparse warnings like: sound/isa/sb/sb16_csp.c:743:22: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- include/sound/sb16_csp.h | 2 +- sound/isa/sb/sb16_csp.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h index c7c7788005e4..7817e88bd08d 100644 --- a/include/sound/sb16_csp.h +++ b/include/sound/sb16_csp.h @@ -46,7 +46,7 @@ enum { struct snd_sb_csp_ops { int (*csp_use) (struct snd_sb_csp * p); int (*csp_unuse) (struct snd_sb_csp * p); - int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode); + int (*csp_autoload) (struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode); int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels); int (*csp_stop) (struct snd_sb_csp * p); int (*csp_qsound_transfer) (struct snd_sb_csp * p); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index fa5780bb0c68..2210e7c72787 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -93,7 +93,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, struct snd_sb_csp_microcode __user * code); static int snd_sb_csp_unload(struct snd_sb_csp * p); static int snd_sb_csp_load_user(struct snd_sb_csp * p, const unsigned char __user *buf, int size, int load_flags); -static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode); +static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode); static int snd_sb_csp_check_version(struct snd_sb_csp * p); static int snd_sb_csp_use(struct snd_sb_csp * p); @@ -726,7 +726,7 @@ static int snd_sb_csp_firmware_load(struct snd_sb_csp *p, int index, int flags) * autoload hardware codec if necessary * return 0 if CSP is loaded and ready to run (p->running != 0) */ -static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode) +static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode) { unsigned long flags; int err = 0; @@ -736,7 +736,7 @@ static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec return -EBUSY; /* autoload microcode only if requested hardware codec is not already loaded */ - if (((1 << pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) { + if (((1U << (__force int)pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) { p->running = SNDRV_SB_CSP_ST_AUTO; } else { switch (pcm_sfmt) { From 3ac14b3960185d4c8a2f14b84042aa1aa8531d88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:19:45 +0200 Subject: [PATCH 377/529] ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits The open codes with the bit shift in xen_snd_front_alsa.c give sparse warnings as the PCM format type is with __bitwise. There is already a standard macro to get the format bits, so let's use it instead. This fixes sparse warnings like: sound/xen/xen_snd_front_alsa.c:191:47: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai --- sound/xen/xen_snd_front_alsa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 5a2bd70a2fa1..129180e17db1 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -188,7 +188,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) - if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) + if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats) mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; return mask; @@ -202,7 +202,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) - mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; + mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa); return mask; } From d3c637632da95d7646053c64b855641cd917960e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:01 +0200 Subject: [PATCH 378/529] ALSA: ymfpci: Proper endian notations The bank values are all little-endians, so they should be defined with __le32. This fixes lots of sparse warnings like: sound/pci/ymfpci/ymfpci_main.c:315:23: warning: cast to restricted __le32 sound/pci/ymfpci/ymfpci_main.c:342:32: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci.h | 78 +++++++++++++++++----------------- sound/pci/ymfpci/ymfpci_main.c | 6 +-- 2 files changed, 42 insertions(+), 42 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h index aa9bb065f385..e2fa7e360d79 100644 --- a/sound/pci/ymfpci/ymfpci.h +++ b/sound/pci/ymfpci/ymfpci.h @@ -185,50 +185,50 @@ */ struct snd_ymfpci_playback_bank { - u32 format; - u32 loop_default; - u32 base; /* 32-bit address */ - u32 loop_start; /* 32-bit offset */ - u32 loop_end; /* 32-bit offset */ - u32 loop_frac; /* 8-bit fraction - loop_start */ - u32 delta_end; /* pitch delta end */ - u32 lpfK_end; - u32 eg_gain_end; - u32 left_gain_end; - u32 right_gain_end; - u32 eff1_gain_end; - u32 eff2_gain_end; - u32 eff3_gain_end; - u32 lpfQ; - u32 status; - u32 num_of_frames; - u32 loop_count; - u32 start; - u32 start_frac; - u32 delta; - u32 lpfK; - u32 eg_gain; - u32 left_gain; - u32 right_gain; - u32 eff1_gain; - u32 eff2_gain; - u32 eff3_gain; - u32 lpfD1; - u32 lpfD2; + __le32 format; + __le32 loop_default; + __le32 base; /* 32-bit address */ + __le32 loop_start; /* 32-bit offset */ + __le32 loop_end; /* 32-bit offset */ + __le32 loop_frac; /* 8-bit fraction - loop_start */ + __le32 delta_end; /* pitch delta end */ + __le32 lpfK_end; + __le32 eg_gain_end; + __le32 left_gain_end; + __le32 right_gain_end; + __le32 eff1_gain_end; + __le32 eff2_gain_end; + __le32 eff3_gain_end; + __le32 lpfQ; + __le32 status; + __le32 num_of_frames; + __le32 loop_count; + __le32 start; + __le32 start_frac; + __le32 delta; + __le32 lpfK; + __le32 eg_gain; + __le32 left_gain; + __le32 right_gain; + __le32 eff1_gain; + __le32 eff2_gain; + __le32 eff3_gain; + __le32 lpfD1; + __le32 lpfD2; }; struct snd_ymfpci_capture_bank { - u32 base; /* 32-bit address */ - u32 loop_end; /* 32-bit offset */ - u32 start; /* 32-bit offset */ - u32 num_of_loops; /* counter */ + __le32 base; /* 32-bit address */ + __le32 loop_end; /* 32-bit offset */ + __le32 start; /* 32-bit offset */ + __le32 num_of_loops; /* counter */ }; struct snd_ymfpci_effect_bank { - u32 base; /* 32-bit address */ - u32 loop_end; /* 32-bit offset */ - u32 start; /* 32-bit offset */ - u32 temp; + __le32 base; /* 32-bit address */ + __le32 loop_end; /* 32-bit offset */ + __le32 start; /* 32-bit offset */ + __le32 temp; }; struct snd_ymfpci_pcm; @@ -316,7 +316,7 @@ struct snd_ymfpci { dma_addr_t work_base_addr; struct snd_dma_buffer ac3_tmp_base; - u32 *ctrl_playback; + __le32 *ctrl_playback; struct snd_ymfpci_playback_bank *bank_playback[YDSXG_PLAYBACK_VOICES][2]; struct snd_ymfpci_capture_bank *bank_capture[YDSXG_CAPTURE_VOICES][2]; struct snd_ymfpci_effect_bank *bank_effect[YDSXG_EFFECT_VOICES][2]; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 6f81396aadc9..a4926fb03991 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -336,7 +336,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ unsigned int subs = ypcm->substream->number; unsigned int next_bank = 1 - chip->active_bank; struct snd_ymfpci_playback_bank *bank; - u32 volume; + __le32 volume; bank = &voice->bank[next_bank]; volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15); @@ -505,7 +505,7 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int u32 lpfK = snd_ymfpci_calc_lpfK(runtime->rate); struct snd_ymfpci_playback_bank *bank; unsigned int nbank; - u32 vol_left, vol_right; + __le32 vol_left, vol_right; u8 use_left, use_right; unsigned long flags; @@ -2135,7 +2135,7 @@ static int snd_ymfpci_memalloc(struct snd_ymfpci *chip) chip->bank_base_playback = ptr; chip->bank_base_playback_addr = ptr_addr; - chip->ctrl_playback = (u32 *)ptr; + chip->ctrl_playback = (__le32 *)ptr; chip->ctrl_playback[0] = cpu_to_le32(YDSXG_PLAYBACK_VOICES); ptr += ALIGN(playback_ctrl_size, 0x100); ptr_addr += ALIGN(playback_ctrl_size, 0x100); From 752089fea357b36ba9dd477db594bdc677110579 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:02 +0200 Subject: [PATCH 379/529] ALSA: trident: Proper endian notations The TLB entries in Trident driver are represented in little-endian, hence they should be declared as __le32. This patch fixes the sparse warnings like: sound/pci/trident/trident_memory.c:226:17: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/trident/trident.h | 2 +- sound/pci/trident/trident_main.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h index 9624e5937719..2d62c1921255 100644 --- a/sound/pci/trident/trident.h +++ b/sound/pci/trident/trident.h @@ -264,7 +264,7 @@ struct snd_trident_memblk_arg { }; struct snd_trident_tlb { - unsigned int * entries; /* 16k-aligned TLB table */ + __le32 *entries; /* 16k-aligned TLB table */ dma_addr_t entries_dmaaddr; /* 16k-aligned PCI address to TLB table */ unsigned long * shadow_entries; /* shadow entries with virtual addresses */ struct snd_dma_buffer buffer; diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 49c64fae3466..5523e193d556 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3359,7 +3359,7 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) dev_err(trident->card->dev, "unable to allocate TLB buffer\n"); return -ENOMEM; } - trident->tlb.entries = (unsigned int*)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4); + trident->tlb.entries = (__le32 *)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4); trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4); /* allocate shadow TLB page table (virtual addresses) */ trident->tlb.shadow_entries = From 7362b0fca5de9570b5ba7bce5034105028daa599 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:03 +0200 Subject: [PATCH 380/529] ALSA: hda: Proper endian notations for BDL pointers The BDL pointer used in snd_hdac_dsp_prepare() should be declared as __le32, as warned by sparse: sound/hda/hdac_stream.c:655:47: warning: incorrect type in argument 4 (different base types) sound/hda/hdac_stream.c:655:47: expected restricted __le32 [usertype] **bdlp sound/hda/hdac_stream.c:655:47: got unsigned int [usertype] ** Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index e1472c7ab6c1..eee422390d8e 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -621,7 +621,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, unsigned int byte_size, struct snd_dma_buffer *bufp) { struct hdac_bus *bus = azx_dev->bus; - u32 *bdl; + __le32 *bdl; int err; snd_hdac_dsp_lock(azx_dev); @@ -651,7 +651,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); azx_dev->frags = 0; - bdl = (u32 *)azx_dev->bdl.area; + bdl = (__le32 *)azx_dev->bdl.area; err = setup_bdle(bus, bufp, azx_dev, &bdl, 0, byte_size, 0); if (err < 0) goto error; From be05e3de3a933156d472127f659d4473c461dcc5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:04 +0200 Subject: [PATCH 381/529] ALSA: riptide: Properly endian notations The SG descriptor of Riptide contains the little-endian values, hence we need to define with __le32 properly. This fixes sparse warnings like: sound/pci/riptide/riptide.c:1112:40: warning: cast to restricted __le32 Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 9d2b2ef15c6b..23017e3bc76c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -470,10 +470,10 @@ struct snd_riptide { }; struct sgd { /* scatter gather desriptor */ - u32 dwNextLink; - u32 dwSegPtrPhys; - u32 dwSegLen; - u32 dwStat_Ctl; + __le32 dwNextLink; + __le32 dwSegPtrPhys; + __le32 dwSegLen; + __le32 dwStat_Ctl; }; struct pcmhw { /* pcm descriptor */ From 0e7ca66a97c3bbf0b7665010d1b1d6a8c8e3811e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:05 +0200 Subject: [PATCH 382/529] ALSA: mixart: Proper endian notations The miXart driver deals with big-endian values as raw data, while it declares most of variables as u32. This leads to sparse warnings like sound/pci/mixart/mixart.c:1203:23: warning: cast to restricted __be32 Fix them by properly defining the structs and add the explicit cast to macros. Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart_core.c | 4 ++-- sound/pci/mixart/mixart_hwdep.c | 42 ++++++++++++++++----------------- sound/pci/mixart/mixart_hwdep.h | 8 +++---- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 8bf2ce32d4a8..46c292b52fd6 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -107,7 +107,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, #ifndef __BIG_ENDIAN size /= 4; /* u32 size */ for(i=0; i < size; i++) { - ((u32*)resp->data)[i] = be32_to_cpu(((u32*)resp->data)[i]); + ((u32*)resp->data)[i] = be32_to_cpu(((__be32*)resp->data)[i]); } #endif @@ -519,7 +519,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) /* Traces are text: the swapped msg_data has to be swapped back ! */ int i; for(i=0; i<(resp.size/4); i++) { - (mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]); + ((__be32*)mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]); } #endif ((char*)mixart_msg_data)[resp.size - 1] = 0; diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 5bfd3ac80db5..bc92758de82c 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -73,30 +73,30 @@ static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr, */ struct snd_mixart_elf32_ehdr { u8 e_ident[16]; - u16 e_type; - u16 e_machine; - u32 e_version; - u32 e_entry; - u32 e_phoff; - u32 e_shoff; - u32 e_flags; - u16 e_ehsize; - u16 e_phentsize; - u16 e_phnum; - u16 e_shentsize; - u16 e_shnum; - u16 e_shstrndx; + __be16 e_type; + __be16 e_machine; + __be32 e_version; + __be32 e_entry; + __be32 e_phoff; + __be32 e_shoff; + __be32 e_flags; + __be16 e_ehsize; + __be16 e_phentsize; + __be16 e_phnum; + __be16 e_shentsize; + __be16 e_shnum; + __be16 e_shstrndx; }; struct snd_mixart_elf32_phdr { - u32 p_type; - u32 p_offset; - u32 p_vaddr; - u32 p_paddr; - u32 p_filesz; - u32 p_memsz; - u32 p_flags; - u32 p_align; + __be32 p_type; + __be32 p_offset; + __be32 p_vaddr; + __be32 p_paddr; + __be32 p_filesz; + __be32 p_memsz; + __be32 p_flags; + __be32 p_align; }; static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp ) diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h index 812e288ef2e7..2794cd385b8e 100644 --- a/sound/pci/mixart/mixart_hwdep.h +++ b/sound/pci/mixart/mixart_hwdep.h @@ -26,19 +26,19 @@ #include #ifndef readl_be -#define readl_be(x) be32_to_cpu(__raw_readl(x)) +#define readl_be(x) be32_to_cpu((__force __be32)__raw_readl(x)) #endif #ifndef writel_be -#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr) +#define writel_be(data,addr) __raw_writel((__force u32)cpu_to_be32(data),addr) #endif #ifndef readl_le -#define readl_le(x) le32_to_cpu(__raw_readl(x)) +#define readl_le(x) le32_to_cpu((__force __le32)__raw_readl(x)) #endif #ifndef writel_le -#define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr) +#define writel_le(data,addr) __raw_writel((__force u32)cpu_to_le32(data),addr) #endif #define MIXART_MEM(mgr,x) ((mgr)->mem[0].virt + (x)) From 0d9a26fc74578289e0edeac36c6d562596f8a72e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:06 +0200 Subject: [PATCH 383/529] ALSA: lola: Proper endian notations The BDL entries in lola driver are little-endian while we code them as u32. This leads to sparse warnings like: sound/pci/lola/lola.c:105:40: warning: incorrect type in assignment (different base types) sound/pci/lola/lola.c:105:40: expected unsigned int [unsigned] [usertype] sound/pci/lola/lola.c:105:40: got restricted __le32 [usertype] This patch fixes the declarations to the proper __le32 type. Also, there was a typo in the original code, where __user was used that was intended as __iomem. This was caused also by sparse: sound/pci/lola/lola_mixer.c:132:27: warning: incorrect type in assignment (different address spaces) Fixed in this patch as well. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 4 ++-- sound/pci/lola/lola.h | 4 ++-- sound/pci/lola/lola_pcm.c | 8 ++++---- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 9ff600084973..254f24366892 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -369,9 +369,9 @@ static int setup_corb_rirb(struct lola *chip) return err; chip->corb.addr = chip->rb.addr; - chip->corb.buf = (u32 *)chip->rb.area; + chip->corb.buf = (__le32 *)chip->rb.area; chip->rirb.addr = chip->rb.addr + 2048; - chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + chip->rirb.buf = (__le32 *)(chip->rb.area + 2048); /* disable ringbuffer DMAs */ lola_writeb(chip, BAR0, RIRBCTL, 0); diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index f0b100059efd..bd852fed8bb6 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -220,7 +220,7 @@ struct lola_bar { /* CORB/RIRB */ struct lola_rb { - u32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ + __le32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ dma_addr_t addr; /* physical address of CORB/RIRB buffer */ unsigned short rp, wp; /* read/write pointers */ int cmds; /* number of pending requests */ @@ -275,7 +275,7 @@ struct lola_mixer_array { struct lola_mixer_widget { unsigned int nid; unsigned int caps; - struct lola_mixer_array __user *array; + struct lola_mixer_array __iomem *array; struct lola_mixer_array *array_saved; unsigned int src_stream_outs; unsigned int src_phys_ins; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 310b26a756c9..e70276c3ea20 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -316,10 +316,10 @@ static int lola_pcm_hw_free(struct snd_pcm_substream *substream) * set up a BDL entry */ static int setup_bdle(struct snd_pcm_substream *substream, - struct lola_stream *str, u32 **bdlp, + struct lola_stream *str, __le32 **bdlp, int ofs, int size) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; while (size > 0) { dma_addr_t addr; @@ -355,14 +355,14 @@ static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm, struct snd_pcm_substream *substream, struct lola_stream *str) { - u32 *bdl; + __le32 *bdl; int i, ofs, periods, period_bytes; period_bytes = str->period_bytes; periods = str->bufsize / period_bytes; /* program the initial BDL entries */ - bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); + bdl = (__le32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); ofs = 0; str->frags = 0; for (i = 0; i < periods; i++) { From 3c164e2ce601bd7abf76d9f37b7f4afa6fa988a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:07 +0200 Subject: [PATCH 384/529] ALSA: intel8x0: Proper endian notations The BD address tables in intel8x0 driver are in little-endian, hence they should be represented as __le32 instead u32. Spotted by sparse, warnings like: sound/pci/intel8x0.c:688:40: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 4c24346340f4..5ee468d1aefe 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -351,7 +351,7 @@ enum { struct ichdev { unsigned int ichd; /* ich device number */ unsigned long reg_offset; /* offset to bmaddr */ - u32 *bdbar; /* CPU address (32bit) */ + __le32 *bdbar; /* CPU address (32bit) */ unsigned int bdbar_addr; /* PCI bus address (32bit) */ struct snd_pcm_substream *substream; unsigned int physbuf; /* physical address (32bit) */ @@ -677,7 +677,7 @@ static void snd_intel8x0_ali_codec_write(struct snd_ac97 *ac97, unsigned short r static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev) { int idx; - u32 *bdbar = ichdev->bdbar; + __le32 *bdbar = ichdev->bdbar; unsigned long port = ichdev->reg_offset; iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr); @@ -3143,7 +3143,7 @@ static int snd_intel8x0_create(struct snd_card *card, int_sta_masks = 0; for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; - ichdev->bdbar = ((u32 *)chip->bdbars.area) + + ichdev->bdbar = ((__le32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2); From 7752a7de2596097c8f702fd652cba05118f83bcc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:08 +0200 Subject: [PATCH 385/529] ALSA: intel8x0m: Proper endian notations The BD address tables in intel8x0m driver are in little-endian, hence they should be represented as __le32 instead u32. Spotted by sparse, warnings like: sound/pci/intel8x0m.c:406:40: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/intel8x0m.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3a4769a97d29..943a726b1c1b 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -168,7 +168,7 @@ enum { ALID_MDMIN, ALID_MDMOUT, ALID_MDMLAST = ALID_MDMOUT }; struct ichdev { unsigned int ichd; /* ich device number */ unsigned long reg_offset; /* offset to bmaddr */ - u32 *bdbar; /* CPU address (32bit) */ + __le32 *bdbar; /* CPU address (32bit) */ unsigned int bdbar_addr; /* PCI bus address (32bit) */ struct snd_pcm_substream *substream; unsigned int physbuf; /* physical address (32bit) */ @@ -395,7 +395,7 @@ static unsigned short snd_intel8x0m_codec_read(struct snd_ac97 *ac97, static void snd_intel8x0m_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) { int idx; - u32 *bdbar = ichdev->bdbar; + __le32 *bdbar = ichdev->bdbar; unsigned long port = ichdev->reg_offset; iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr); @@ -1217,7 +1217,7 @@ static int snd_intel8x0m_create(struct snd_card *card, int_sta_masks = 0; for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; - ichdev->bdbar = ((u32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); + ichdev->bdbar = ((__le32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2); int_sta_masks |= ichdev->int_sta_mask; } From 8c0ab942e05941ade5f1fca59d29b7034fdf164c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:09 +0200 Subject: [PATCH 386/529] ALSA: maestro3: Proper endian notations The ASSP data passed to maestro3 driver is in little-endian format, hence the data pointer should be with __le16. Spotted by sparse, warnings like: sound/pci/maestro3.c:2128:35: warning: cast to restricted __le16 Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 224e942f556d..62962178a9d7 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2103,7 +2103,7 @@ static const u16 minisrc_lpf[MINISRC_LPF_LEN] = { static void snd_m3_assp_init(struct snd_m3 *chip) { unsigned int i; - const u16 *data; + const __le16 *data; /* zero kernel data */ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++) @@ -2121,7 +2121,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) KDATA_DMA_XFER0); /* write kernel into code memory.. */ - data = (const u16 *)chip->assp_kernel_image->data; + data = (const __le16 *)chip->assp_kernel_image->data; for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, REV_B_CODE_MEMORY_BEGIN + i, @@ -2134,7 +2134,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) * drop it there. It seems that the minisrc doesn't * need vectors, so we won't bother with them.. */ - data = (const u16 *)chip->assp_minisrc_image->data; + data = (const __le16 *)chip->assp_minisrc_image->data; for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 0x400 + i, le16_to_cpu(data[i])); From 2a833a02a12b1dbb605739d589d11b4400c2078c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:10 +0200 Subject: [PATCH 387/529] ALSA: echoaudio: Proper endian notations Many data fields defined in echoaudio drivers are in little-endian, hence they should be defined with __le16 or __le32. This makes it easier to catch the forgotten conversions. Spotted by sparse, a warning like: sound/pci/echoaudio/echoaudio_dsp.c:990:36: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.h | 2 +- sound/pci/echoaudio/echoaudio_3g.c | 14 ++++---- sound/pci/echoaudio/echoaudio_dsp.c | 6 ++-- sound/pci/echoaudio/echoaudio_dsp.h | 50 ++++++++++++++--------------- sound/pci/echoaudio/echoaudio_gml.c | 8 +++-- 5 files changed, 42 insertions(+), 38 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 44b390a667d5..be4d0489394a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -294,7 +294,7 @@ struct audiopipe { - volatile u32 *dma_counter; /* Commpage register that contains + volatile __le32 *dma_counter; /* Commpage register that contains * the current dma position * (lower 32 bits only) */ diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 22c786b8a889..cc3c79387194 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -73,19 +73,21 @@ register. write_control_reg sends the new control register value to the DSP. */ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq, char force) { + __le32 ctl_reg, frq_reg; + if (wait_handshake(chip)) return -EIO; dev_dbg(chip->card->dev, "WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq); - ctl = cpu_to_le32(ctl); - frq = cpu_to_le32(frq); + ctl_reg = cpu_to_le32(ctl); + frq_reg = cpu_to_le32(frq); - if (ctl != chip->comm_page->control_register || - frq != chip->comm_page->e3g_frq_register || force) { - chip->comm_page->e3g_frq_register = frq; - chip->comm_page->control_register = ctl; + if (ctl_reg != chip->comm_page->control_register || + frq_reg != chip->comm_page->e3g_frq_register || force) { + chip->comm_page->e3g_frq_register = frq_reg; + chip->comm_page->control_register = ctl_reg; clear_handshake(chip); return send_vector(chip, DSP_VC_WRITE_CONTROL_REG); } diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 15aae2fad8e4..b181752b8481 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -679,7 +679,7 @@ static int restore_dsp_rettings(struct echoaudio *chip) /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; - chip->comm_page->handshake = 0xffffffff; + chip->comm_page->handshake = cpu_to_le32(0xffffffff); /* Restore output busses */ for (i = 0; i < num_busses_out(chip); i++) { @@ -989,7 +989,7 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init the comm page */ chip->comm_page->comm_size = cpu_to_le32(sizeof(struct comm_page)); - chip->comm_page->handshake = 0xffffffff; + chip->comm_page->handshake = cpu_to_le32(0xffffffff); chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); @@ -1087,7 +1087,7 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, /* The counter register is where the DSP writes the 32 bit DMA position for a pipe. The DSP is constantly updating this value as it moves data. The DMA counter is in units of bytes, not samples. */ - pipe->dma_counter = &chip->comm_page->position[pipe_index]; + pipe->dma_counter = (__le32 *)&chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; return pipe_index; } diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index cb7d75a0a503..aa9129519795 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -627,8 +627,8 @@ sg_entry struct is read by the DSP, so all values must be little-endian. */ #define MAX_SGLIST_ENTRIES 512 struct sg_entry { - u32 addr; - u32 size; + __le32 addr; + __le32 size; }; @@ -643,18 +643,18 @@ struct sg_entry { ****************************************************************************/ struct comm_page { /* Base Length*/ - u32 comm_size; /* size of this object 0x000 4 */ - u32 flags; /* See Appendix A below 0x004 4 */ - u32 unused; /* Unused entry 0x008 4 */ - u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */ - u32 handshake; /* DSP command handshake 0x010 4 */ - u32 cmd_start; /* Chs. to start mask 0x014 4 */ - u32 cmd_stop; /* Chs. to stop mask 0x018 4 */ - u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */ - u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */ + __le32 comm_size; /* size of this object 0x000 4 */ + __le32 flags; /* See Appendix A below 0x004 4 */ + __le32 unused; /* Unused entry 0x008 4 */ + __le32 sample_rate; /* Card sample rate in Hz 0x00c 4 */ + __le32 handshake; /* DSP command handshake 0x010 4 */ + __le32 cmd_start; /* Chs. to start mask 0x014 4 */ + __le32 cmd_stop; /* Chs. to stop mask 0x018 4 */ + __le32 cmd_reset; /* Chs. to reset mask 0x01c 4 */ + __le16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */ struct sg_entry sglist_addr[DSP_MAXPIPES]; /* Chs. Physical sglist addrs 0x060 32*8 */ - u32 position[DSP_MAXPIPES]; + __le32 position[DSP_MAXPIPES]; /* Positions for ea. ch. 0x160 32*4 */ s8 vu_meter[DSP_MAXPIPES]; /* VU meters 0x1e0 32*1 */ @@ -666,28 +666,28 @@ struct comm_page { /* Base Length*/ /* Input gain 0x230 16*1 */ s8 monitors[MONITOR_ARRAY_SIZE]; /* Monitor map 0x240 0x180 */ - u32 play_coeff[MAX_PLAY_TAPS]; + __le32 play_coeff[MAX_PLAY_TAPS]; /* Gina/Darla play filters - obsolete 0x3c0 168*4 */ - u32 rec_coeff[MAX_REC_TAPS]; + __le32 rec_coeff[MAX_REC_TAPS]; /* Gina/Darla record filters - obsolete 0x660 192*4 */ - u16 midi_input[MIDI_IN_BUFFER_SIZE]; + __le16 midi_input[MIDI_IN_BUFFER_SIZE]; /* MIDI input data transfer buffer 0x960 256*2 */ u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */ u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */ u8 gd_resampler_state; /* Should always be 3 0xb62 1 */ u8 filler2; /* 0xb63 1 */ - u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */ - u16 input_clock; /* Chg. Input clock state 0xb68 2 */ - u16 output_clock; /* Chg. Output clock state 0xb6a 2 */ - u32 status_clocks; /* Current Input clock state 0xb6c 4 */ - u32 ext_box_status; /* External box status 0xb70 4 */ - u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */ - u32 midi_out_free_count; + __le32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */ + __le16 input_clock; /* Chg. Input clock state 0xb68 2 */ + __le16 output_clock; /* Chg. Output clock state 0xb6a 2 */ + __le32 status_clocks; /* Current Input clock state 0xb6c 4 */ + __le32 ext_box_status; /* External box status 0xb70 4 */ + __le32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */ + __le32 midi_out_free_count; /* # of bytes free in MIDI output FIFO 0xb78 4 */ - u32 unused2; /* Cyclic pipes 0xb7c 4 */ - u32 control_register; + __le32 unused2; /* Cyclic pipes 0xb7c 4 */ + __le32 control_register; /* Mona, Gina24, Layla24, 3G ctrl reg 0xb80 4 */ - u32 e3g_frq_register; /* 3G frequency register 0xb84 4 */ + __le32 e3g_frq_register; /* 3G frequency register 0xb84 4 */ u8 filler[24]; /* filler 0xb88 24*1 */ s8 vmixer[VMIXER_ARRAY_SIZE]; /* Vmixer levels 0xba0 64*1 */ diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c index 834b39e97db7..eea6fe530ab4 100644 --- a/sound/pci/echoaudio/echoaudio_gml.c +++ b/sound/pci/echoaudio/echoaudio_gml.c @@ -63,6 +63,8 @@ the control register. write_control_reg sends the new control register value to the DSP. */ static int write_control_reg(struct echoaudio *chip, u32 value, char force) { + __le32 reg_value; + /* Handle the digital input auto-mute */ if (chip->digital_in_automute) value |= GML_DIGITAL_IN_AUTO_MUTE; @@ -72,11 +74,11 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force) dev_dbg(chip->card->dev, "write_control_reg: 0x%x\n", value); /* Write the control register */ - value = cpu_to_le32(value); - if (value != chip->comm_page->control_register || force) { + reg_value = cpu_to_le32(value); + if (reg_value != chip->comm_page->control_register || force) { if (wait_handshake(chip)) return -EIO; - chip->comm_page->control_register = value; + chip->comm_page->control_register = reg_value; clear_handshake(chip); return send_vector(chip, DSP_VC_WRITE_CONTROL_REG); } From 58578d1894490c62bf64c3293cb06e0fcdc86a31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:11 +0200 Subject: [PATCH 388/529] ALSA: bt87x: Proper endian notations The RISC data in bt87x is in little-endian, hence we should define it with __le32 properly. Spotted by sparse, a warning like: sound/pci/bt87x.c:240:17: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/bt87x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index d8ade8771a32..ba971042f871 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -228,14 +228,14 @@ static int snd_bt87x_create_risc(struct snd_bt87x *chip, struct snd_pcm_substrea unsigned int periods, unsigned int period_bytes) { unsigned int i, offset; - u32 *risc; + __le32 *risc; if (chip->dma_risc.area == NULL) { if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), PAGE_ALIGN(MAX_RISC_SIZE), &chip->dma_risc) < 0) return -ENOMEM; } - risc = (u32 *)chip->dma_risc.area; + risc = (__le32 *)chip->dma_risc.area; offset = 0; *risc++ = cpu_to_le32(RISC_SYNC | RISC_SYNC_FM1); *risc++ = cpu_to_le32(0); From c44a81a40af0e1aa52b88d1c60682e30c411fb23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:12 +0200 Subject: [PATCH 389/529] ALSA: atiixp: Proper endian notations The DMA address table in atiixp driver is in little-endian, hence we should define it with __le32 properly. Spotted by sparse, a warning like: sound/pci/atiixp.c:393:28: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7ae63d452bba..a1e4944dcfe8 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -207,10 +207,10 @@ struct atiixp; */ struct atiixp_dma_desc { - u32 addr; /* DMA buffer address */ + __le32 addr; /* DMA buffer address */ u16 status; /* status bits */ u16 size; /* size of the packet in dwords */ - u32 next; /* address of the next packet descriptor */ + __le32 next; /* address of the next packet descriptor */ }; /* From 7e49aadf64992c138b354a2e9ae35a3ae0399415 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:13 +0200 Subject: [PATCH 390/529] ALSA: atiixp_modem: Proper endian notations The DMA address table in atiixp modem driver is in little-endian, hence we should define it with __le32 properly. Spotted by sparse, a warning like: sound/pci/atiixp_modem.c:360:28: warning: incorrect type in assignment (different base types) Signed-off-by: Takashi Iwai --- sound/pci/atiixp_modem.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index a586635664e0..dc1de860cedf 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -183,10 +183,10 @@ struct atiixp_modem; */ struct atiixp_dma_desc { - u32 addr; /* DMA buffer address */ + __le32 addr; /* DMA buffer address */ u16 status; /* status bits */ u16 size; /* size of the packet in dwords */ - u32 next; /* address of the next packet descriptor */ + __le32 next; /* address of the next packet descriptor */ }; /* From 13e9a3edb4b702b701b5a9553b830aa3d2a3e3e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:24:14 +0200 Subject: [PATCH 391/529] ALSA: sb: Proper endian notations The data types defined in SB CSP driver code are all in little-endian, hence the proper type like __le32 should be used. Spotted by sparse, a warning like: sound/isa/sb/sb16_csp.c:330:14: warning: cast to restricted __le32 Signed-off-by: Takashi Iwai --- sound/isa/sb/sb16_csp.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 2210e7c72787..b9d67a7065cd 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -60,18 +60,18 @@ MODULE_FIRMWARE("sb16/ima_adpcm_capture.csp"); * RIFF data format */ struct riff_header { - __u32 name; - __u32 len; + __le32 name; + __le32 len; }; struct desc_header { struct riff_header info; - __u16 func_nr; - __u16 VOC_type; - __u16 flags_play_rec; - __u16 flags_16bit_8bit; - __u16 flags_stereo_mono; - __u16 flags_rates; + __le16 func_nr; + __le16 VOC_type; + __le16 flags_play_rec; + __le16 flags_16bit_8bit; + __le16 flags_stereo_mono; + __le16 flags_rates; }; /* @@ -314,7 +314,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, unsigned short func_nr = 0; struct riff_header file_h, item_h, code_h; - __u32 item_type; + __le32 item_type; struct desc_header funcdesc_h; unsigned long flags; @@ -326,7 +326,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&file_h, data_ptr, sizeof(file_h))) return -EFAULT; - if ((file_h.name != RIFF_HEADER) || + if ((le32_to_cpu(file_h.name) != RIFF_HEADER) || (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) { snd_printd("%s: Invalid RIFF header\n", __func__); return -EINVAL; @@ -336,7 +336,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; - if (item_type != CSP__HEADER) { + if (le32_to_cpu(item_type) != CSP__HEADER) { snd_printd("%s: Invalid RIFF file type\n", __func__); return -EINVAL; } @@ -346,12 +346,12 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_h, data_ptr, sizeof(item_h))) return -EFAULT; data_ptr += sizeof(item_h); - if (item_h.name != LIST_HEADER) + if (le32_to_cpu(item_h.name) != LIST_HEADER) continue; if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; - switch (item_type) { + switch (le32_to_cpu(item_type)) { case FUNC_HEADER: if (copy_from_user(&funcdesc_h, data_ptr + sizeof(item_type), sizeof(funcdesc_h))) return -EFAULT; @@ -378,7 +378,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; /* init microcode blocks */ - if (code_h.name != INIT_HEADER) + if (le32_to_cpu(code_h.name) != INIT_HEADER) break; data_ptr += sizeof(code_h); err = snd_sb_csp_load_user(p, data_ptr, le32_to_cpu(code_h.len), @@ -391,7 +391,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&code_h, data_ptr, sizeof(code_h))) return -EFAULT; - if (code_h.name != MAIN_HEADER) { + if (le32_to_cpu(code_h.name) != MAIN_HEADER) { snd_printd("%s: Missing 'main' microcode\n", __func__); return -EINVAL; } From df3f0347fd856272ca9fdbb6e691b7b512b7acb4 Mon Sep 17 00:00:00 2001 From: Jia-Ju Bai Date: Fri, 27 Jul 2018 16:55:28 +0800 Subject: [PATCH 392/529] ALSA: usb-audio: quirks: Replace mdelay() with msleep() and usleep_range() snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and snd_usb_ctl_msg_quirk() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep() and usleep_range(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index dde87d64bc32..8a945ece9869 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1213,7 +1213,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, if (err < 0) return err; - mdelay(20); /* Delay needed after setting the interface */ + msleep(20); /* Delay needed after setting the interface */ /* Vendor mode switch cmd is required. */ if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) { @@ -1234,7 +1234,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, return err; } - mdelay(20); + msleep(20); } return 0; } @@ -1281,7 +1281,7 @@ void snd_usb_set_interface_quirk(struct usb_device *dev) switch (USB_ID_VENDOR(chip->usb_id)) { case 0x23ba: /* Playback Design */ case 0x0644: /* TEAC Corp. */ - mdelay(50); + msleep(50); break; } } @@ -1301,7 +1301,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* * "TEAC Corp." products need a 20ms delay after each @@ -1309,14 +1309,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x0644 && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ if (is_itf_usb_dsd_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, * otherwise requests like get/set frequency return as failed despite @@ -1326,7 +1326,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, chip->usb_id == USB_ID(0x046d, 0x0a46) || chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + usleep_range(1000, 2000); } /* From 08fd8325d94eeddfe49ef8191337e5f54553f6b0 Mon Sep 17 00:00:00 2001 From: Jia-Ju Bai Date: Fri, 27 Jul 2018 16:57:56 +0800 Subject: [PATCH 393/529] ALSA:: ctxfi: cthw20k1: Replace mdelay() with msleep() hw_pll_init(), hw_reset_dac() and hw_card_init() are never called in atomic context. They calls mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 8e6eb9d7984b..6a051a1c3724 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1319,7 +1319,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) break; hw_write_20kx(hw, PLLCTL, pllctl); - mdelay(40); + msleep(40); } if (i >= 3) { dev_alert(hw->card->dev, "PLL initialization failed!!!