From 101f6f4ba7109aafd7b3f2eafd7124f854dda451 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2007 12:03:09 +0200 Subject: [PATCH 001/102] [ALSA] Fix unfreed pnp driver in opti9xx ISA driver - Added the missing pnp_unregister_card_driver() in the case ISA PnP isn't found, which caused an error at kobject_add with -EEXIST Jun 11 09:07:31 rain kernel: kobject_add failed for opti9xx with -EEXIST, don't try to register things with the same name in the same directory. Jun 11 09:07:31 rain kernel: [] kobject_shadow_add+0x12a/0x1c0 Jun 11 09:07:31 rain kernel: [] kobject_register+0x21/0x50 Jun 11 09:07:31 rain kernel: [] bus_add_driver+0x72/0x1b0 Jun 11 09:07:31 rain kernel: [] pnp_register_card_driver+0x4f/0xc0 Jun 11 09:07:31 rain kernel: [] alsa_card_opti9xx_init+0xa/0x25 [snd_opti92x_ad1848] Jun 11 09:07:31 rain kernel: [] sys_init_module+0x157/0x1610 Jun 11 09:07:31 rain kernel: [] syscall_call+0x7/0xb - Fixed the probe behavior when no PnP is set up. Now it behaves like the former version. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 60c120ffb9de..049d479ce2b3 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -1927,10 +1927,12 @@ static struct snd_card *snd_opti9xx_card_new(void) static int __devinit snd_opti9xx_isa_match(struct device *devptr, unsigned int dev) { +#ifdef CONFIG_PNP if (snd_opti9xx_pnp_is_probed) return 0; if (isapnp) return 0; +#endif return 1; } @@ -2096,6 +2098,7 @@ static int __init alsa_card_opti9xx_init(void) pnp_register_card_driver(&opti9xx_pnpc_driver); if (snd_opti9xx_pnp_is_probed) return 0; + pnp_unregister_card_driver(&opti9xx_pnpc_driver); #endif return isa_register_driver(&snd_opti9xx_driver, 1); } From 500bec3d2ba4787e6ab110d159b2926093c27304 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 10 May 2007 08:52:19 +0200 Subject: [PATCH 002/102] [ALSA] usb-audio: another Logitech QuickCam ID This patch adds the ID for another quickcam microphone, reported by freqmod on ALSA ticket #0003040 I'm going to submit a USB patch separately to provide a macro to simplify these entries, as suggested by Alan Stern. We could switch to using that in future. Signed-off-by: Daniel Drake Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 374fbf657a2d..59f07e874e52 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -52,6 +52,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x0850, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | From c111b8de975a357ff58e73047e9af48afacf2347 Mon Sep 17 00:00:00 2001 From: Richard Knutsson Date: Mon, 14 May 2007 10:38:57 +0200 Subject: [PATCH 003/102] [ALSA] usbusx2yaudio: kfree(NULL) is valid if (!x) kfree(x); is not needed since kfree(NULL) is valid. Signed-off-by: Richard Knutsson Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usx2y/usbusx2yaudio.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 0a352e46862f..48e9aa3f18c9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -935,10 +935,9 @@ static struct snd_pcm_ops snd_usX2Y_pcm_ops = */ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream) { - if (NULL != usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]) { - kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); - usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; - } + kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); + usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; + kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]); usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL; } From 050f05eaec1c7c5434c78d010ada3cfeb7d0b3b3 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 14 May 2007 11:02:51 +0200 Subject: [PATCH 004/102] [ALSA] ASoC S3C24xx machine drivers - s3c2443-AC97 This patch adds AC97 support to the Samsung S3C2443 CPU. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c2443-ac97.c | 401 +++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx-ac97.h | 25 ++ 2 files changed, 426 insertions(+) create mode 100644 sound/soc/s3c24xx/s3c2443-ac97.c create mode 100644 sound/soc/s3c24xx/s3c24xx-ac97.h diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c new file mode 100644 index 000000000000..75acf7ef5528 --- /dev/null +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -0,0 +1,401 @@ +/* + * s3c2443-ac97.c -- ALSA Soc Audio Layer + * + * (c) 2007 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * Copyright (C) 2005, Sean Choi + * 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. + * + * Revision history + * 21st Mar 2007 Initial Version + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +struct s3c24xx_ac97_info { + void __iomem *regs; + struct clk *ac97_clk; +}; +static struct s3c24xx_ac97_info s3c24xx_ac97; + +DECLARE_COMPLETION(ac97_completion); +static u32 codec_ready; +static DECLARE_MUTEX(ac97_mutex); + +static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl; + u32 ac_codec_cmd; + u32 stat, addr, data; + + down(&ac97_mutex); + + codec_ready = S3C_AC97_GLBSTAT_CODECREADY; + ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + + wait_for_completion(&ac97_completion); + + stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," + " rep addr = %02x\n", reg, addr); + + up(&ac97_mutex); + + return (unsigned short)data; +} + +static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl; + u32 ac_codec_cmd; + + down(&ac97_mutex); + + codec_ready = S3C_AC97_GLBSTAT_CODECREADY; + ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + + wait_for_completion(&ac97_completion); + + ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); + + up(&ac97_mutex); + +} + +static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl; + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = 0; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl; + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = 0; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA | + S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); +} + +static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id) +{ + int status; + u32 ac_glbctrl; + + status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready; + + if (status) { + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + complete(&ac97_completion); + } + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c2443_ac97_read, + .write = s3c2443_ac97_write, + .warm_reset = s3c2443_ac97_warm_reset, + .reset = s3c2443_ac97_cold_reset, +}; + +static struct s3c2410_dma_client s3c2443_dma_client_out = { + .name = "AC97 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2443_dma_client_in = { + .name = "AC97 PCM Stereo in" +}; + +static struct s3c2410_dma_client s3c2443_dma_client_micin = { + .name = "AC97 Mic Mono in" +}; + +static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { + .client = &s3c2443_dma_client_out, + .channel = DMACH_PCM_OUT, + .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, + .dma_size = 4, +}; + +static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { + .client = &s3c2443_dma_client_in, + .channel = DMACH_PCM_IN, + .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, + .dma_size = 4, +}; + +static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { + .client = &s3c2443_dma_client_micin, + .channel = DMACH_MIC_IN, + .