\n"); @@ -1407,7 +1407,7 @@ static int hw_reset_dac(struct hw *hw) /* To be effective, need to reset the DAC twice. */ for (i = 0; i < 2; i++) { /* set gpio */ - mdelay(100); + msleep(100); gpioorg = (u16)hw_read_20kx(hw, GPIO); gpioorg &= 0xfffd; hw_write_20kx(hw, GPIO, gpioorg); @@ -2030,7 +2030,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, GIE, 0); /* Reset all SRC pending interrupts */ hw_write_20kx(hw, SRCIP, 0); - mdelay(30); + msleep(30); /* Detect the card ID and configure GPIO accordingly. */ switch (hw->model) { From fad56c895f1f33f9063da558067307b00d44d40d Mon Sep 17 00:00:00 2001 From: Jia-Ju Bai Date: Fri, 27 Jul 2018 17:01:43 +0800 Subject: [PATCH 394/529] ALSA: ctxfi: cthw20k2: Replace mdelay() with msleep() and usleep_range() hw_pll_init(), hw_dac_stop(), hw_dac_start() and hw_adc_init() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k2.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index b866d6b2c923..3c966fafc754 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1316,12 +1316,12 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4); set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); pllctl = hw_read_20kx(hw, PLL_CTL); set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); for (i = 0; i < 1000; i++) { pllstat = hw_read_20kx(hw, PLL_STAT); @@ -1584,7 +1584,7 @@ static void hw_dac_stop(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data &= 0xFFFFFFFD; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(10); + usleep_range(10000, 11000); } static void hw_dac_start(struct hw *hw) @@ -1593,7 +1593,7 @@ static void hw_dac_start(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data |= 0x2; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); } static void hw_dac_reset(struct hw *hw) @@ -1864,11 +1864,11 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw_write_20kx(hw, GPIO_DATA, data); } - mdelay(10); + usleep_range(10000, 11000); /* Return the ADC to normal operation. */ data |= (0x1 << 15); hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); /* I2C write to register offset 0x0B to set ADC LRCLK polarity */ /* invert bit, interface format to I2S, word length to 24-bit, */ From d77a4b4a5b0b2ebcbc9840995d91311ef28302ab Mon Sep 17 00:00:00 2001 From: Park Ju Hyung Date: Sat, 28 Jul 2018 03:16:21 +0900 Subject: [PATCH 395/529] ALSA: hda - Turn CX8200 into D3 as well upon reboot As an equivalent codec with CX20724, CX8200 is also subject to the reboot bug. Late 2017 and 2018 LG Gram and some HP Spectre laptops are known victims to this issue, causing extremely loud noises upon reboot. Now that we know that this bug is subject to multiple codecs, fix the comment as well. Signed-off-by: Park Ju Hyung Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f641c20095f7..909a880f5e01 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -211,6 +211,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x14f12008: /* CX8200 */ case 0x14f150f2: /* CX20722 */ case 0x14f150f4: /* CX20724 */ break; @@ -218,7 +219,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) return; } - /* Turn the CX20722 codec into D3 to avoid spurious noises + /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); From f59cf9a0551dd954ad8b752461cf19d9789f4b1d Mon Sep 17 00:00:00 2001 From: Park Ju Hyung Date: Sat, 28 Jul 2018 03:16:42 +0900 Subject: [PATCH 396/529] ALSA: hda - Sleep for 10ms after entering D3 on Conexant codecs On rare occasions, we are still noticing that the internal speaker spitting out spurious noises even after adding the problematic codec to the list. Adding a 10ms artificial delay before rebooting fixes the issue entirely. Patch for Realtek codecs also adds the same amount of delay after entering D3. Signed-off-by: Park Ju Hyung Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 909a880f5e01..1a8a2d440fbd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -226,6 +226,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); } static void cx_auto_free(struct hda_codec *codec) From f69548ffafcc4942022f16f2f192b24143de1dba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jul 2018 23:06:51 +0200 Subject: [PATCH 397/529] ALSA: hda/hdmi: Use single mutex unlock in error paths Instead of calling mutex_unlock() at each error path multiple times, take the standard goto-and-a-single-unlock approach. This will simplify the code and make easier to find the unbalanced mutex locks. No functional changes, but only the code readability improvement as a preliminary work for further changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 67 +++++++++++++++++++------------------- 1 file changed, 33 insertions(+), 34 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ee56359be9ee..cb587dce67a9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -339,13 +339,13 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, if (!per_pin) { /* no pin is bound to the pcm */ uinfo->count = 0; - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } eld = &per_pin->sink_eld; uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&spec->pcm_lock); + unlock: + mutex_unlock(&spec->pcm_lock); return 0; } @@ -357,6 +357,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; int pcm_idx; + int err = 0; pcm_idx = kcontrol->private_value; mutex_lock(&spec->pcm_lock); @@ -365,16 +366,15 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, /* no pin is bound to the pcm */ memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } - eld = &per_pin->sink_eld; + eld = &per_pin->sink_eld; if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || eld->eld_size > ELD_MAX_SIZE) { - mutex_unlock(&spec->pcm_lock); snd_BUG(); - return -EINVAL; + err = -EINVAL; + goto unlock; } memset(ucontrol->value.bytes.data, 0, @@ -382,9 +382,10 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&spec->pcm_lock); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } static const struct snd_kcontrol_new eld_bytes_ctl = { @@ -1209,8 +1210,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(codec, hinfo); if (!spec->dyn_pcm_assign) { if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } } else { /* no pin is assigned to the PCM @@ -1218,16 +1219,13 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, */ if (pin_idx < 0) { err = hdmi_pcm_open_no_pin(hinfo, codec, substream); - mutex_unlock(&spec->pcm_lock); - return err; + goto unlock; } } err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); - if (err < 0) { - mutex_unlock(&spec->pcm_lock); - return err; - } + if (err < 0) + goto unlock; per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ @@ -1264,12 +1262,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - mutex_unlock(&spec->pcm_lock); - return -ENODEV; + err = -ENODEV; + goto unlock; } } - mutex_unlock(&spec->pcm_lock); /* Store the updated parameters */ runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; @@ -1278,7 +1275,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } /* @@ -1867,7 +1866,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_runtime *runtime = substream->runtime; bool non_pcm; int pinctl; - int err; + int err = 0; mutex_lock(&spec->pcm_lock); pin_idx = hinfo_to_pin_index(codec, hinfo); @@ -1879,13 +1878,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; @@ -1924,6 +1922,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* snd_hda_set_dev_select() has been called before */ err = spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + unlock: mutex_unlock(&spec->pcm_lock); return err; } @@ -1945,6 +1944,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; int pinctl; + int err = 0; if (hinfo->nid) { pcm_idx = hinfo_to_pcm_index(codec, hinfo); @@ -1963,14 +1963,12 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (spec->dyn_pcm_assign && pin_idx < 0) { - mutex_unlock(&spec->pcm_lock); - return 0; - } + if (spec->dyn_pcm_assign && pin_idx < 0) + goto unlock; if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); @@ -1989,10 +1987,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin->setup = false; per_pin->channels = 0; mutex_unlock(&per_pin->lock); + unlock: mutex_unlock(&spec->pcm_lock); } - return 0; + return err; } static const struct hda_pcm_ops generic_ops = { From 279fef50b607f9cee94f10bae84f6730e97ccd7c Mon Sep 17 00:00:00 2001 From: Edward Cragg Date: Fri, 27 Jul 2018 13:59:28 +0100 Subject: [PATCH 398/529] ASoC: tegra: i2s: Fix typo/broken macro Fix typo in macro TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK. Signed-off-by: Edward Cragg Signed-off-by: Jorge Sanjuan Reviewed-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 774fc6ad2026..2e561e946de2 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -173,7 +173,7 @@ /* Number of slots in frame, minus 1 */ #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 -#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT) /* TDM mode slot enable bitmask */ #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 From c9c9780d8fa526a124933134a7e4041fa09662f6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl Date: Sat, 28 Jul 2018 14:30:17 +0200 Subject: [PATCH 399/529] ASoC: wm8988: fix typo in rate constraints Remove duplicated entry and add missing zero in rate constraints. Signed-off-by: Ladislav Michl Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; From 345a9ca37aa69bb3133c1a8390a71f993abcef0c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jul 2018 08:46:40 +0200 Subject: [PATCH 400/529] ALSA: memalloc: Fix missing PAGE_SIZE definition The recent fix moved the inline snd_sgbuf_aligned_pages() outside the ifdef, and this triggered a build error on some architectures due to the undefined PAGE_SIZE, as spotted by 0day bot. Fix it by adding the missing header inclusion. Fixes: 4cae99d9b530 ("ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally") Reported-by: kbuild test robot Cc: Pierre-Louis Bossart Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/memalloc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c669900e6cbe..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include + struct device; /* From fe209b97184f577b70ae0799c2b149be36092f84 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:51:41 +0000 Subject: [PATCH 401/529] ASoC: ak4642: convert to SPDX identifiers As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 605055964529..353237025514 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); From 7a968dc66aecaaea3424ca2525b758687b39bc0e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:52:15 +0000 Subject: [PATCH 402/529] ASoC: ak4554: convert to SPDX identifiers As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee13406d93..2fa83a1a84cf 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto #include #include @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto "); From c0ca5604d43233b0f5cfb793be9638af21f5535d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:52:46 +0000 Subject: [PATCH 403/529] ASoC: da7210: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111b7184..e172913d04a4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include #include From e028937c77f44bb38cdbc464afce2bf41d96006b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:53:08 +0000 Subject: [PATCH 404/529] ASoC: ak4613: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9351cf..c1181a20714d 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include #include From 7464d3faf62a5adcb99f4f8ea9baabda78aed4aa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 08:00:22 +0000 Subject: [PATCH 405/529] ASoC: sh: Kconfig: convert to SPDX identifiers By default all files without license information are under the default license of the kernel, which is GPL version 2. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800bf3a8..dc20f0f7080a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST From 4321723648b0abb456f7a9af51bb09a4ec60799d Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 28 Jul 2018 00:06:59 +0300 Subject: [PATCH 406/529] ASoC: tegra_alc5632: fix device_node refcounting tegra_alc5632_probe() increments reference count of device nodes with of_parse_phandle(), but there is no code decrementing them in the driver. The patch adds of_node_put() to tegra_alc5632_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Acked-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } From ae1c696a480c67c45fb23b35162183f72c6be0e1 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 26 Jul 2018 15:49:10 -0500 Subject: [PATCH 407/529] ASoC: sirf: Fix potential NULL pointer dereference There is a potential execution path in which function platform_get_resource() returns NULL. If this happens, we will end up having a NULL pointer dereference. Fix this by replacing devm_ioremap with devm_ioremap_resource, which has the NULL check and the memory region request. This code was detected with the help of Coccinelle. Cc: stable@vger.kernel.org Fixes: 2bd8d1d5cf89 ("ASoC: sirf: Add audio usp interface driver") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/sirf/sirf-usp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf969d0..d70fcd4a1adf 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) From 8fc9983db199bb397d48e32a6400765b70f1995a Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 27 Jul 2018 11:37:28 +0900 Subject: [PATCH 408/529] ASoC: uniphier: add support for multichannel output This patch adds multichannel PCM output support for LD11/LD20. Currently driver tested and supported only 2ch, 6ch, and 8ch. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-core.c | 78 ++++++++++++++++++++++++++++++++--- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-reg.h | 1 + sound/soc/uniphier/aio.h | 3 ++ 4 files changed, 78 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..8b09bbb0f8d0 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -264,6 +264,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } } +/** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + /** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..23a5c3c68658 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * From d8504acca759ae672c6d34d49a33d54ace094cbb Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 27 Jul 2018 11:37:44 +0900 Subject: [PATCH 409/529] ASoC: uniphier: change functions to static This patch changes some functions that are not used by other objects to static. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-core.c | 6 +++--- sound/soc/uniphier/aio.h | 3 --- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 8b09bbb0f8d0..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -327,7 +327,7 @@ static int aio_port_set_ch(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -446,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -511,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 23a5c3c68658..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -325,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); From f7debfe54090d1a1c38e1f070be20d83bb70a8e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 29 Jul 2018 23:03:05 +0200 Subject: [PATCH 410/529] ALSA: seq: virmidi: Offload the output event processing The virmidi sequencer stuff tries to translate the rawmidi bytes to sequencer events and deliver the packets at trigger callback. The amount of the whole process of these translations and deliveries depends on the incoming rawmidi bytes, and we have no limit for that; this was the cause of a CPU soft lockup that had been reported and fixed recently. Although we've fixed the soft lockup by putting the temporary unlock and cond_resched(), it's rather a quick band aid. In this patch, meanwhile, the event parsing and delivery process is offloaded to a dedicated work, and the trigger callback just kicks it off. It has three merits, at least: - The processing is always done in a sleepable context, which can assure the event delivery with non-atomic flag without hackish is_atomic() usage. - Other relevant codes can be simplified, reducing the lines - It makes me happier Signed-off-by: Takashi Iwai --- include/sound/seq_virmidi.h | 1 + sound/core/seq/seq_virmidi.c | 102 ++++++++++++++++------------------- 2 files changed, 48 insertions(+), 55 deletions(-) diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index 695257ae64ac..d488dcfa3a4e 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -41,6 +41,7 @@ struct snd_virmidi { struct snd_seq_event event; struct snd_virmidi_dev *rdev; struct snd_rawmidi_substream *substream; + struct work_struct output_work; }; #define SNDRV_VIRMIDI_SUBSCRIBE (1<<0) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 8ebbca554e99..67ea5d62cebc 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -154,68 +154,56 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i } } +/* process rawmidi bytes and send events; + * we need no lock here for vmidi->event since it's handled only in this work + */ +static void snd_vmidi_output_work(struct work_struct *work) +{ + struct snd_virmidi *vmidi; + struct snd_rawmidi_substream *substream; + unsigned char input; + int ret; + + vmidi = container_of(work, struct snd_virmidi, output_work); + substream = vmidi->substream; + + /* discard the outputs in dispatch mode unless subscribed */ + if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && + !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { + while (!snd_rawmidi_transmit_empty(substream)) + snd_rawmidi_transmit_ack(substream, 1); + return; + } + + while (vmidi->trigger) { + if (snd_rawmidi_transmit(substream, &input, 1) != 1) + break; + if (snd_midi_event_encode_byte(vmidi->parser, input, + &vmidi->event) <= 0) + continue; + if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { + ret = snd_seq_kernel_client_dispatch(vmidi->client, + &vmidi->event, + false, 0); + vmidi->event.type = SNDRV_SEQ_EVENT_NONE; + if (ret < 0) + break; + } + /* rawmidi input might be huge, allow to have a break */ + cond_resched(); + } +} + /* * trigger rawmidi stream for output */ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_virmidi *vmidi = substream->runtime->private_data; - int count, res; - unsigned char buf[32], *pbuf; - unsigned long flags; - bool check_resched = !in_atomic(); - if (up) { - vmidi->trigger = 1; - if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && - !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - while (snd_rawmidi_transmit(substream, buf, - sizeof(buf)) > 0) { - /* ignored */ - } - return; - } - spin_lock_irqsave(&substream->runtime->lock, flags); - if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; - vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - while (1) { - count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf)); - if (count <= 0) - break; - pbuf = buf; - while (count > 0) { - res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event); - if (res < 0) { - snd_midi_event_reset_encode(vmidi->parser); - continue; - } - __snd_rawmidi_transmit_ack(substream, res); - pbuf += res; - count -= res; - if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; - vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - } - if (!check_resched) - continue; - /* do temporary unlock & cond_resched() for avoiding - * CPU soft lockup, which may happen via a write from - * a huge rawmidi buffer - */ - spin_unlock_irqrestore(&substream->runtime->lock, flags); - cond_resched(); - spin_lock_irqsave(&substream->runtime->lock, flags); - } - out: - spin_unlock_irqrestore(&substream->runtime->lock, flags); - } else { - vmidi->trigger = 0; - } + vmidi->trigger = !!up; + if (up) + queue_work(system_highpri_wq, &vmidi->output_work); } /* @@ -270,6 +258,7 @@ static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream) vmidi->port = rdev->port; snd_virmidi_init_event(vmidi, &vmidi->event); vmidi->rdev = rdev; + INIT_WORK(&vmidi->output_work, snd_vmidi_output_work); runtime->private_data = vmidi; return 0; } @@ -299,6 +288,9 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream) static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_virmidi *vmidi = substream->runtime->private_data; + + vmidi->trigger = 0; /* to be sure */ + cancel_work_sync(&vmidi->output_work); snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; kfree(vmidi); From 89b4ab213feb11a5bece544cfa12374f66c2c47c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jul 2018 14:48:29 +0200 Subject: [PATCH 411/529] ALSA: seq: virmidi: Use READ_ONCE/WRITE_ONCE() macros The trigger flag in vmidi object can be referred in different contexts concurrently, hence it's better to be put with READ_ONCE() and WRITE_ONCE() macros to assure the accesses. Signed-off-by: Takashi Iwai --- include/sound/seq_virmidi.h | 2 +- sound/core/seq/seq_virmidi.c | 14 +++++--------- 2 files changed, 6 insertions(+), 10 deletions(-) diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index d488dcfa3a4e..796ce7772213 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -36,7 +36,7 @@ struct snd_virmidi { int seq_mode; int client; int port; - unsigned int trigger: 1; + bool trigger; struct snd_midi_event *parser; struct snd_seq_event event; struct snd_virmidi_dev *rdev; diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 67ea5d62cebc..03ac5e72dbe6 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -89,7 +89,7 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, else down_read(&rdev->filelist_sem); list_for_each_entry(vmidi, &rdev->filelist, list) { - if (!vmidi->trigger) + if (!READ_ONCE(vmidi->trigger)) continue; if (ev->type == SNDRV_SEQ_EVENT_SYSEX) { if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE) @@ -147,11 +147,7 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i { struct snd_virmidi *vmidi = substream->runtime->private_data; - if (up) { - vmidi->trigger = 1; - } else { - vmidi->trigger = 0; - } + WRITE_ONCE(vmidi->trigger, !!up); } /* process rawmidi bytes and send events; @@ -175,7 +171,7 @@ static void snd_vmidi_output_work(struct work_struct *work) return; } - while (vmidi->trigger) { + while (READ_ONCE(vmidi->trigger)) { if (snd_rawmidi_transmit(substream, &input, 1) != 1) break; if (snd_midi_event_encode_byte(vmidi->parser, input, @@ -201,7 +197,7 @@ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, { struct snd_virmidi *vmidi = substream->runtime->private_data; - vmidi->trigger = !!up; + WRITE_ONCE(vmidi->trigger, !!up); if (up) queue_work(system_highpri_wq, &vmidi->output_work); } @@ -289,7 +285,7 @@ static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_virmidi *vmidi = substream->runtime->private_data; - vmidi->trigger = 0; /* to be sure */ + WRITE_ONCE(vmidi->trigger, false); /* to be sure */ cancel_work_sync(&vmidi->output_work); snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; From 11785ef53228d23ec386f5fe4a34601536f0c891 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Tue, 31 Jul 2018 13:28:42 +0100 Subject: [PATCH 412/529] ALSA: usb-audio: Initial Power Domain support Thee USB Audio Class 3 (UAC3) introduces Power Domains as a new feature to let a host turn individual parts of an audio function to different power states via USB requests. This lets the device get to know a bit amore about what the host is up to in order to optimize power consumption efficiently. The Power Domains are optional for UAC3 configuration but all UAC3 devices shall include at least one BADD configuration where the support for Power Domains is compulsory. This patch adds a set of features/helpers to parse these power domains and change their status. Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v3.h | 4 ++ sound/usb/Makefile | 1 + sound/usb/power.c | 104 +++++++++++++++++++++++++++++++++++ sound/usb/power.h | 19 +++++++ 4 files changed, 128 insertions(+) create mode 100644 sound/usb/power.c diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h index 334bfa6dfb47..6b708434b7f9 100644 --- a/include/linux/usb/audio-v3.h +++ b/include/linux/usb/audio-v3.h @@ -447,4 +447,8 @@ struct uac3_interrupt_data_msg { /* BADD sample rate is always fixed to 48kHz */ #define UAC3_BADD_SAMPLING_RATE 48000 +/* BADD power domains recovery times in 50us increments */ +#define UAC3_BADD_PD_RECOVER_D1D0 0x0258 /* 30ms */ +#define UAC3_BADD_PD_RECOVER_D2D0 0x1770 /* 300ms */ + #endif /* __LINUX_USB_AUDIO_V3_H */ diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 05440e2df8d9..d330f74c90e6 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_us16x08.o \ pcm.o \ + power.o \ proc.o \ quirks.o \ stream.o diff --git a/sound/usb/power.c b/sound/usb/power.c new file mode 100644 index 000000000000..bd303a1ba1b7 --- /dev/null +++ b/sound/usb/power.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * UAC3 Power Domain state management functions + */ + +#include +#include +#include +#include +#include + +#include "usbaudio.h" +#include "helper.h" +#include "power.h" + +struct snd_usb_power_domain * +snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, + unsigned char id) +{ + struct snd_usb_power_domain *pd; + void *p; + + pd = kzalloc(sizeof(*pd), GFP_KERNEL); + if (!pd) + return NULL; + + p = NULL; + while ((p = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + p, UAC3_POWER_DOMAIN)) != NULL) { + struct uac3_power_domain_descriptor *pd_desc = p; + int i; + + for (i = 0; i < pd_desc->bNrEntities; i++) { + if (pd_desc->baEntityID[i] == id) { + pd->pd_id = pd_desc->bPowerDomainID; + pd->pd_d1d0_rec = + le16_to_cpu(pd_desc->waRecoveryTime1); + pd->pd_d2d0_rec = + le16_to_cpu(pd_desc->waRecoveryTime2); + return pd; + } + } + } + + kfree(pd); + return NULL; +} + +int snd_usb_power_domain_set(struct snd_usb_audio *chip, + struct snd_usb_power_domain *pd, + unsigned char state) +{ + struct usb_device *dev = chip->dev; + unsigned char current_state; + int err, idx; + + idx = snd_usb_ctrl_intf(chip) | (pd->pd_id << 8); + + err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx, + ¤t_state, sizeof(current_state)); + if (err < 0) { + dev_err(&dev->dev, "Can't get UAC3 power state for id %d\n", + pd->pd_id); + return err; + } + + if (current_state == state) { + dev_dbg(&dev->dev, "UAC3 power domain id %d already in state %d\n", + pd->pd_id, state); + return 0; + } + + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx, + &state, sizeof(state)); + if (err < 0) { + dev_err(&dev->dev, "Can't set UAC3 power state to %d for id %d\n", + state, pd->pd_id); + return err; + } + + if (state == UAC3_PD_STATE_D0) { + switch (current_state) { + case UAC3_PD_STATE_D2: + udelay(pd->pd_d2d0_rec * 50); + break; + case UAC3_PD_STATE_D1: + udelay(pd->pd_d1d0_rec * 50); + break; + default: + return -EINVAL; + } + } + + dev_dbg(&dev->dev, "UAC3 power domain id %d change to state %d\n", + pd->pd_id, state); + + return 0; +} diff --git a/sound/usb/power.h b/sound/usb/power.h index b2e25f60c5a2..6004231a7c75 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -2,6 +2,25 @@ #ifndef __USBAUDIO_POWER_H #define __USBAUDIO_POWER_H +struct snd_usb_power_domain { + int pd_id; /* UAC3 Power Domain ID */ + int pd_d1d0_rec; /* D1 to D0 recovery time */ + int pd_d2d0_rec; /* D2 to D0 recovery time */ +}; + +enum { + UAC3_PD_STATE_D0, + UAC3_PD_STATE_D1, + UAC3_PD_STATE_D2, +}; + +int snd_usb_power_domain_set(struct snd_usb_audio *chip, + struct snd_usb_power_domain *pd, + unsigned char state); +struct snd_usb_power_domain * +snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, + unsigned char id); + #ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip); void snd_usb_autosuspend(struct snd_usb_audio *chip); From 7edf3b5e6a4544b42d3572a7058f8ffe96349ee8 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Tue, 31 Jul 2018 13:28:43 +0100 Subject: [PATCH 413/529] ALSA: usb-audio: AudioStreaming Power Domain parsing Power Domains in the UAC3 spec are mainly intended to be associated to an Input or Output Terminal so the host changes the power state of the entire capture or playback path within the topology. This patch adds support for finding Power Domains associated to an Audio Streaming Interface (bTerminalLink) and adds a reference to them in the usb audio substreams (snd_usb_substream). Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 ++ sound/usb/stream.c | 66 ++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 60 insertions(+), 8 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 9b41b7dda84f..ac785d15ced4 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -37,6 +37,7 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_endpoint; +struct snd_usb_power_domain; struct snd_urb_ctx { struct urb *urb; @@ -115,6 +116,7 @@ struct snd_usb_substream { int interface; /* current interface */ int endpoint; /* assigned endpoint */ struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ + struct snd_usb_power_domain *str_pd; /* UAC3 Power Domain for streaming path */ snd_pcm_format_t pcm_format; /* current audio format (for hw_params callback) */ unsigned int channels; /* current number of channels (for hw_params callback) */ unsigned int channels_max; /* max channels in the all audiofmts */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 729afd808cc4..c0567fa1e84b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -37,6 +37,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "power.h" /* * free a substream @@ -53,6 +54,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + kfree(subs->str_pd); } @@ -82,7 +84,8 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) static void snd_usb_init_substream(struct snd_usb_stream *as, int stream, - struct audioformat *fp) + struct audioformat *fp, + struct snd_usb_power_domain *pd) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -107,6 +110,9 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + if (pd) + subs->str_pd = pd; + snd_usb_preallocate_buffer(subs); } @@ -468,9 +474,11 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor * fmt_list and will be freed on the chip instance release. do not free * fp or do remove it from the substream fmt_list to avoid double-free. */ -int snd_usb_add_audio_stream(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp) +static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp, + struct snd_usb_power_domain *pd) + { struct snd_usb_stream *as; struct snd_usb_substream *subs; @@ -498,7 +506,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, stream, fp, pd); return add_chmap(as->pcm, stream, subs); } @@ -526,7 +534,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, stream, fp, pd); /* * Keep using head insertion for M-Audio Audiophile USB (tm) which has a @@ -544,6 +552,21 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, return add_chmap(pcm, stream, &as->substream[stream]); } +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp) +{ + return __snd_usb_add_audio_stream(chip, stream, fp, NULL); +} + +static int snd_usb_add_audio_stream_v3(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp, + struct snd_usb_power_domain *pd) +{ + return __snd_usb_add_audio_stream(chip, stream, fp, pd); +} + static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct usb_host_interface *alts, int protocol, int iface_no) @@ -819,6 +842,7 @@ found_clock: static struct audioformat * snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, struct usb_host_interface *alts, + struct snd_usb_power_domain **pd_out, int iface_no, int altset_idx, int altno, int stream) { @@ -829,6 +853,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, struct uac3_as_header_descriptor *as = NULL; struct uac3_hc_descriptor_header hc_header; struct snd_pcm_chmap_elem *chmap; + struct snd_usb_power_domain *pd; unsigned char badd_profile; u64 badd_formats = 0; unsigned int num_channels; @@ -1008,12 +1033,28 @@ found_clock: fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + pd = kzalloc(sizeof(pd), GFP_KERNEL); + if (!pd) { + kfree(fp->rate_table); + kfree(fp); + return NULL; + } + pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + UAC3_BADD_PD_ID10 : UAC3_BADD_PD_ID11; + pd->pd_d1d0_rec = UAC3_BADD_PD_RECOVER_D1D0; + pd->pd_d2d0_rec = UAC3_BADD_PD_RECOVER_D2D0; + } else { fp->attributes = parse_uac_endpoint_attributes(chip, alts, UAC_VERSION_3, iface_no); + + pd = snd_usb_find_power_domain(chip->ctrl_intf, + as->bTerminalLink); + /* ok, let's parse further... */ if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { + kfree(pd); kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); @@ -1021,6 +1062,9 @@ found_clock: } } + if (pd) + *pd_out = pd; + return fp; } @@ -1032,6 +1076,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) struct usb_interface_descriptor *altsd; int i, altno, err, stream; struct audioformat *fp = NULL; + struct snd_usb_power_domain *pd = NULL; int num, protocol; dev = chip->dev; @@ -1114,7 +1159,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) break; } case UAC_VERSION_3: - fp = snd_usb_get_audioformat_uac3(chip, alts, + fp = snd_usb_get_audioformat_uac3(chip, alts, &pd, iface_no, i, altno, stream); break; } @@ -1125,9 +1170,14 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return PTR_ERR(fp); dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); - err = snd_usb_add_audio_stream(chip, stream, fp); + if (protocol == UAC_VERSION_3) + err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd); + else + err = snd_usb_add_audio_stream(chip, stream, fp); + if (err < 0) { list_del(&fp->list); /* unlink for avoiding double-free */ + kfree(pd); kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); From 3f59aa11c6776da8d0f9f50c741ef02bfc4a8766 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Tue, 31 Jul 2018 13:28:44 +0100 Subject: [PATCH 414/529] ALSA: usb-audio: Add UAC3 Power Domains to suspend/resume Set the UAC3 Power Domain state for an Audio Streaming interface to D2 state before suspending the device (usb_driver callback). This lets the device know there is no intention to use any of the Units in the Audio Function and that the host is not going to even listen for wake-up events (interrupts) on the units. When the usb_driver gets resumed, the state D0 (fully powered) will be set. This ties up the UAC3 Power Domains to the runtime PM. Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/card.c | 9 +++++++++ sound/usb/pcm.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/pcm.h | 2 ++ 3 files changed, 59 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index a1ed798a1c6b..2bfe4e80a6b9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -809,6 +809,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_pcm_suspend_all(as->pcm); + snd_usb_pcm_suspend(as); as->substream[0].need_setup_ep = as->substream[1].need_setup_ep = true; } @@ -824,6 +825,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); + struct snd_usb_stream *as; struct usb_mixer_interface *mixer; struct list_head *p; int err = 0; @@ -834,6 +836,13 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) return 0; atomic_inc(&chip->active); /* avoid autopm */ + + list_for_each_entry(as, &chip->pcm_list, list) { + err = snd_usb_pcm_resume(as); + if (err < 0) + goto err_out; + } + /* * ALSA leaves material resumption to user space * we just notify and restart the mixers diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4b930fa47277..99ec9d5caa58 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -711,6 +711,54 @@ static int configure_endpoint(struct snd_usb_substream *subs) return ret; } +static int snd_usb_pcm_change_state(struct snd_usb_substream *subs, int state) +{ + int ret; + + if (!subs->str_pd) + return 0; + + ret = snd_usb_power_domain_set(subs->stream->chip, subs->str_pd, state); + if (ret < 0) { + dev_err(&subs->dev->dev, + "Cannot change Power Domain ID: %d to state: %d. Err: %d\n", + subs->str_pd->pd_id, state, ret); + return ret; + } + + return 0; +} + +int snd_usb_pcm_suspend(struct snd_usb_stream *as) +{ + int ret; + + ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D2); + if (ret < 0) + return ret; + + ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D2); + if (ret < 0) + return ret; + + return 0; +} + +int snd_usb_pcm_resume(struct snd_usb_stream *as) +{ + int ret; + + ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D0); + if (ret < 0) + return ret; + + ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D0); + if (ret < 0) + return ret; + + return 0; +} + /* * hw_params callback * diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index f77ec58bf1a1..9833627c1eca 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -6,6 +6,8 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, unsigned int rate); void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); +int snd_usb_pcm_suspend(struct snd_usb_stream *as); +int snd_usb_pcm_resume(struct snd_usb_stream *as); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, From a0a4959eb4e94ce98ee5549dd7d1296d41162ca8 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Tue, 31 Jul 2018 13:28:45 +0100 Subject: [PATCH 415/529] ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks Make use of UAC3 Power Domains associated to an Audio Streaming path within the PCM's logic. This means, when there is no audio being transferred (pcm is closed), the host will set the Power Domain associated to that substream to state D1. When audio is being transferred (from hw_params onwards), the Power Domain will be set to D0 state. This is the way the host lets the device know which Terminal is going to be actively used and it is for the device to manage its own internal resources on that UAC3 Power Domain. Note the resume method now sets the Power Domain to D1 state as resuming the device doesn't mean audio streaming will occur. Signed-off-by: Jorge Sanjuan Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 26 ++++++++++++++++++++------ sound/usb/stream.c | 6 +++++- 2 files changed, 25 insertions(+), 7 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 99ec9d5caa58..bbc7116c9543 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -748,11 +748,11 @@ int snd_usb_pcm_resume(struct snd_usb_stream *as) { int ret; - ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D0); + ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D1); if (ret < 0) return ret; - ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D0); + ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D1); if (ret < 0) return ret; @@ -803,16 +803,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) return ret; - ret = set_format(subs, fmt); - snd_usb_unlock_shutdown(subs->stream->chip); + + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); if (ret < 0) - return ret; + goto unlock; + + ret = set_format(subs, fmt); + if (ret < 0) + goto unlock; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; - return 0; + unlock: + snd_usb_unlock_shutdown(subs->stream->chip); + return ret; } /* @@ -869,6 +875,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock; @@ -1313,6 +1323,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; + int ret; stop_endpoints(subs, true); @@ -1321,7 +1332,10 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) !snd_usb_lock_shutdown(subs->stream->chip)) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D1); snd_usb_unlock_shutdown(subs->stream->chip); + if (ret < 0) + return ret; } subs->pcm_substream = NULL; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c0567fa1e84b..8fe3b0e00e45 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -110,8 +110,12 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; - if (pd) + if (pd) { subs->str_pd = pd; + /* Initialize Power Domain to idle status D1 */ + snd_usb_power_domain_set(subs->stream->chip, pd, + UAC3_PD_STATE_D1); + } snd_usb_preallocate_buffer(subs); } From 1877c9fda1b7ba2ee2d9b108b9c9298e69b5b488 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 27 Jul 2018 13:17:57 +0100 Subject: [PATCH 416/529] ASoC: dt-bindings: add dt bindings for wcd9335 audio codec This patch adds bindings for wcd9335 audio codec which can support both SLIMbus and I2S/I2C interface. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- .../bindings/sound/qcom,wcd9335.txt | 123 ++++++++++++++++++ 1 file changed, 123 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt new file mode 100644 index 000000000000..1d8d49e30af7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt @@ -0,0 +1,123 @@ +QCOM WCD9335 Codec + +Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports +Qualcomm Technologies, Inc. (QTI) multimedia solutions, including +the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built +Soundwire controller, interrupt mux. It supports both I2S/I2C and +SLIMbus audio interfaces. + +Required properties with SLIMbus Interface: + +- compatible: + Usage: required + Value type: + Definition: For SLIMbus interface it should be "slimMID,PID", + textual representation of Manufacturer ID, Product Code, + shall be in lower case hexadecimal with leading zeroes + suppressed. Refer to slimbus/bus.txt for details. + Should be: + "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus. + +- reg + Usage: required + Value type: + Definition: Should be ('Device index', 'Instance ID') + +- interrupts + Usage: required + Value type: + Definition: Interrupts via WCD INTR1 and INTR2 pins + +- interrupt-names: + Usage: required + Value type: + Definition: Interrupt names of WCD INTR1 and INTR2 + Should be: "intr1", "intr2" + +- reset-gpio: + Usage: required + Value type: + Definition: Reset gpio line + +- qcom,ifd: + Usage: required + Value type: + Definition: SLIM interface device + +- clocks: + Usage: required + Value type: + Definition: See clock-bindings.txt section "consumers". List of + three clock specifiers for mclk, mclk2 and slimbus clock. + +- clock-names: + Usage: required + Value type: + Definition: Must contain "mclk", "mclk2" and "slimbus" strings. + +- vdd-buck-supply: + Usage: required + Value type: + Definition: Should contain a reference to the 1.8V buck supply + +- vdd-buck-sido-supply: + Usage: required + Value type: + Definition: Should contain a reference to the 1.8V SIDO buck supply + +- vdd-rx-supply: + Usage: required + Value type: + Definition: Should contain a reference to the 1.8V rx supply + +- vdd-tx-supply: + Usage: required + Value type: + Definition: Should contain a reference to the 1.8V tx supply + +- vdd-vbat-supply: + Usage: Optional + Value type: + Definition: Should contain a reference to the vbat supply + +- vdd-micbias-supply: + Usage: required + Value type: + Definition: Should contain a reference to the micbias supply + +- vdd-io-supply: + Usage: required + Value type: + Definition: Should contain a reference to the 1.8V io supply + +- interrupt-controller: + Usage: required + Definition: Indicating that this is a interrupt controller + +- #interrupt-cells: + Usage: required + Value type: + Definition: should be 1 + +#sound-dai-cells + Usage: required + Value type: + Definition: Must be 1 + +codec@1{ + compatible = "slim217,1a0"; + reg = <1 0>; + interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>; + interrupt-names = "intr2" + reset-gpio = <&msmgpio 64 0>; + qcom,ifd = <&wc9335_ifd>; + clock-names = "mclk", "native"; + clocks = <&rpmcc RPM_SMD_DIV_CLK1>, + <&rpmcc RPM_SMD_BB_CLK1>; + vdd-buck-supply = <&pm8994_s4>; + vdd-rx-supply = <&pm8994_s4>; + vdd-buck-sido-supply = <&pm8994_s4>; + vdd-tx-supply = <&pm8994_s4>; + vdd-io-supply = <&pm8994_s4>; + #sound-dai-cells = <1>; +} From e57d4ca882e289a2ddc844e82fa33ad1453e9871 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 27 Jul 2018 13:18:00 +0100 Subject: [PATCH 417/529] ASoC: wcd9335: add support to wcd9335 codec Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, It supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Seven DACs and nine dedicated interpolators, Seven (six audio ADCs, and one VBAT ADC), Multibutton headset control (MBHC), Active noise cancellation and Sidetone paths and processing. This patchset adds very basic support for playback and capture via the 9 interpolators and ADC respectively. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd9335.c | 1154 ++++++++++++++++++++++++++++++++++++ 3 files changed, 1161 insertions(+) create mode 100644 sound/soc/codecs/wcd9335.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..cb09abf18dde 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1066,6 +1066,11 @@ config SND_SOC_UDA1380 tristate depends on I2C +config SND_SOC_WCD9335 + tristate "WCD9335 Codec" + depends on MFD_WCD9335 + tristate + config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..01410b63daac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,6 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o @@ -451,6 +452,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c new file mode 100644 index 000000000000..bd9de5d45fa9 --- /dev/null +++ b/sound/soc/codecs/wcd9335.c @@ -0,0 +1,1154 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +// Copyright (c) 2017-2018, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +/* Fractional Rates */ +#define WCD9335_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100) +#define WCD9335_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* slave port water mark level + * (0: 6bytes, 1: 9bytes, 2: 12 bytes, 3: 15 bytes) + */ +#define SLAVE_PORT_WATER_MARK_6BYTES 0 +#define SLAVE_PORT_WATER_MARK_9BYTES 1 +#define SLAVE_PORT_WATER_MARK_12BYTES 2 +#define SLAVE_PORT_WATER_MARK_15BYTES 3 +#define SLAVE_PORT_WATER_MARK_SHIFT 1 +#define SLAVE_PORT_ENABLE 1 +#define SLAVE_PORT_DISABLE 0 +#define WCD9335_SLIM_WATER_MARK_VAL \ + ((SLAVE_PORT_WATER_MARK_12BYTES << SLAVE_PORT_WATER_MARK_SHIFT) | \ + (SLAVE_PORT_ENABLE)) + +#define WCD9335_SLIM_NUM_PORT_REG 3 +#define WCD9335_SLIM_PGD_PORT_INT_TX_EN0 (WCD9335_SLIM_PGD_PORT_INT_EN0 + 2) + +#define WCD9335_MCLK_CLK_12P288MHZ 12288000 +#define WCD9335_MCLK_CLK_9P6MHZ 9600000 + +#define WCD9335_SLIM_CLOSE_TIMEOUT 1000 +#define WCD9335_SLIM_IRQ_OVERFLOW (1 << 0) +#define WCD9335_SLIM_IRQ_UNDERFLOW (1 << 1) +#define WCD9335_SLIM_IRQ_PORT_CLOSED (1 << 2) + +#define WCD9335_NUM_INTERPOLATORS 9 +#define WCD9335_RX_START 16 +#define WCD9335_SLIM_CH_START 128 + +#define WCD9335_SLIM_RX_CH(p) \ + {.port = p + WCD9335_RX_START, .shift = p,} + +/* vout step value */ +#define WCD9335_CALCULATE_VOUT_D(req_mv) (((req_mv - 650) * 10) / 25) + +enum { + WCD9335_RX0 = 0, + WCD9335_RX1, + WCD9335_RX2, + WCD9335_RX3, + WCD9335_RX4, + WCD9335_RX5, + WCD9335_RX6, + WCD9335_RX7, + WCD9335_RX8, + WCD9335_RX9, + WCD9335_RX10, + WCD9335_RX11, + WCD9335_RX12, + WCD9335_RX_MAX, +}; + +enum { + SIDO_SOURCE_INTERNAL = 0, + SIDO_SOURCE_RCO_BG, +}; + +enum wcd9335_sido_voltage { + SIDO_VOLTAGE_SVS_MV = 950, + SIDO_VOLTAGE_NOMINAL_MV = 1100, +}; + +enum { + AIF1_PB = 0, + AIF1_CAP, + AIF2_PB, + AIF2_CAP, + AIF3_PB, + AIF3_CAP, + AIF4_PB, + NUM_CODEC_DAIS, +}; + +enum { + INTn_2_INP_SEL_ZERO = 0, + INTn_2_INP_SEL_RX0, + INTn_2_INP_SEL_RX1, + INTn_2_INP_SEL_RX2, + INTn_2_INP_SEL_RX3, + INTn_2_INP_SEL_RX4, + INTn_2_INP_SEL_RX5, + INTn_2_INP_SEL_RX6, + INTn_2_INP_SEL_RX7, + INTn_2_INP_SEL_PROXIMITY, +}; + +enum { + INTn_1_MIX_INP_SEL_ZERO = 0, + INTn_1_MIX_INP_SEL_DEC0, + INTn_1_MIX_INP_SEL_DEC1, + INTn_1_MIX_INP_SEL_IIR0, + INTn_1_MIX_INP_SEL_IIR1, + INTn_1_MIX_INP_SEL_RX0, + INTn_1_MIX_INP_SEL_RX1, + INTn_1_MIX_INP_SEL_RX2, + INTn_1_MIX_INP_SEL_RX3, + INTn_1_MIX_INP_SEL_RX4, + INTn_1_MIX_INP_SEL_RX5, + INTn_1_MIX_INP_SEL_RX6, + INTn_1_MIX_INP_SEL_RX7, + +}; + +enum wcd_clock_type { + WCD_CLK_OFF, + WCD_CLK_RCO, + WCD_CLK_MCLK, +}; + +struct wcd9335_slim_ch { + u32 ch_num; + u16 port; + u16 shift; + struct list_head list; +}; + +struct wcd_slim_codec_dai_data { + struct list_head slim_ch_list; + struct slim_stream_config sconfig; + struct slim_stream_runtime *sruntime; +}; + +struct wcd9335_codec { + struct device *dev; + struct clk *mclk; + struct clk *native_clk; + u32 mclk_rate; + u8 intf_type; + u8 version; + + struct slim_device *slim; + struct slim_device *slim_ifd; + struct regmap *regmap; + struct regmap *if_regmap; + struct regmap_irq_chip_data *irq_data; + + struct wcd9335_slim_ch rx_chs[WCD9335_RX_MAX]; + u32 num_rx_port; + + int sido_input_src; + enum wcd9335_sido_voltage sido_voltage; + + struct wcd_slim_codec_dai_data dai[NUM_CODEC_DAIS]; + struct snd_soc_component *component; + + int master_bias_users; + int clk_mclk_users; + int clk_rco_users; + int sido_ccl_cnt; + enum wcd_clock_type clk_type; + + u32 hph_mode; +}; + +static const struct wcd9335_slim_ch wcd9335_rx_chs[WCD9335_RX_MAX] = { + WCD9335_SLIM_RX_CH(0), /* 16 */ + WCD9335_SLIM_RX_CH(1), /* 17 */ + WCD9335_SLIM_RX_CH(2), + WCD9335_SLIM_RX_CH(3), + WCD9335_SLIM_RX_CH(4), + WCD9335_SLIM_RX_CH(5), + WCD9335_SLIM_RX_CH(6), + WCD9335_SLIM_RX_CH(7), + WCD9335_SLIM_RX_CH(8), + WCD9335_SLIM_RX_CH(9), + WCD9335_SLIM_RX_CH(10), + WCD9335_SLIM_RX_CH(11), + WCD9335_SLIM_RX_CH(12), +}; + +struct interp_sample_rate { + int rate; + int rate_val; +}; + +static struct interp_sample_rate int_mix_rate_val[] = { + {48000, 0x4}, /* 48K */ + {96000, 0x5}, /* 96K */ + {192000, 0x6}, /* 192K */ +}; + +static struct interp_sample_rate int_prim_rate_val[] = { + {8000, 0x0}, /* 8K */ + {16000, 0x1}, /* 16K */ + {24000, -EINVAL},/* 24K */ + {32000, 0x3}, /* 32K */ + {48000, 0x4}, /* 48K */ + {96000, 0x5}, /* 96K */ + {192000, 0x6}, /* 192K */ + {384000, 0x7}, /* 384K */ + {44100, 0x8}, /* 44.1K */ +}; + +struct wcd9335_reg_mask_val { + u16 reg; + u8 mask; + u8 val; +}; + +static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_val_2_0[] = { + {WCD9335_RCO_CTRL_2, 0x0F, 0x08}, + {WCD9335_RX_BIAS_FLYB_MID_RST, 0xF0, 0x10}, + {WCD9335_FLYBACK_CTRL_1, 0x20, 0x20}, + {WCD9335_HPH_OCP_CTL, 0xFF, 0x5A}, + {WCD9335_HPH_L_TEST, 0x01, 0x01}, + {WCD9335_HPH_R_TEST, 0x01, 0x01}, + {WCD9335_CDC_BOOST0_BOOST_CFG1, 0x3F, 0x12}, + {WCD9335_CDC_BOOST0_BOOST_CFG2, 0x1C, 0x08}, + {WCD9335_CDC_COMPANDER7_CTL7, 0x1E, 0x18}, + {WCD9335_CDC_BOOST1_BOOST_CFG1, 0x3F, 0x12}, + {WCD9335_CDC_BOOST1_BOOST_CFG2, 0x1C, 0x08}, + {WCD9335_CDC_COMPANDER8_CTL7, 0x1E, 0x18}, + {WCD9335_CDC_TX0_TX_PATH_SEC7, 0xFF, 0x45}, + {WCD9335_CDC_RX0_RX_PATH_SEC0, 0xFC, 0xF4}, + {WCD9335_HPH_REFBUFF_LP_CTL, 0x08, 0x08}, + {WCD9335_HPH_REFBUFF_LP_CTL, 0x06, 0x02}, +}; + +static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_common_val[] = { + /* Rbuckfly/R_EAR(32) */ + {WCD9335_CDC_CLSH_K2_MSB, 0x0F, 0x00}, + {WCD9335_CDC_CLSH_K2_LSB, 0xFF, 0x60}, + {WCD9335_CPE_SS_DMIC_CFG, 0x80, 0x00}, + {WCD9335_CDC_BOOST0_BOOST_CTL, 0x70, 0x50}, + {WCD9335_CDC_BOOST1_BOOST_CTL, 0x70, 0x50}, + {WCD9335_CDC_RX7_RX_PATH_CFG1, 0x08, 0x08}, + {WCD9335_CDC_RX8_RX_PATH_CFG1, 0x08, 0x08}, + {WCD9335_ANA_LO_1_2, 0x3C, 0X3C}, + {WCD9335_DIFF_LO_COM_SWCAP_REFBUF_FREQ, 0x70, 0x00}, + {WCD9335_DIFF_LO_COM_PA_FREQ, 0x70, 0x40}, + {WCD9335_SOC_MAD_AUDIO_CTL_2, 0x03, 0x03}, + {WCD9335_CDC_TOP_TOP_CFG1, 0x02, 0x02}, + {WCD9335_CDC_TOP_TOP_CFG1, 0x01, 0x01}, + {WCD9335_EAR_CMBUFF, 0x08, 0x00}, + {WCD9335_CDC_TX9_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX10_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX11_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX12_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_COMPANDER7_CTL3, 0x80, 0x80}, + {WCD9335_CDC_COMPANDER8_CTL3, 0x80, 0x80}, + {WCD9335_CDC_COMPANDER7_CTL7, 0x01, 0x01}, + {WCD9335_CDC_COMPANDER8_CTL7, 0x01, 0x01}, + {WCD9335_CDC_RX0_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX1_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX2_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX3_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX4_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX5_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX6_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX7_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX8_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX0_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX1_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX2_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX3_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX4_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX5_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX6_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX7_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX8_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_VBADC_IBIAS_FE, 0x0C, 0x08}, +}; + +static int wcd9335_set_mix_interpolator_rate(struct snd_soc_dai *dai, + int rate_val, + u32 rate) +{ + struct snd_soc_component *component = dai->component; + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + struct wcd9335_slim_ch *ch; + int val, j; + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { + for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { + val = snd_soc_component_read32(component, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)) & + WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + + if (val == (ch->shift + INTn_2_INP_SEL_RX0)) + snd_soc_component_update_bits(component, + WCD9335_CDC_RX_PATH_MIX_CTL(j), + WCD9335_CDC_MIX_PCM_RATE_MASK, + rate_val); + } + } + + return 0; +} + +static int wcd9335_set_prim_interpolator_rate(struct snd_soc_dai *dai, + u8 rate_val, + u32 rate) +{ + struct snd_soc_component *comp = dai->component; + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + struct wcd9335_slim_ch *ch; + u8 cfg0, cfg1, inp0_sel, inp1_sel, inp2_sel; + int inp, j; + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { + inp = ch->shift + INTn_1_MIX_INP_SEL_RX0; + /* + * Loop through all interpolator MUX inputs and find out + * to which interpolator input, the slim rx port + * is connected + */ + for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { + cfg0 = snd_soc_component_read32(comp, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG0(j)); + cfg1 = snd_soc_component_read32(comp, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)); + + inp0_sel = cfg0 & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + inp1_sel = (cfg0 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + inp2_sel = (cfg1 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + + if ((inp0_sel == inp) || (inp1_sel == inp) || + (inp2_sel == inp)) { + /* rate is in Hz */ + if ((j == 0) && (rate == 44100)) + dev_info(wcd->dev, + "Cannot set 44.1KHz on INT0\n"); + else + snd_soc_component_update_bits(comp, + WCD9335_CDC_RX_PATH_CTL(j), + WCD9335_CDC_MIX_PCM_RATE_MASK, + rate_val); + } + } + } + + return 0; +} + +static int wcd9335_set_interpolator_rate(struct snd_soc_dai *dai, u32 rate) +{ + int i; + + /* set mixing path rate */ + for (i = 0; i < ARRAY_SIZE(int_mix_rate_val); i++) { + if (rate == int_mix_rate_val[i].rate) { + wcd9335_set_mix_interpolator_rate(dai, + int_mix_rate_val[i].rate_val, rate); + break; + } + } + + /* set primary path sample rate */ + for (i = 0; i < ARRAY_SIZE(int_prim_rate_val); i++) { + if (rate == int_prim_rate_val[i].rate) { + wcd9335_set_prim_interpolator_rate(dai, + int_prim_rate_val[i].rate_val, rate); + break; + } + } + + return 0; +} + +static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, + struct wcd_slim_codec_dai_data *dai_data, + int direction) +{ + struct list_head *slim_ch_list = &dai_data->slim_ch_list; + struct slim_stream_config *cfg = &dai_data->sconfig; + struct wcd9335_slim_ch *ch; + u16 payload = 0; + int ret, i; + + cfg->ch_count = 0; + cfg->direction = direction; + cfg->port_mask = 0; + + /* Configure slave interface device */ + list_for_each_entry(ch, slim_ch_list, list) { + cfg->ch_count++; + payload |= 1 << ch->shift; + cfg->port_mask |= BIT(ch->port); + } + + cfg->chs = kcalloc(cfg->ch_count, sizeof(unsigned int), GFP_KERNEL); + if (!cfg->chs) + return -ENOMEM; + + i = 0; + list_for_each_entry(ch, slim_ch_list, list) { + cfg->chs[i++] = ch->ch_num; + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + /* write to interface device */ + ret = regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), + payload); + + if (ret < 0) + goto err; + + /* configure the slave port for water mark and enable*/ + ret = regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_RX_PORT_CFG(ch->port), + WCD9335_SLIM_WATER_MARK_VAL); + if (ret < 0) + goto err; + } + } + + dai_data->sruntime = slim_stream_allocate(wcd->slim, "WCD9335-SLIM"); + slim_stream_prepare(dai_data->sruntime, cfg); + + return 0; + +err: + dev_err(wcd->dev, "Error Setting slim hw params\n"); + kfree(cfg->chs); + cfg->chs = NULL; + + return ret; +} + +static int wcd9335_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wcd9335_codec *wcd; + int ret; + + wcd = snd_soc_component_get_drvdata(dai->component); + + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ret = wcd9335_set_interpolator_rate(dai, params_rate(params)); + if (ret) { + dev_err(wcd->dev, "cannot set sample rate: %u\n", + params_rate(params)); + return ret; + } + switch (params_width(params)) { + case 16 ... 24: + wcd->dai[dai->id].sconfig.bps = params_width(params); + break; + default: + dev_err(wcd->dev, "%s: Invalid format 0x%x\n", + __func__, params_width(params)); + return -EINVAL; + } + break; + default: + dev_err(wcd->dev, "Invalid stream type %d\n", + substream->stream); + return -EINVAL; + }; + + wcd->dai[dai->id].sconfig.rate = params_rate(params); + wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); + + return 0; +} + +static int wcd9335_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wcd_slim_codec_dai_data *dai_data; + struct wcd9335_codec *wcd; + + wcd = snd_soc_component_get_drvdata(dai->component); + dai_data = &wcd->dai[dai->id]; + slim_stream_enable(dai_data->sruntime); + + return 0; +} + +static int wcd9335_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct wcd9335_codec *wcd; + int i; + + wcd = snd_soc_component_get_drvdata(dai->component); + + if (!tx_slot || !rx_slot) { + dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", + tx_slot, rx_slot); + return -EINVAL; + } + + if (wcd->rx_chs) { + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); + } + } + + return 0; +} + +static int wcd9335_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + struct wcd9335_slim_ch *ch; + struct wcd9335_codec *wcd; + int i = 0; + + wcd = snd_soc_component_get_drvdata(dai->component); + + switch (dai->id) { + case AIF1_PB: + case AIF2_PB: + case AIF3_PB: + case AIF4_PB: + if (!rx_slot || !rx_num) { + dev_err(wcd->dev, "Invalid rx_slot %p or rx_num %p\n", + rx_slot, rx_num); + return -EINVAL; + } + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) + rx_slot[i++] = ch->ch_num; + + *rx_num = i; + break; + default: + dev_err(wcd->dev, "Invalid DAI ID %x\n", dai->id); + break; + } + + return 0; +} + +static struct snd_soc_dai_ops wcd9335_dai_ops = { + .hw_params = wcd9335_hw_params, + .prepare = wcd9335_prepare, + .set_channel_map = wcd9335_set_channel_map, + .get_channel_map = wcd9335_get_channel_map, +}; + +static struct snd_soc_dai_driver wcd9335_slim_dais[] = { + [0] = { + .name = "wcd9335_rx1", + .id = AIF1_PB, + .playback = { + .stream_name = "AIF1 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_max = 192000, + .rate_min = 8000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [1] = { + .name = "wcd9335_tx1", + .id = AIF1_CAP, + .capture = { + .stream_name = "AIF1 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [2] = { + .name = "wcd9335_rx2", + .id = AIF2_PB, + .playback = { + .stream_name = "AIF2 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [3] = { + .name = "wcd9335_tx2", + .id = AIF2_CAP, + .capture = { + .stream_name = "AIF2 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [4] = { + .name = "wcd9335_rx3", + .id = AIF3_PB, + .playback = { + .stream_name = "AIF3 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [5] = { + .name = "wcd9335_tx3", + .id = AIF3_CAP, + .capture = { + .stream_name = "AIF3 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [6] = { + .name = "wcd9335_rx4", + .id = AIF4_PB, + .playback = { + .stream_name = "AIF4 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, +}; + +static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) +{ + struct wcd9335_codec *wcd = data; + unsigned long status = 0; + int i, j, port_id; + unsigned int val, int_val = 0; + bool tx; + unsigned short reg = 0; + + for (i = WCD9335_SLIM_PGD_PORT_INT_STATUS_RX_0, j = 0; + i <= WCD9335_SLIM_PGD_PORT_INT_STATUS_TX_1; i++, j++) { + regmap_read(wcd->if_regmap, i, &val); + status |= ((u32)val << (8 * j)); + } + + for_each_set_bit(j, &status, 32) { + tx = (j >= 16 ? true : false); + port_id = (tx ? j - 16 : j); + regmap_read(wcd->if_regmap, + WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); + if (val) { + if (!tx) + reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + + (port_id / 8); + else + reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + + (port_id / 8); + regmap_read( + wcd->if_regmap, reg, &int_val); + /* + * Ignore interrupts for ports for which the + * interrupts are not specifically enabled. + */ + if (!(int_val & (1 << (port_id % 8)))) + continue; + } + if (val & WCD9335_SLIM_IRQ_OVERFLOW) + dev_err_ratelimited(wcd->dev, + "%s: overflow error on %s port %d, value %x\n", + __func__, (tx ? "TX" : "RX"), port_id, val); + if (val & WCD9335_SLIM_IRQ_UNDERFLOW) + dev_err_ratelimited(wcd->dev, + "%s: underflow error on %s port %d, value %x\n", + __func__, (tx ? "TX" : "RX"), port_id, val); + if ((val & WCD9335_SLIM_IRQ_OVERFLOW) || + (val & WCD9335_SLIM_IRQ_UNDERFLOW)) { + if (!tx) + reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + + (port_id / 8); + else + reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + + (port_id / 8); + regmap_read( + wcd->if_regmap, reg, &int_val); + if (int_val & (1 << (port_id % 8))) { + int_val = int_val ^ (1 << (port_id % 8)); + regmap_write(wcd->if_regmap, + reg, int_val); + } + } + + regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_PORT_INT_CLR_RX_0 + (j / 8), + BIT(j % 8)); + } + + return IRQ_HANDLED; +} + +static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) +{ + int slim_irq = regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS); + int i, ret = 0; + + ret = request_threaded_irq(slim_irq, NULL, wcd9335_slimbus_irq, + IRQF_TRIGGER_RISING, "SLIMBus Slave", wcd); + if (ret) { + dev_err(wcd->dev, "Failed to request SLIMBUS irq\n"); + return ret; + } + + /* enable interrupts on all slave ports */ + for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) + regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, + 0xFF); + + return ret; +} + +static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, + bool ccl_flag) +{ + struct snd_soc_component *comp = wcd->component; + + if (ccl_flag) { + if (++wcd->sido_ccl_cnt == 1) + snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, + WCD9335_SIDO_SIDO_CCL_DEF_VALUE); + } else { + if (wcd->sido_ccl_cnt == 0) { + dev_err(wcd->dev, "sido_ccl already disabled\n"); + return; + } + if (--wcd->sido_ccl_cnt == 0) + snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, + WCD9335_SIDO_SIDO_CCL_10_ICHARG_PWR_SEL_C320FF); + } +} + +static int wcd9335_enable_master_bias(struct wcd9335_codec *wcd) +{ + wcd->master_bias_users++; + if (wcd->master_bias_users == 1) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_EN_MASK, + WCD9335_ANA_BIAS_ENABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_EN_MASK, + WCD9335_ANA_BIAS_PRECHRG_ENABLE); + /* + * 1ms delay is required after pre-charge is enabled + * as per HW requirement + */ + usleep_range(1000, 1100); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_EN_MASK, + WCD9335_ANA_BIAS_PRECHRG_DISABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); + } + + return 0; +} + +static int wcd9335_enable_mclk(struct wcd9335_codec *wcd) +{ + /* Enable mclk requires master bias to be enabled first */ + if (wcd->master_bias_users <= 0) + return -EINVAL; + + if (((wcd->clk_mclk_users == 0) && (wcd->clk_type == WCD_CLK_MCLK)) || + ((wcd->clk_mclk_users > 0) && (wcd->clk_type != WCD_CLK_MCLK))) { + dev_err(wcd->dev, "Error enabling MCLK, clk_type: %d\n", + wcd->clk_type); + return -EINVAL; + } + + if (++wcd->clk_mclk_users == 1) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, + WCD9335_ANA_CLK_EXT_CLKBUF_ENABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_SRC_MASK, + WCD9335_ANA_CLK_MCLK_SRC_EXTERNAL); + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_EN_MASK, + WCD9335_ANA_CLK_MCLK_ENABLE); + regmap_update_bits(wcd->regmap, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_CONTROL, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_EN_MASK, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE); + regmap_update_bits(wcd->regmap, + WCD9335_CDC_CLK_RST_CTRL_MCLK_CONTROL, + WCD9335_CDC_CLK_RST_CTRL_MCLK_EN_MASK, + WCD9335_CDC_CLK_RST_CTRL_MCLK_ENABLE); + /* + * 10us sleep is required after clock is enabled + * as per HW requirement + */ + usleep_range(10, 15); + } + + wcd->clk_type = WCD_CLK_MCLK; + + return 0; +} + +static int wcd9335_disable_mclk(struct wcd9335_codec *wcd) +{ + if (wcd->clk_mclk_users <= 0) + return -EINVAL; + + if (--wcd->clk_mclk_users == 0) { + if (wcd->clk_rco_users > 0) { + /* MCLK to RCO switch */ + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_SRC_MASK, + WCD9335_ANA_CLK_MCLK_SRC_RCO); + wcd->clk_type = WCD_CLK_RCO; + } else { + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_EN_MASK, + WCD9335_ANA_CLK_MCLK_DISABLE); + wcd->clk_type = WCD_CLK_OFF; + } + + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, + WCD9335_ANA_CLK_EXT_CLKBUF_DISABLE); + } + + return 0; +} + +static int wcd9335_disable_master_bias(struct wcd9335_codec *wcd) +{ + if (wcd->master_bias_users <= 0) + return -EINVAL; + + wcd->master_bias_users--; + if (wcd->master_bias_users == 0) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_EN_MASK, + WCD9335_ANA_BIAS_DISABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); + } + return 0; +} + +static int wcd9335_cdc_req_mclk_enable(struct wcd9335_codec *wcd, + bool enable) +{ + int ret = 0; + + if (enable) { + wcd9335_cdc_sido_ccl_enable(wcd, true); + ret = clk_prepare_enable(wcd->mclk); + if (ret) { + dev_err(wcd->dev, "%s: ext clk enable failed\n", + __func__); + goto err; + } + /* get BG */ + wcd9335_enable_master_bias(wcd); + /* get MCLK */ + wcd9335_enable_mclk(wcd); + + } else { + /* put MCLK */ + wcd9335_disable_mclk(wcd); + /* put BG */ + wcd9335_disable_master_bias(wcd); + clk_disable_unprepare(wcd->mclk); + wcd9335_cdc_sido_ccl_enable(wcd, false); + } +err: + return ret; +} + +static void wcd9335_codec_apply_sido_voltage(struct wcd9335_codec *wcd, + enum wcd9335_sido_voltage req_mv) +{ + struct snd_soc_component *comp = wcd->component; + int vout_d_val; + + if (req_mv == wcd->sido_voltage) + return; + + /* compute the vout_d step value */ + vout_d_val = WCD9335_CALCULATE_VOUT_D(req_mv) & + WCD9335_ANA_BUCK_VOUT_MASK; + snd_soc_component_write(comp, WCD9335_ANA_BUCK_VOUT_D, vout_d_val); + snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, + WCD9335_ANA_BUCK_CTL_RAMP_START_ENABLE); + + /* 1 msec sleep required after SIDO Vout_D voltage change */ + usleep_range(1000, 1100); + wcd->sido_voltage = req_mv; + snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, + WCD9335_ANA_BUCK_CTL_RAMP_START_DISABLE); +} + +static int wcd9335_codec_update_sido_voltage(struct wcd9335_codec *wcd, + enum wcd9335_sido_voltage req_mv) +{ + int ret = 0; + + /* enable mclk before setting SIDO voltage */ + ret = wcd9335_cdc_req_mclk_enable(wcd, true); + if (ret) { + dev_err(wcd->dev, "Ext clk enable failed\n"); + goto err; + } + + wcd9335_codec_apply_sido_voltage(wcd, req_mv); + wcd9335_cdc_req_mclk_enable(wcd, false); + +err: + return ret; +} + +static int _wcd9335_codec_enable_mclk(struct snd_soc_component *component, + int enable) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int ret; + + if (enable) { + ret = wcd9335_cdc_req_mclk_enable(wcd, true); + if (ret) + return ret; + + wcd9335_codec_apply_sido_voltage(wcd, + SIDO_VOLTAGE_NOMINAL_MV); + } else { + wcd9335_codec_update_sido_voltage(wcd, + wcd->sido_voltage); + wcd9335_cdc_req_mclk_enable(wcd, false); + } + + return 0; +} + +static void wcd9335_enable_sido_buck(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + + snd_soc_component_update_bits(component, WCD9335_ANA_RCO, + WCD9335_ANA_RCO_BG_EN_MASK, + WCD9335_ANA_RCO_BG_ENABLE); + snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_MASK, + WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_EXT); + /* 100us sleep needed after IREF settings */ + usleep_range(100, 110); + snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_MASK, + WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_EXT); + /* 100us sleep needed after VREF settings */ + usleep_range(100, 110); + wcd->sido_input_src = SIDO_SOURCE_RCO_BG; +} + +static int wcd9335_enable_efuse_sensing(struct snd_soc_component *comp) +{ + _wcd9335_codec_enable_mclk(comp, true); + snd_soc_component_update_bits(comp, + WCD9335_CHIP_TIER_CTRL_EFUSE_CTL, + WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK, + WCD9335_CHIP_TIER_CTRL_EFUSE_ENABLE); + /* + * 5ms sleep required after enabling efuse control + * before checking the status. + */ + usleep_range(5000, 5500); + + if (!(snd_soc_component_read32(comp, + WCD9335_CHIP_TIER_CTRL_EFUSE_STATUS) & + WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK)) + WARN(1, "%s: Efuse sense is not complete\n", __func__); + + wcd9335_enable_sido_buck(comp); + _wcd9335_codec_enable_mclk(comp, false); + + return 0; +} + +static void wcd9335_codec_init(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int i; + + /* ungate MCLK and set clk rate */ + regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_GATE, + WCD9335_CODEC_RPM_CLK_GATE_MCLK_GATE_MASK, 0); + + regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); + + for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_common_val); i++) + snd_soc_component_update_bits(component, + wcd9335_codec_reg_init_common_val[i].reg, + wcd9335_codec_reg_init_common_val[i].mask, + wcd9335_codec_reg_init_common_val[i].val); + + if (WCD9335_IS_2_0(wcd->version)) { + for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_val_2_0); i++) + snd_soc_component_update_bits(component, + wcd9335_codec_reg_init_val_2_0[i].reg, + wcd9335_codec_reg_init_val_2_0[i].mask, + wcd9335_codec_reg_init_val_2_0[i].val); + } + + wcd9335_enable_efuse_sensing(component); +} + +static int wcd9335_codec_probe(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int i; + + snd_soc_component_init_regmap(component, wcd->regmap); + wcd->component = component; + + wcd9335_codec_init(component); + + for (i = 0; i < NUM_CODEC_DAIS; i++) + INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); + + return wcd9335_setup_irqs(wcd); +} + +static void wcd9335_codec_remove(struct snd_soc_component *comp) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + + free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); +} + +static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, + int clk_id, int source, + unsigned int freq, int dir) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + + wcd->mclk_rate = freq; + + if (wcd->mclk_rate == WCD9335_MCLK_CLK_12P288MHZ) + snd_soc_component_update_bits(comp, + WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_12P288MHZ); + else if (wcd->mclk_rate == WCD9335_MCLK_CLK_9P6MHZ) + snd_soc_component_update_bits(comp, + WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); + + return clk_set_rate(wcd->mclk, freq); +} + +static const struct snd_soc_component_driver wcd9335_component_drv = { + .probe = wcd9335_codec_probe, + .remove = wcd9335_codec_remove, + .set_sysclk = wcd9335_codec_set_sysclk, +}; + +static int wcd9335_probe(struct platform_device *pdev) +{ + struct wcd9335 *pdata = dev_get_drvdata(pdev->dev.parent); + struct device *dev = &pdev->dev; + struct wcd9335_codec *wcd; + + wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); + if (!wcd) + return -ENOMEM; + + dev_set_drvdata(dev, wcd); + + memcpy(wcd->rx_chs, wcd9335_rx_chs, sizeof(wcd9335_rx_chs)); + + wcd->regmap = pdata->regmap; + wcd->if_regmap = pdata->ifd_regmap; + wcd->slim = pdata->slim; + wcd->slim_ifd = pdata->slim_ifd; + wcd->irq_data = pdata->irq_data; + wcd->version = pdata->version; + wcd->intf_type = pdata->intf_type; + wcd->dev = dev; + wcd->mclk = pdata->mclk; + wcd->native_clk = pdata->native_clk; + wcd->sido_input_src = SIDO_SOURCE_INTERNAL; + wcd->sido_voltage = SIDO_VOLTAGE_NOMINAL_MV; + + return devm_snd_soc_register_component(dev, &wcd9335_component_drv, + wcd9335_slim_dais, + ARRAY_SIZE(wcd9335_slim_dais)); +} + +static struct platform_driver wcd9335_codec_driver = { + .probe = wcd9335_probe, + .driver = { + .name = "wcd9335-codec", + }, +}; +module_platform_driver(wcd9335_codec_driver); +MODULE_DESCRIPTION("WCD9335 Codec driver"); +MODULE_LICENSE("GPL v2"); From c8cb5f775c8dacb605e628a320ded42be3bd9453 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 27 Jul 2018 13:18:01 +0100 Subject: [PATCH 418/529] ASoC: wcd9335: add CLASS-H Controller support CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series. This patchset adds basic CLASS-H controller apis for WCD codecs after wcd9335 to use. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wcd-clsh.c | 605 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wcd-clsh.h | 49 +++ sound/soc/codecs/wcd9335.c | 10 + 4 files changed, 665 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 01410b63daac..e3a3d4694e15 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd9335.o +snd-soc-wcd9335-objs := wcd-clsh.o wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd-clsh.c b/sound/soc/codecs/wcd-clsh.c new file mode 100644 index 000000000000..2393456cbd97 --- /dev/null +++ b/sound/soc/codecs/wcd-clsh.c @@ -0,0 +1,605 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +// Copyright (c) 2017-2018, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include "wcd-clsh.h" + +struct wcd_clsh_ctrl { + int state; + int mode; + int flyback_users; + int buck_users; + int clsh_users; + int codec_version; + struct snd_soc_component *comp; +}; + +/* Class-H registers for codecs from and above WCD9335 */ +#define WCD9XXX_A_CDC_RX0_RX_PATH_CFG0 WCD9335_REG(0xB, 0x42) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK BIT(6) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE BIT(6) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE 0 +#define WCD9XXX_A_CDC_RX1_RX_PATH_CFG0 WCD9335_REG(0xB, 0x56) +#define WCD9XXX_A_CDC_RX2_RX_PATH_CFG0 WCD9335_REG(0xB, 0x6A) +#define WCD9XXX_A_CDC_CLSH_K1_MSB WCD9335_REG(0xC, 0x08) +#define WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK GENMASK(3, 0) +#define WCD9XXX_A_CDC_CLSH_K1_LSB WCD9335_REG(0xC, 0x09) +#define WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK GENMASK(7, 0) +#define WCD9XXX_A_ANA_RX_SUPPLIES WCD9335_REG(0x6, 0x08) +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK BIT(1) +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H 0 +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB BIT(1) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK BIT(2) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA BIT(2) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT 0 +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK BIT(3) +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA BIT(3) +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT 0 +#define WCD9XXX_A_ANA_RX_VNEG_EN_MASK BIT(6) +#define WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT 6 +#define WCD9XXX_A_ANA_RX_VNEG_ENABLE BIT(6) +#define WCD9XXX_A_ANA_RX_VNEG_DISABLE 0 +#define WCD9XXX_A_ANA_RX_VPOS_EN_MASK BIT(7) +#define WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT 7 +#define WCD9XXX_A_ANA_RX_VPOS_ENABLE BIT(7) +#define WCD9XXX_A_ANA_RX_VPOS_DISABLE 0 +#define WCD9XXX_A_ANA_HPH WCD9335_REG(0x6, 0x09) +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK GENMASK(3, 2) +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA 0x08 +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP 0x04 +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL 0x0 +#define WCD9XXX_A_CDC_CLSH_CRC WCD9335_REG(0xC, 0x01) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK BIT(0) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_ENABLE BIT(0) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_DISABLE 0 +#define WCD9XXX_FLYBACK_EN WCD9335_REG(0x6, 0xA4) +#define WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK GENMASK(6, 5) +#define WCD9XXX_FLYBACK_EN_DELAY_26P25_US 0x40 +#define WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK BIT(4) +#define WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY BIT(4) +#define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY 0 +#define WCD9XXX_RX_BIAS_FLYB_BUFF WCD9335_REG(0x6, 0xC7) +#define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK GENMASK(7, 4) +#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK GENMASK(0, 3) +#define WCD9XXX_HPH_L_EN WCD9335_REG(0x6, 0xD3) +#define WCD9XXX_HPH_CONST_SEL_L_MASK GENMASK(7, 3) +#define WCD9XXX_HPH_CONST_SEL_BYPASS 0 +#define WCD9XXX_HPH_CONST_SEL_LP_PATH 0x40 +#define WCD9XXX_HPH_CONST_SEL_HQ_PATH 0x80 +#define WCD9XXX_HPH_R_EN WCD9335_REG(0x6, 0xD6) +#define WCD9XXX_HPH_REFBUFF_UHQA_CTL WCD9335_REG(0x6, 0xDD) +#define WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK GENMASK(2, 0) +#define WCD9XXX_CLASSH_CTRL_VCL_2 WCD9335_REG(0x6, 0x9B) +#define WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK GENMASK(5, 4) +#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM 0x20 +#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM 0x0 +#define WCD9XXX_CDC_RX1_RX_PATH_CTL WCD9335_REG(0xB, 0x55) +#define WCD9XXX_CDC_RX2_RX_PATH_CTL WCD9335_REG(0xB, 0x69) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_CONTROL WCD9335_REG(0xD, 0x41) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_EN_MASK BIT(0) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_11P3_EN_MASK BIT(1) +#define WCD9XXX_CLASSH_CTRL_CCL_1 WCD9335_REG(0x6, 0x9C) +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK GENMASK(7, 4) +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA 0x50 +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA 0x30 + +#define CLSH_REQ_ENABLE true +#define CLSH_REQ_DISABLE false +#define WCD_USLEEP_RANGE 50 + +enum { + DAC_GAIN_0DB = 0, + DAC_GAIN_0P2DB, + DAC_GAIN_0P4DB, + DAC_GAIN_0P6DB, + DAC_GAIN_0P8DB, + DAC_GAIN_M0P2DB, + DAC_GAIN_M0P4DB, + DAC_GAIN_M0P6DB, +}; + +static inline void wcd_enable_clsh_block(struct wcd_clsh_ctrl *ctrl, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + if ((enable && ++ctrl->clsh_users == 1) || + (!enable && --ctrl->clsh_users == 0)) + snd_soc_component_update_bits(comp, WCD9XXX_A_CDC_CLSH_CRC, + WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK, + enable); + if (ctrl->clsh_users < 0) + ctrl->clsh_users = 0; +} + +static inline bool wcd_clsh_enable_status(struct snd_soc_component *comp) +{ + return snd_soc_component_read32(comp, WCD9XXX_A_CDC_CLSH_CRC) & + WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK; +} + +static inline void wcd_clsh_set_buck_mode(struct snd_soc_component *comp, + int mode) +{ + /* set to HIFI */ + if (mode == CLS_H_HIFI) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT); +} + +static inline void wcd_clsh_set_flyback_mode(struct snd_soc_component *comp, + int mode) +{ + /* set to HIFI */ + if (mode == CLS_H_HIFI) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT); +} + +static void wcd_clsh_buck_ctrl(struct wcd_clsh_ctrl *ctrl, + int mode, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + /* enable/disable buck */ + if ((enable && (++ctrl->buck_users == 1)) || + (!enable && (--ctrl->buck_users == 0))) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_EN_MASK, + enable << WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT); + /* + * 500us sleep is required after buck enable/disable + * as per HW requirement + */ + usleep_range(500, 500 + WCD_USLEEP_RANGE); +} + +static void wcd_clsh_flyback_ctrl(struct wcd_clsh_ctrl *ctrl, + int mode, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + int vneg[] = {0x00, 0x40}; + + /* enable/disable flyback */ + if ((enable && (++ctrl->flyback_users == 1)) || + (!enable && (--ctrl->flyback_users == 0))) { + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_EN_MASK, + enable << WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT); + /* 100usec delay is needed as per HW requirement */ + usleep_range(100, 110); + + if (enable && (WCD9335_IS_1_1(ctrl->codec_version))) { + wcd_clsh_set_flyback_mode(comp, CLS_H_HIFI); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK, + WCD9XXX_FLYBACK_EN_DELAY_26P25_US); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, + WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY); + vneg[0] = snd_soc_component_read32(comp, + WCD9XXX_A_ANA_RX_SUPPLIES); + vneg[0] &= ~(0x40); + vneg[1] = vneg[0] | 0x40; + + snd_soc_component_write(comp, + WCD9XXX_A_ANA_RX_SUPPLIES, vneg[0]); + snd_soc_component_write(comp, + WCD9XXX_A_ANA_RX_SUPPLIES, vneg[1]); + /* 500usec delay is needed as per HW requirement */ + usleep_range(500, 510); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, + WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY); + wcd_clsh_set_flyback_mode(comp, mode); + } + + } + /* + * 500us sleep is required after flyback enable/disable + * as per HW requirement + */ + usleep_range(500, 500 + WCD_USLEEP_RANGE); +} + +static void wcd_clsh_set_gain_path(struct wcd_clsh_ctrl *ctrl, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + int val = 0; + + switch (mode) { + case CLS_H_NORMAL: + case CLS_AB: + val = WCD9XXX_HPH_CONST_SEL_BYPASS; + break; + case CLS_H_HIFI: + val = WCD9XXX_HPH_CONST_SEL_HQ_PATH; + break; + case CLS_H_LP: + val = WCD9XXX_HPH_CONST_SEL_LP_PATH; + break; + }; + + snd_soc_component_update_bits(comp, WCD9XXX_HPH_L_EN, + WCD9XXX_HPH_CONST_SEL_L_MASK, + val); + + snd_soc_component_update_bits(comp, WCD9XXX_HPH_R_EN, + WCD9XXX_HPH_CONST_SEL_L_MASK, + val); +} + +static void wcd_clsh_set_hph_mode(struct snd_soc_component *comp, + int mode) +{ + int val = 0, gain = 0, res_val; + int ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + + res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM; + switch (mode) { + case CLS_H_NORMAL: + res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM; + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; + gain = DAC_GAIN_0DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_AB: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; + gain = DAC_GAIN_0DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_H_HIFI: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA; + gain = DAC_GAIN_M0P2DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_H_LP: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA; + break; + }; + + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_HPH, + WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK, val); + snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_VCL_2, + WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK, + res_val); + if (mode != CLS_H_LP) + snd_soc_component_update_bits(comp, + WCD9XXX_HPH_REFBUFF_UHQA_CTL, + WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK, + gain); + snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_CCL_1, + WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK, + ipeak); +} + +static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, + int mode) +{ + + snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, + WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK, 0x0A); + snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, + WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK, 0x0A); + /* Sleep needed to avoid click and pop as per HW requirement */ + usleep_range(100, 110); +} + +static void wcd_clsh_set_buck_regulator_mode(struct snd_soc_component *comp, + int mode) +{ + if (mode == CLS_AB) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H); +} + +static void wcd_clsh_state_lo(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode != CLS_AB) { + dev_err(comp->dev, "%s: LO cannot be in this mode: %d\n", + __func__, mode); + return; + } + + if (is_enable) { + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + } else { + wcd_clsh_buck_ctrl(ctrl, mode, false); + wcd_clsh_flyback_ctrl(ctrl, mode, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_hph_r(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode == CLS_H_NORMAL) { + dev_err(comp->dev, "%s: Normal mode not applicable for hph_r\n", + __func__); + return; + } + + if (is_enable) { + if (mode != CLS_AB) { + wcd_enable_clsh_block(ctrl, true); + /* + * These K1 values depend on the Headphone Impedance + * For now it is assumed to be 16 ohm + */ + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_MSB, + WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, + 0x00); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_LSB, + WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, + 0xC0); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + } + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + wcd_clsh_set_hph_mode(comp, mode); + wcd_clsh_set_gain_path(ctrl, mode); + } else { + wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); + + if (mode != CLS_AB) { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + } + /* buck and flyback set to default mode and disable */ + wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_hph_l(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode == CLS_H_NORMAL) { + dev_err(comp->dev, "%s: Normal mode not applicable for hph_l\n", + __func__); + return; + } + + if (is_enable) { + if (mode != CLS_AB) { + wcd_enable_clsh_block(ctrl, true); + /* + * These K1 values depend on the Headphone Impedance + * For now it is assumed to be 16 ohm + */ + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_MSB, + WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, + 0x00); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_LSB, + WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, + 0xC0); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + } + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + wcd_clsh_set_hph_mode(comp, mode); + wcd_clsh_set_gain_path(ctrl, mode); + } else { + wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); + + if (mode != CLS_AB) { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + } + /* set buck and flyback to Default Mode */ + wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_ear(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode != CLS_H_NORMAL) { + dev_err(comp->dev, "%s: mode: %d cannot be used for EAR\n", + __func__, mode); + return; + } + + if (is_enable) { + wcd_enable_clsh_block(ctrl, true); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + } else { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + wcd_clsh_buck_ctrl(ctrl, mode, false); + wcd_clsh_flyback_ctrl(ctrl, mode, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + } +} + +static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + switch (req_state) { + case WCD_CLSH_STATE_EAR: + wcd_clsh_state_ear(ctrl, req_state, is_enable, mode); + break; + case WCD_CLSH_STATE_HPHL: + wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); + break; + case WCD_CLSH_STATE_HPHR: + wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); + break; + break; + case WCD_CLSH_STATE_LO: + wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); + break; + default: + break; + } + + return 0; +} + +/* + * Function: wcd_clsh_is_state_valid + * Params: state + * Description: + * Provides information on valid states of Class H configuration + */ +static bool wcd_clsh_is_state_valid(int state) +{ + switch (state) { + case WCD_CLSH_STATE_IDLE: + case WCD_CLSH_STATE_EAR: + case WCD_CLSH_STATE_HPHL: + case WCD_CLSH_STATE_HPHR: + case WCD_CLSH_STATE_LO: + return true; + default: + return false; + }; +} + +/* + * Function: wcd_clsh_fsm + * Params: ctrl, req_state, req_type, clsh_event + * Description: + * This function handles PRE DAC and POST DAC conditions of different devices + * and updates class H configuration of different combination of devices + * based on validity of their states. ctrl will contain current + * class h state information + */ +int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, + enum wcd_clsh_event clsh_event, + int nstate, + enum wcd_clsh_mode mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (nstate == ctrl->state) + return 0; + + if (!wcd_clsh_is_state_valid(nstate)) { + dev_err(comp->dev, "Class-H not a valid new state:\n"); + return -EINVAL; + } + + switch (clsh_event) { + case WCD_CLSH_EVENT_PRE_DAC: + _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_ENABLE, mode); + break; + case WCD_CLSH_EVENT_POST_PA: + _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_DISABLE, mode); + break; + }; + + ctrl->state = nstate; + ctrl->mode = mode; + + return 0; +} + +int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) +{ + return ctrl->state; +} + +struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, + int version) +{ + struct wcd_clsh_ctrl *ctrl; + + ctrl = kzalloc(sizeof(*ctrl), GFP_KERNEL); + if (!ctrl) + return ERR_PTR(-ENOMEM); + + ctrl->state = WCD_CLSH_STATE_IDLE; + ctrl->comp = comp; + + return ctrl; +} + +void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) +{ + kfree(ctrl); +} diff --git a/sound/soc/codecs/wcd-clsh.h b/sound/soc/codecs/wcd-clsh.h new file mode 100644 index 000000000000..a902f9893467 --- /dev/null +++ b/sound/soc/codecs/wcd-clsh.h @@ -0,0 +1,49 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef _WCD_CLSH_V2_H_ +#define _WCD_CLSH_V2_H_ +#include + +enum wcd_clsh_event { + WCD_CLSH_EVENT_PRE_DAC = 1, + WCD_CLSH_EVENT_POST_PA, +}; + +/* + * Basic states for Class H state machine. + * represented as a bit mask within a u8 data type + * bit 0: EAR mode + * bit 1: HPH Left mode + * bit 2: HPH Right mode + * bit 3: Lineout mode + */ +#define WCD_CLSH_STATE_IDLE 0 +#define WCD_CLSH_STATE_EAR BIT(0) +#define WCD_CLSH_STATE_HPHL BIT(1) +#define WCD_CLSH_STATE_HPHR BIT(2) +#define WCD_CLSH_STATE_LO BIT(3) +#define WCD_CLSH_STATE_MAX 4 +#define NUM_CLSH_STATES_V2 BIT(WCD_CLSH_STATE_MAX) + +enum wcd_clsh_mode { + CLS_H_NORMAL = 0, /* Class-H Default */ + CLS_H_HIFI, /* Class-H HiFi */ + CLS_H_LP, /* Class-H Low Power */ + CLS_AB, /* Class-AB */ + CLS_H_LOHIFI, /* LoHIFI */ + CLS_NONE, /* None of the above modes */ +}; + +struct wcd_clsh_ctrl; + +extern struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc( + struct snd_soc_component *component, + int version); +extern void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl); +extern int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl); +extern int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, + enum wcd_clsh_event event, + int state, + enum wcd_clsh_mode mode); + +#endif /* _WCD_CLSH_V2_H_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index bd9de5d45fa9..06c73699f16f 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -21,6 +21,7 @@ #include #include #include +#include "wcd-clsh.h" #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -181,6 +182,7 @@ struct wcd9335_codec { int sido_ccl_cnt; enum wcd_clock_type clk_type; + struct wcd_clsh_ctrl *clsh_ctrl; u32 hph_mode; }; @@ -1066,6 +1068,13 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) int i; snd_soc_component_init_regmap(component, wcd->regmap); + /* Class-H Init*/ + wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); + if (IS_ERR(wcd->clsh_ctrl)) + return PTR_ERR(wcd->clsh_ctrl); + + /* Default HPH Mode to Class-H HiFi */ + wcd->hph_mode = CLS_H_HIFI; wcd->component = component; wcd9335_codec_init(component); @@ -1080,6 +1089,7 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) { struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + wcd_clsh_ctrl_free(wcd->clsh_ctrl); free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); } From 00bc22e3eea0346b594442331df53d0d1e7104dd Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:06 +0530 Subject: [PATCH 419/529] ASoC: qcom: dt-bindings: Add sdm845 machine bindings Add devicetree bindings documentation file for SDM845 sound card. Reviewed-by: Rob Herring Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 +++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt new file mode 100644 index 000000000000..408c4837e6d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt @@ -0,0 +1,80 @@ +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver + +This binding describes the SDM845 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: + Definition: must be "qcom,sdm845-sndcard" + +- audio-routing: + Usage: Optional + Value type: + Definition: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, MicBias + of codec and the jacks on the board. + +- model: + Usage: required + Value type: + Definition: The user-visible name of this sound card. + += dailinks +Each subnode of sndcard represents either a dailink, and subnodes of each +dailinks would be cpu/codec/platform dais. + +- link-name: + Usage: required + Value type: + Definition: User friendly name for dai link + += CPU, PLATFORM, CODEC dais subnodes +- cpu: + Usage: required + Value type: + Definition: cpu dai sub-node + +- codec: + Usage: required + Value type: + Definition: codec dai sub-node + +- platform: + Usage: Optional + Value type: + Definition: platform dai sub-node + +- sound-dai: + Usage: required + Value type: + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Example: + +audio { + compatible = "qcom,sdm845-sndcard"; + model = "sdm845-snd-card"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>; + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + pri-mi2s-dai-link { + link-name = "PRI MI2S Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; +}; From 0c901e8cea15ba1e318d8f4342f7dc27a80b5978 Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:07 +0530 Subject: [PATCH 420/529] ASoC: dt-bindings: Update dt binding name for apq8096 Remove qcom prefix from machine driver dt bindings of apq8096 SoC. Acked-by: Srinivas Kandagatla Reviewed-by: Rob Herring Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt index aa54e49fc8a2..20bc034f6c01 100644 --- a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt +++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt @@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. Value type: Definition: must be "qcom,apq8096-sndcard" -- qcom,audio-routing: +- audio-routing: Usage: Optional Value type: Definition: A list of the connections between audio components. @@ -49,6 +49,12 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. "DMIC1" "DMIC2" "DMIC3" + +- model: + Usage: required + Value type: + Definition: The user-visible name of this sound card. + = dailinks Each subnode of sndcard represents either a dailink, and subnodes of each dailinks would be cpu/codec/platform dais. @@ -79,11 +85,16 @@ dailinks would be cpu/codec/platform dais. Value type: Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. +Obsolete: + qcom,model: String for soundcard name (Use model instead) + qcom,audio-routing: A list of the connections between audio components. + (Use audio-routing instead) + Example: audio { compatible = "qcom,apq8096-sndcard"; - qcom,model = "DB820c"; + model = "DB820c"; mm1-dai-link { link-name = "MultiMedia1"; From c25e295cd77b37903ddc9ee27384e17aad08f27c Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:08 +0530 Subject: [PATCH 421/529] ASoC: qcom: Add support to parse common audio device nodes This adds support to parse cpu, platform and codec device nodes and add them in dai-links. Also, update apq8096 machine driver to use the common API. Acked-by: Srinivas Kandagatla Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Makefile | 2 +- sound/soc/qcom/apq8096.c | 111 ++++---------------------------------- sound/soc/qcom/common.c | 112 +++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 12 +++++ 4 files changed, 136 insertions(+), 101 deletions(-) create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 206945bb9ba1..fefecc072265 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,7 +13,7 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o +snd-soc-apq8096-objs := apq8096.o common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index a56156281c8d..1e4a90d59228 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -9,6 +9,7 @@ #include #include #include +#include "common.h" static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -24,109 +25,16 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int apq8096_sbc_parse_of(struct snd_soc_card *card) +static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct device_node *np; - struct device_node *codec = NULL; - struct device_node *platform = NULL; - struct device_node *cpu = NULL; - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - int ret, num_links; + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; - ret = snd_soc_of_parse_card_name(card, "qcom,model"); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - - /* DAPM routes */ - if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, - "qcom,audio-routing"); - if (ret) - return ret; - } - - /* Populate links */ - num_links = of_get_child_count(dev->of_node); - - /* Allocate the DAI link array */ - card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; - - card->num_links = num_links; - link = card->dai_link; - - for_each_child_of_node(dev->of_node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); - ret = -EINVAL; - goto err; - } - - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); - if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); - goto err; - } - - platform = of_get_child_by_name(np, "platform"); - codec = of_get_child_by_name(np, "codec"); - if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, - "sound-dai", - 0); - if (!link->platform_of_node) { - dev_err(card->dev, "platform dai not found\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); - goto err; - } - link->no_pcm = 1; - link->ignore_pmdown_time = 1; + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; - link->dynamic = 1; - } - - link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - - link->dpcm_playback = 1; - link->dpcm_capture = 1; - link->stream_name = link->name; link++; } - - return 0; -err: - of_node_put(cpu); - of_node_put(codec); - of_node_put(platform); - kfree(card->dai_link); - return ret; } static int apq8096_platform_probe(struct platform_device *pdev) @@ -142,18 +50,21 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; card->auto_bind = true; dev_set_drvdata(dev, card); - ret = apq8096_sbc_parse_of(card); + ret = qcom_snd_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); goto err; } + apq8096_add_be_ops(card); ret = snd_soc_register_card(card); if (ret) - goto err; + goto err_card_register; return 0; +err_card_register: + kfree(card->dai_link); err: kfree(card); return ret; diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c new file mode 100644 index 000000000000..eb1b9da05dd4 --- /dev/null +++ b/sound/soc/qcom/common.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include +#include "common.h" + +int qcom_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "Can't find cpu DT node\n"); + ret = -EINVAL; + goto err; + } + + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + if (!link->cpu_of_node) { + dev_err(card->dev, "error getting cpu phandle\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + if (ret) { + dev_err(card->dev, "error getting cpu dai name\n"); + goto err; + } + + platform = of_get_child_by_name(np, "platform"); + codec = of_get_child_by_name(np, "codec"); + if (codec && platform) { + link->platform_of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platform_of_node) { + dev_err(card->dev, "platform dai not found\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "codec dai not found\n"); + goto err; + } + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + } else { + link->platform_of_node = link->cpu_of_node; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + link->dynamic = 1; + } + + link->ignore_suspend = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + link->dpcm_playback = 1; + link->dpcm_capture = 1; + link->stream_name = link->name; + link++; + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} +EXPORT_SYMBOL(qcom_snd_parse_of); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h new file mode 100644 index 000000000000..ad5d2cf27459 --- /dev/null +++ b/sound/soc/qcom/common.h @@ -0,0 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_COMMON_H__ +#define __QCOM_SND_COMMON_H__ + +#include +#include + +int qcom_snd_parse_of(struct snd_soc_card *card); + +#endif From 6b1687bf76ef84cb1e31386c4871a01fe66937bf Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:09 +0530 Subject: [PATCH 422/529] ASoC: qcom: add sdm845 sound card support This patch adds sdm845 audio machine driver support. Acked-by: Srinivas Kandagatla Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 8 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sdm845.c | 286 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 296 insertions(+) create mode 100644 sound/soc/qcom/sdm845.c diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 87838fa27997..350730839c6f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -90,3 +90,11 @@ config SND_SOC_MSM8996 Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. Say Y if you want to use audio device on this SoCs + +config SND_SOC_SDM845 + tristate "SoC Machine driver for SDM845 boards" + select SND_SOC_QDSP6 + help + To add support for audio on Qualcomm Technologies Inc. + SDM845 SoC-based systems. + Say Y if you want to use audio device on this SoCs. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index fefecc072265..f0e94d48ba98 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -14,10 +14,12 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o snd-soc-apq8096-objs := apq8096.o common.o +snd-soc-sdm845-objs := sdm845.o common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o +obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c new file mode 100644 index 000000000000..bf4ec4646906 --- /dev/null +++ b/sound/soc/qcom/sdm845.c @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2018, The Linux Foundation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "common.h" +#include "qdsp6/q6afe.h" + +#define DEFAULT_SAMPLE_RATE_48K 48000 +#define DEFAULT_MCLK_RATE 24576000 +#define DEFAULT_BCLK_RATE 12288000 + +struct sdm845_snd_data { + struct snd_soc_card *card; + uint32_t pri_mi2s_clk_count; + uint32_t quat_tdm_clk_count; +}; + +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; + +static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + int channels, slot_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 32; + break; + default: + dev_err(rtd->dev, "%s: invalid param format 0x%x\n", + __func__, params_format(params)); + return -EINVAL; + } + + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, + channels, tdm_slot_offset); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } else { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, channels, + tdm_slot_offset, 0, NULL); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } +end: + return ret; +} + +static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + switch (cpu_dai->id) { + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + ret = sdm845_tdm_snd_hw_params(substream, params); + break; + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return ret; +} + +static int sdm845_snd_startup(struct snd_pcm_substream *substream) +{ + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (++(data->pri_mi2s_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(cpu_dai, fmt); + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (++(data->quat_tdm_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return 0; +} + +static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (--(data->pri_mi2s_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + 0, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + }; + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (--(data->quat_tdm_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } +} + +static struct snd_soc_ops sdm845_be_ops = { + .hw_params = sdm845_snd_hw_params, + .startup = sdm845_snd_startup, + .shutdown = sdm845_snd_shutdown, +}; + +static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K; + channels->min = channels->max = 2; + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static void sdm845_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; + + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) { + link->ops = &sdm845_be_ops; + link->be_hw_params_fixup = sdm845_be_hw_params_fixup; + } + link++; + } +} + +static int sdm845_snd_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sdm845_snd_data *data; + struct device *dev = &pdev->dev; + int ret; + + card = kzalloc(sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + /* Allocate the private data */ + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + card->dev = dev; + card->auto_bind = true; + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); + if (ret) { + dev_err(dev, "Error parsing OF data\n"); + goto parse_dt_fail; + } + + data->card = card; + snd_soc_card_set_drvdata(card, data); + + sdm845_add_be_ops(card); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(dev, "Sound card registration failed\n"); + goto register_card_fail; + } + return ret; + +register_card_fail: + kfree(card->dai_link); +parse_dt_fail: + kfree(data); + kfree(card); + return ret; +} + +static int sdm845_snd_platform_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + + card->auto_bind = false; + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(data); + kfree(card); + return 0; +} + +static const struct of_device_id sdm845_snd_device_id[] = { + { .compatible = "qcom,sdm845-sndcard" }, + {}, +}; +MODULE_DEVICE_TABLE(of, sdm845_snd_device_id); + +static struct platform_driver sdm845_snd_driver = { + .probe = sdm845_snd_platform_probe, + .remove = sdm845_snd_platform_remove, + .driver = { + .name = "msm-snd-sdm845", + .of_match_table = sdm845_snd_device_id, + }, +}; +module_platform_driver(sdm845_snd_driver); + +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); From d101f9b96ee08f0454989bc3adb10e6cf7f3f953 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:39 +0100 Subject: [PATCH 423/529] ASoC: nau8540: remove redundant variable osrate Variable osrate is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'osrate' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/nau8540.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 17104f8dc1a9..e3c8cd17daf2 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) return -EINVAL; - osrate = osr_adc_sel[osr].osr; if (rate * osr > CLK_ADC_MAX) { dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); From 18127744cf446f113ca33f07e5cea893388f781a Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:40 +0100 Subject: [PATCH 424/529] ASoC: stm32: remove redundant pointers 'priv' and 'rtd' Pointer 'priv' is assigned and not used, removing this allows the removal of pointer 'rtd'. Cleans up clang warning: warning: variable 'priv' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 0e9373064032..706ff005234f 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd) static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; - struct snd_soc_pcm_runtime *rtd; - struct stm32_adfsdm_priv *priv; substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - if (substream) { - rtd = substream->private_data; - priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - + if (substream) snd_pcm_lib_preallocate_free_for_all(pcm); - } } static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { From 96963dedd000605a0e84e2ac6e41263a50f05953 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:34 +0100 Subject: [PATCH 425/529] ALSA: asihpi: remove redundant variable max_streams Variable max_streams is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'max_streams' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 8d5abfa4e24b..2864698436a5 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -635,7 +635,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, { struct hpi_message hm; struct hpi_response hr; - u32 max_streams; HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n"); memset(&hm, 0, sizeof(hm)); @@ -660,10 +659,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, pao->type = hr.u.ax.info.adapter_type; pao->index = hr.u.ax.info.adapter_index; - max_streams = - hr.u.ax.info.num_outstreams + - hr.u.ax.info.num_instreams; - HPI_DEBUG_LOG(VERBOSE, "got adapter info type %x index %d serial %d\n", hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index, From 45bf41005ac0d7cae0c1caa85d06cb35976823fa Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:35 +0100 Subject: [PATCH 426/529] ALSA: cs5535audio: remove redundant pointer 'dma' Pointer 'dma' is being assigned but is never used hence it is redundant and can be removed. Cleans up two clang warnings: warning: variable 'dma' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index de409cda50aa..4590086d9cd8 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -192,8 +192,6 @@ static void process_bm0_irq(struct cs5535audio *cs5535au) bm_stat = cs_readb(cs5535au, ACC_BM0_STATUS); spin_unlock(&cs5535au->reg_lock); if (bm_stat & EOP) { - struct cs5535audio_dma *dma; - dma = cs5535au->playback_substream->runtime->private_data; snd_pcm_period_elapsed(cs5535au->playback_substream); } else { dev_err(cs5535au->card->dev, @@ -208,11 +206,8 @@ static void process_bm1_irq(struct cs5535audio *cs5535au) spin_lock(&cs5535au->reg_lock); bm_stat = cs_readb(cs5535au, ACC_BM1_STATUS); spin_unlock(&cs5535au->reg_lock); - if (bm_stat & EOP) { - struct cs5535audio_dma *dma; - dma = cs5535au->capture_substream->runtime->private_data; + if (bm_stat & EOP) snd_pcm_period_elapsed(cs5535au->capture_substream); - } } static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id) From de42b4b96ebe29058ce1cb59a1f98d58b8abd132 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:36 +0100 Subject: [PATCH 427/529] ALSA: emu10k1: remove redundant variable attn Variable attn is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'attn' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 69f9b100bd24..26f6eda3e766 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -290,7 +290,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, struct snd_pcm_runtime *runtime = substream->runtime; unsigned int silent_page, tmp; int voice, stereo, w_16; - unsigned char attn, send_amount[8]; + unsigned char send_amount[8]; unsigned char send_routing[8]; unsigned long flags; unsigned int pitch_target; @@ -313,7 +313,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* volume parameters */ if (extra) { - attn = 0; memset(send_routing, 0, sizeof(send_routing)); send_routing[0] = 0; send_routing[1] = 1; From 3b0cbc7812d759ffb38d881e75c0318512283d00 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:37 +0100 Subject: [PATCH 428/529] ALSA: ens137x: remove redundant array pcm_devs The array pcm_devs is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'pcm_devs' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 39f79a6b5283..727eb3da1fda 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2392,7 +2392,7 @@ static int snd_audiopci_probe(struct pci_dev *pci, static int dev; struct snd_card *card; struct ensoniq *ensoniq; - int err, pcm_devs[2]; + int err; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -2412,7 +2412,6 @@ static int snd_audiopci_probe(struct pci_dev *pci, } card->private_data = ensoniq; - pcm_devs[0] = 0; pcm_devs[1] = 1; #ifdef CHIP1370 if ((err = snd_ensoniq_1370_mixer(ensoniq)) < 0) { snd_card_free(card); From 0d00085b905c904d6de4296aa6a1e3613d0a4d67 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:38 +0100 Subject: [PATCH 429/529] ALSA: sonicvibes: remove redundant pointer 'dir' Pointer 'dir' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'dir' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/sonicvibes.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7fbdb703bfcd..7218f38b59db 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1433,14 +1433,12 @@ static int snd_sonicvibes_midi(struct sonicvibes *sonic, { struct snd_mpu401 * mpu = rmidi->private_data; struct snd_card *card = sonic->card; - struct snd_rawmidi_str *dir; unsigned int idx; int err; mpu->private_data = sonic; mpu->open_input = snd_sonicvibes_midi_input_open; mpu->close_input = snd_sonicvibes_midi_input_close; - dir = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; for (idx = 0; idx < ARRAY_SIZE(snd_sonicvibes_midi_controls); idx++) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_sonicvibes_midi_controls[idx], sonic))) < 0) return err; From d36455a38ed82e7d1e6807573935874f0b279f45 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:41 +0100 Subject: [PATCH 430/529] ALSA: usb-audio: remove redundant pointer 'urb' Pointer 'urb' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'urb' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c90607ebe155..d86be8bfe412 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -325,7 +325,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) unsigned long flags; struct snd_usb_packet_info *uninitialized_var(packet); struct snd_urb_ctx *ctx = NULL; - struct urb *urb; int err, i; spin_lock_irqsave(&ep->lock, flags); @@ -345,7 +344,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) return; list_del_init(&ctx->ready_list); - urb = ctx->urb; /* copy over the length information */ for (i = 0; i < packet->packets; i++) From b0a39d356ae1478a5876bb02ba08af155a3a5554 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2018 14:18:50 +0100 Subject: [PATCH 431/529] ASoC: wcd9335: Fix build due to CLASS-H Controller support This reverts commit c8cb5f775c8dac (ASoC: vert "ASoC: wcd9335: add CLASS-H Controller support) due to missing dependencies. Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wcd-clsh.c | 605 ------------------------------------ sound/soc/codecs/wcd-clsh.h | 49 --- sound/soc/codecs/wcd9335.c | 10 - 4 files changed, 1 insertion(+), 665 deletions(-) delete mode 100644 sound/soc/codecs/wcd-clsh.c delete mode 100644 sound/soc/codecs/wcd-clsh.h diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e3a3d4694e15..01410b63daac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd-clsh.o wcd9335.o +snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd-clsh.c b/sound/soc/codecs/wcd-clsh.c deleted file mode 100644 index 2393456cbd97..000000000000 --- a/sound/soc/codecs/wcd-clsh.c +++ /dev/null @@ -1,605 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include -#include -#include -#include -#include -#include -#include "wcd-clsh.h" - -struct wcd_clsh_ctrl { - int state; - int mode; - int flyback_users; - int buck_users; - int clsh_users; - int codec_version; - struct snd_soc_component *comp; -}; - -/* Class-H registers for codecs from and above WCD9335 */ -#define WCD9XXX_A_CDC_RX0_RX_PATH_CFG0 WCD9335_REG(0xB, 0x42) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE 0 -#define WCD9XXX_A_CDC_RX1_RX_PATH_CFG0 WCD9335_REG(0xB, 0x56) -#define WCD9XXX_A_CDC_RX2_RX_PATH_CFG0 WCD9335_REG(0xB, 0x6A) -#define WCD9XXX_A_CDC_CLSH_K1_MSB WCD9335_REG(0xC, 0x08) -#define WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK GENMASK(3, 0) -#define WCD9XXX_A_CDC_CLSH_K1_LSB WCD9335_REG(0xC, 0x09) -#define WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK GENMASK(7, 0) -#define WCD9XXX_A_ANA_RX_SUPPLIES WCD9335_REG(0x6, 0x08) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK BIT(1) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H 0 -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB BIT(1) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VNEG_EN_MASK BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT 6 -#define WCD9XXX_A_ANA_RX_VNEG_ENABLE BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_DISABLE 0 -#define WCD9XXX_A_ANA_RX_VPOS_EN_MASK BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT 7 -#define WCD9XXX_A_ANA_RX_VPOS_ENABLE BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_DISABLE 0 -#define WCD9XXX_A_ANA_HPH WCD9335_REG(0x6, 0x09) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK GENMASK(3, 2) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA 0x08 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP 0x04 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL 0x0 -#define WCD9XXX_A_CDC_CLSH_CRC WCD9335_REG(0xC, 0x01) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_ENABLE BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_DISABLE 0 -#define WCD9XXX_FLYBACK_EN WCD9335_REG(0x6, 0xA4) -#define WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK GENMASK(6, 5) -#define WCD9XXX_FLYBACK_EN_DELAY_26P25_US 0x40 -#define WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY 0 -#define WCD9XXX_RX_BIAS_FLYB_BUFF WCD9335_REG(0x6, 0xC7) -#define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK GENMASK(7, 4) -#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK GENMASK(0, 3) -#define WCD9XXX_HPH_L_EN WCD9335_REG(0x6, 0xD3) -#define WCD9XXX_HPH_CONST_SEL_L_MASK GENMASK(7, 3) -#define WCD9XXX_HPH_CONST_SEL_BYPASS 0 -#define WCD9XXX_HPH_CONST_SEL_LP_PATH 0x40 -#define WCD9XXX_HPH_CONST_SEL_HQ_PATH 0x80 -#define WCD9XXX_HPH_R_EN WCD9335_REG(0x6, 0xD6) -#define WCD9XXX_HPH_REFBUFF_UHQA_CTL WCD9335_REG(0x6, 0xDD) -#define WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK GENMASK(2, 0) -#define WCD9XXX_CLASSH_CTRL_VCL_2 WCD9335_REG(0x6, 0x9B) -#define WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK GENMASK(5, 4) -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM 0x20 -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM 0x0 -#define WCD9XXX_CDC_RX1_RX_PATH_CTL WCD9335_REG(0xB, 0x55) -#define WCD9XXX_CDC_RX2_RX_PATH_CTL WCD9335_REG(0xB, 0x69) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_CONTROL WCD9335_REG(0xD, 0x41) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_EN_MASK BIT(0) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_11P3_EN_MASK BIT(1) -#define WCD9XXX_CLASSH_CTRL_CCL_1 WCD9335_REG(0x6, 0x9C) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK GENMASK(7, 4) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA 0x50 -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA 0x30 - -#define CLSH_REQ_ENABLE true -#define CLSH_REQ_DISABLE false -#define WCD_USLEEP_RANGE 50 - -enum { - DAC_GAIN_0DB = 0, - DAC_GAIN_0P2DB, - DAC_GAIN_0P4DB, - DAC_GAIN_0P6DB, - DAC_GAIN_0P8DB, - DAC_GAIN_M0P2DB, - DAC_GAIN_M0P4DB, - DAC_GAIN_M0P6DB, -}; - -static inline void wcd_enable_clsh_block(struct wcd_clsh_ctrl *ctrl, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - if ((enable && ++ctrl->clsh_users == 1) || - (!enable && --ctrl->clsh_users == 0)) - snd_soc_component_update_bits(comp, WCD9XXX_A_CDC_CLSH_CRC, - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK, - enable); - if (ctrl->clsh_users < 0) - ctrl->clsh_users = 0; -} - -static inline bool wcd_clsh_enable_status(struct snd_soc_component *comp) -{ - return snd_soc_component_read32(comp, WCD9XXX_A_CDC_CLSH_CRC) & - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK; -} - -static inline void wcd_clsh_set_buck_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT); -} - -static inline void wcd_clsh_set_flyback_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT); -} - -static void wcd_clsh_buck_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - /* enable/disable buck */ - if ((enable && (++ctrl->buck_users == 1)) || - (!enable && (--ctrl->buck_users == 0))) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT); - /* - * 500us sleep is required after buck enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_flyback_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - int vneg[] = {0x00, 0x40}; - - /* enable/disable flyback */ - if ((enable && (++ctrl->flyback_users == 1)) || - (!enable && (--ctrl->flyback_users == 0))) { - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT); - /* 100usec delay is needed as per HW requirement */ - usleep_range(100, 110); - - if (enable && (WCD9335_IS_1_1(ctrl->codec_version))) { - wcd_clsh_set_flyback_mode(comp, CLS_H_HIFI); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK, - WCD9XXX_FLYBACK_EN_DELAY_26P25_US); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY); - vneg[0] = snd_soc_component_read32(comp, - WCD9XXX_A_ANA_RX_SUPPLIES); - vneg[0] &= ~(0x40); - vneg[1] = vneg[0] | 0x40; - - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[0]); - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[1]); - /* 500usec delay is needed as per HW requirement */ - usleep_range(500, 510); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY); - wcd_clsh_set_flyback_mode(comp, mode); - } - - } - /* - * 500us sleep is required after flyback enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_set_gain_path(struct wcd_clsh_ctrl *ctrl, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - int val = 0; - - switch (mode) { - case CLS_H_NORMAL: - case CLS_AB: - val = WCD9XXX_HPH_CONST_SEL_BYPASS; - break; - case CLS_H_HIFI: - val = WCD9XXX_HPH_CONST_SEL_HQ_PATH; - break; - case CLS_H_LP: - val = WCD9XXX_HPH_CONST_SEL_LP_PATH; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_L_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_R_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); -} - -static void wcd_clsh_set_hph_mode(struct snd_soc_component *comp, - int mode) -{ - int val = 0, gain = 0, res_val; - int ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM; - switch (mode) { - case CLS_H_NORMAL: - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM; - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_AB: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_HIFI: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA; - gain = DAC_GAIN_M0P2DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_LP: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_HPH, - WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK, val); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_VCL_2, - WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK, - res_val); - if (mode != CLS_H_LP) - snd_soc_component_update_bits(comp, - WCD9XXX_HPH_REFBUFF_UHQA_CTL, - WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK, - gain); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_CCL_1, - WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK, - ipeak); -} - -static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, - int mode) -{ - - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK, 0x0A); - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK, 0x0A); - /* Sleep needed to avoid click and pop as per HW requirement */ - usleep_range(100, 110); -} - -static void wcd_clsh_set_buck_regulator_mode(struct snd_soc_component *comp, - int mode) -{ - if (mode == CLS_AB) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H); -} - -static void wcd_clsh_state_lo(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_AB) { - dev_err(comp->dev, "%s: LO cannot be in this mode: %d\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_r(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_r\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* buck and flyback set to default mode and disable */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_l(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_l\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* set buck and flyback to Default Mode */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_ear(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_H_NORMAL) { - dev_err(comp->dev, "%s: mode: %d cannot be used for EAR\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_enable_clsh_block(ctrl, true); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - } -} - -static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - switch (req_state) { - case WCD_CLSH_STATE_EAR: - wcd_clsh_state_ear(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHL: - wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHR: - wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); - break; - break; - case WCD_CLSH_STATE_LO: - wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); - break; - default: - break; - } - - return 0; -} - -/* - * Function: wcd_clsh_is_state_valid - * Params: state - * Description: - * Provides information on valid states of Class H configuration - */ -static bool wcd_clsh_is_state_valid(int state) -{ - switch (state) { - case WCD_CLSH_STATE_IDLE: - case WCD_CLSH_STATE_EAR: - case WCD_CLSH_STATE_HPHL: - case WCD_CLSH_STATE_HPHR: - case WCD_CLSH_STATE_LO: - return true; - default: - return false; - }; -} - -/* - * Function: wcd_clsh_fsm - * Params: ctrl, req_state, req_type, clsh_event - * Description: - * This function handles PRE DAC and POST DAC conditions of different devices - * and updates class H configuration of different combination of devices - * based on validity of their states. ctrl will contain current - * class h state information - */ -int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event clsh_event, - int nstate, - enum wcd_clsh_mode mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (nstate == ctrl->state) - return 0; - - if (!wcd_clsh_is_state_valid(nstate)) { - dev_err(comp->dev, "Class-H not a valid new state:\n"); - return -EINVAL; - } - - switch (clsh_event) { - case WCD_CLSH_EVENT_PRE_DAC: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_ENABLE, mode); - break; - case WCD_CLSH_EVENT_POST_PA: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_DISABLE, mode); - break; - }; - - ctrl->state = nstate; - ctrl->mode = mode; - - return 0; -} - -int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) -{ - return ctrl->state; -} - -struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, - int version) -{ - struct wcd_clsh_ctrl *ctrl; - - ctrl = kzalloc(sizeof(*ctrl), GFP_KERNEL); - if (!ctrl) - return ERR_PTR(-ENOMEM); - - ctrl->state = WCD_CLSH_STATE_IDLE; - ctrl->comp = comp; - - return ctrl; -} - -void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) -{ - kfree(ctrl); -} diff --git a/sound/soc/codecs/wcd-clsh.h b/sound/soc/codecs/wcd-clsh.h deleted file mode 100644 index a902f9893467..000000000000 --- a/sound/soc/codecs/wcd-clsh.h +++ /dev/null @@ -1,49 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ - -#ifndef _WCD_CLSH_V2_H_ -#define _WCD_CLSH_V2_H_ -#include - -enum wcd_clsh_event { - WCD_CLSH_EVENT_PRE_DAC = 1, - WCD_CLSH_EVENT_POST_PA, -}; - -/* - * Basic states for Class H state machine. - * represented as a bit mask within a u8 data type - * bit 0: EAR mode - * bit 1: HPH Left mode - * bit 2: HPH Right mode - * bit 3: Lineout mode - */ -#define WCD_CLSH_STATE_IDLE 0 -#define WCD_CLSH_STATE_EAR BIT(0) -#define WCD_CLSH_STATE_HPHL BIT(1) -#define WCD_CLSH_STATE_HPHR BIT(2) -#define WCD_CLSH_STATE_LO BIT(3) -#define WCD_CLSH_STATE_MAX 4 -#define NUM_CLSH_STATES_V2 BIT(WCD_CLSH_STATE_MAX) - -enum wcd_clsh_mode { - CLS_H_NORMAL = 0, /* Class-H Default */ - CLS_H_HIFI, /* Class-H HiFi */ - CLS_H_LP, /* Class-H Low Power */ - CLS_AB, /* Class-AB */ - CLS_H_LOHIFI, /* LoHIFI */ - CLS_NONE, /* None of the above modes */ -}; - -struct wcd_clsh_ctrl; - -extern struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc( - struct snd_soc_component *component, - int version); -extern void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event event, - int state, - enum wcd_clsh_mode mode); - -#endif /* _WCD_CLSH_V2_H_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 06c73699f16f..bd9de5d45fa9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -21,7 +21,6 @@ #include #include #include -#include "wcd-clsh.h" #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -182,7 +181,6 @@ struct wcd9335_codec { int sido_ccl_cnt; enum wcd_clock_type clk_type; - struct wcd_clsh_ctrl *clsh_ctrl; u32 hph_mode; }; @@ -1068,13 +1066,6 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) int i; snd_soc_component_init_regmap(component, wcd->regmap); - /* Class-H Init*/ - wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); - if (IS_ERR(wcd->clsh_ctrl)) - return PTR_ERR(wcd->clsh_ctrl); - - /* Default HPH Mode to Class-H HiFi */ - wcd->hph_mode = CLS_H_HIFI; wcd->component = component; wcd9335_codec_init(component); @@ -1089,7 +1080,6 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) { struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - wcd_clsh_ctrl_free(wcd->clsh_ctrl); free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); } From b74fd69043262003c5b7ce545e59f5a7234ab290 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2018 14:22:56 +0100 Subject: [PATCH 432/529] ASoC: wcd9335: Fix build This reverts commit e57d4ca882e28 (ASoC: wcd9335: add support to wcd9335 codec) due to build failures caused by missing dependencies. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/wcd9335.c | 1154 ------------------------------------ 3 files changed, 1161 deletions(-) delete mode 100644 sound/soc/codecs/wcd9335.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb09abf18dde..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1066,11 +1066,6 @@ config SND_SOC_UDA1380 tristate depends on I2C -config SND_SOC_WCD9335 - tristate "WCD9335 Codec" - depends on MFD_WCD9335 - tristate - config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 01410b63daac..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,6 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o @@ -452,7 +451,6 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o -obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c deleted file mode 100644 index bd9de5d45fa9..000000000000 --- a/sound/soc/codecs/wcd9335.c +++ /dev/null @@ -1,1154 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) -/* Fractional Rates */ -#define WCD9335_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100) -#define WCD9335_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) - -/* slave port water mark level - * (0: 6bytes, 1: 9bytes, 2: 12 bytes, 3: 15 bytes) - */ -#define SLAVE_PORT_WATER_MARK_6BYTES 0 -#define SLAVE_PORT_WATER_MARK_9BYTES 1 -#define SLAVE_PORT_WATER_MARK_12BYTES 2 -#define SLAVE_PORT_WATER_MARK_15BYTES 3 -#define SLAVE_PORT_WATER_MARK_SHIFT 1 -#define SLAVE_PORT_ENABLE 1 -#define SLAVE_PORT_DISABLE 0 -#define WCD9335_SLIM_WATER_MARK_VAL \ - ((SLAVE_PORT_WATER_MARK_12BYTES << SLAVE_PORT_WATER_MARK_SHIFT) | \ - (SLAVE_PORT_ENABLE)) - -#define WCD9335_SLIM_NUM_PORT_REG 3 -#define WCD9335_SLIM_PGD_PORT_INT_TX_EN0 (WCD9335_SLIM_PGD_PORT_INT_EN0 + 2) - -#define WCD9335_MCLK_CLK_12P288MHZ 12288000 -#define WCD9335_MCLK_CLK_9P6MHZ 9600000 - -#define WCD9335_SLIM_CLOSE_TIMEOUT 1000 -#define WCD9335_SLIM_IRQ_OVERFLOW (1 << 0) -#define WCD9335_SLIM_IRQ_UNDERFLOW (1 << 1) -#define WCD9335_SLIM_IRQ_PORT_CLOSED (1 << 2) - -#define WCD9335_NUM_INTERPOLATORS 9 -#define WCD9335_RX_START 16 -#define WCD9335_SLIM_CH_START 128 - -#define WCD9335_SLIM_RX_CH(p) \ - {.port = p + WCD9335_RX_START, .shift = p,} - -/* vout step value */ -#define WCD9335_CALCULATE_VOUT_D(req_mv) (((req_mv - 650) * 10) / 25) - -enum { - WCD9335_RX0 = 0, - WCD9335_RX1, - WCD9335_RX2, - WCD9335_RX3, - WCD9335_RX4, - WCD9335_RX5, - WCD9335_RX6, - WCD9335_RX7, - WCD9335_RX8, - WCD9335_RX9, - WCD9335_RX10, - WCD9335_RX11, - WCD9335_RX12, - WCD9335_RX_MAX, -}; - -enum { - SIDO_SOURCE_INTERNAL = 0, - SIDO_SOURCE_RCO_BG, -}; - -enum wcd9335_sido_voltage { - SIDO_VOLTAGE_SVS_MV = 950, - SIDO_VOLTAGE_NOMINAL_MV = 1100, -}; - -enum { - AIF1_PB = 0, - AIF1_CAP, - AIF2_PB, - AIF2_CAP, - AIF3_PB, - AIF3_CAP, - AIF4_PB, - NUM_CODEC_DAIS, -}; - -enum { - INTn_2_INP_SEL_ZERO = 0, - INTn_2_INP_SEL_RX0, - INTn_2_INP_SEL_RX1, - INTn_2_INP_SEL_RX2, - INTn_2_INP_SEL_RX3, - INTn_2_INP_SEL_RX4, - INTn_2_INP_SEL_RX5, - INTn_2_INP_SEL_RX6, - INTn_2_INP_SEL_RX7, - INTn_2_INP_SEL_PROXIMITY, -}; - -enum { - INTn_1_MIX_INP_SEL_ZERO = 0, - INTn_1_MIX_INP_SEL_DEC0, - INTn_1_MIX_INP_SEL_DEC1, - INTn_1_MIX_INP_SEL_IIR0, - INTn_1_MIX_INP_SEL_IIR1, - INTn_1_MIX_INP_SEL_RX0, - INTn_1_MIX_INP_SEL_RX1, - INTn_1_MIX_INP_SEL_RX2, - INTn_1_MIX_INP_SEL_RX3, - INTn_1_MIX_INP_SEL_RX4, - INTn_1_MIX_INP_SEL_RX5, - INTn_1_MIX_INP_SEL_RX6, - INTn_1_MIX_INP_SEL_RX7, - -}; - -enum wcd_clock_type { - WCD_CLK_OFF, - WCD_CLK_RCO, - WCD_CLK_MCLK, -}; - -struct wcd9335_slim_ch { - u32 ch_num; - u16 port; - u16 shift; - struct list_head list; -}; - -struct wcd_slim_codec_dai_data { - struct list_head slim_ch_list; - struct slim_stream_config sconfig; - struct slim_stream_runtime *sruntime; -}; - -struct wcd9335_codec { - struct device *dev; - struct clk *mclk; - struct clk *native_clk; - u32 mclk_rate; - u8 intf_type; - u8 version; - - struct slim_device *slim; - struct slim_device *slim_ifd; - struct regmap *regmap; - struct regmap *if_regmap; - struct regmap_irq_chip_data *irq_data; - - struct wcd9335_slim_ch rx_chs[WCD9335_RX_MAX]; - u32 num_rx_port; - - int sido_input_src; - enum wcd9335_sido_voltage sido_voltage; - - struct wcd_slim_codec_dai_data dai[NUM_CODEC_DAIS]; - struct snd_soc_component *component; - - int master_bias_users; - int clk_mclk_users; - int clk_rco_users; - int sido_ccl_cnt; - enum wcd_clock_type clk_type; - - u32 hph_mode; -}; - -static const struct wcd9335_slim_ch wcd9335_rx_chs[WCD9335_RX_MAX] = { - WCD9335_SLIM_RX_CH(0), /* 16 */ - WCD9335_SLIM_RX_CH(1), /* 17 */ - WCD9335_SLIM_RX_CH(2), - WCD9335_SLIM_RX_CH(3), - WCD9335_SLIM_RX_CH(4), - WCD9335_SLIM_RX_CH(5), - WCD9335_SLIM_RX_CH(6), - WCD9335_SLIM_RX_CH(7), - WCD9335_SLIM_RX_CH(8), - WCD9335_SLIM_RX_CH(9), - WCD9335_SLIM_RX_CH(10), - WCD9335_SLIM_RX_CH(11), - WCD9335_SLIM_RX_CH(12), -}; - -struct interp_sample_rate { - int rate; - int rate_val; -}; - -static struct interp_sample_rate int_mix_rate_val[] = { - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ -}; - -static struct interp_sample_rate int_prim_rate_val[] = { - {8000, 0x0}, /* 8K */ - {16000, 0x1}, /* 16K */ - {24000, -EINVAL},/* 24K */ - {32000, 0x3}, /* 32K */ - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ - {384000, 0x7}, /* 384K */ - {44100, 0x8}, /* 44.1K */ -}; - -struct wcd9335_reg_mask_val { - u16 reg; - u8 mask; - u8 val; -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_val_2_0[] = { - {WCD9335_RCO_CTRL_2, 0x0F, 0x08}, - {WCD9335_RX_BIAS_FLYB_MID_RST, 0xF0, 0x10}, - {WCD9335_FLYBACK_CTRL_1, 0x20, 0x20}, - {WCD9335_HPH_OCP_CTL, 0xFF, 0x5A}, - {WCD9335_HPH_L_TEST, 0x01, 0x01}, - {WCD9335_HPH_R_TEST, 0x01, 0x01}, - {WCD9335_CDC_BOOST0_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST0_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_BOOST1_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST1_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_TX0_TX_PATH_SEC7, 0xFF, 0x45}, - {WCD9335_CDC_RX0_RX_PATH_SEC0, 0xFC, 0xF4}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x08, 0x08}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x06, 0x02}, -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_common_val[] = { - /* Rbuckfly/R_EAR(32) */ - {WCD9335_CDC_CLSH_K2_MSB, 0x0F, 0x00}, - {WCD9335_CDC_CLSH_K2_LSB, 0xFF, 0x60}, - {WCD9335_CPE_SS_DMIC_CFG, 0x80, 0x00}, - {WCD9335_CDC_BOOST0_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_BOOST1_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_RX7_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_CDC_RX8_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_ANA_LO_1_2, 0x3C, 0X3C}, - {WCD9335_DIFF_LO_COM_SWCAP_REFBUF_FREQ, 0x70, 0x00}, - {WCD9335_DIFF_LO_COM_PA_FREQ, 0x70, 0x40}, - {WCD9335_SOC_MAD_AUDIO_CTL_2, 0x03, 0x03}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x02, 0x02}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x01, 0x01}, - {WCD9335_EAR_CMBUFF, 0x08, 0x00}, - {WCD9335_CDC_TX9_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX10_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX11_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX12_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER7_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER8_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_VBADC_IBIAS_FE, 0x0C, 0x08}, -}; - -static int wcd9335_set_mix_interpolator_rate(struct snd_soc_dai *dai, - int rate_val, - u32 rate) -{ - struct snd_soc_component *component = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - struct wcd9335_slim_ch *ch; - int val, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - val = snd_soc_component_read32(component, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)) & - WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if (val == (ch->shift + INTn_2_INP_SEL_RX0)) - snd_soc_component_update_bits(component, - WCD9335_CDC_RX_PATH_MIX_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - - return 0; -} - -static int wcd9335_set_prim_interpolator_rate(struct snd_soc_dai *dai, - u8 rate_val, - u32 rate) -{ - struct snd_soc_component *comp = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - struct wcd9335_slim_ch *ch; - u8 cfg0, cfg1, inp0_sel, inp1_sel, inp2_sel; - int inp, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - inp = ch->shift + INTn_1_MIX_INP_SEL_RX0; - /* - * Loop through all interpolator MUX inputs and find out - * to which interpolator input, the slim rx port - * is connected - */ - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - cfg0 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG0(j)); - cfg1 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)); - - inp0_sel = cfg0 & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp1_sel = (cfg0 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp2_sel = (cfg1 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if ((inp0_sel == inp) || (inp1_sel == inp) || - (inp2_sel == inp)) { - /* rate is in Hz */ - if ((j == 0) && (rate == 44100)) - dev_info(wcd->dev, - "Cannot set 44.1KHz on INT0\n"); - else - snd_soc_component_update_bits(comp, - WCD9335_CDC_RX_PATH_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - } - - return 0; -} - -static int wcd9335_set_interpolator_rate(struct snd_soc_dai *dai, u32 rate) -{ - int i; - - /* set mixing path rate */ - for (i = 0; i < ARRAY_SIZE(int_mix_rate_val); i++) { - if (rate == int_mix_rate_val[i].rate) { - wcd9335_set_mix_interpolator_rate(dai, - int_mix_rate_val[i].rate_val, rate); - break; - } - } - - /* set primary path sample rate */ - for (i = 0; i < ARRAY_SIZE(int_prim_rate_val); i++) { - if (rate == int_prim_rate_val[i].rate) { - wcd9335_set_prim_interpolator_rate(dai, - int_prim_rate_val[i].rate_val, rate); - break; - } - } - - return 0; -} - -static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, - struct wcd_slim_codec_dai_data *dai_data, - int direction) -{ - struct list_head *slim_ch_list = &dai_data->slim_ch_list; - struct slim_stream_config *cfg = &dai_data->sconfig; - struct wcd9335_slim_ch *ch; - u16 payload = 0; - int ret, i; - - cfg->ch_count = 0; - cfg->direction = direction; - cfg->port_mask = 0; - - /* Configure slave interface device */ - list_for_each_entry(ch, slim_ch_list, list) { - cfg->ch_count++; - payload |= 1 << ch->shift; - cfg->port_mask |= BIT(ch->port); - } - - cfg->chs = kcalloc(cfg->ch_count, sizeof(unsigned int), GFP_KERNEL); - if (!cfg->chs) - return -ENOMEM; - - i = 0; - list_for_each_entry(ch, slim_ch_list, list) { - cfg->chs[i++] = ch->ch_num; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - /* write to interface device */ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), - payload); - - if (ret < 0) - goto err; - - /* configure the slave port for water mark and enable*/ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_CFG(ch->port), - WCD9335_SLIM_WATER_MARK_VAL); - if (ret < 0) - goto err; - } - } - - dai_data->sruntime = slim_stream_allocate(wcd->slim, "WCD9335-SLIM"); - slim_stream_prepare(dai_data->sruntime, cfg); - - return 0; - -err: - dev_err(wcd->dev, "Error Setting slim hw params\n"); - kfree(cfg->chs); - cfg->chs = NULL; - - return ret; -} - -static int wcd9335_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct wcd9335_codec *wcd; - int ret; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (substream->stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - ret = wcd9335_set_interpolator_rate(dai, params_rate(params)); - if (ret) { - dev_err(wcd->dev, "cannot set sample rate: %u\n", - params_rate(params)); - return ret; - } - switch (params_width(params)) { - case 16 ... 24: - wcd->dai[dai->id].sconfig.bps = params_width(params); - break; - default: - dev_err(wcd->dev, "%s: Invalid format 0x%x\n", - __func__, params_width(params)); - return -EINVAL; - } - break; - default: - dev_err(wcd->dev, "Invalid stream type %d\n", - substream->stream); - return -EINVAL; - }; - - wcd->dai[dai->id].sconfig.rate = params_rate(params); - wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); - - return 0; -} - -static int wcd9335_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wcd_slim_codec_dai_data *dai_data; - struct wcd9335_codec *wcd; - - wcd = snd_soc_component_get_drvdata(dai->component); - dai_data = &wcd->dai[dai->id]; - slim_stream_enable(dai_data->sruntime); - - return 0; -} - -static int wcd9335_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct wcd9335_codec *wcd; - int i; - - wcd = snd_soc_component_get_drvdata(dai->component); - - if (!tx_slot || !rx_slot) { - dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", - tx_slot, rx_slot); - return -EINVAL; - } - - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } - } - - return 0; -} - -static int wcd9335_get_channel_map(struct snd_soc_dai *dai, - unsigned int *tx_num, unsigned int *tx_slot, - unsigned int *rx_num, unsigned int *rx_slot) -{ - struct wcd9335_slim_ch *ch; - struct wcd9335_codec *wcd; - int i = 0; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (dai->id) { - case AIF1_PB: - case AIF2_PB: - case AIF3_PB: - case AIF4_PB: - if (!rx_slot || !rx_num) { - dev_err(wcd->dev, "Invalid rx_slot %p or rx_num %p\n", - rx_slot, rx_num); - return -EINVAL; - } - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) - rx_slot[i++] = ch->ch_num; - - *rx_num = i; - break; - default: - dev_err(wcd->dev, "Invalid DAI ID %x\n", dai->id); - break; - } - - return 0; -} - -static struct snd_soc_dai_ops wcd9335_dai_ops = { - .hw_params = wcd9335_hw_params, - .prepare = wcd9335_prepare, - .set_channel_map = wcd9335_set_channel_map, - .get_channel_map = wcd9335_get_channel_map, -}; - -static struct snd_soc_dai_driver wcd9335_slim_dais[] = { - [0] = { - .name = "wcd9335_rx1", - .id = AIF1_PB, - .playback = { - .stream_name = "AIF1 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_max = 192000, - .rate_min = 8000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [1] = { - .name = "wcd9335_tx1", - .id = AIF1_CAP, - .capture = { - .stream_name = "AIF1 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [2] = { - .name = "wcd9335_rx2", - .id = AIF2_PB, - .playback = { - .stream_name = "AIF2 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [3] = { - .name = "wcd9335_tx2", - .id = AIF2_CAP, - .capture = { - .stream_name = "AIF2 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [4] = { - .name = "wcd9335_rx3", - .id = AIF3_PB, - .playback = { - .stream_name = "AIF3 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [5] = { - .name = "wcd9335_tx3", - .id = AIF3_CAP, - .capture = { - .stream_name = "AIF3 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [6] = { - .name = "wcd9335_rx4", - .id = AIF4_PB, - .playback = { - .stream_name = "AIF4 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, -}; - -static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -{ - struct wcd9335_codec *wcd = data; - unsigned long status = 0; - int i, j, port_id; - unsigned int val, int_val = 0; - bool tx; - unsigned short reg = 0; - - for (i = WCD9335_SLIM_PGD_PORT_INT_STATUS_RX_0, j = 0; - i <= WCD9335_SLIM_PGD_PORT_INT_STATUS_TX_1; i++, j++) { - regmap_read(wcd->if_regmap, i, &val); - status |= ((u32)val << (8 * j)); - } - - for_each_set_bit(j, &status, 32) { - tx = (j >= 16 ? true : false); - port_id = (tx ? j - 16 : j); - regmap_read(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); - if (val) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - /* - * Ignore interrupts for ports for which the - * interrupts are not specifically enabled. - */ - if (!(int_val & (1 << (port_id % 8)))) - continue; - } - if (val & WCD9335_SLIM_IRQ_OVERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: overflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if (val & WCD9335_SLIM_IRQ_UNDERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: underflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if ((val & WCD9335_SLIM_IRQ_OVERFLOW) || - (val & WCD9335_SLIM_IRQ_UNDERFLOW)) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - if (int_val & (1 << (port_id % 8))) { - int_val = int_val ^ (1 << (port_id % 8)); - regmap_write(wcd->if_regmap, - reg, int_val); - } - } - - regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_CLR_RX_0 + (j / 8), - BIT(j % 8)); - } - - return IRQ_HANDLED; -} - -static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) -{ - int slim_irq = regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS); - int i, ret = 0; - - ret = request_threaded_irq(slim_irq, NULL, wcd9335_slimbus_irq, - IRQF_TRIGGER_RISING, "SLIMBus Slave", wcd); - if (ret) { - dev_err(wcd->dev, "Failed to request SLIMBUS irq\n"); - return ret; - } - - /* enable interrupts on all slave ports */ - for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) - regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, - 0xFF); - - return ret; -} - -static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, - bool ccl_flag) -{ - struct snd_soc_component *comp = wcd->component; - - if (ccl_flag) { - if (++wcd->sido_ccl_cnt == 1) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_DEF_VALUE); - } else { - if (wcd->sido_ccl_cnt == 0) { - dev_err(wcd->dev, "sido_ccl already disabled\n"); - return; - } - if (--wcd->sido_ccl_cnt == 0) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_10_ICHARG_PWR_SEL_C320FF); - } -} - -static int wcd9335_enable_master_bias(struct wcd9335_codec *wcd) -{ - wcd->master_bias_users++; - if (wcd->master_bias_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_ENABLE); - /* - * 1ms delay is required after pre-charge is enabled - * as per HW requirement - */ - usleep_range(1000, 1100); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - - return 0; -} - -static int wcd9335_enable_mclk(struct wcd9335_codec *wcd) -{ - /* Enable mclk requires master bias to be enabled first */ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - if (((wcd->clk_mclk_users == 0) && (wcd->clk_type == WCD_CLK_MCLK)) || - ((wcd->clk_mclk_users > 0) && (wcd->clk_type != WCD_CLK_MCLK))) { - dev_err(wcd->dev, "Error enabling MCLK, clk_type: %d\n", - wcd->clk_type); - return -EINVAL; - } - - if (++wcd->clk_mclk_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_EXTERNAL); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_MCLK_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_MCLK_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_MCLK_ENABLE); - /* - * 10us sleep is required after clock is enabled - * as per HW requirement - */ - usleep_range(10, 15); - } - - wcd->clk_type = WCD_CLK_MCLK; - - return 0; -} - -static int wcd9335_disable_mclk(struct wcd9335_codec *wcd) -{ - if (wcd->clk_mclk_users <= 0) - return -EINVAL; - - if (--wcd->clk_mclk_users == 0) { - if (wcd->clk_rco_users > 0) { - /* MCLK to RCO switch */ - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_RCO); - wcd->clk_type = WCD_CLK_RCO; - } else { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_DISABLE); - wcd->clk_type = WCD_CLK_OFF; - } - - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_DISABLE); - } - - return 0; -} - -static int wcd9335_disable_master_bias(struct wcd9335_codec *wcd) -{ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - wcd->master_bias_users--; - if (wcd->master_bias_users == 0) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - return 0; -} - -static int wcd9335_cdc_req_mclk_enable(struct wcd9335_codec *wcd, - bool enable) -{ - int ret = 0; - - if (enable) { - wcd9335_cdc_sido_ccl_enable(wcd, true); - ret = clk_prepare_enable(wcd->mclk); - if (ret) { - dev_err(wcd->dev, "%s: ext clk enable failed\n", - __func__); - goto err; - } - /* get BG */ - wcd9335_enable_master_bias(wcd); - /* get MCLK */ - wcd9335_enable_mclk(wcd); - - } else { - /* put MCLK */ - wcd9335_disable_mclk(wcd); - /* put BG */ - wcd9335_disable_master_bias(wcd); - clk_disable_unprepare(wcd->mclk); - wcd9335_cdc_sido_ccl_enable(wcd, false); - } -err: - return ret; -} - -static void wcd9335_codec_apply_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - struct snd_soc_component *comp = wcd->component; - int vout_d_val; - - if (req_mv == wcd->sido_voltage) - return; - - /* compute the vout_d step value */ - vout_d_val = WCD9335_CALCULATE_VOUT_D(req_mv) & - WCD9335_ANA_BUCK_VOUT_MASK; - snd_soc_component_write(comp, WCD9335_ANA_BUCK_VOUT_D, vout_d_val); - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_ENABLE); - - /* 1 msec sleep required after SIDO Vout_D voltage change */ - usleep_range(1000, 1100); - wcd->sido_voltage = req_mv; - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_DISABLE); -} - -static int wcd9335_codec_update_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - int ret = 0; - - /* enable mclk before setting SIDO voltage */ - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) { - dev_err(wcd->dev, "Ext clk enable failed\n"); - goto err; - } - - wcd9335_codec_apply_sido_voltage(wcd, req_mv); - wcd9335_cdc_req_mclk_enable(wcd, false); - -err: - return ret; -} - -static int _wcd9335_codec_enable_mclk(struct snd_soc_component *component, - int enable) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int ret; - - if (enable) { - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) - return ret; - - wcd9335_codec_apply_sido_voltage(wcd, - SIDO_VOLTAGE_NOMINAL_MV); - } else { - wcd9335_codec_update_sido_voltage(wcd, - wcd->sido_voltage); - wcd9335_cdc_req_mclk_enable(wcd, false); - } - - return 0; -} - -static void wcd9335_enable_sido_buck(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - - snd_soc_component_update_bits(component, WCD9335_ANA_RCO, - WCD9335_ANA_RCO_BG_EN_MASK, - WCD9335_ANA_RCO_BG_ENABLE); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_EXT); - /* 100us sleep needed after IREF settings */ - usleep_range(100, 110); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_EXT); - /* 100us sleep needed after VREF settings */ - usleep_range(100, 110); - wcd->sido_input_src = SIDO_SOURCE_RCO_BG; -} - -static int wcd9335_enable_efuse_sensing(struct snd_soc_component *comp) -{ - _wcd9335_codec_enable_mclk(comp, true); - snd_soc_component_update_bits(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_CTL, - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK, - WCD9335_CHIP_TIER_CTRL_EFUSE_ENABLE); - /* - * 5ms sleep required after enabling efuse control - * before checking the status. - */ - usleep_range(5000, 5500); - - if (!(snd_soc_component_read32(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_STATUS) & - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK)) - WARN(1, "%s: Efuse sense is not complete\n", __func__); - - wcd9335_enable_sido_buck(comp); - _wcd9335_codec_enable_mclk(comp, false); - - return 0; -} - -static void wcd9335_codec_init(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - /* ungate MCLK and set clk rate */ - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_GATE, - WCD9335_CODEC_RPM_CLK_GATE_MCLK_GATE_MASK, 0); - - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_common_val); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_common_val[i].reg, - wcd9335_codec_reg_init_common_val[i].mask, - wcd9335_codec_reg_init_common_val[i].val); - - if (WCD9335_IS_2_0(wcd->version)) { - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_val_2_0); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_val_2_0[i].reg, - wcd9335_codec_reg_init_val_2_0[i].mask, - wcd9335_codec_reg_init_val_2_0[i].val); - } - - wcd9335_enable_efuse_sensing(component); -} - -static int wcd9335_codec_probe(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - snd_soc_component_init_regmap(component, wcd->regmap); - wcd->component = component; - - wcd9335_codec_init(component); - - for (i = 0; i < NUM_CODEC_DAIS; i++) - INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); - - return wcd9335_setup_irqs(wcd); -} - -static void wcd9335_codec_remove(struct snd_soc_component *comp) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); -} - -static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, - int clk_id, int source, - unsigned int freq, int dir) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - wcd->mclk_rate = freq; - - if (wcd->mclk_rate == WCD9335_MCLK_CLK_12P288MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_12P288MHZ); - else if (wcd->mclk_rate == WCD9335_MCLK_CLK_9P6MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - return clk_set_rate(wcd->mclk, freq); -} - -static const struct snd_soc_component_driver wcd9335_component_drv = { - .probe = wcd9335_codec_probe, - .remove = wcd9335_codec_remove, - .set_sysclk = wcd9335_codec_set_sysclk, -}; - -static int wcd9335_probe(struct platform_device *pdev) -{ - struct wcd9335 *pdata = dev_get_drvdata(pdev->dev.parent); - struct device *dev = &pdev->dev; - struct wcd9335_codec *wcd; - - wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); - if (!wcd) - return -ENOMEM; - - dev_set_drvdata(dev, wcd); - - memcpy(wcd->rx_chs, wcd9335_rx_chs, sizeof(wcd9335_rx_chs)); - - wcd->regmap = pdata->regmap; - wcd->if_regmap = pdata->ifd_regmap; - wcd->slim = pdata->slim; - wcd->slim_ifd = pdata->slim_ifd; - wcd->irq_data = pdata->irq_data; - wcd->version = pdata->version; - wcd->intf_type = pdata->intf_type; - wcd->dev = dev; - wcd->mclk = pdata->mclk; - wcd->native_clk = pdata->native_clk; - wcd->sido_input_src = SIDO_SOURCE_INTERNAL; - wcd->sido_voltage = SIDO_VOLTAGE_NOMINAL_MV; - - return devm_snd_soc_register_component(dev, &wcd9335_component_drv, - wcd9335_slim_dais, - ARRAY_SIZE(wcd9335_slim_dais)); -} - -static struct platform_driver wcd9335_codec_driver = { - .probe = wcd9335_probe, - .driver = { - .name = "wcd9335-codec", - }, -}; -module_platform_driver(wcd9335_codec_driver); -MODULE_DESCRIPTION("WCD9335 Codec driver"); -MODULE_LICENSE("GPL v2"); From 9fb4c2bf130b922c77c16a8368732699799c40de Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 1 Aug 2018 15:37:33 +0530 Subject: [PATCH 433/529] ASoC: soc-pcm: Use delay set in component pointer function Take into account the base delay set in pointer callback. There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Now, Total delay = base delay + cpu_dai delay + codec_dai delay Signed-off-by: Akshu Agrawal Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4019bc10897c..9833e53754cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1179,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous total delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1190,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) offset = component->driver->ops->pointer(substream); break; } + /* base delay if assigned in pointer callback */ + delay = runtime->delay; if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); From 5a6cd13d4faef48bdcb7ae9c1e98175332ced7cd Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 10:58:25 -0500 Subject: [PATCH 434/529] ALSA: pcm: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357375 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 85bab922ce69..0391cb1a4f19 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -353,6 +353,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } + /* fall through */ default: return (__force snd_pcm_format_t)-EINVAL; } From d5e77fca87e636dda510710689b4c2ca93765598 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 11:14:16 -0500 Subject: [PATCH 435/529] ALSA: usb: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115084 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bbc7116c9543..382847154227 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,6 +1694,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; From ef965ad5a7697ff16e3be01954f5c57208e36c22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2018 14:38:18 +0200 Subject: [PATCH 436/529] ALSA: seq: Minor cleanup of MIDI event parser helpers snd_midi_event_encode_byte() can never fail, and it can return rather true/false. Change the return type to bool, adjust the argument to receive a MIDI byte as unsigned char, and adjust the comment accordingly. This allows callers to drop error checks, which simplifies the code. Meanwhile, snd_midi_event_encode() helper is used only in seq_midi.c, and it can be better folded into it. This will reduce the total amount of lines in the end. Signed-off-by: Takashi Iwai --- include/sound/seq_midi_event.h | 6 ++--- sound/core/seq/oss/seq_oss_midi.c | 2 +- sound/core/seq/seq_midi.c | 24 ++++++++---------- sound/core/seq/seq_midi_event.c | 42 ++++++------------------------- sound/core/seq/seq_virmidi.c | 4 +-- 5 files changed, 22 insertions(+), 56 deletions(-) diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h index e40f43e6fc7b..2f135bccf457 100644 --- a/include/sound/seq_midi_event.h +++ b/include/sound/seq_midi_event.h @@ -43,10 +43,8 @@ void snd_midi_event_free(struct snd_midi_event *dev); void snd_midi_event_reset_encode(struct snd_midi_event *dev); void snd_midi_event_reset_decode(struct snd_midi_event *dev); void snd_midi_event_no_status(struct snd_midi_event *dev, int on); -/* encode from byte stream - return number of written bytes if success */ -long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count, - struct snd_seq_event *ev); -int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev); +bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, + struct snd_seq_event *ev); /* decode from event to bytes - return number of written bytes if success */ long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count, struct snd_seq_event *ev); diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 9debd1b8fd28..0d5f8b16d057 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -637,7 +637,7 @@ snd_seq_oss_midi_putc(struct seq_oss_devinfo *dp, int dev, unsigned char c, stru if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (snd_midi_event_encode_byte(mdev->coder, c, ev) > 0) { + if (snd_midi_event_encode_byte(mdev->coder, c, ev)) { snd_seq_oss_fill_addr(dp, ev, mdev->client, mdev->port); snd_use_lock_free(&mdev->use_lock); return 0; diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 5dd0ee258359..9e0dabd3ce5f 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -78,7 +78,7 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream) struct seq_midisynth *msynth; struct snd_seq_event ev; char buf[16], *pbuf; - long res, count; + long res; if (substream == NULL) return; @@ -94,19 +94,15 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream) if (msynth->parser == NULL) continue; pbuf = buf; - while (res > 0) { - count = snd_midi_event_encode(msynth->parser, pbuf, res, &ev); - if (count < 0) - break; - pbuf += count; - res -= count; - if (ev.type != SNDRV_SEQ_EVENT_NONE) { - ev.source.port = msynth->seq_port; - ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; - snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0); - /* clear event and reset header */ - memset(&ev, 0, sizeof(ev)); - } + while (res-- > 0) { + if (!snd_midi_event_encode_byte(msynth->parser, + *pbuf++, &ev)) + continue; + ev.source.port = msynth->seq_port; + ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; + snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0); + /* clear event and reset header */ + memset(&ev, 0, sizeof(ev)); } } } diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..53c0dfab90d6 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -214,45 +214,17 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) } #endif /* 0 */ -/* - * read bytes and encode to sequencer event if finished - * return the size of encoded bytes - */ -long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count, - struct snd_seq_event *ev) -{ - long result = 0; - int rc; - - ev->type = SNDRV_SEQ_EVENT_NONE; - - while (count-- > 0) { - rc = snd_midi_event_encode_byte(dev, *buf++, ev); - result++; - if (rc < 0) - return rc; - else if (rc > 0) - return result; - } - - return result; -} -EXPORT_SYMBOL(snd_midi_event_encode); - /* * read one byte and encode to sequencer event: - * return 1 if MIDI bytes are encoded to an event - * 0 data is not finished - * negative for error + * return true if MIDI bytes are encoded to an event + * false data is not finished */ -int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, - struct snd_seq_event *ev) +bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, + struct snd_seq_event *ev) { - int rc = 0; + bool rc = false; unsigned long flags; - c &= 0xff; - if (c >= MIDI_CMD_COMMON_CLOCK) { /* real-time event */ ev->type = status_event[ST_SPECIAL + c - 0xf0].event; @@ -293,7 +265,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, status_event[dev->type].encode(dev, ev); if (dev->type >= ST_SPECIAL) dev->type = ST_INVALID; - rc = 1; + rc = true; } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { @@ -306,7 +278,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, dev->read = 0; /* continue to parse */ else reset_encode(dev); /* all parsed */ - rc = 1; + rc = true; } } diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 03ac5e72dbe6..0c84926eb726 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -174,8 +174,8 @@ static void snd_vmidi_output_work(struct work_struct *work) while (READ_ONCE(vmidi->trigger)) { if (snd_rawmidi_transmit(substream, &input, 1) != 1) break; - if (snd_midi_event_encode_byte(vmidi->parser, input, - &vmidi->event) <= 0) + if (!snd_midi_event_encode_byte(vmidi->parser, input, + &vmidi->event)) continue; if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { ret = snd_seq_kernel_client_dispatch(vmidi->client, From fc4bfd9a35f3d9cbf5ad6a20faedca71d1d9ed52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2018 14:59:07 +0200 Subject: [PATCH 437/529] ALSA: seq: Remove dead codes There are a few functions that have been commented out for ages. And also there are functions that do nothing but placeholders. Let's kill them. Signed-off-by: Takashi Iwai --- sound/core/seq/seq.c | 7 ----- sound/core/seq/seq_memory.c | 12 --------- sound/core/seq/seq_memory.h | 6 ----- sound/core/seq/seq_midi_emul.c | 12 --------- sound/core/seq/seq_midi_event.c | 45 --------------------------------- sound/core/seq/seq_virmidi.c | 33 ------------------------ 6 files changed, 115 deletions(-) diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 639544b4fb04..e685eccdc741 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -87,10 +87,6 @@ static int __init alsa_seq_init(void) if ((err = client_init_data()) < 0) goto error; - /* init memory, room for selected events */ - if ((err = snd_sequencer_memory_init()) < 0) - goto error; - /* init event queues */ if ((err = snd_seq_queues_init()) < 0) goto error; @@ -126,9 +122,6 @@ static void __exit alsa_seq_exit(void) /* unregister sequencer device */ snd_sequencer_device_done(); - /* release event memory */ - snd_sequencer_memory_done(); - snd_seq_autoload_exit(); } diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index a4c8543176b2..5b0388202bac 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -504,18 +504,6 @@ int snd_seq_pool_delete(struct snd_seq_pool **ppool) return 0; } -/* initialize sequencer memory */ -int __init snd_sequencer_memory_init(void) -{ - return 0; -} - -/* release sequencer memory */ -void __exit snd_sequencer_memory_done(void) -{ -} - - /* exported to seq_clientmgr.c */ void snd_seq_info_pool(struct snd_info_buffer *buffer, struct snd_seq_pool *pool, char *space) diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h index 3abe306c394a..1292fe91f02e 100644 --- a/sound/core/seq/seq_memory.h +++ b/sound/core/seq/seq_memory.h @@ -94,12 +94,6 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize); /* remove pool */ int snd_seq_pool_delete(struct snd_seq_pool **pool); -/* init memory */ -int snd_sequencer_memory_init(void); - -/* release event memory */ -void snd_sequencer_memory_done(void); - /* polling */ int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file, poll_table *wait); diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 288f839a554b..f9f21331aeea 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -728,15 +728,3 @@ void snd_midi_channel_free_set(struct snd_midi_channel_set *chset) kfree(chset); } EXPORT_SYMBOL(snd_midi_channel_free_set); - -static int __init alsa_seq_midi_emul_init(void) -{ - return 0; -} - -static void __exit alsa_seq_midi_emul_exit(void) -{ -} - -module_init(alsa_seq_midi_emul_init) -module_exit(alsa_seq_midi_emul_exit) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 53c0dfab90d6..b11419537062 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -175,45 +175,12 @@ void snd_midi_event_reset_decode(struct snd_midi_event *dev) } EXPORT_SYMBOL(snd_midi_event_reset_decode); -#if 0 -void snd_midi_event_init(struct snd_midi_event *dev) -{ - snd_midi_event_reset_encode(dev); - snd_midi_event_reset_decode(dev); -} -#endif /* 0 */ - void snd_midi_event_no_status(struct snd_midi_event *dev, int on) { dev->nostat = on ? 1 : 0; } EXPORT_SYMBOL(snd_midi_event_no_status); -/* - * resize buffer - */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) -{ - unsigned char *new_buf, *old_buf; - unsigned long flags; - - if (bufsize == dev->bufsize) - return 0; - new_buf = kmalloc(bufsize, GFP_KERNEL); - if (new_buf == NULL) - return -ENOMEM; - spin_lock_irqsave(&dev->lock, flags); - old_buf = dev->buf; - dev->buf = new_buf; - dev->bufsize = bufsize; - reset_encode(dev); - spin_unlock_irqrestore(&dev->lock, flags); - kfree(old_buf); - return 0; -} -#endif /* 0 */ - /* * read one byte and encode to sequencer event: * return true if MIDI bytes are encoded to an event @@ -503,15 +470,3 @@ static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf, } return idx; } - -static int __init alsa_seq_midi_event_init(void) -{ - return 0; -} - -static void __exit alsa_seq_midi_event_exit(void) -{ -} - -module_init(alsa_seq_midi_event_init) -module_exit(alsa_seq_midi_event_exit) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 0c84926eb726..a2f1c6b58693 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -109,23 +109,6 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, return 0; } -/* - * receive an event from the remote virmidi port - * - * for rawmidi inputs, you can call this function from the event - * handler of a remote port which is attached to the virmidi via - * SNDRV_VIRMIDI_SEQ_ATTACH. - */ -#if 0 -int snd_virmidi_receive(struct snd_rawmidi *rmidi, struct snd_seq_event *ev) -{ - struct snd_virmidi_dev *rdev; - - rdev = rmidi->private_data; - return snd_virmidi_dev_receive_event(rdev, ev, true); -} -#endif /* 0 */ - /* * event handler of virmidi port */ @@ -544,19 +527,3 @@ int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmi return 0; } EXPORT_SYMBOL(snd_virmidi_new); - -/* - * ENTRY functions - */ - -static int __init alsa_virmidi_init(void) -{ - return 0; -} - -static void __exit alsa_virmidi_exit(void) -{ -} - -module_init(alsa_virmidi_init) -module_exit(alsa_virmidi_exit) From 00976ad5271999ba06d24319fd1031b178aff832 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2018 16:37:02 +0200 Subject: [PATCH 438/529] ALSA: seq: Fix leftovers at probe error path The sequencer core module doesn't call some destructors in the error path of the init code, which may leave some resources. This patch mainly fix these leaks by calling the destructors appropriately at alsa_seq_init(). Also the patch brings a few cleanups along with it, namely: - Expand the old "if ((err = xxx) < 0)" coding style - Get rid of empty seq_queue_init() and its caller - Change snd_seq_info_done() to void Last but not least, a couple of functions lose __exit annotation since they are called also in alsa_seq_init(). No functional changes but minor code cleanups. Signed-off-by: Takashi Iwai --- sound/core/seq/seq.c | 26 ++++++++++++++++---------- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_info.c | 10 ++-------- sound/core/seq/seq_info.h | 6 +++--- sound/core/seq/seq_queue.c | 12 +----------- sound/core/seq/seq_queue.h | 3 --- 6 files changed, 23 insertions(+), 36 deletions(-) diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index e685eccdc741..7de98d71f2aa 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -84,26 +84,32 @@ static int __init alsa_seq_init(void) { int err; - if ((err = client_init_data()) < 0) - goto error; - - /* init event queues */ - if ((err = snd_seq_queues_init()) < 0) + err = client_init_data(); + if (err < 0) goto error; /* register sequencer device */ - if ((err = snd_sequencer_device_init()) < 0) + err = snd_sequencer_device_init(); + if (err < 0) goto error; /* register proc interface */ - if ((err = snd_seq_info_init()) < 0) - goto error; + err = snd_seq_info_init(); + if (err < 0) + goto error_device; /* register our internal client */ - if ((err = snd_seq_system_client_init()) < 0) - goto error; + err = snd_seq_system_client_init(); + if (err < 0) + goto error_info; snd_seq_autoload_init(); + return 0; + + error_info: + snd_seq_info_done(); + error_device: + snd_sequencer_device_done(); error: return err; } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6fd4b074b206..a0b768e2f697 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2543,7 +2543,7 @@ int __init snd_sequencer_device_init(void) /* * unregister sequencer device */ -void __exit snd_sequencer_device_done(void) +void snd_sequencer_device_done(void) { snd_unregister_device(&seq_dev); put_device(&seq_dev); diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c index 97015447b9b3..b27fedd435b6 100644 --- a/sound/core/seq/seq_info.c +++ b/sound/core/seq/seq_info.c @@ -50,7 +50,7 @@ create_info_entry(char *name, void (*read)(struct snd_info_entry *, return entry; } -static void free_info_entries(void) +void snd_seq_info_done(void) { snd_info_free_entry(queues_entry); snd_info_free_entry(clients_entry); @@ -70,12 +70,6 @@ int __init snd_seq_info_init(void) return 0; error: - free_info_entries(); + snd_seq_info_done(); return -ENOMEM; } - -int __exit snd_seq_info_done(void) -{ - free_info_entries(); - return 0; -} diff --git a/sound/core/seq/seq_info.h b/sound/core/seq/seq_info.h index f8549f81a645..2cdf8f6e63f5 100644 --- a/sound/core/seq/seq_info.h +++ b/sound/core/seq/seq_info.h @@ -30,11 +30,11 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, struct snd_info_buff #ifdef CONFIG_SND_PROC_FS -int snd_seq_info_init( void ); -int snd_seq_info_done( void ); +int snd_seq_info_init(void); +void snd_seq_info_done(void); #else static inline int snd_seq_info_init(void) { return 0; } -static inline int snd_seq_info_done(void) { return 0; } +static inline void snd_seq_info_done(void) {} #endif #endif diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index b377f5048352..3b3ac96f1f5f 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -159,18 +159,8 @@ static void queue_delete(struct snd_seq_queue *q) /*----------------------------------------------------------------*/ -/* setup queues */ -int __init snd_seq_queues_init(void) -{ - /* - memset(queue_list, 0, sizeof(queue_list)); - num_queues = 0; - */ - return 0; -} - /* delete all existing queues */ -void __exit snd_seq_queues_delete(void) +void snd_seq_queues_delete(void) { int i; diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 719093489a2c..76db43b79a2b 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -63,9 +63,6 @@ struct snd_seq_queue { /* get the number of current queues */ int snd_seq_queue_get_cur_queues(void); -/* init queues structure */ -int snd_seq_queues_init(void); - /* delete queues */ void snd_seq_queues_delete(void); From 04702e8d0092832eaeeacc6b1bfbf81a66f242c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2018 16:42:29 +0200 Subject: [PATCH 439/529] ALSA: seq: Use no intrruptible mutex_lock All usages of mutex in ALSA sequencer core would take too long, hence we don't have to care about the user interruption that makes things complicated. Let's replace them with simpler mutex_lock(). Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index a0b768e2f697..92e6524a3a9d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -311,10 +311,9 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (err < 0) return err; - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; + mutex_lock(®ister_mutex); client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); - if (client == NULL) { + if (!client) { mutex_unlock(®ister_mutex); return -ENOMEM; /* failure code */ } @@ -1704,10 +1703,7 @@ static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client, if (queue == NULL) return -EINVAL; - if (mutex_lock_interruptible(&queue->timer_mutex)) { - queuefree(queue); - return -ERESTARTSYS; - } + mutex_lock(&queue->timer_mutex); tmr = queue->timer; memset(timer, 0, sizeof(*timer)); timer->queue = queue->queue; @@ -1741,10 +1737,7 @@ static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client, q = queueptr(timer->queue); if (q == NULL) return -ENXIO; - if (mutex_lock_interruptible(&q->timer_mutex)) { - queuefree(q); - return -ERESTARTSYS; - } + mutex_lock(&q->timer_mutex); tmr = q->timer; snd_seq_queue_timer_close(timer->queue); tmr->type = timer->type; @@ -2180,8 +2173,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, if (card == NULL && client_index >= SNDRV_SEQ_GLOBAL_CLIENTS) return -EINVAL; - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; + mutex_lock(®ister_mutex); if (card) { client_index += SNDRV_SEQ_GLOBAL_CLIENTS @@ -2522,19 +2514,15 @@ int __init snd_sequencer_device_init(void) snd_device_initialize(&seq_dev, NULL); dev_set_name(&seq_dev, "seq"); - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; - + mutex_lock(®ister_mutex); err = snd_register_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0, &snd_seq_f_ops, NULL, &seq_dev); + mutex_unlock(®ister_mutex); if (err < 0) { - mutex_unlock(®ister_mutex); put_device(&seq_dev); return err; } - mutex_unlock(®ister_mutex); - return 0; } From 93ce1b12966d9d60ee5583ffbde822a22909568e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2018 16:43:56 +0200 Subject: [PATCH 440/529] ALSA: seq: Drop unused 64bit division macros The old ugly macros remained in the code without usage. Rip them off. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_queue.h | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 76db43b79a2b..e006fc8e3a36 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -109,28 +109,4 @@ int snd_seq_queue_is_used(int queueid, int client); int snd_seq_control_queue(struct snd_seq_event *ev, int atomic, int hop); -/* - * 64bit division - for sync stuff.. - */ -#if defined(i386) || defined(i486) - -#define udiv_qrnnd(q, r, n1, n0, d) \ - __asm__ ("divl %4" \ - : "=a" ((u32)(q)), \ - "=d" ((u32)(r)) \ - : "0" ((u32)(n0)), \ - "1" ((u32)(n1)), \ - "rm" ((u32)(d))) - -#define u64_div(x,y,q) do {u32 __tmp; udiv_qrnnd(q, __tmp, (x)>>32, x, y);} while (0) -#define u64_mod(x,y,r) do {u32 __tmp; udiv_qrnnd(__tmp, q, (x)>>32, x, y);} while (0) -#define u64_divmod(x,y,q,r) udiv_qrnnd(q, r, (x)>>32, x, y) - -#else -#define u64_div(x,y,q) ((q) = (u32)((u64)(x) / (u64)(y))) -#define u64_mod(x,y,r) ((r) = (u32)((u64)(x) % (u64)(y))) -#define u64_divmod(x,y,q,r) (u64_div(x,y,q), u64_mod(x,y,r)) -#endif - - #endif From 11175556eec58b27a1398e80f8b1302314f2242c Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 2 Aug 2018 04:31:02 +0000 Subject: [PATCH 441/529] ALSA: usb-audio: Fix invalid use of sizeof in parse_uac_endpoint_attributes() sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wei Yongjun Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8fe3b0e00e45..67cf849aa16b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1037,7 +1037,7 @@ found_clock: fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; - pd = kzalloc(sizeof(pd), GFP_KERNEL); + pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { kfree(fp->rate_table); kfree(fp); From 789b7f43858f1c74a1c528e8b8f78c78624800d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Aug 2018 07:40:19 +0200 Subject: [PATCH 442/529] ALSA: sb: Fix a typo There was a typo of COPY_USER in the dead code (that is disabled as default). Fixes: 4b83eff81c81 ("ALSA: sb: Convert to the new PCM ops") Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index bc5af71d3bdb..f46f6ec3ea0c 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -470,7 +470,7 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, /* convert to word unit */ pos = (pos << 1) + rec->loop_start[voice]; count <<= 1; - LOOP_WRITE(rec, pos, src, count, COPY_UESR); + LOOP_WRITE(rec, pos, src, count, COPY_USER); return 0; } From 9a73f6a235c2351adc07471519d4980e2fb33bbc Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:47:09 -0500 Subject: [PATCH 443/529] ASoC: wm8961: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1271173 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index f70f563d59f3..68b4cadc308f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { From 065dcc270af66acb4226b49f635b41a054b663d4 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:49:13 -0500 Subject: [PATCH 444/529] ASoC: rt5640: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056547 ("Missing break in switch") Addresses-Coverity-ID: 1056548 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8bf8d360c25f..27770143ae8f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; From 43a26bd026dab09a8b28c40e94ba534a52375b20 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:50:20 -0500 Subject: [PATCH 445/529] ASoC: rt5677: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1271174 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..922becfee59b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; + /* fall through */ case 24: val |= (2 << 8); break; From 85e7e77079f3201799467803f6c8533a9921e32d Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:51:18 -0500 Subject: [PATCH 446/529] ASoC: wm8955: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115047 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ba44e3d6c1e0..cd204f79647d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; From 3eb7dbc6d844d2033271434d02ff90ac9b19e393 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:51:59 -0500 Subject: [PATCH 447/529] ASoC: wm8960: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115041 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c30f5aa392c6..8dc1f3d6a988 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } + /* fall through */ default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); From da41787b9f319a7a23fcc8cf65c6ad646803962e Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:52:41 -0500 Subject: [PATCH 448/529] ASoC: wm8904: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115042 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9037a35b931d..1965635ec07c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; From 42ef3c94ff6e8315377c04504f1bce50d1f21738 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:53:19 -0500 Subject: [PATCH 449/529] ASoC: wm8996: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 146354 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d9d206046f8c..78a408236cfb 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + /* fall through */ case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, From a9531ab151117e976544e0dc74a9bc8cbd01145f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:54:03 -0500 Subject: [PATCH 450/529] ASoC: wm8962: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115043 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a11e9d6bf950..efd8910b1ff7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; From af5d1d5d4ba71164711025e077511ad48d5690e9 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:54:45 -0500 Subject: [PATCH 451/529] ASoC: wm8995: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115045 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 60e227832331..68c99fe37097 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; From 7a2235ef507822679ac89007a6a37b23f30eed56 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:55:26 -0500 Subject: [PATCH 452/529] ASoC: wm9081: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357430 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5a0ea7b3c149..399255d1f78a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; From 2cea1542859bc812f1ec51ea71c06e927e5b922e Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 1 Aug 2018 14:56:16 -0500 Subject: [PATCH 453/529] ASoC: wm8994: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115050 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7fdfdf3f6e67..62f8c5b9ba92 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } + /* fall through */ default: return -EINVAL; From 73edbe42582207ff8f47168f05124376394aa643 Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Thu, 2 Aug 2018 12:32:59 +0530 Subject: [PATCH 454/529] ASoC: qcom: Fix unmet dependency warning for SND_SOC_SDM845 Add DEPENDS_ON QCOM_APR for SND_SOC_SDM845 to fix the warning: unmet direct dependencies detected for SND_SOC_QDSP6. Reported-by: Stephen Rothwell Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 350730839c6f..5f03312ef007 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -93,6 +93,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" + depends on QCOM_APR select SND_SOC_QDSP6 help To add support for audio on Qualcomm Technologies Inc. From fdec79c18b08c68cfa079f2d3ee23c5a120a2eda Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 2 Aug 2018 01:47:30 +0000 Subject: [PATCH 455/529] ASoC: fsi: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/sh_fsi.h | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 7a9710b4b799..89eafe23ef88 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -1,16 +1,13 @@ -#ifndef __SOUND_FSI_H -#define __SOUND_FSI_H - -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Fifo-attached Serial Interface (FSI) support for SH7724 * * Copyright (C) 2009 Renesas Solutions Corp. * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + #include #include From 8e82a728792bf66b9f0a29c9d4c4b0630f7b9c79 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 2 Aug 2018 14:04:45 +0200 Subject: [PATCH 456/529] ALSA: hda: Correct Asrock B85M-ITX power_save blacklist entry I added the subsys product-id for the HDMI HDA device rather then for the PCH one, this commit fixes this. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Cc: stable@vger.kernel.org Signed-off-by: Hans de Goede Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..647ae1a71e10 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2207,7 +2207,7 @@ out_free: */ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ - SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ From 1eb576881ff884dd6d10272b96cc336d156492c2 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:36 +0100 Subject: [PATCH 457/529] ASoC: apq8096: remove auto rebinding Remove auto rebinding support, as component framework can deadlock in few usecases if we are trying to add new/remove component within a bind/unbind callbacks. Card rebinding is ASoC core feature so all the previous component framework stuff in q6dsp remains removed. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1e4a90d59228..6ee7e66cbaaa 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -48,7 +48,6 @@ static int apq8096_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; - card->auto_bind = true; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) { @@ -74,7 +73,6 @@ static int apq8096_platform_remove(struct platform_device *pdev) { struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); - card->auto_bind = false; snd_soc_unregister_card(card); kfree(card->dai_link); kfree(card); From 62121debfb31a8700e387bd2987779b3a98bc520 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:37 +0100 Subject: [PATCH 458/529] ASoC: smd845: remove auto rebinding Remove auto rebinding support, as component framework can deadlock in few usecases if we are trying to add new/remove component within a bind/unbind callbacks. Card rebinding is ASoC core feature so all the previous component framework stuff in q6dsp remains removed. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index bf4ec4646906..be0cb1122036 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -226,7 +226,6 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; - card->auto_bind = true; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) { @@ -258,7 +257,6 @@ static int sdm845_snd_platform_remove(struct platform_device *pdev) struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - card->auto_bind = false; snd_soc_unregister_card(card); kfree(card->dai_link); kfree(data); From 611cbc8799b672f41b6ba7afed758ad9efb959a7 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:38 +0100 Subject: [PATCH 459/529] ASoC: core: remove support for card rebind using component framework DRM based audio components get registered inside the component framework bind callback. However component framework has a big mutex lock taken for every call to component_add, component_del and bind, unbind callbacks. This can lead to deadlock situation if we are trying to add new/remove component within a bind/unbind callbacks. Which is what was happening with bcm2837 rpi 3. Revert this change till we sort out the mutex issue. Reported-by: Guillaume Tucker Reported-by: Stefan Wahren Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ----- sound/soc/soc-core.c | 62 -------------------------------------------- 2 files changed, 69 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ace474e8649e..41cec42fb456 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include @@ -1091,12 +1090,6 @@ struct snd_soc_card { struct work_struct deferred_resume_work; - /* component framework related */ - bool components_added; - /* set in machine driver to enable/disable auto re-binding */ - bool auto_bind; - struct component_match *match; - /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 81b27923303d..82eb3868be67 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -279,28 +279,11 @@ static inline void snd_soc_debugfs_exit(void) #endif -static int snd_soc_card_comp_compare(struct device *dev, void *data) -{ - struct snd_soc_component *component; - - lockdep_assert_held(&client_mutex); - list_for_each_entry(component, &component_list, list) { - if (dev == component->dev) { - if (!strcmp(component->name, data)) - return 1; - break; - } - } - - return 0; -} - static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, struct snd_soc_component *component) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_rtdcom_list *new_rtdcom; - char *cname; for_each_rtdcom(rtd, rtdcom) { /* already connected */ @@ -317,13 +300,6 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, list_add_tail(&new_rtdcom->list, &rtd->component_list); - if (rtd->card->auto_bind && !rtd->card->components_added) { - cname = devm_kasprintf(rtd->card->dev, GFP_KERNEL, - "%s", component->name); - component_match_add(rtd->card->dev, &rtd->card->match, - snd_soc_card_comp_compare, cname); - } - return 0; } @@ -859,25 +835,6 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, return false; } -static int snd_soc_card_comp_bind(struct device *dev) -{ - struct snd_soc_card *card = dev_get_drvdata(dev); - - if (card->instantiated) - return 0; - - return snd_soc_register_card(card); -} - -static void snd_soc_card_comp_unbind(struct device *dev) -{ -} - -static const struct component_master_ops snd_soc_card_comp_ops = { - .bind = snd_soc_card_comp_bind, - .unbind = snd_soc_card_comp_unbind, -}; - static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2169,12 +2126,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); - if (card->auto_bind && !card->components_added) { - component_master_add_with_match(card->dev, - &snd_soc_card_comp_ops, - card->match); - card->components_added = true; - } mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2820,9 +2771,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card) dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } - if (!card->auto_bind && card->components_added) - component_master_del(card->dev, &snd_soc_card_comp_ops); - return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); @@ -3235,17 +3183,8 @@ int snd_soc_add_component(struct device *dev, snd_soc_component_add(component); - ret = component_add(dev, NULL); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to add Component: %d\n", ret); - goto err_comp; - } - return 0; -err_comp: - soc_remove_component(component); - snd_soc_unregister_dais(component); err_cleanup: snd_soc_component_cleanup(component); err_free: @@ -3293,7 +3232,6 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_unlock(&client_mutex); if (found) { - component_del(dev, NULL); snd_soc_component_cleanup(component); } From 26a6dce8ef99b0199e7682d1c203cf7d0b5fd5b0 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Thu, 2 Aug 2018 18:21:31 -0500 Subject: [PATCH 460/529] ASoC: Intel: bxt: Use refcap device for mono recording The refcap capture device supports mono recording only, this patch adds the channel constraints. Signed-off-by: Yong Zhi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index be6e4b40bf03..6f052fc8d1e2 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); From a9fe47e5e96be6401738f6c1e087edbcff6a5ceb Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 2 Aug 2018 15:40:37 -0500 Subject: [PATCH 461/529] ALSA: galaxy: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1468367 ("Missing break in switch") Addresses-Coverity-ID: 115037 ("Missing break in switch") Addresses-Coverity-ID: 115038 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/isa/galaxy/galaxy.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index b9994cc9f5fb..af9eea41379f 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -260,6 +260,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; + /* Fall through */ case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -304,6 +305,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; + /* Fall through */ default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -333,6 +335,7 @@ mpu: break; case 2: mpu_irq[n] = 9; + /* Fall through */ case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; From 3e313f34720ea1bb876636263e30a779ff65836e Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 2 Aug 2018 15:41:21 -0500 Subject: [PATCH 462/529] ALSA: opti92x: mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1165394 ("Missing break in switch") Addresses-Coverity-ID: 1167851 ("Missing break in switch") Addresses-Coverity-ID: 402015 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 505cd81e19fa..ac0ab6eb40f0 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -261,6 +261,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } + /* Fall through */ case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -303,6 +304,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } + /* Fall through */ case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -350,6 +352,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); + /* Fall through */ case OPTi9XX_HW_82C925: /* enable WSS mode */ From 734be97b967c7e2bab72b34da87534ccd854c72d Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 2 Aug 2018 15:41:55 -0500 Subject: [PATCH 463/529] ALSA: opti9xx: mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 402016 ("Missing break in switch") Addresses-Coverity-ID: 1056542 ("Missing break in switch") Addresses-Coverity-ID: 1339579 ("Missing break in switch") Addresses-Coverity-ID: 1369526 ("Missing break in switch") Addresses-Coverity-ID: 1369529 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 8894c7c18ad6..c6136c6b0214 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -176,10 +176,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; + /* fall through */ case 10 ... 19: out /= 10; + /* fall through */ case 20 ... 