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, + .dma_size = 4, +}; + +static int s3c2443_ac97_probe(struct platform_device *pdev) +{ + int ret; + u32 ac_glbctrl; + + s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100); + if (s3c24xx_ac97.regs == NULL) + return -ENXIO; + + s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (s3c24xx_ac97.ac97_clk == NULL) { + printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n"); + iounmap(s3c24xx_ac97.regs); + return -ENODEV; + } + clk_enable(s3c24xx_ac97.ac97_clk); + + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO); + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = 0; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + + ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); + clk_disable(s3c24xx_ac97.ac97_clk); + clk_put(s3c24xx_ac97.ac97_clk); + iounmap(s3c24xx_ac97.regs); + } + return ret; +} + +static void s3c2443_ac97_remove(struct platform_device *pdev) +{ + free_irq(IRQ_S3C2443_AC97, NULL); + clk_disable(s3c24xx_ac97.ac97_clk); + clk_put(s3c24xx_ac97.ac97_clk); + iounmap(s3c24xx_ac97.regs); +} + +static int s3c2443_ac97_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; + else + cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in; + + return 0; +} + +static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) +{ + u32 ac_glbctrl; + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + switch(cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + break; + } + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + + return 0; +} + +static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in; + + return 0; +} + +static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + u32 ac_glbctrl; + + ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + switch(cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + } + writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + + return 0; +} + +#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +{ + .name = "s3c2443-ac97", + .id = 0, + .type = SND_SOC_DAI_AC97, + .probe = s3c2443_ac97_probe, + .remove = s3c2443_ac97_remove, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = s3c2443_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = s3c2443_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = s3c2443_ac97_hw_params, + .trigger = s3c2443_ac97_trigger}, +}, +{ + .name = "pxa2xx-ac97-mic", + .id = 1, + .type = SND_SOC_DAI_AC97, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = s3c2443_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = s3c2443_ac97_hw_mic_params, + .trigger = s3c2443_ac97_mic_trigger,}, +}, +}; + +EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +MODULE_AUTHOR("Graeme Gregory"); +MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h new file mode 100644 index 000000000000..2b835e8260fa --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -0,0 +1,25 @@ +/* + * s3c24xx-ac97.c -- ALSA Soc Audio Layer + * + * (c) 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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. + * + * Revision history + * 10th Nov 2006 Initial version. + */ + +#ifndef S3C24XXAC97_H_ +#define S3C24XXAC97_H_ + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; + +#endif /*S3C24XXAC97_H_*/ From 74930bb6db56bcc9899723c6c79fe681524e5b62 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 14 May 2007 11:03:52 +0200 Subject: [PATCH 005/102] [ALSA] ASoC S3C24xx machine drivers - Openmoko Neo1973 This patch adds ALSA support for the Openmoko Neo1973 phone. Features:- * HiFi Playback and capture. * Phone calls supported. * Support for BT PCM in WM8753 voice interface. * Support for LM4857 audio amp. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 670 +++++++++++++++++++++++++++++ 1 file changed, 670 insertions(+) create mode 100644 sound/soc/s3c24xx/neo1973_wm8753.c diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c new file mode 100644 index 000000000000..d5a8fc2cf8d6 --- /dev/null +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -0,0 +1,670 @@ +/* + * neo1973_wm8753.c -- SoC audio for Neo1973 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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. + * + * Revision history + * 20th Jan 2007 Initial version. + * 05th Feb 2007 Rename all to Neo1973 + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/wm8753.h" +#include "lm4857.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +/* define the scenarios */ +#define NEO_AUDIO_OFF 0 +#define NEO_GSM_CALL_AUDIO_HANDSET 1 +#define NEO_GSM_CALL_AUDIO_HEADSET 2 +#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 +#define NEO_STEREO_TO_SPEAKERS 4 +#define NEO_STEREO_TO_HEADPHONES 5 +#define NEO_CAPTURE_HANDSET 6 +#define NEO_CAPTURE_HEADSET 7 +#define NEO_CAPTURE_BLUETOOTH 8 + +static struct snd_soc_machine neo1973; +static struct i2c_client *i2c; + +static int neo1973_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS ); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4,4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_hifi_ops = { + .hw_params = neo1973_hifi_hw_params, + .hw_free = neo1973_hifi_hw_free, +}; + +static int neo1973_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); +} + +static struct snd_soc_ops neo1973_voice_ops = { + .hw_params = neo1973_voice_hw_params, + .hw_free = neo1973_voice_hw_free, +}; + +static int neo1973_scenario = 0; + +static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = neo1973_scenario; + return 0; +} + +static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) +{ + switch(neo1973_scenario) { + case NEO_AUDIO_OFF: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_GSM_CALL_AUDIO_HANDSET: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + break; + case NEO_GSM_CALL_AUDIO_HEADSET: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_GSM_CALL_AUDIO_BLUETOOTH: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_STEREO_TO_SPEAKERS: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_STEREO_TO_HEADPHONES: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_CAPTURE_HANDSET: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + break; + case NEO_CAPTURE_HEADSET: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + case NEO_CAPTURE_BLUETOOTH: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + break; + default: + snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + } + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (neo1973_scenario == ucontrol->value.integer.value[0]) + return 0; + + neo1973_scenario = ucontrol->value.integer.value[0]; + set_scenario_endpoints(codec, neo1973_scenario); + return 1; +} + +static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; + +static void lm4857_write_regs(void) +{ + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) + printk(KERN_ERR "lm4857: i2c write failed\n"); +} + +static int lm4857_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int reg=kcontrol->private_value & 0xFF; + int shift = (kcontrol->private_value >> 8) & 0x0F; + int mask = (kcontrol->private_value >> 16) & 0xFF; + + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; + return 0; +} + +static int lm4857_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int reg = kcontrol->private_value & 0xFF; + int shift = (kcontrol->private_value >> 8) & 0x0F; + int mask = (kcontrol->private_value >> 16) & 0xFF; + + if (((lm4857_regs[reg] >> shift ) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + lm4857_regs[reg] &= ~ (mask << shift); + lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; + lm4857_write_regs(); + return 1; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + + if (value) + value -= 5; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + if (value) + value += 5; + + if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) + return 0; + + lm4857_regs[LM4857_CTRL] &= 0xF0; + lm4857_regs[LM4857_CTRL] |= value; + lm4857_write_regs(); + return 1; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Call Mic", NULL), +}; + + +/* example machine audio_mapnections */ +static const char* audio_map[][3] = { + + /* Connections to the lm4857 amp */ + {"Audio Out", NULL, "LOUT1"}, + {"Audio Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Call Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, + + {NULL, NULL, NULL}, +}; + +static const char *lm4857_mode[] = { + "Off", + "Call Speaker", + "Stereo Speakers", + "Stereo Speakers + Headphones", + "Headphones" +}; + +static const struct soc_enum lm4857_mode_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), +}; + +static const char *neo_scenarios[] = { + "Off", + "GSM Handset", + "GSM Headset", + "GSM Bluetooth", + "Speakers", + "Headphones", + "Capture Handset", + "Capture Headset", + "Capture Bluetooth" +}; + +static const struct soc_enum neo_scenario_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios), +}; + +static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], + lm4857_get_mode, lm4857_set_mode), + SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], + neo1973_get_scenario, neo1973_set_scenario), + SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, + lm4857_get_reg, lm4857_set_reg), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 II. It is missing logic to detect hp/mic insertions and logic + * to re-route the audio in such an event. + */ +static int neo1973_wm8753_init(struct snd_soc_codec *codec) +{ + int i, err; + + /* set up NC codec pins */ + snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); + snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); + snd_soc_dapm_set_endpoint(codec, "OUT3", 0); + snd_soc_dapm_set_endpoint(codec, "OUT4", 0); + snd_soc_dapm_set_endpoint(codec, "LINE1", 0); + snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + + + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + + /* Add neo1973 specific widgets */ + for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); + + /* add neo1973 specific controls */ + for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8753_neo1973_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + /* set up neo1973 specific audio path audio_mapnects */ + for (i = 0; audio_map[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + } + + snd_soc_dapm_sync_endpoints(codec); + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_cpu_dai bt_dai = +{ .name = "Bluetooth", + .id = 0, + .type = SND_SOC_DAI_PCM, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_wm8753_init, + .ops = &neo1973_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_voice_ops, +}, +}; + +static struct snd_soc_machine neo1973 = { + .name = "neo1973", + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), +}; + +static struct wm8753_setup_data neo1973_wm8753_setup = { + .i2c_address = 0x1a, +}; + +static struct snd_soc_device neo1973_snd_devdata = { + .machine = &neo1973, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8753, + .codec_data = &neo1973_wm8753_setup, +}; + +static struct i2c_client client_template; + +static unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) +{ + int ret; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) + return -ENOMEM; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "LM4857 failed to attach at addr %x\n", addr); + goto exit_err; + } + + lm4857_write_regs(); + return ret; + +exit_err: + kfree(i2c); + return ret; +} + +static int lm4857_i2c_detach(struct i2c_client *client) +{ + i2c_detach_client(client); + kfree(client); + return 0; +} + +static int lm4857_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, lm4857_amp_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "LM4857 I2C Amp", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_LM4857, + .attach_adapter = lm4857_i2c_attach, + .detach_client = lm4857_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "LM4857", + .driver = &lm4857_i2c_driver, +}; + +static struct platform_device *neo1973_snd_device; + +static int __init neo1973_init(void) +{ + int ret; + + neo1973_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); + neo1973_snd_devdata.dev = &neo1973_snd_device->dev; + ret = platform_device_add(neo1973_snd_device); + + if (ret) + platform_device_put(neo1973_snd_device); + + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + + return ret; +} + +static void __exit neo1973_exit(void) +{ + platform_device_unregister(neo1973_snd_device); +} + +module_init(neo1973_init); +module_exit(neo1973_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); +MODULE_LICENSE("GPL"); From 438b76ad816ef7d8e523d7b10d113eeb0b9932b2 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 14 May 2007 11:04:34 +0200 Subject: [PATCH 006/102] [ALSA] ASoC S3C24xx machine drivers - SMDK 2443 This patch adds ALSA support for the SMDK2443 reference board. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/smdk2443_wm9710.c | 85 +++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) create mode 100644 sound/soc/s3c24xx/smdk2443_wm9710.c diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c new file mode 100644 index 000000000000..d46cd811ceb3 --- /dev/null +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -0,0 +1,85 @@ +/* + * smdk2443_wm9710.c -- SoC audio for smdk2443 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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. + * + * Revision history + * 8th Mar 2007 Initial version. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +static struct snd_soc_machine smdk2443; + +static struct snd_soc_dai_link smdk2443_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c2443_ac97_dai[0], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_machine smdk2443 = { + .name = "SMDK2443", + .dai_link = smdk2443_dai, + .num_links = ARRAY_SIZE(smdk2443_dai), +}; + +static struct snd_soc_device smdk2443_snd_ac97_devdata = { + .machine = &smdk2443, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *smdk2443_snd_ac97_device; + +static int __init smdk2443_init(void) +{ + int ret; + + smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk2443_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk2443_snd_ac97_device, + &smdk2443_snd_ac97_devdata); + smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; + ret = platform_device_add(smdk2443_snd_ac97_device); + + if (ret) + platform_device_put(smdk2443_snd_ac97_device); + + return ret; +} + +static void __exit smdk2443_exit(void) +{ + platform_device_unregister(smdk2443_snd_ac97_device); +} + +module_init(smdk2443_init); +module_exit(smdk2443_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); +MODULE_LICENSE("GPL"); From 7a05f067c0da139613cbe74583bb7d208a5f87b9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 14 May 2007 11:05:09 +0200 Subject: [PATCH 007/102] [ALSA] ASoC S3C24xx machine drivers - I2C ID for LM4857 This patch adds I2C ID for the LM4857 audio amp and corrects the spacing of the WM8731, WM8750 and WM8753 ID's. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/linux/i2c-id.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index aa83d4163096..b69014865714 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -115,9 +115,10 @@ #define I2C_DRIVERID_KS0127 86 /* Samsung ks0127 video decoder */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ #define I2C_DRIVERID_ISL1208 88 /* Intersil ISL1208 RTC */ -#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ -#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ -#define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ +#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ +#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ +#define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ +#define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ From 80ab1c0e9ea90467e34dd3187b1d8162e8be314b Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 14 May 2007 11:06:11 +0200 Subject: [PATCH 008/102] [ALSA] ASoC S3C24xx machine drivers - Kconfig This patch adds Kconfig and build support for the Neo1973, SMDK2443 and S3C2443 AC97 ALSA audio drivers. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/Kconfig | 27 +++++++++++++++++++++++++++ sound/soc/s3c24xx/Makefile | 9 +++++++++ 2 files changed, 36 insertions(+) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 044a3712077a..e97c68306a9a 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,7 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" depends on ARCH_S3C2410 && SND_SOC + select SND_PCM help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -8,3 +9,29 @@ config SND_S3C24XX_SOC config SND_S3C24XX_SOC_I2S tristate + +config SND_S3C2443_SOC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + +config SND_S3C24XX_SOC_NEO1973_WM8753 + tristate "SoC I2S Audio support for NEO1973 - WM8753" + depends on SND_S3C24XX_SOC && MACH_GTA01 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on smdk2440 + with the WM8753. + +config SND_S3C24XX_SOC_SMDK2443_WM9710 + tristate "SoC AC97 Audio support for SMDK2443 - WM9710" + depends on SND_S3C24XX_SOC && MACH_SMDK2443 + select SND_S3C2443_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y if you want to add support for SoC audio on smdk2443 + with the WM9710. + + diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 6f0fffcb30f5..13c92f0fa1e4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,6 +1,15 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o +obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o + +# S3C24XX Machine Support +snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o + +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o From aef3b06ac69783d6a6d1e4357c62bab46dd16141 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 14 May 2007 18:40:07 +0200 Subject: [PATCH 009/102] [ALSA] SH7760 ASoC support ALSA ASoC support for SH7760 This patch adds ALSA ASoC drivers for the Audio interfaces of the SH7760 SoC: Add driver for the SH7760 DMA engine (dmabrg) Add AC97 driver for HAC unit(s) found on SH7760/SH7780 Add I2S driver for SSI unit(s) found on SH7760/SH7780 Add a generic SH7760-AC97 machine driver. Hook it all up with the build system. Signed-off-by: Manuel Lauss Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/sh/Kconfig | 39 ++++ sound/soc/sh/Makefile | 14 ++ sound/soc/sh/dma-sh7760.c | 354 ++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 322 +++++++++++++++++++++++++++++ sound/soc/sh/sh7760-ac97.c | 92 +++++++++ sound/soc/sh/ssi.c | 400 +++++++++++++++++++++++++++++++++++++ 8 files changed, 1223 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sh/Kconfig create mode 100644 sound/soc/sh/Makefile create mode 100644 sound/soc/sh/dma-sh7760.c create mode 100644 sound/soc/sh/hac.c create mode 100644 sound/soc/sh/sh7760-ac97.c create mode 100644 sound/soc/sh/ssi.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 10cffc087181..97b255233175 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -27,6 +27,7 @@ config SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/sh/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0ae2e49036f9..304140377632 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 000000000000..a332e51d6887 --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,39 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +config SND_SH7760_AC97 + tristate "SH7760 AC97 sound support" + depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760 + select SND_SOC_SH4_HAC + select SND_SOC_AC97_CODEC + help + This option enables generic sound support for the first + AC97 unit of the SH7760. + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 000000000000..a8e8ab81cc6a --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,14 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards +snd-soc-sh7760-ac97-objs := sh7760-ac97.o + +obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 000000000000..cdee374b843e --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,354 @@ +/* + * 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 + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of a + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + pr_debug("PCM data: addr 0x%08ulx len %d\n", + (u32)runtime->dma_addr, runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * position at the time the IRQ was triggered, which results in + * fast-playback sound in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-pcm", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 000000000000..8e3f03908cdb --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,322 @@ +/* + * 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! + */ + +/* 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 + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "hac: reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + pr_debug("hac: invalid depth %d bit\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c new file mode 100644 index 000000000000..5563f14511fa --- /dev/null +++ b/sound/soc/sh/sh7760-ac97.c @@ -0,0 +1,92 @@ +/* + * Generic AC97 sound support for SH7760 + * + * (c) 2007 Manuel Lauss + * + * Licensed under the GPLv2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" + +#define IPSEL 0xFE400034 + +/* platform specific structs can be declared here */ +extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_platform sh7760_soc_platform; + +static int machine_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync_endpoints(codec); + return 0; +} + +static struct snd_soc_dai_link sh7760_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &sh4_hac_dai[0], /* HAC0 */ + .codec_dai = &ac97_dai, + .init = machine_init, + .ops = NULL, +}; + +static struct snd_soc_machine sh7760_ac97_soc_machine = { + .name = "SH7760 AC97", + .dai_link = &sh7760_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device sh7760_ac97_snd_devdata = { + .machine = &sh7760_ac97_soc_machine, + .platform = &sh7760_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *sh7760_ac97_snd_device; + +static int __init sh7760_ac97_init(void) +{ + int ret; + unsigned short ipsel; + + /* enable both AC97 controllers in pinmux reg */ + ipsel = ctrl_inw(IPSEL); + ctrl_outw(ipsel | (3 << 10), IPSEL); + + ret = -ENOMEM; + sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1); + if (!sh7760_ac97_snd_device) + goto out; + + platform_set_drvdata(sh7760_ac97_snd_device, + &sh7760_ac97_snd_devdata); + sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev; + ret = platform_device_add(sh7760_ac97_snd_device); + + if (ret) + platform_device_put(sh7760_ac97_snd_device); + +out: + return ret; +} + +static void __exit sh7760_ac97_exit(void) +{ + platform_device_unregister(sh7760_ac97_snd_device); +} + +module_init(sh7760_ac97_init); +module_exit(sh7760_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 000000000000..b72bc316cb8e --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,400 @@ +/* + * 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! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT) +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK (7 << CR_DWL_SHIFT) +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK (7 << CR_SWL_SHIFT) +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_SHIFT 4 +#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */ +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + pr_debug("ssi: already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); + pr_debug("bits: %d channels: %d\n", bits, channels); + + ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | + CR_SWL_MASK); + + /* direction (send/receive) */ + if (!recv) + ssicr |= CR_TRMD; /* transmit */ + + /* channels */ + if ((channels < 2) || (channels > 8) || (channels & 1)) { + pr_debug("ssi: invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK; + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + pr_debug("ssi: invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP | + CR_SWS_MASTER | CR_SCK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + pr_debug("ssi: unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + pr_debug("ssi: invalid inversion\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + pr_debug("ssi: invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss "); From 198de43d758ca2700e2b52b49c0b189b4931466c Mon Sep 17 00:00:00 2001 From: Adrian McMenamin Date: Fri, 18 May 2007 14:26:59 +0200 Subject: [PATCH 010/102] [ALSA] Add ALSA support for the SEGA Dreamcast PCM device ALSA support for the SEGA Dreamcast Yamaha AICA sound device (pcm) This patch adds ALSA sound support for pcm playback on two channels on the SEGA Dreamcast built-in sound device (the Yamaha AICA) Add driver for the AICA sound device built into the SEGA Dreamcast Hook it all up with