30: out /= 10; + /* fall through */ default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -834,6 +837,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } + /* fall through */ case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -863,6 +867,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } + /* fall through */ case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); From 13a01635823c409a0911304ee50987cb851317da Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 2 Aug 2018 15:42:39 -0500 Subject: [PATCH 464/529] ALSA: es18xx: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115075 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 2a6960c3e2a4..0d103d6f805e 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1024,6 +1024,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem val = 3; } else retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00; + /* fall through */ /* 4 source chips */ case 0x1868: case 0x1878: From 9038820cef3c3824cdd846d69894746a73c053db Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Fri, 3 Aug 2018 14:47:56 +0100 Subject: [PATCH 465/529] ALSA: gus: fix spelling mistake "acumulator" -> "accumulator" Trivial spelling mistake fix in debug message Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index ca79878d8d8c..2fd32ef22c30 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -461,7 +461,7 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus) printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16)); printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d)); - printk(KERN_INFO " -%i- GFA1 effect acumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); + printk(KERN_INFO " -%i- GFA1 effect accumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); } if (mode & 0x20) { printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4); From a6403299893982a2c0b0a6f43261976c43e1598e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:40:25 +0200 Subject: [PATCH 466/529] ALSA: compress: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4b01a37c836e..26b5e245b074 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1160,18 +1160,6 @@ int snd_compress_deregister(struct snd_compr *device) } EXPORT_SYMBOL_GPL(snd_compress_deregister); -static int __init snd_compress_init(void) -{ - return 0; -} - -static void __exit snd_compress_exit(void) -{ -} - -module_init(snd_compress_init); -module_exit(snd_compress_exit); - MODULE_DESCRIPTION("ALSA Compressed offload framework"); MODULE_AUTHOR("Vinod Koul "); MODULE_LICENSE("GPL v2"); From 969686ee0e0ff62ece428e8e02b07f81ac88a84d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:42:46 +0200 Subject: [PATCH 467/529] ALSA: drivers: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/drivers/mpu401/mpu401_uart.c | 16 ---------------- sound/drivers/opl3/opl3_lib.c | 16 ---------------- sound/drivers/opl4/opl4_lib.c | 12 ------------ sound/drivers/vx/vx_core.c | 15 --------------- 4 files changed, 59 deletions(-) diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 3e745f47dd2f..dae26e856b26 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -617,19 +617,3 @@ free_device: } EXPORT_SYMBOL(snd_mpu401_uart_new); - -/* - * INIT part - */ - -static int __init alsa_mpu401_uart_init(void) -{ - return 0; -} - -static void __exit alsa_mpu401_uart_exit(void) -{ -} - -module_init(alsa_mpu401_uart_init) -module_exit(alsa_mpu401_uart_exit) diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 1a5355b747ec..cf86c36c7c3b 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -538,19 +538,3 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, } EXPORT_SYMBOL(snd_opl3_hwdep_new); - -/* - * INIT part - */ - -static int __init alsa_opl3_init(void) -{ - return 0; -} - -static void __exit alsa_opl3_exit(void) -{ -} - -module_init(alsa_opl3_init) -module_exit(alsa_opl3_exit) diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index db76a5bf2bd2..819d2dce2a19 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -263,15 +263,3 @@ int snd_opl4_create(struct snd_card *card, } EXPORT_SYMBOL(snd_opl4_create); - -static int __init alsa_opl4_init(void) -{ - return 0; -} - -static void __exit alsa_opl4_exit(void) -{ -} - -module_init(alsa_opl4_init) -module_exit(alsa_opl4_exit) diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 121357397a6d..04368dd59a4c 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -815,18 +815,3 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, } EXPORT_SYMBOL(snd_vx_create); - -/* - * module entries - */ -static int __init alsa_vx_core_init(void) -{ - return 0; -} - -static void __exit alsa_vx_core_exit(void) -{ -} - -module_init(alsa_vx_core_init) -module_exit(alsa_vx_core_exit) From 498aaa9152eef3e101ec897f9c9b51662f700830 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:44:15 +0200 Subject: [PATCH 468/529] ALSA: isa: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688_lib.c | 16 ---------------- sound/isa/gus/gus_main.c | 16 ---------------- sound/isa/sb/sb16_csp.c | 16 ---------------- sound/isa/sb/sb16_main.c | 16 ---------------- sound/isa/sb/sb8_main.c | 16 ---------------- sound/isa/sb/sb_common.c | 16 ---------------- sound/isa/wss/wss_lib.c | 16 ---------------- 7 files changed, 112 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index f9c0662e9a22..50cdce0e8946 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -1029,19 +1029,3 @@ EXPORT_SYMBOL(snd_es1688_mixer_write); EXPORT_SYMBOL(snd_es1688_create); EXPORT_SYMBOL(snd_es1688_pcm); EXPORT_SYMBOL(snd_es1688_mixer); - -/* - * INIT part - */ - -static int __init alsa_es1688_init(void) -{ - return 0; -} - -static void __exit alsa_es1688_exit(void) -{ -} - -module_init(alsa_es1688_init) -module_exit(alsa_es1688_exit) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 3cf9b13c780a..3b8a0c880db5 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -465,19 +465,3 @@ EXPORT_SYMBOL(snd_gf1_mem_alloc); EXPORT_SYMBOL(snd_gf1_mem_xfree); EXPORT_SYMBOL(snd_gf1_mem_free); EXPORT_SYMBOL(snd_gf1_mem_lock); - -/* - * INIT part - */ - -static int __init alsa_gus_init(void) -{ - return 0; -} - -static void __exit alsa_gus_exit(void) -{ -} - -module_init(alsa_gus_init) -module_exit(alsa_gus_exit) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index b9d67a7065cd..bf3db0d2ea12 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1185,19 +1185,3 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff /* */ EXPORT_SYMBOL(snd_sb_csp_new); - -/* - * INIT part - */ - -static int __init alsa_sb_csp_init(void) -{ - return 0; -} - -static void __exit alsa_sb_csp_exit(void) -{ -} - -module_init(alsa_sb_csp_init) -module_exit(alsa_sb_csp_exit) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 11ed4a6e5bf1..37e6ce7b0b13 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -900,19 +900,3 @@ EXPORT_SYMBOL(snd_sb16dsp_pcm); EXPORT_SYMBOL(snd_sb16dsp_get_pcm_ops); EXPORT_SYMBOL(snd_sb16dsp_configure); EXPORT_SYMBOL(snd_sb16dsp_interrupt); - -/* - * INIT part - */ - -static int __init alsa_sb16_init(void) -{ - return 0; -} - -static void __exit alsa_sb16_exit(void) -{ -} - -module_init(alsa_sb16_init) -module_exit(alsa_sb16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 80e7dcaa551f..481797744b3c 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -621,19 +621,3 @@ EXPORT_SYMBOL(snd_sb8dsp_interrupt); /* sb8_midi.c */ EXPORT_SYMBOL(snd_sb8dsp_midi_interrupt); EXPORT_SYMBOL(snd_sb8dsp_midi); - -/* - * INIT part - */ - -static int __init alsa_sb8_init(void) -{ - return 0; -} - -static void __exit alsa_sb8_exit(void) -{ -} - -module_init(alsa_sb8_init) -module_exit(alsa_sb8_exit) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 787a4ade4afd..90b254aaef74 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -305,19 +305,3 @@ EXPORT_SYMBOL(snd_sbmixer_add_ctl); EXPORT_SYMBOL(snd_sbmixer_suspend); EXPORT_SYMBOL(snd_sbmixer_resume); #endif - -/* - * INIT part - */ - -static int __init alsa_sb_common_init(void) -{ - return 0; -} - -static void __exit alsa_sb_common_exit(void) -{ -} - -module_init(alsa_sb_common_init) -module_exit(alsa_sb_common_exit) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index d23cc8abe1ef..32453f81b95a 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2279,19 +2279,3 @@ const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction) &snd_wss_playback_ops : &snd_wss_capture_ops; } EXPORT_SYMBOL(snd_wss_get_pcm_ops); - -/* - * INIT part - */ - -static int __init alsa_wss_init(void) -{ - return 0; -} - -static void __exit alsa_wss_exit(void) -{ -} - -module_init(alsa_wss_init); -module_exit(alsa_wss_exit); From 3b23dc52da90c340c51273414f4e2d13e07e594c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:48:26 +0200 Subject: [PATCH 469/529] ALSA: i2c: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/i2c/cs8427.c | 12 ------------ sound/i2c/i2c.c | 13 ------------- sound/i2c/other/ak4xxx-adda.c | 12 ------------ sound/i2c/tea6330t.c | 16 ---------------- 4 files changed, 53 deletions(-) diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 7e21621e492a..2647309bc675 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -621,15 +621,3 @@ int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate) } EXPORT_SYMBOL(snd_cs8427_iec958_pcm); - -static int __init alsa_cs8427_module_init(void) -{ - return 0; -} - -static void __exit alsa_cs8427_module_exit(void) -{ -} - -module_init(alsa_cs8427_module_init) -module_exit(alsa_cs8427_module_exit) diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index ef2a9afe9e19..c4a232f18a79 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -338,16 +338,3 @@ static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) snd_i2c_bit_stop(bus); return err; } - - -static int __init alsa_i2c_init(void) -{ - return 0; -} - -static void __exit alsa_i2c_exit(void) -{ -} - -module_init(alsa_i2c_init) -module_exit(alsa_i2c_exit) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index bf377dc192aa..7f2761a2e7c8 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -911,15 +911,3 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) return 0; } EXPORT_SYMBOL(snd_akm4xxx_build_controls); - -static int __init alsa_akm4xxx_module_init(void) -{ - return 0; -} - -static void __exit alsa_akm4xxx_module_exit(void) -{ -} - -module_init(alsa_akm4xxx_module_init) -module_exit(alsa_akm4xxx_module_exit) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 2d22310dce05..239c4822427f 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -368,19 +368,3 @@ int snd_tea6330t_update_mixer(struct snd_card *card, EXPORT_SYMBOL(snd_tea6330t_detect); EXPORT_SYMBOL(snd_tea6330t_update_mixer); - -/* - * INIT part - */ - -static int __init alsa_tea6330t_init(void) -{ - return 0; -} - -static void __exit alsa_tea6330t_exit(void) -{ -} - -module_init(alsa_tea6330t_init) -module_exit(alsa_tea6330t_exit) From a7da09fecf35cc1b328f0229c019757f6582f6bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:48:41 +0200 Subject: [PATCH 470/529] ALSA: pci: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 16 ---------------- sound/pci/ice1712/ak4xxx.c | 12 ------------ 2 files changed, 28 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 1ef7cdf1d3e8..f4459d1a9d67 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2941,19 +2941,3 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, } EXPORT_SYMBOL(snd_ac97_tune_hardware); - -/* - * INIT part - */ - -static int __init alsa_ac97_init(void) -{ - return 0; -} - -static void __exit alsa_ac97_exit(void) -{ -} - -module_init(alsa_ac97_init) -module_exit(alsa_ac97_exit) diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index 179ef7a5f0d1..a553897a4c4f 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -179,18 +179,6 @@ int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice) return 0; } -static int __init alsa_ice1712_akm4xxx_module_init(void) -{ - return 0; -} - -static void __exit alsa_ice1712_akm4xxx_module_exit(void) -{ -} - -module_init(alsa_ice1712_akm4xxx_module_init) -module_exit(alsa_ice1712_akm4xxx_module_exit) - EXPORT_SYMBOL(snd_ice1712_akm4xxx_init); EXPORT_SYMBOL(snd_ice1712_akm4xxx_free); EXPORT_SYMBOL(snd_ice1712_akm4xxx_build_controls); From c000c4f1d96534885bc7a8ef5185fd8c0de5e827 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2018 15:48:54 +0200 Subject: [PATCH 471/529] ALSA: synth: Remove empty init and exit For a sake of code simplification, remove the init and the exit entries that do nothing. Notes for readers: actually it's OK to remove *both* init and exit, but not OK to remove the exit entry. By removing only the exit while keeping init, the module becomes permanently loaded; i.e. you cannot unload it any longer! Signed-off-by: Takashi Iwai --- sound/synth/emux/emux.c | 17 ----------------- sound/synth/util_mem.c | 16 ---------------- 2 files changed, 33 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index b840ff2dcfbb..64f3141a3e1b 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -163,20 +163,3 @@ int snd_emux_free(struct snd_emux *emu) } EXPORT_SYMBOL(snd_emux_free); - - -/* - * INIT part - */ - -static int __init alsa_emux_init(void) -{ - return 0; -} - -static void __exit alsa_emux_exit(void) -{ -} - -module_init(alsa_emux_init) -module_exit(alsa_emux_exit) diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c index 8e34bc4e07ec..4bd1e98200d2 100644 --- a/sound/synth/util_mem.c +++ b/sound/synth/util_mem.c @@ -193,19 +193,3 @@ EXPORT_SYMBOL(snd_util_mem_avail); EXPORT_SYMBOL(__snd_util_mem_alloc); EXPORT_SYMBOL(__snd_util_mem_free); EXPORT_SYMBOL(__snd_util_memblk_new); - -/* - * INIT part - */ - -static int __init alsa_util_mem_init(void) -{ - return 0; -} - -static void __exit alsa_util_mem_exit(void) -{ -} - -module_init(alsa_util_mem_init) -module_exit(alsa_util_mem_exit) From 8530ebf1079ccc84ffa32d970cdcae168b2f3684 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 08:42:11 -0500 Subject: [PATCH 472/529] ASoC: smd845: fix memory leak In case memory resources for *card* were allocated, release them before return. Addresses-Coverity-ID: 1472244 ("Resource leak") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index be0cb1122036..c1adb77230eb 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -222,8 +222,10 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) /* Allocate the private data */ data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } card->dev = dev; dev_set_drvdata(dev, card); @@ -248,6 +250,7 @@ register_card_fail: kfree(card->dai_link); parse_dt_fail: kfree(data); +data_alloc_fail: kfree(card); return ret; } From 3b7c88fcc2873eba64f9e8cef34c4af0cba20887 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:28:24 -0500 Subject: [PATCH 473/529] ASoC: davinci-i2s: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1364478 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 807040bb3921..a3206e65e5e5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * rate is lowered. */ inv_fs = true; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; From 85c81941d5033fc8a68646823d3157840c9de8b3 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:32:52 -0500 Subject: [PATCH 474/529] ASoC: omap-mcpdm: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1369526 ("Missing break in switch") Addresses-Coverity-ID: 1369529 ("Missing break in switch") Addresses-Coverity-ID: 1451415 ("Missing break in switch") Addresses-Coverity-ID: 115103 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e97360f9890..4c1be36c2207 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; + /* fall through */ case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; + /* fall through */ case 3: link_mask |= 1 << 2; + /* fall through */ case 2: link_mask |= 1 << 1; + /* fall through */ case 1: link_mask |= 1 << 0; break; From 1a12d5dc7dd1ffd985503f9770b736fb03db2e3f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:34:30 -0500 Subject: [PATCH 475/529] ASoC: core: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 146568 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 82eb3868be67..9cfe10d8040c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -528,6 +528,7 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + /* fall through */ case SND_SOC_BIAS_OFF: if (component->driver->suspend) From 16bbeb2b43c3f5d69e1348477e75a24ae6d55d5a Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:29:53 -0500 Subject: [PATCH 476/529] ASoC: fsl_esai: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1222121 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 8f43110373b8..c1d1d06783e5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; + /* fall through */ case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; From a773c3b6be185d171e2755ac715e8b1ea099ebbc Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:31:48 -0500 Subject: [PATCH 477/529] ASoC: omap-dmic: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1468847 ("Missing break in switch") Addresses-Coverity-ID: 1468849 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 51dd7c65096b..fe966272bd0c 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, switch (channels) { case 6: dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; + /* fall through */ case 4: dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; + /* fall through */ case 2: dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; break; From 5019027a566de4986a7f66017cf0d6d794fc155f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 11:33:57 -0500 Subject: [PATCH 478/529] ASoC: samsung: i2s: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1381093 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f914ed45db7d..d6c62aa13041 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; + /* fall through */ case 4: val |= MOD_DC1_EN; break; From 2f3b94e539a46052f8eba1f295ff5646e227578a Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 15:52:33 -0500 Subject: [PATCH 479/529] ALSA: seq: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/core/seq/seq_midi_emul.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index f9f21331aeea..c1975dd31871 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -318,7 +318,7 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* go through here */ + /* fall through */ case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); From eb2caeb88c181904110f8e74995c7b0bb7fd39f6 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 3 Aug 2018 15:53:54 -0500 Subject: [PATCH 480/529] ALSA: seq_oss: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Warning level 2 was used: -Wimplicit-fallthrough=2 Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index 4f24ea9fad93..ba127c22539a 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* continue to next */ + /* fall through and continue to next */ case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; From 56e40eb6d656194e55ce2012fee9d5a496270aaa Mon Sep 17 00:00:00 2001 From: Alexandru Gagniuc Date: Sat, 4 Aug 2018 11:44:44 -0500 Subject: [PATCH 481/529] ALSA: hda/realtek - Add mute LED quirk for HP Spectre x360 This device has the same issues as the HP x360 wrt the MUTE LED and the front speakers not working. This patch fixes the MUTE LED issue, but doesn't touch the HDA verbs. The fix for the x360 does not work on the Spectre. Signed-off-by: Alexandru Gagniuc Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2d0b2f3bd30..b20974ef1e13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6474,6 +6474,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), From ac69c2f578bf51a3804c5e96467571ea5be0e882 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 15:11:03 -0500 Subject: [PATCH 482/529] ALSA: mixart: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a74f1ad7e7b8..9cd297a42f24 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -182,6 +182,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; + /* fall through */ default: if(rate == 0) return 0; /* nothing to do */ From 9d5a289a86a15a1a4248022f9338517f7e62832b Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 15:12:09 -0500 Subject: [PATCH 483/529] ALSA: emu10k1: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 26f6eda3e766..9f2b6097f486 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -778,7 +778,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -928,7 +928,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); From ef0075280cfe09ef076ce1b85f0a2294c5ed86f3 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 15:13:26 -0500 Subject: [PATCH 484/529] ALSA: echoaudio: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115156 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 358ef7dcf410..907cf1a46712 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -713,6 +713,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; + /* fall through */ case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -764,6 +765,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; + /* fall through */ case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; From 627661ced8246c8e833f3bc3817070e934cd79ba Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 6 Aug 2018 16:14:06 +0900 Subject: [PATCH 485/529] ALSA: dice: fix wrong copy to rx parameters for Alesis iO26 A commit 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') adds wrong copy to rx parameters instead of tx parameters for Alesis iO26. This commit fixes the bug for v4.18-rc8. Fixes: 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') Signed-off-by: Takashi Sakamoto Cc: # v4.18 Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-alesis.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index b2efb1c71a98..218292bdace6 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -37,7 +37,7 @@ int snd_dice_detect_alesis_formats(struct snd_dice *dice) MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } else { - memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs, + memcpy(dice->tx_pcm_chs, alesis_io26_tx_pcm_chs, MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } From 038541dae968dbef99edca4b22bbb2a7b4afdc83 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:49:02 -0500 Subject: [PATCH 486/529] ASoC: max9850: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 74d7f52c7e73..6e6134589588 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case MAX9850_STATUSA: case MAX9850_STATUSB: - return 1; + return true; default: - return 0; + return false; } } From 508e8641f89cc89554adbec3ce59bcd28a4ced4b Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:49:55 -0500 Subject: [PATCH 487/529] ASoC: rt5631: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index e52e4670cf65..865f49ac38dd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg) case RT5631_INDEX_ADD: case RT5631_INDEX_DATA: case RT5631_EQ_CTRL: - return 1; + return true; default: - return 0; + return false; } } @@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg) case RT5631_VENDOR_ID: case RT5631_VENDOR_ID1: case RT5631_VENDOR_ID2: - return 1; + return true; default: - return 0; + return false; } } From 10754bfc051256ab922b818de3ddd11391862a5f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:50:27 -0500 Subject: [PATCH 488/529] ASoC: tda7419: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/tda7419.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, From 064ee5a370150534e12e5bac4ff81f86528e27ca Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:51:01 -0500 Subject: [PATCH 489/529] ASoC: wm8990: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 411b9eee88c2..457bc437ce54 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: - return 1; + return true; default: - return 0; + return false; } } From c34c4515286f27e7e99d8a0011b4322d1de6c9bc Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:52:01 -0500 Subject: [PATCH 490/529] ASoC: cs4270: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2a7a4168c072..3c266eeb89bf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) - return 1; + return true; return reg == CS4270_CHIPID; } From eb086306bc6b42dc2c538ec2350d44a82d1a835b Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:52:35 -0500 Subject: [PATCH 491/529] ASoC: wm8996: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 78a408236cfb..91711f8958c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg) case WM8996_RIGHT_PDM_SPEAKER: case WM8996_PDM_SPEAKER_MUTE_SEQUENCE: case WM8996_PDM_SPEAKER_VOLUME: - return 1; + return true; default: - return 0; + return false; } } @@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) case WM8996_MIC_DETECT_3: case WM8996_HEADPHONE_DETECT_1: case WM8996_HEADPHONE_DETECT_2: - return 1; + return true; default: - return 0; + return false; } } From 965afd3c1dbaf293bfa9452aaf0a7e53d78d07f3 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:53:10 -0500 Subject: [PATCH 492/529] ASoC: da7219: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d..c0144f2f8174 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2143,9 +2143,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg) case DA7219_ACCDET_IRQ_EVENT_B: case DA7219_ACCDET_CONFIG_8: case DA7219_SYSTEM_STATUS: - return 1; + return true; default: - return 0; + return false; } } From bc94c8884e5a94914ec950dfc615563a1253e3f3 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:53:38 -0500 Subject: [PATCH 493/529] ASoC: twl6040: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bfd1abd72253..94675da514c8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component, case TWL6040_REG_HFRCTL: return priv->dl2_unmuted; default: - return 1; + return true; } } From 1752a35acd8e837463fb7b7d9492b18e2a2ed5e3 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:54:10 -0500 Subject: [PATCH 494/529] ASoC: da7213: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 54cb5f24969f..92d006a5283e 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg) case DA7213_ALC_OFFSET_AUTO_M_R: case DA7213_ALC_OFFSET_AUTO_U_R: case DA7213_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } From 380ae4ec420304aa59adb36776eaf78aef37192c Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:54:54 -0500 Subject: [PATCH 495/529] ASoC: wm5100-tables: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } From e1ec62b147c2f2deae66e69ee4f7347dc80123ae Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:55:28 -0500 Subject: [PATCH 496/529] ASoC: da9055: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da9055.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index afdf90c78884..f6a7bf9560e7 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev, case DA9055_HP_R_GAIN_STATUS: case DA9055_LINE_GAIN_STATUS: case DA9055_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } From bee7d3c9f89a1bbe3d5f88ac9d6946de4705731b Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Sat, 4 Aug 2018 16:56:02 -0500 Subject: [PATCH 497/529] ASoC: wm8903: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7b8b6ef2f632..6cb3c153ba19 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg) case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: case WM8903_DC_SERVO_READBACK_4: - return 1; + return true; default: - return 0; + return false; } } From 8e3684f66e15c354dba5544c2af8ce973ed40cf6 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:05 +0100 Subject: [PATCH 498/529] ASoC: qcom: make common.c as proper module This patch converts common helper functions in to proper module and also fixes below warning. WARNING: sound/soc/qcom/snd-soc-sdm845: 'qcom_snd_parse_of' exported twice. Previous export was in sound/soc/qcom/snd-soc-apq8096.ko Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 5 +++++ sound/soc/qcom/Makefile | 6 ++++-- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5f03312ef007..2a4c912d1e48 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. +config SND_SOC_QCOM_COMMON + tristate + config SND_SOC_QDSP6_COMMON tristate @@ -86,6 +89,7 @@ config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. @@ -95,6 +99,7 @@ config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index f0e94d48ba98..41b2c7a23a4d 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,13 +13,15 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o common.o -snd-soc-sdm845-objs := sdm845.o common.o +snd-soc-apq8096-objs := apq8096.o +snd-soc-sdm845-objs := sdm845.o +snd-soc-qcom-common-objs := common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o +obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ From e9d244b14dd5cf9a78ab256a4463d946e580ca63 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:06 +0100 Subject: [PATCH 499/529] ASoC: apq8096: remove unused header files This patch removes unused header files from the driver. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 6ee7e66cbaaa..1543e85629f8 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -1,9 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2018, Linaro Limited -#include #include -#include #include #include #include From 846b2c96808cc7cdf4e0619d00604b3edd15b35a Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:07 +0100 Subject: [PATCH 500/529] ASoC: sdm845: remove unused header files This patch removes unused header files from the driver. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index c1adb77230eb..2a781d87ee65 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -5,11 +5,9 @@ #include #include -#include #include #include #include -#include #include "common.h" #include "qdsp6/q6afe.h" From d72117d0c89a3f5657ef91d5eef31337964e3cb2 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:08 +0100 Subject: [PATCH 501/529] ASoC: qcom: remove unused header files from common.h This patch removes unused header files from common.h. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/common.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h index ad5d2cf27459..f05c05b12bd7 100644 --- a/sound/soc/qcom/common.h +++ b/sound/soc/qcom/common.h @@ -4,7 +4,6 @@ #ifndef __QCOM_SND_COMMON_H__ #define __QCOM_SND_COMMON_H__ -#include #include int qcom_snd_parse_of(struct snd_soc_card *card); From 0961503412e3e13d82f885f9fb3af527edd58e47 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:09 +0100 Subject: [PATCH 502/529] ASoC: qdsp6: q6afe-dai: add SLIM tx AIF_IN dapm Add missing AIF_IN dapm for slim tx ports. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index e988692a3ced..139d24be3fb9 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1117,6 +1117,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", From ad0eaee6195db1db1749dd46b9e6f4466793d178 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Mon, 6 Aug 2018 07:14:51 -0500 Subject: [PATCH 503/529] ASoC: wm8994: Fix missing break in switch Add missing break statement in order to prevent the code from falling through to the default case. Addresses-Coverity-ID: 115050 ("Missing break in switch") Reported-by: Valdis Kletnieks Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 62f8c5b9ba92..14f1b0c0d286 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,7 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } - /* fall through */ + break; default: return -EINVAL; From bbdb7012b0736cda0b9b00a2949e9207cf2f892f Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 6 Aug 2018 12:57:14 +0530 Subject: [PATCH 504/529] ASoC: AMD: Make ACP->SYSMEM DMA non circular In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69008df..816abd65a6ed 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: From 662fb3efe7ee835f0eeba6bc63b81e82a97fc312 Mon Sep 17 00:00:00 2001 From: "Mukunda, Vijendar" Date: Mon, 6 Aug 2018 12:57:15 +0530 Subject: [PATCH 505/529] ASoC: AMD: Modified DMA pointer for capture Give position on ACP->SYSMEM DMA channel for the number of bytes that have been transferred on the basis of current descriptor under service. Signed-off-by: Vijendar Mukunda Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 31 ++++++++++++++++++------------- sound/soc/amd/acp.h | 1 + 2 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 816abd65a6ed..32f27c5e4d93 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -922,10 +922,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -945,10 +942,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } } @@ -1002,6 +996,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 buffersize; u32 pos = 0; u64 bytescount = 0; + u16 dscr; + u32 period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1009,11 +1005,20 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + period_bytes = frames_to_bytes(runtime, runtime->period_size); + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } else { + buffersize = frames_to_bytes(runtime, runtime->buffer_size); + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + pos = do_div(bytescount, buffersize); + } return bytes_to_frames(runtime, pos); } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0a2240bff62e..be3963e8f4fa 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -138,6 +138,7 @@ struct audio_substream_data { u32 sram_bank; u32 byte_cnt_high_reg_offset; u32 byte_cnt_low_reg_offset; + u32 dma_curr_dscr; uint64_t size; u64 bytescount; void __iomem *acp_mmio; From c21c834adb5bc81e7081aa93ac50619c6d060506 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 6 Aug 2018 12:57:16 +0530 Subject: [PATCH 506/529] ASoC: AMD: Set delay value for the capture case ACP->SYSMEM DMA happens at every I2S->SYSMEM period completion. Thus, there is delay of x frames till I2S->SYSMEM reaches a period length. This delay is communicated to user space. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 32f27c5e4d93..e359938e3d7e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -922,6 +922,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; @@ -942,6 +946,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = + mmACP_I2S_RECEIVED_BYTE_CNT_LOW; rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } @@ -997,7 +1005,7 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 pos = 0; u64 bytescount = 0; u16 dscr; - u32 period_bytes; + u32 period_bytes, delay; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1012,6 +1020,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) pos = period_bytes; else pos = 0; + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); From d2f884612c42850db0b3521b74a05636a5fc035f Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:42 +0100 Subject: [PATCH 507/529] ALSA: intel_hdmi: remove redundant variable cfg_val Variable cfg_val is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'cfg_val' set but not used [-Wunused-but-set-variable] [ Background info about val_bit field from alsa-devel ML thread: tiwai: Actually this made me wonder what is the definition of val_bit. It seems always 1 in the current code after the commit 964ca8083c02. Pierre? pbossart: This val_bit is only there for debug/test, it should be set to one by default and has nothing to do with the lpcm_id. This variable was set even in patches before upstream submission and was never needed, I guess it must be a 9-yr old issue. Good catch! ] Signed-off-by: Colin Ian King Acked-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/x86/intel_hdmi_audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index edc9f5a34eff..fa7dca5a68c8 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -290,7 +290,6 @@ static void had_reset_audio(struct snd_intelhad *intelhaddata) static int had_prog_status_reg(struct snd_pcm_substream *substream, struct snd_intelhad *intelhaddata) { - union aud_cfg cfg_val = {.regval = 0}; union aud_ch_status_0 ch_stat0 = {.regval = 0}; union aud_ch_status_1 ch_stat1 = {.regval = 0}; @@ -298,7 +297,6 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, IEC958_AES0_NONAUDIO) >> 1; ch_stat0.regx.clk_acc = (intelhaddata->aes_bits & IEC958_AES3_CON_CLOCK) >> 4; - cfg_val.regx.val_bit = ch_stat0.regx.lpcm_id; switch (substream->runtime->rate) { case AUD_SAMPLE_RATE_32: From 0b0722e191757ca6851382d78c15277ca9c5c211 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 3 Aug 2018 13:30:03 +0100 Subject: [PATCH 508/529] ASoC: compress: make BE and FE order inline with dpcm For some reason order of startup/hw_params/prepare are reversed in dynamic compress usecase when compared to dpcm usecase. This is a issue with platforms like QCOM where it expects the BE to be initialized before FE. Interestingly the compress trigger callback order is inline with dpcm. Am not 100% sure why the compress audio case has been reversed. This patch is making the order inline with dpcm. If the reverse ordering is just co-incendental then this change makes sense and will avoid inventing some new mechanism to cope with ordering. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 96 +++++++++++++++++++--------------------- 1 file changed, 46 insertions(+), 50 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b9e1673fea51..409d082e80d1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -140,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) stream = SNDRV_PCM_STREAM_CAPTURE; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto be_err; + else if (ret == 0) + dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + fe->dpcm[stream].runtime = fe_substream->runtime; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto out; + } if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -153,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = soc_compr_components_open(cstream, &component); if (ret < 0) - goto machine_err; + goto open_err; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { ret = fe->dai_link->compr_ops->startup(cstream); @@ -164,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) } } - fe->dpcm[stream].runtime = fe_substream->runtime; - - ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) - goto fe_err; - else if (ret == 0) - dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", - fe->dai_link->name, stream ? "capture" : "playback"); - - /* calculate valid and active FE <-> BE dpcms */ - dpcm_process_paths(fe, stream, &list, 1); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { - /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - goto path_err; - } - dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); @@ -201,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; -path_err: - dpcm_path_put(&list); -fe_err: - if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) - fe->dai_link->compr_ops->shutdown(cstream); machine_err: soc_compr_components_free(cstream, component); - +open_err: if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: + dpcm_path_put(&list); +be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); return ret; @@ -551,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) { ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai); if (ret < 0) @@ -577,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, goto out; } - /* - * Create an empty hw_params for the BE as the machine driver must - * fix this up to match DSP decoder and ASRC configuration. - * I.e. machine driver fixup for compressed BE is mandatory. - */ - memset(&fe->dpcm[fe_substream->stream].hw_params, 0, - sizeof(struct snd_pcm_hw_params)); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) - goto out; - - ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) - goto out; - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; From f861e3e28a3016a2064d9f600eaa92a530b732b4 Mon Sep 17 00:00:00 2001 From: Matthias Kaehlcke Date: Tue, 7 Aug 2018 10:19:40 -0700 Subject: [PATCH 509/529] ASoC: rt5677: Fix initialization of rt5677_of_match.data The driver expects to find the device id in rt5677_of_match.data, however it is currently assigned to rt5677_of_match.type. Fix this. The problem was found with the help of clang: sound/soc/codecs/rt5677.c:5010:36: warning: expression which evaluates to zero treated as a null pointer constant of type 'const void *' [-Wnon-literal-null-conversion] { .compatible = "realtek,rt5677", RT5677 }, ^~~~~~ Fixes: ddc9e69b9dc2 ("ASoC: rt5677: Hide platform data in the module sources") Signed-off-by: Matthias Kaehlcke Reviewed-by: Guenter Roeck Acked-by: Andy Shevchenko Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 922becfee59b..9b7a1833d331 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5008,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", .data = (const void *)RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); From b1470d4ce77c2d60661c7d5325d4fb8063e15ff8 Mon Sep 17 00:00:00 2001 From: Ajit Pandey Date: Tue, 7 Aug 2018 18:30:42 +0100 Subject: [PATCH 510/529] ASoC: wm_adsp: Correct DSP pointer for preloader control The offset of the DSP core needs to be taken into account for the DSP preloader control get and put. Currently the dsp->preloaded variable will only ever be read/updated on the first DSP, whilst this doesn't affect the operation of the control the readback will be incorrect. Signed-off-by: Ajit Pandey Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4e7f8525449e..08dc82770273 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2643,7 +2643,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; ucontrol->value.integer.value[0] = dsp->preloaded; @@ -2655,10 +2658,11 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); From 0717edbdfed61b4c1e8291140f78882d3a481042 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Tue, 7 Aug 2018 20:06:38 -0700 Subject: [PATCH 511/529] ASoC: max98373: Added software reset register to readable registers Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a92586106932..92b7125ea169 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, static bool max98373_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case MAX98373_R2000_SW_RESET: case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3: case MAX98373_R2010_IRQ_CTRL: case MAX98373_R2014_THERM_WARN_THRESH From 0a047f07525fecfa8f6fccc5d30afff7e816de8d Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 Aug 2018 17:13:38 +0100 Subject: [PATCH 512/529] ASoC: wm_adsp: Declare firmware controls from codec driver To allow for more flexibility in naming of DSP-type cores move the creation of the firmware controls to the codec drivers instead of having a hardcoded list in wm_adsp. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm2200.c | 10 ++++------ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 5 +++++ sound/soc/codecs/wm_adsp.c | 35 +++++++++-------------------------- sound/soc/codecs/wm_adsp.h | 10 +++++++++- 6 files changed, 32 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0da52ead91e0..45e50fe3bf25 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 3663b9fd4d65..deff65161504 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { static int wm2200_probe(struct snd_soc_component *component) { struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component); - int ret; wm2200->component = component; - ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2); - if (ret != 0) - return ret; - - return ret; + return 0; } static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a01a0c0e01eb..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 00c735c585d9..b0789a03d699 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), +WM_ADSP_FW_CONTROL("DSP4", 3), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e39b0e0b04df..fbd0515c49d7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,8 +684,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) } #endif -static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -695,9 +695,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, return 0; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_get); -static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -721,8 +722,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_put); -static const struct soc_enum wm_adsp_fw_enum[] = { +const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), @@ -731,24 +733,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; - -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6], - wm_adsp_fw_get, wm_adsp_fw_put), -}; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2884,9 +2869,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp dsp->component = component; - return snd_soc_add_component_controls(component, - &wm_adsp_fw_controls[dsp->num - 1], - 1); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_component_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index bc6d359f0533..8d58cb9d9bb9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -121,7 +121,11 @@ struct wm_adsp { .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +#define WM_ADSP_FW_CONTROL(dspname, num) \ + SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \ + wm_adsp_fw_get, wm_adsp_fw_put) + +extern const struct soc_enum wm_adsp_fw_enum[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); @@ -144,6 +148,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); From 605391d0f4bfdff2f2c6c5477ce0ccf776d8d5c0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 Aug 2018 17:13:39 +0100 Subject: [PATCH 513/529] ASoC: wm_adsp: Make DSP name configurable by codec driver Instead of harcoding that a core must always be called "DSPn" add a name member to struct wm_adsp so that the owning codec driver can provide a custom name. This allows for re-use of the wm_adsp driver with parts where the processing cores are named differently. If no name is provided the default DSPn name is used. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 69 ++++++++++++++++++++++++++------------ sound/soc/codecs/wm_adsp.h | 2 ++ 2 files changed, 50 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1c12c78dbcce..f61656070225 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include @@ -35,15 +36,15 @@ #include "wm_adsp.h" #define adsp_crit(_dsp, fmt, ...) \ - dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_err(_dsp, fmt, ...) \ - dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_warn(_dsp, fmt, ...) \ - dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_info(_dsp, fmt, ...) \ - dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_dbg(_dsp, fmt, ...) \ - dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define ADSP1_CONTROL_1 0x00 #define ADSP1_CONTROL_2 0x02 @@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, struct snd_soc_component *component) { struct dentry *root = NULL; - char *root_name; int i; if (!component->debugfs_root) { @@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, goto err; } - root_name = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!root_name) - goto err; - - snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num); - root = debugfs_create_dir(root_name, component->debugfs_root); - kfree(root_name); + root = debugfs_create_dir(dsp->name, component->debugfs_root); if (!root) goto err; @@ -1315,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, switch (dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x", - dsp->num, region_name, alg_region->alg); + snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", + dsp->name, region_name, alg_region->alg); break; default: ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "DSP%d%c %.12s %x", dsp->num, *region_name, + "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); /* Truncate the subname from the start if it is too long */ @@ -1648,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -2226,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -2398,8 +2392,38 @@ out: return ret; } +static int wm_adsp_create_name(struct wm_adsp *dsp) +{ + char *p; + + if (!dsp->name) { + dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d", + dsp->num); + if (!dsp->name) + return -ENOMEM; + } + + if (!dsp->fwf_name) { + p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL); + if (!p) + return -ENOMEM; + + dsp->fwf_name = p; + for (; *p != 0; ++p) + *p = tolower(*p); + } + + return 0; +} + int wm_adsp1_init(struct wm_adsp *dsp) { + int ret; + + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + INIT_LIST_HEAD(&dsp->alg_regions); mutex_init(&dsp->pwr_lock); @@ -2672,7 +2696,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); dsp->preloaded = ucontrol->value.integer.value[0]; @@ -2867,8 +2891,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp { char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); - + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); snd_soc_component_disable_pin(component, preload); wm_adsp2_init_debugfs(dsp, component); @@ -2891,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp) { int ret; + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + switch (dsp->rev) { case 0: /* diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 8d58cb9d9bb9..4b8778b0b06c 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -57,6 +57,8 @@ struct wm_adsp_compr_buf; struct wm_adsp { const char *part; + const char *name; + const char *fwf_name; int rev; int num; int type; From a62e4739473a29646af4e37a5da289795cde6dc0 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:12 -0400 Subject: [PATCH 514/529] ALSA: hda/ca0132 - Create mmio gpio function to make code clearer This patch adds a new function, ca0132_mmio_gpio_set, to clear up what is going on with writes to mmio region 0x320. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 62 ++++++++++++++++++++++-------------- 1 file changed, 38 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 27d3388cd2a2..665142ef1186 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3072,6 +3072,24 @@ static bool dspload_wait_loaded(struct hda_codec *codec) * Setup GPIO for the other variants of Core3D. */ +/* + * For cards with PCI-E region2 (Sound Blaster Z/ZxR, Recon3D, and AE-5) + * the mmio address 0x320 is used to set GPIO pins. The format for the data + * The first eight bits are just the number of the pin. So far, I've only seen + * this number go to 7. + */ +static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin, + bool enable) +{ + struct ca0132_spec *spec = codec->spec; + unsigned short gpio_data; + + gpio_data = gpio_pin & 0xF; + gpio_data |= ((enable << 8) & 0x100); + + writew(gpio_data, spec->mem_base + 0x320); +} + /* * Sets up the GPIO pins so that they are discoverable. If this isn't done, * the card shows as having no GPIO pins. @@ -3947,9 +3965,9 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /*speaker out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0007, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0101, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, true); chipio_set_control_param(codec, 0x0D, 0x18); break; case QUIRK_R3DI: @@ -3983,9 +4001,9 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /* Headphone out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0107, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0001, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, true); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, false); chipio_set_control_param(codec, 0x0D, 0x12); break; case QUIRK_R3DI: @@ -4025,9 +4043,9 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /* Surround out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0007, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0101, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, true); chipio_set_control_param(codec, 0x0D, 0x18); break; case QUIRK_R3DI: @@ -4291,7 +4309,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case REAR_MIC: switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0000, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 0, false); tmp = FLOAT_THREE; break; case QUIRK_R3DI: @@ -4323,7 +4341,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) ca0132_mic_boost_set(codec, 0); switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0000, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 0, false); break; case QUIRK_R3DI: r3di_gpio_mic_set(codec, R3DI_REAR_MIC); @@ -4349,8 +4367,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case FRONT_MIC: switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0100, spec->mem_base + 0x320); - writew(0x0005, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 0, true); + ca0132_mmio_gpio_set(codec, 5, false); tmp = FLOAT_THREE; break; case QUIRK_R3DI: @@ -6890,16 +6908,12 @@ static void sbz_region2_exit(struct hda_codec *codec) writeb(0x0, spec->mem_base + 0x100); for (i = 0; i < 8; i++) writeb(0xb3, spec->mem_base + 0x304); - /* - * I believe these are GPIO, with the right most hex digit being the - * gpio pin, and the second digit being on or off. We see this more in - * the input/output select functions. - */ - writew(0x0000, spec->mem_base + 0x320); - writew(0x0001, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0005, spec->mem_base + 0x320); - writew(0x0007, spec->mem_base + 0x320); + + ca0132_mmio_gpio_set(codec, 0, false); + ca0132_mmio_gpio_set(codec, 1, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 5, false); + ca0132_mmio_gpio_set(codec, 7, false); } static void sbz_set_pin_ctl_default(struct hda_codec *codec) @@ -7236,7 +7250,7 @@ static int ca0132_init(struct hda_codec *codec) ca0132_refresh_widget_caps(codec); if (spec->quirk == QUIRK_SBZ) - writew(0x0107, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, true); switch (spec->quirk) { case QUIRK_R3DI: From d97420d2b0379e498adc3fae5db8fa70945b5d56 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:13 -0400 Subject: [PATCH 515/529] ALSA: hda/ca0132 - Clean up ca0132_init function. This patch cleans up ca0132_init by removing unnecessary commands and ordering things better. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 665142ef1186..8b98d18d97ac 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7249,14 +7249,12 @@ static int ca0132_init(struct hda_codec *codec) ca0132_refresh_widget_caps(codec); - if (spec->quirk == QUIRK_SBZ) - ca0132_mmio_gpio_set(codec, 7, true); - switch (spec->quirk) { case QUIRK_R3DI: r3di_setup_defaults(codec); break; case QUIRK_SBZ: + sbz_setup_defaults(codec); break; default: ca0132_setup_defaults(codec); @@ -7287,20 +7285,12 @@ static int ca0132_init(struct hda_codec *codec) ca0132_gpio_setup(codec); snd_hda_sequence_write(codec, spec->spec_init_verbs); - switch (spec->quirk) { - case QUIRK_SBZ: - sbz_setup_defaults(codec); + if (spec->use_alt_functions) { ca0132_alt_select_out(codec); ca0132_alt_select_in(codec); - break; - case QUIRK_R3DI: - ca0132_alt_select_out(codec); - ca0132_alt_select_in(codec); - break; - default: + } else { ca0132_select_out(codec); ca0132_select_mic(codec); - break; } snd_hda_jack_report_sync(codec); From a1b7f016a1ae5e51f0e11a70cf1a5875d3ccee73 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:14 -0400 Subject: [PATCH 516/529] ALSA: hda/ca0132 - Add alt_functions unsolicited response This patch fixes a previous oversight where the microphone unsolicited response would use the wrong input selection function. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8b98d18d97ac..3e43d5686207 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6744,7 +6744,12 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { - ca0132_select_mic(codec); + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) + ca0132_alt_select_in(codec); + else + ca0132_select_mic(codec); } static void ca0132_init_unsol(struct hda_codec *codec) From 8f8c523c4604afe231196920bf08310141a4f0ba Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:15 -0400 Subject: [PATCH 517/529] ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D This patch adds the PCI subsys ID for the Recon3D that has been tested, and adds the QUIRK_R3D enumeration. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 3e43d5686207..8a4be5fcba55 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -994,6 +994,7 @@ enum { QUIRK_ALIENWARE_M17XR4, QUIRK_SBZ, QUIRK_R3DI, + QUIRK_R3D, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -1050,6 +1051,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), + SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), {} }; From 7f73df95401f7a2392ccf1880ba1e54cfed62779 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:16 -0400 Subject: [PATCH 518/529] ALSA: hda/ca0132 - Add Recon3D pincfg This patch adds pin configs from the Recon3D, taken from the Window's driver. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8a4be5fcba55..4d2b79e19516 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1026,6 +1026,21 @@ static const struct hda_pintbl sbz_pincfgs[] = { {} }; +/* Recon3D pin configs taken from Windows Driver */ +static const struct hda_pintbl r3d_pincfgs[] = { + { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c520f0 }, /* SPDIF In */ + { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + /* Recon3D integrated pin configs taken from Windows Driver */ static const struct hda_pintbl r3di_pincfgs[] = { { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ @@ -7396,8 +7411,15 @@ static void ca0132_config(struct hda_codec *codec) spec->unsol_tag_amic1 = 0x11; break; case QUIRK_SBZ: - codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); - snd_hda_apply_pincfgs(codec, sbz_pincfgs); + case QUIRK_R3D: + if (spec->quirk == QUIRK_SBZ) { + codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + } + if (spec->quirk == QUIRK_R3D) { + codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3d_pincfgs); + } spec->num_outputs = 2; spec->out_pins[0] = 0x0B; /* Line out */ From 08eca6b1f1468a4021bac7b3929fd3eb491e2629 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:17 -0400 Subject: [PATCH 519/529] ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio This patch adds the ability to choose whether or not to map the pci region2, which is used for things such as GPIO on the Recon3D and Sound Blaster Z. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 32 +++++++++++++++++++------------- 1 file changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4d2b79e19516..989770797a00 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -965,9 +965,11 @@ struct ca0132_spec { long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif /* - * Sound Blaster Z PCI region 2 iomem, used for input and output - * switching, and other unknown commands. + * The Recon3D, Sound Blaster Z, Sound Blaster ZxR, and Sound Blaster + * AE-5 all use PCI region 2 to toggle GPIO and other currently unknown + * things. */ + bool use_pci_mmio; void __iomem *mem_base; /* @@ -7562,16 +7564,6 @@ static int patch_ca0132(struct hda_codec *codec) else spec->quirk = QUIRK_NONE; - /* Setup BAR Region 2 for Sound Blaster Z */ - if (spec->quirk == QUIRK_SBZ) { - spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); - if (spec->mem_base == NULL) { - codec_warn(codec, "pci_iomap failed!"); - codec_info(codec, "perhaps this is not an SBZ?"); - spec->quirk = QUIRK_NONE; - } - } - spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; @@ -7590,19 +7582,33 @@ static int patch_ca0132(struct hda_codec *codec) break; } - /* Setup whether or not to use alt functions/controls */ + /* Setup whether or not to use alt functions/controls/pci_mmio */ switch (spec->quirk) { case QUIRK_SBZ: + spec->use_alt_controls = true; + spec->use_alt_functions = true; + spec->use_pci_mmio = true; + break; case QUIRK_R3DI: spec->use_alt_controls = true; spec->use_alt_functions = true; + spec->use_pci_mmio = false; break; default: spec->use_alt_controls = false; spec->use_alt_functions = false; + spec->use_pci_mmio = false; break; } + if (spec->use_pci_mmio) { + spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); + if (spec->mem_base == NULL) { + codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); + spec->quirk = QUIRK_NONE; + } + } + spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; From e42c7c7313e41f121d252711e35deae7964c95ad Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:18 -0400 Subject: [PATCH 520/529] ALSA: hda/ca0132 - Add Recon3D startup functions and setup This patch adds functions for Recon3D startup, and sets values for things such as use_pci_mmio. It also renames some functions and tables from the sbz prefix into ca0132, as the Recon3D uses them as well. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 49 ++++++++++++++++++++++++++---------- 1 file changed, 36 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 989770797a00..c938298cb103 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -897,7 +897,7 @@ struct ca0132_spec { const struct hda_verb *base_init_verbs; const struct hda_verb *base_exit_verbs; const struct hda_verb *chip_init_verbs; - const struct hda_verb *sbz_init_verbs; + const struct hda_verb *desktop_init_verbs; struct hda_verb *spec_init_verbs; struct auto_pin_cfg autocfg; @@ -6839,8 +6839,8 @@ static struct hda_verb ca0132_init_verbs0[] = { {} }; -/* Extra init verbs for SBZ */ -static struct hda_verb sbz_init_verbs[] = { +/* Extra init verbs for desktop cards. */ +static struct hda_verb ca0132_init_verbs1[] = { {0x15, 0x70D, 0x20}, {0x15, 0x70E, 0x19}, {0x15, 0x707, 0x00}, @@ -7135,9 +7135,27 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); } -/* - * Extra commands that don't really fit anywhere else. - */ +static void r3d_pre_dsp_setup(struct hda_codec *codec) +{ + + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); + + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); +} + static void r3di_pre_dsp_setup(struct hda_codec *codec) { chipio_write(codec, 0x18b0a4, 0x000000c2); @@ -7162,13 +7180,12 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); } - /* * These are sent before the DSP is downloaded. Not sure * what they do, or if they're necessary. Could possibly * be removed. Figure they're better to leave in. */ -static void sbz_region2_startup(struct hda_codec *codec) +static void ca0132_mmio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; @@ -7208,7 +7225,7 @@ static void ca0132_alt_init(struct hda_codec *codec) ca0132_gpio_init(codec); sbz_pre_dsp_setup(codec); snd_hda_sequence_write(codec, spec->chip_init_verbs); - snd_hda_sequence_write(codec, spec->sbz_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); break; case QUIRK_R3DI: codec_dbg(codec, "R3DI alt_init"); @@ -7219,6 +7236,11 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4); break; + case QUIRK_R3D: + r3d_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); + break; } } @@ -7255,8 +7277,8 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - if (spec->quirk == QUIRK_SBZ) - sbz_region2_startup(codec); + if (spec->use_pci_mmio) + ca0132_mmio_init(codec); snd_hda_power_up_pm(codec); @@ -7507,8 +7529,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; spec->chip_init_verbs = ca0132_init_verbs0; - if (spec->quirk == QUIRK_SBZ) - spec->sbz_init_verbs = sbz_init_verbs; + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) + spec->desktop_init_verbs = ca0132_init_verbs1; spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, sizeof(struct hda_verb), GFP_KERNEL); @@ -7585,6 +7607,7 @@ static int patch_ca0132(struct hda_codec *codec) /* Setup whether or not to use alt functions/controls/pci_mmio */ switch (spec->quirk) { case QUIRK_SBZ: + case QUIRK_R3D: spec->use_alt_controls = true; spec->use_alt_functions = true; spec->use_pci_mmio = true; From c986f50ca974397f8726bf6776ad8938d6808848 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:19 -0400 Subject: [PATCH 521/529] ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D The Recon3D can use many of the same functions as the Recon3Di, so many of the r3di prefix function remain the same, but change their names to the more generic r3d prefix. This patch does this, and adds quirk checks for things specific to the Recon3Di. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c938298cb103..8edd7675fb77 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6223,10 +6223,10 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* - * Recon3Di r3di_setup_defaults sub functions. + * Recon3D r3d_setup_defaults sub functions. */ -static void r3di_dsp_scp_startup(struct hda_codec *codec) +static void r3d_dsp_scp_startup(struct hda_codec *codec) { unsigned int tmp; @@ -6247,7 +6247,7 @@ static void r3di_dsp_scp_startup(struct hda_codec *codec) } -static void r3di_dsp_initial_mic_setup(struct hda_codec *codec) +static void r3d_dsp_initial_mic_setup(struct hda_codec *codec) { unsigned int tmp; @@ -6457,10 +6457,10 @@ static void ca0132_setup_defaults(struct hda_codec *codec) } /* - * Setup default parameters for Recon3Di DSP. + * Setup default parameters for Recon3D/Recon3Di DSP. */ -static void r3di_setup_defaults(struct hda_codec *codec) +static void r3d_setup_defaults(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; unsigned int tmp; @@ -6470,9 +6470,9 @@ static void r3di_setup_defaults(struct hda_codec *codec) if (spec->dsp_state != DSP_DOWNLOADED) return; - r3di_dsp_scp_startup(codec); + r3d_dsp_scp_startup(codec); - r3di_dsp_initial_mic_setup(codec); + r3d_dsp_initial_mic_setup(codec); /*remove DSP headroom*/ tmp = FLOAT_ZERO; @@ -6486,7 +6486,8 @@ static void r3di_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + if (spec->quirk == QUIRK_R3DI) + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); /* Setup effect defaults */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; @@ -6498,7 +6499,6 @@ static void r3di_setup_defaults(struct hda_codec *codec) ca0132_effects[idx].def_vals[i]); } } - } /* @@ -7297,7 +7297,8 @@ static int ca0132_init(struct hda_codec *codec) switch (spec->quirk) { case QUIRK_R3DI: - r3di_setup_defaults(codec); + case QUIRK_R3D: + r3d_setup_defaults(codec); break; case QUIRK_SBZ: sbz_setup_defaults(codec); From 42aa3a169062c48e5cbb1f3a6523f8b7c892b699 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:20 -0400 Subject: [PATCH 522/529] ALSA: hda/ca0132 - Add Recon3D input and output select commands This patch adds commands to the alternative input and output select commands to support the Recon3D. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8edd7675fb77..d46695e133c0 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3993,6 +3993,10 @@ static int ca0132_alt_select_out(struct hda_codec *codec) chipio_set_control_param(codec, 0x0D, 0x24); r3di_gpio_out_set(codec, R3DI_LINE_OUT); break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0D, 0x24); + ca0132_mmio_gpio_set(codec, 1, true); + break; } /* disable headphone node */ @@ -4029,6 +4033,10 @@ static int ca0132_alt_select_out(struct hda_codec *codec) chipio_set_control_param(codec, 0x0D, 0x21); r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0D, 0x21); + ca0132_mmio_gpio_set(codec, 0x1, false); + break; } snd_hda_codec_write(codec, spec->out_pins[0], 0, @@ -4071,6 +4079,10 @@ static int ca0132_alt_select_out(struct hda_codec *codec) chipio_set_control_param(codec, 0x0D, 0x24); r3di_gpio_out_set(codec, R3DI_LINE_OUT); break; + case QUIRK_R3D: + ca0132_mmio_gpio_set(codec, 1, true); + chipio_set_control_param(codec, 0x0D, 0x24); + break; } /* enable line out node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, @@ -4328,6 +4340,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case REAR_MIC: switch (spec->quirk) { case QUIRK_SBZ: + case QUIRK_R3D: ca0132_mmio_gpio_set(codec, 0, false); tmp = FLOAT_THREE; break; @@ -4360,6 +4373,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) ca0132_mic_boost_set(codec, 0); switch (spec->quirk) { case QUIRK_SBZ: + case QUIRK_R3D: ca0132_mmio_gpio_set(codec, 0, false); break; case QUIRK_R3DI: @@ -4386,6 +4400,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case FRONT_MIC: switch (spec->quirk) { case QUIRK_SBZ: + case QUIRK_R3D: ca0132_mmio_gpio_set(codec, 0, true); ca0132_mmio_gpio_set(codec, 5, false); tmp = FLOAT_THREE; From e25e3445049c353223752fd1bacead9d413b0a5a Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:21 -0400 Subject: [PATCH 523/529] ALSA: hda/ca0132 - Change mixer controls for Recon3D This patch adds changes to setup the Recon3D's mixer controls. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d46695e133c0..601efaa5c610 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5780,11 +5780,11 @@ static const struct snd_kcontrol_new ca0132_mixer[] = { }; /* - * SBZ specific control mixer. Removes auto-detect for mic, and adds surround - * controls. Also sets both the Front Playback and Capture Volume controls to - * alt so they set the DSP's decibel level. + * Desktop specific control mixer. Removes auto-detect for mic, and adds + * surround controls. Also sets both the Front Playback and Capture Volume + * controls to alt so they set the DSP's decibel level. */ -static const struct snd_kcontrol_new sbz_mixer[] = { +static const struct snd_kcontrol_new desktop_mixer[] = { CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), @@ -5855,8 +5855,8 @@ static int ca0132_build_controls(struct hda_codec *codec) */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { - /* SBZ breaks if Echo Cancellation is used */ - if (spec->quirk == QUIRK_SBZ) { + /* SBZ and R3D break if Echo Cancellation is used. */ + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) { if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + OUT_EFFECTS_COUNT)) continue; @@ -7608,9 +7608,13 @@ static int patch_ca0132(struct hda_codec *codec) /* Set which mixers each quirk uses. */ switch (spec->quirk) { case QUIRK_SBZ: - spec->mixers[0] = sbz_mixer; + spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound Blaster Z"); break; + case QUIRK_R3D: + spec->mixers[0] = desktop_mixer; + snd_hda_codec_set_name(codec, "Recon3D"); + break; case QUIRK_R3DI: spec->mixers[0] = r3di_mixer; snd_hda_codec_set_name(codec, "Recon3Di"); From 2f295f91b740f0055735a7528f8f4cf8b3111239 Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Wed, 8 Aug 2018 13:34:22 -0400 Subject: [PATCH 524/529] ALSA: hda/ca0132 - Add exit commands for Recon3D This patch adds exit functions for the Recon3D, and cleans up the current exit function. Signed-off-by: Connor McAdams Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 601efaa5c610..0166a3d7cd55 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6968,7 +6968,7 @@ static void sbz_set_pin_ctl_default(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); } -static void sbz_clear_unsolicited(struct hda_codec *codec) +static void ca0132_clear_unsolicited(struct hda_codec *codec) { hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13}; unsigned int i; @@ -7021,21 +7021,22 @@ static void sbz_exit_chip(struct hda_codec *codec) chipio_set_control_param(codec, 0x0D, 0x24); - sbz_clear_unsolicited(codec); + ca0132_clear_unsolicited(codec); sbz_set_pin_ctl_default(codec); snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00); - if (dspload_is_loaded(codec)) - dsp_reset(codec); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00); - sbz_region2_exit(codec); } +static void r3d_exit_chip(struct hda_codec *codec) +{ + ca0132_clear_unsolicited(codec); + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -7381,16 +7382,17 @@ static void ca0132_free(struct hda_codec *codec) case QUIRK_SBZ: sbz_exit_chip(codec); break; + case QUIRK_R3D: + r3d_exit_chip(codec); + break; case QUIRK_R3DI: r3di_gpio_shutdown(codec); - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); - break; - default: - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); break; } + + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + snd_hda_power_down(codec); if (spec->mem_base) iounmap(spec->mem_base); From 0c93c5ce107659069d16fa34ddce465acdf9d996 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 8 Aug 2018 14:25:16 -0500 Subject: [PATCH 525/529] ALSA: opl3: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 114878 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_midi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 471916ca0b6b..a33cb744e96c 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -368,6 +368,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } + /* fall through */ default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; From 725097323bbcbbca51ede20b542c6229a2869445 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 8 Aug 2018 17:11:48 -0500 Subject: [PATCH 526/529] ALSA: mixart: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Addresses-Coverity-ID: 114889 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 46c292b52fd6..71776bfe0485 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -540,7 +540,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* no break, continue ! */ + /* fall through */ case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); From 91c6e15efc1756b068dc6e945c1626397e60a119 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 8 Aug 2018 17:23:44 -0500 Subject: [PATCH 527/529] ALSA: usb-audio: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357413 ("Missing break in switch") Addresses-Coverity-ID: 114917 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 73e811f86a95..c63c84b54969 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -856,12 +856,14 @@ static int check_input_term(struct mixer_build *state, int id, term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ else /* UAC_VERSION_2 */ term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + /* fall through */ case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ if (protocol == UAC_VERSION_1 && !term->type) term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ else if (protocol == UAC_VERSION_2 && !term->type) term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + /* fall through */ case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; From 17c81d2f5a59929c73a2a19fd49fe0b068fda76f Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 9 Aug 2018 10:48:50 +0100 Subject: [PATCH 528/529] ASoC: da7219: Add delays to capture path to remove DC offset noise On some platforms it has been noted that a pop noise can be witnessed when capturing audio, mainly for first time after a headset jack has been inserted. This is due to a DC offset in the Mic PGA and so to avoid this delays are required when powering up the capture path. This commit rectifies the problem by adding delays post Mic PGA and post Mixin PGA. The post Mic PGA delay is determined based on Mic Bias voltage, and is only applied the first time after a headset jack is inserted. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 5 ++++ sound/soc/codecs/da7219.c | 44 ++++++++++++++++++++++++++++++----- sound/soc/codecs/da7219.h | 8 +++++-- 3 files changed, 49 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a49ab751a036..2c7d5088e6f2 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work) container_of(work, struct da7219_aad_priv, btn_det_work); struct snd_soc_component *component = da7219_aad->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 statusa, micbias_ctrl; bool micbias_up = false; int retries = 0; @@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work) if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) dev_warn(component->dev, "Mic bias status check timed out"); + da7219->micbias_on_event = true; + /* * Mic bias pulse required to enable mic, must be done before enabling * button detection to prevent erroneous button readings. @@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219->micbias_on_event = false; + /* Disable mic bias */ snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c0144f2f8174..e46e9f4bc994 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { * DAPM Events */ +static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7219->micbias_on_event) { + /* + * Delay only for first capture after bias enabled to + * avoid possible DC offset related noise. + */ + da7219->micbias_on_event = false; + msleep(da7219->mic_pga_delay); + } + break; + default: + break; + } + + return 0; +} + static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), /* Input PGAs */ - SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, - DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, - DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), + SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU), /* Input Filters */ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, @@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component) snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + /* + * Calculate delay required to compensate for DC offset in + * Mic PGA, based on Mic Bias voltage. + */ + da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY + + (pdata->micbias_lvl * + DA7219_MIC_PGA_OFFSET_DELAY); + /* Mic */ switch (pdata->mic_amp_in_sel) { case DA7219_MIC_AMP_IN_SEL_DIFF: diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1b00023e33cd..3a006862f0e7 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -781,8 +781,10 @@ #define DA7219_SYS_STAT_CHECK_DELAY 50 /* Power up/down Delays */ -#define DA7219_SETTLING_DELAY 40 -#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_MIC_PGA_BASE_DELAY 100 +#define DA7219_MIC_PGA_OFFSET_DELAY 40 enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, @@ -828,6 +830,8 @@ struct da7219_priv { bool master; bool alc_en; + bool micbias_on_event; + unsigned int mic_pga_delay; u8 gain_ramp_ctrl; }; From f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 8 Aug 2018 14:19:33 -0500 Subject: [PATCH 529/529] ASoC: adav80x: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056531 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index db21ecbe0762..8b9ca7e7a682 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } + /* fall through */ default: return -EINVAL; }