the build system. Signed-off-by: Adrian McMenamin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/Kconfig | 2 + sound/Makefile | 2 +- sound/sh/Kconfig | 14 + sound/sh/Makefile | 8 + sound/sh/aica.c | 675 ++++++++++++++++++++++++++++++++++++++++++++++ sound/sh/aica.h | 80 ++++++ 6 files changed, 780 insertions(+), 1 deletion(-) create mode 100644 sound/sh/Kconfig create mode 100644 sound/sh/Makefile create mode 100644 sound/sh/aica.c create mode 100644 sound/sh/aica.h diff --git a/sound/Kconfig b/sound/Kconfig index 9ea473823418..e48b9b37d228 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -65,6 +65,8 @@ source "sound/arm/Kconfig" source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index b7c7fb7c24c8..3ead922bd9c6 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig new file mode 100644 index 000000000000..b7e08ef22a94 --- /dev/null +++ b/sound/sh/Kconfig @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff --git a/sound/sh/Makefile b/sound/sh/Makefile new file mode 100644 index 000000000000..8fdcb6e26f00 --- /dev/null +++ b/sound/sh/Makefile @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o diff --git a/sound/sh/aica.c b/sound/sh/aica.c new file mode 100644 index 000000000000..97bb86a58622 --- /dev/null +++ b/sound/sh/aica.c @@ -0,0 +1,675 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006, 2007 +* +* Requires firmware (BSD licenced) available from: +* http://linuxdc.cvs.sourceforge.net/linuxdc/linux-sh-dc/sound/oss/aica/firmware/ +* or the maintainer +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License 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. +* +* 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 +* +*/ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin "); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + { + snd_printk("WARNING: G2 FIFO appears to be blocked.\n"); + break; + } + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(u32 toi, u32 what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + for (i = 0; i < length; i++) { + if (!(i % 8)) + spu_write_wait(); + writel(what, toi + SPU_MEMORY_BASE); + toi++; + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(u32 toi, void *from, int length) +{ + u32 *froml = from; + u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); + int i; + u32 val; + length = DIV_ROUND_UP(length, 4); + spu_write_wait(); + for (i = 0; i < length; i++) { + if (!(i % 8)) + spu_write_wait(); + val = *froml; + writel(val, to); + froml++; + to++; + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + u32 regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + u32 regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +static void run_spu_dma(struct work_struct *work) +{ + int buffer_size; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_card_aica *dreamcastcard; + struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); + substream = holder-> sspointer; + dreamcastcard = substream->pcm->private_data; + runtime = substream->runtime; + if (unlikely(dreamcastcard->dma_check == 0)) { + buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels > 1) + dreamcastcard->channel->flags |= 0x01; + aica_dma_transfer(runtime->channels, buffer_size, substream); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) + { + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + } + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer function - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + mod_timer(&(dreamcastcard->timer), jiffies + 1); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Use queue to do the heavy lifting */ + spu_working.sspointer = substream; + INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); + queue_work(aica_queue, &(spu_working.spu_dma_work)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream + *substream, int cmd) +{ + struct snd_card_aica *dreamcastcard; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + spu_begin_dma(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + dreamcastcard = substream->pcm->private_data; + if (dreamcastcard->timer.data) + del_timer(&dreamcastcard->timer); + aica_chn_halt(); + break; + default: + return -EINVAL; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue(CARD_NAME); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + return load_aica_firmware(); +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff --git a/sound/sh/aica.h b/sound/sh/aica.h new file mode 100644 index 000000000000..0603b5b0376b --- /dev/null +++ b/sound/sh/aica.h @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License 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. + * + * 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 + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; From 2bac647c45ed9b397ace9109e6935c1364690f8a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 May 2007 18:21:41 +0200 Subject: [PATCH 011/102] [ALSA] hda-codec - Add AD1884 / AD1984 codec support Added the support of AD1884 and AD1984 codec chips. Also experimental quirks for Thinkpad T61/X61 laptops with AD1984. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 10 + sound/pci/hda/patch_analog.c | 345 +++++++++++++++++- 2 files changed, 352 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 355ff0a2bb7c..297ff0de7392 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -864,12 +864,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1884 + N/A + AD1981 basic 3-jack (default) hp HP nx6320 thinkpad Lenovo Thinkpad T60/X60/Z60 toshiba Toshiba U205 + AD1983 + N/A + + AD1984 + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + AD1986A 6stack 6-jack, separate surrounds (default) 3stack 3-stack, shared surrounds diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0e1a879663fa..30248cd562c5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,7 +1,8 @@ /* - * HD audio interface patch for AD1981HD, AD1983, AD1986A, AD1988 + * HD audio interface patch for AD1884, AD1981HD, AD1983, AD1984, AD1986A, + * AD1988 * - * Copyright (c) 2005 Takashi Iwai + * Copyright (c) 2005-2007 Takashi Iwai * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -61,7 +62,7 @@ struct ad198x_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; @@ -2774,12 +2775,350 @@ static int patch_ad1988(struct hda_codec *codec) } +/* + * AD1884 / AD1984 + * + * port-B - front line/mic-in + * port-E - aux in/out + * port-F - aux in/out + * port-C - rear line/mic-in + * port-D - rear line/hp-out + * port-A - front line/hp-out + * + * AD1984 = AD1884 + two digital mic-ins + * + * FIXME: + * For simplicity, we share the single DAC for both HP and line-outs + * right now. The inidividual playbacks could be easily implemented, + * but no build-up framework is given, so far. + */ + +static hda_nid_t ad1884_dac_nids[1] = { + 0x04, +}; + +static hda_nid_t ad1884_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1884_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1884_SPDIF_OUT 0x02 + +static struct hda_input_mux ad1884_capture_source = { + .num_items = 4, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x1 }, + { "CD", 0x2 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884_base_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), + /* + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + */ + HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984_dmic_mixers[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, + HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, + HDA_OUTPUT), + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1884_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-A (HP) mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x22, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono selector */ + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C (rear mic) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +static int patch_ad1884(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); + spec->multiout.dac_nids = ad1884_dac_nids; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids); + spec->adc_nids = ad1884_adc_nids; + spec->capsrc_nids = ad1884_capsrc_nids; + spec->input_mux = &ad1884_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1884_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1884_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + return 0; +} + +/* + * Lenovo Thinkpad T61/X61 + */ +static struct hda_input_mux ad1984_thinkpad_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* additional verbs */ +static struct hda_verb ad1984_thinkpad_init_verbs[] = { + /* Port-E (docking station mic) pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* docking mic boost */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer - docking mic; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } /* end */ +}; + +/* Digial MIC ADC NID 0x05 + 0x06 */ +static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + stream_tag, 0, format); + return 0; +} + +static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + 0, 0, 0); + return 0; +} + +static struct hda_pcm_stream ad1984_pcm_dmic_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x05, + .ops = { + .prepare = ad1984_pcm_dmic_prepare, + .cleanup = ad1984_pcm_dmic_cleanup + }, +}; + +static int ad1984_build_pcms(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + struct hda_pcm *info; + int err; + + err = ad198x_build_pcms(codec); + if (err < 0) + return err; + + info = spec->pcm_rec + codec->num_pcms; + codec->num_pcms++; + info->name = "AD1984 Digital Mic"; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture; + return 0; +} + +/* models */ +enum { + AD1984_BASIC, + AD1984_THINKPAD, + AD1984_MODELS +}; + +static const char *ad1984_models[AD1984_MODELS] = { + [AD1984_BASIC] = "basic", + [AD1984_THINKPAD] = "thinkpad", +}; + +static struct snd_pci_quirk ad1984_cfg_tbl[] = { + /* Lenovo Thinkpad T61/X61 */ + SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + {} +}; + +static int patch_ad1984(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config, err; + + err = patch_ad1884(codec); + if (err < 0) + return err; + spec = codec->spec; + board_config = snd_hda_check_board_config(codec, AD1984_MODELS, + ad1984_models, ad1984_cfg_tbl); + switch (board_config) { + case AD1984_BASIC: + /* additional digital mics */ + spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers; + codec->patch_ops.build_pcms = ad1984_build_pcms; + break; + case AD1984_THINKPAD: + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984_thinkpad_capture_source; + spec->mixers[0] = ad1984_thinkpad_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; + break; + } + return 0; +} + + /* * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, + { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, From 959047431a9e65f6ae25705e77a8e460a1aa04c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 19 May 2007 17:06:42 +0200 Subject: [PATCH 012/102] [ALSA] Add description about probe_mask option for snd-hda-intel Added a brief description about probe_mask option for snd-hda-intel. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 297ff0de7392..32124bf00498 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -759,6 +759,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) @@ -966,6 +967,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. from the irq. Remember this is a last resort, and should be avoided as much as possible... + MORE NOTES ON "azx_get_response timeout" PROBLEMS: + On some hardwares, you may need to add a proper probe_mask option + to avoid the "azx_get_response timeout" problem above, instead. + This occurs when the access to non-existing or non-working codec slot + (likely a modem one) causes a stall of the communication via HD-audio + bus. You can see which codec slots are probed by enabling + CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec + proc files. Then limit the slots to probe by probe_mask option. + For example, probe_mask=1 means to probe only the first slot, and + probe_mask=4 means only the third slot. + The power-management is supported. Module snd-hdsp From ea7cfcdfe6439cd98816da9d339a6bc9032d2084 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Sat, 19 May 2007 17:21:04 +0200 Subject: [PATCH 013/102] [ALSA] ice1724 - Add PCM Playback Switch to Revo 7.1 This patch adds the support of mute for front channels of M-Audio Revolution 7.1 (the DAC AK4381 features a mute bit). Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/ak4xxx-adda.h | 1 + sound/i2c/other/ak4xxx-adda.c | 24 ++++++++++++++++++++++-- sound/pci/ice1712/revo.c | 7 ++++++- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h index aa49dda4f410..fd0a6c46f497 100644 --- a/include/sound/ak4xxx-adda.h +++ b/include/sound/ak4xxx-adda.h @@ -43,6 +43,7 @@ struct snd_ak4xxx_ops { struct snd_akm4xxx_dac_channel { char *name; /* mixer volume name */ unsigned int num_channels; + char *switch_name; /* mixer switch*/ }; /* ADC labels and channels */ diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 8805110017a7..fd335159f849 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -481,8 +481,8 @@ static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol, int addr = AK_GET_ADDR(kcontrol->private_value); int shift = AK_GET_SHIFT(kcontrol->private_value); int invert = AK_GET_INVERT(kcontrol->private_value); - unsigned char val = snd_akm4xxx_get(ak, chip, addr); - + /* we observe the (1<value.integer.value[0] = (val & (1<num_dacs; ) { + /* mute control for Revolution 7.1 - AK4381 */ + if (ak->type == SND_AK4381 + && ak->dac_info[mixer_ch].switch_name) { + memset(&knew, 0, sizeof(knew)); + knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + knew.count = 1; + knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + knew.name = ak->dac_info[mixer_ch].switch_name; + knew.info = ak4xxx_switch_info; + knew.get = ak4xxx_switch_get; + knew.put = ak4xxx_switch_put; + knew.access = 0; + /* register 1, bit 0 (SMUTE): 0 = normal operation, + 1 = mute */ + knew.private_value = + AK_COMPOSE(idx/2, 1, 0, 0) | AK_INVERT; + err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); + if (err < 0) + return err; + } memset(&knew, 0, sizeof(knew)); if (! ak->dac_info || ! ak->dac_info[mixer_ch].name) { knew.name = "DAC Volume"; diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 690ceb340644..d18a31e188a9 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -186,7 +186,12 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, #define AK_DAC(xname,xch) { .name = xname, .num_channels = xch } static const struct snd_akm4xxx_dac_channel revo71_front[] = { - AK_DAC("PCM Playback Volume", 2) + { + .name = "PCM Playback Volume", + .num_channels = 2, + /* front channels DAC supports muting */ + .switch_name = "PCM Playback Switch", + }, }; static const struct snd_akm4xxx_dac_channel revo71_surround[] = { From 0dae0f83cd9956d0959f6d6de9f5391da6483274 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 May 2007 12:41:29 +0200 Subject: [PATCH 014/102] [ALSA] hda-codec - Add support of newer version of Intel iMac Added the pin configs for newer version of Intel iMac. The information provided by Ivan N. Zlatev . Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_sigmatel.c | 12 ++++++++++++ 2 files changed, 13 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 32124bf00498..bef13521f6c4 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -923,6 +923,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. macbook-pro-v1 Intel Mac Book Pro 1st generation macbook-pro Intel Mac Book Pro 2nd generation imac-intel Intel iMac + imac-intel-20 Intel iMac (newer version) STAC9202/9250/9251 ref Reference board, base config diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e3964fc4c405..3653f57b7f69 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -64,6 +64,7 @@ enum { STAC_MACBOOK_PRO_V1, STAC_MACBOOK_PRO_V2, STAC_IMAC_INTEL, + STAC_IMAC_INTEL_20, STAC_922X_MODELS }; @@ -567,6 +568,12 @@ static unsigned int imac_intel_pin_configs[10] = { 0x400000fc, 0x400000fb, }; +static unsigned int imac_intel_20_pin_configs[10] = { + 0x0121E21F, 0x90A7012E, 0x9017E110, 0x400000FD, + 0x400000FE, 0x0181E020, 0x1145E230, 0x11C5E240, + 0x400000FC, 0x400000FB, +}; + static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, @@ -576,6 +583,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs, [STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs, [STAC_IMAC_INTEL] = imac_intel_pin_configs, + [STAC_IMAC_INTEL_20] = imac_intel_20_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { @@ -587,6 +595,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1", [STAC_MACBOOK_PRO_V2] = "macbook-pro", [STAC_IMAC_INTEL] = "imac-intel", + [STAC_IMAC_INTEL_20] = "imac-intel-20", }; static struct snd_pci_quirk stac922x_cfg_tbl[] = { @@ -2037,6 +2046,9 @@ static int patch_stac922x(struct hda_codec *codec) case 0x106b0700: /* Intel-based iMac */ spec->board_config = STAC_IMAC_INTEL; break; + case 0x106b1700: /* Intel iMac 20 */ + spec->board_config = STAC_IMAC_INTEL_20; + break; } } From 0ba7962b9f06c02dd1af93002e8d757805d16758 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 May 2007 16:27:32 +0200 Subject: [PATCH 015/102] [ALSA] hda-codec - Fix wrong mixer controls for AD1984 thinkpad model Fixed the wrong mixer controls for AD1984 thinkpad model. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 30248cd562c5..4659fdeec3dc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2979,11 +2979,9 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), From 621887aee9c7b4b613c12b82b83df7e56877f303 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 May 2007 18:46:54 +0200 Subject: [PATCH 016/102] [ALSA] Add support for Cyrix/NatSemi Geode CS5530 (VSA1) Add support for Cyrix/NatSemi Geode SC5530 (VSA1). The driver is snd-cs5530. Signed-off-by Ash Willis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 7 +- include/sound/sb.h | 1 + sound/isa/sb/sb16_main.c | 10 + sound/isa/sb/sb_common.c | 5 +- sound/isa/sb/sb_mixer.c | 3 + sound/pci/Kconfig | 10 + sound/pci/Makefile | 2 + sound/pci/cs5530.c | 306 ++++++++++++++++++ 8 files changed, 342 insertions(+), 2 deletions(-) create mode 100644 sound/pci/cs5530.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index bef13521f6c4..47daca0ecd32 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- diff --git a/include/sound/sb.h b/include/sound/sb.h index 2dd5c8e5b4fe..3ad854b397d2 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 383911b9e74d..5d4d3aafe2d5 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct snd_pcm_substream *substream) __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct snd_pcm_substream *substream) __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f3852167..efa9d5c2558a 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb * chip) minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -168,6 +168,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_DT019X: str = "(DT019X/ALS007)"; break; + case SB_HW_CS5530: + str = "16 (CS5530)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 490b1ca5cf58..3d4befcff28e 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 61e35ecc57b8..fa17786e1587 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -215,6 +215,16 @@ config SND_CS46XX_NEW_DSP This works better than the old code, so say Y. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 diff --git a/sound/pci/Makefile b/sound/pci/Makefile index e06736da9ef1..cd76e0293d06 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -12,6 +12,7 @@ snd-azt3328-objs := azt3328.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_AZT3328) += snd-azt3328.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c new file mode 100644 index 000000000000..240a0a462209 --- /dev/null +++ b/sound/pci/cs5530.c @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis + * (C) Copyright 2003 Red Hat Inc + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * 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, 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. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + From dda369cf2022fb767b158c4ba2fe2f1ca1fba29c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 May 2007 11:50:33 +0200 Subject: [PATCH 017/102] [ALSA] Fix SB-module dependency with PCI drivers A few PCI drivers like ALS4000 and CS5530 require the SB16-codes. This patch fixes / improves the dependency between SB modules and PCI drivers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/Kconfig | 26 ++++++++++++++++++++------ sound/isa/sb/Makefile | 15 +++++++-------- sound/pci/Kconfig | 3 ++- 3 files changed, 29 insertions(+), 15 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index cf3803cd579c..b90ddabae2e9 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -11,6 +11,19 @@ config SND_CS4231_LIB tristate select SND_PCM +config SND_SB_COMMON + tristate + +config SND_SB8_DSP + tristate + select SND_PCM + select SND_SB_COMMON + +config SND_SB16_DSP + tristate + select SND_PCM + select SND_SB_COMMON + config SND_ADLIB tristate "AdLib FM card" depends on SND @@ -55,7 +68,7 @@ config SND_ALS100 select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for soundcards based on Avance Logic ALS100, ALS110, ALS120 and ALS200 chips. @@ -81,6 +94,7 @@ config SND_CMI8330 tristate "C-Media CMI8330" depends on SND select SND_AD1848_LIB + select SND_SB16_DSP help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. @@ -132,7 +146,7 @@ config SND_DT019X select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for soundcards based on the Diamond Technologies DT-019X or Avance Logic ALS-007 chips. @@ -145,7 +159,7 @@ config SND_ES968 depends on SND && PNP && ISA select ISAPNP select SND_MPU401_UART - select SND_PCM + select SND_SB8_DSP help Say Y here to include support for ESS AudioDrive ES968 chips. @@ -321,7 +335,7 @@ config SND_SB8 depends on SND select SND_OPL3_LIB select SND_RAWMIDI - select SND_PCM + select SND_SB8_DSP help Say Y here to include support for Creative Sound Blaster 1.0/ 2.0/Pro (8-bit) or 100% compatible soundcards. @@ -334,7 +348,7 @@ config SND_SB16 depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for Sound Blaster 16 soundcards (including the Plug and Play version). @@ -347,7 +361,7 @@ config SND_SBAWE depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for Sound Blaster AWE soundcards (including the Plug and Play version). diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index fd9d9c5726fc..556e66928029 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -22,14 +22,13 @@ snd-es968-objs := es968.o sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) # Toplevel Module Dependency -obj-$(CONFIG_SND_ALS100) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_CMI8330) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_DT019X) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SB8) += snd-sb8.o snd-sb8-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SB16) += snd-sb16.o snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o +obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o +obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o +obj-$(CONFIG_SND_SB8) += snd-sb8.o +obj-$(CONFIG_SND_SB16) += snd-sb16.o +obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o +obj-$(CONFIG_SND_ES968) += snd-es968.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fa17786e1587..c6b44102aa5b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -33,6 +33,7 @@ config SND_ALS4000 select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_SB_COMMON help Say Y here to include support for soundcards based on Avance Logic ALS4000 chips. @@ -218,7 +219,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on SND && ISA_DMA_API - select SND_SB16 + select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. From 65ff2357a7e40bf4b42858516861c3d0e19f512b Mon Sep 17 00:00:00 2001 From: Ash Willis Date: Tue, 29 May 2007 14:34:17 +0200 Subject: [PATCH 018/102] [ALSA] Disable debugging output for the ALS300 driver Disables debugging output in the ALS300 driver. Also contains a whitespace cleanup and a fix for a potential bug. Signed-off-by: Ash Willis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/als300.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8afcb98ca7bb..48cc39b771d9 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -733,7 +733,8 @@ static int __devinit snd_als300_create(struct snd_card *card, snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + err = snd_als300_ac97(chip); + if (err < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err; From 5d5d3bc3eddf2ad97b2cb090b92580e7fed6cee1 Mon Sep 17 00:00:00 2001 From: "Ivan N. Zlatev" Date: Tue, 29 May 2007 16:03:00 +0200 Subject: [PATCH 019/102] [ALSA] hda-codec - Fix pin configs for Intel Macs * adds the pinconfigs for all 5 Apple boards and 14 Subsystem IDs (support for possibly all iMac, Mac, MacMini etc etc) * adds 'intel-mac-v1' to v5 models which replace the current * reflects changes in Alsa-Configuration.txt Signed-off-by: Ivan N. Zlatev Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 17 +-- sound/pci/hda/patch_sigmatel.c | 101 ++++++++++++------ 2 files changed, 78 insertions(+), 40 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 47daca0ecd32..ca3e6d5e6a53 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -923,12 +923,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF - macmini Intel Mac Mini - macbook Intel Mac Book - macbook-pro-v1 Intel Mac Book Pro 1st generation - macbook-pro Intel Mac Book Pro 2nd generation - imac-intel Intel iMac - imac-intel-20 Intel iMac (newer version) + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) STAC9202/9250/9251 ref Reference board, base config diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3653f57b7f69..dd005ed89f0b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -59,6 +59,12 @@ enum { STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, + STAC_INTEL_MAC_V1, + STAC_INTEL_MAC_V2, + STAC_INTEL_MAC_V3, + STAC_INTEL_MAC_V4, + STAC_INTEL_MAC_V5, + /* for backward compitability */ STAC_MACMINI, STAC_MACBOOK, STAC_MACBOOK_PRO_V1, @@ -550,46 +556,64 @@ static unsigned int d945gtp5_pin_configs[10] = { 0x02a19320, 0x40000100, }; -static unsigned int macbook_pro_v1_pin_configs[10] = { - 0x0321e230, 0x03a1e020, 0x9017e110, 0x01014010, - 0x01a19021, 0x0381e021, 0x1345e240, 0x13c5e22e, - 0x02a19320, 0x400000fb -}; - -static unsigned int macbook_pro_v2_pin_configs[10] = { - 0x0221401f, 0x90a70120, 0x01813024, 0x01014010, - 0x400000fd, 0x01016011, 0x1345e240, 0x13c5e22e, +static unsigned int intel_mac_v1_pin_configs[10] = { + 0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int imac_intel_pin_configs[10] = { - 0x0121e230, 0x90a70120, 0x9017e110, 0x400000fe, - 0x400000fd, 0x0181e021, 0x1145e040, 0x400000fa, +static unsigned int intel_mac_v2_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa, 0x400000fc, 0x400000fb, }; -static unsigned int imac_intel_20_pin_configs[10] = { - 0x0121E21F, 0x90A7012E, 0x9017E110, 0x400000FD, - 0x400000FE, 0x0181E020, 0x1145E230, 0x11C5E240, - 0x400000FC, 0x400000FB, +static unsigned int intel_mac_v3_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240, + 0x400000fc, 0x400000fb, +}; + +static unsigned int intel_mac_v4_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, + 0x400000fc, 0x400000fb, +}; + +static unsigned int intel_mac_v5_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, + 0x400000fc, 0x400000fb, }; static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, - [STAC_MACMINI] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs, - [STAC_IMAC_INTEL] = imac_intel_pin_configs, - [STAC_IMAC_INTEL_20] = imac_intel_20_pin_configs, + [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs, + [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs, + [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, + [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs, + [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs, + /* for backward compitability */ + [STAC_MACMINI] = intel_mac_v3_pin_configs, + [STAC_MACBOOK] = intel_mac_v5_pin_configs, + [STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs, + [STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs, + [STAC_IMAC_INTEL] = intel_mac_v2_pin_configs, + [STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", + [STAC_INTEL_MAC_V1] = "intel-mac-v1", + [STAC_INTEL_MAC_V2] = "intel-mac-v2", + [STAC_INTEL_MAC_V3] = "intel-mac-v3", + [STAC_INTEL_MAC_V4] = "intel-mac-v4", + [STAC_INTEL_MAC_V5] = "intel-mac-v5", + /* for backward compitability */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1", @@ -658,7 +682,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* other systems */ /* Apple Mac Mini (early 2006) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_MACMINI), + "Mac Mini", STAC_INTEL_MAC_V3), {} /* terminator */ }; @@ -2027,27 +2051,36 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, stac922x_cfg_tbl); - if (spec->board_config == STAC_MACMINI) { + if (spec->board_config == STAC_INTEL_MAC_V3) { spec->gpio_mute = 1; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { - case 0x106b0a00: /* MacBook First generatoin */ - spec->board_config = STAC_MACBOOK; + + case 0x106b0800: + spec->board_config = STAC_INTEL_MAC_V1; break; - case 0x106b0200: /* MacBook Pro first generation */ - spec->board_config = STAC_MACBOOK_PRO_V1; + case 0x106b0600: + case 0x106b0700: + spec->board_config = STAC_INTEL_MAC_V2; break; - case 0x106b1e00: /* MacBook Pro second generation */ - spec->board_config = STAC_MACBOOK_PRO_V2; + case 0x106b0e00: + case 0x106b0f00: + case 0x106b1600: + case 0x106b1700: + case 0x106b0200: + case 0x106b1e00: + spec->board_config = STAC_INTEL_MAC_V3; break; - case 0x106b0700: /* Intel-based iMac */ - spec->board_config = STAC_IMAC_INTEL; + case 0x106b1a00: + case 0x00000100: + spec->board_config = STAC_INTEL_MAC_V4; break; - case 0x106b1700: /* Intel iMac 20 */ - spec->board_config = STAC_IMAC_INTEL_20; + case 0x106b0a00: + case 0x106b2200: + spec->board_config = STAC_INTEL_MAC_V5; break; } } From 807a463603a2e014f36e88317aa5e07c15887fcd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 May 2007 19:01:37 +0200 Subject: [PATCH 020/102] [ALSA] hda-codec - Fix STAC922x capture boost level STAC922x provides the capture boost level up to 4, but actually it works only up to 2. Since the range of the mixer is automatically defined from amp-capability bits, we need to override the value beforehand. snd_hda_override_amp_caps() is introduced for this purpose. The function patch_stac922x() calls this for NID 0x12 (Mux Capture Volume). This should fix another recording problem on Intel Macs. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dd005ed89f0b..addcb2a0ae66 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2127,6 +2127,13 @@ static int patch_stac922x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + /* Fix Mux capture level; max to 2 */ + snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, + (0 << AC_AMPCAP_OFFSET_SHIFT) | + (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + return 0; } From ba8c3c37db0d6bf5e6de25016975ebfb558d2aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 May 2007 12:42:31 +0200 Subject: [PATCH 021/102] [ALSA] ali5451 - Fix invalid type of codec->irq field The irq field of struct snd_ali shouldn't be unsigned since it's uninitialized value is -1. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ali5451/ali5451.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 41543a4933e7..e2c8666e06ad 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -239,7 +239,7 @@ struct snd_ali_image { struct snd_ali { - unsigned long irq; + int irq; unsigned long port; unsigned char revision; @@ -2343,7 +2343,7 @@ static int __devinit snd_ali_probe(struct pci_dev *pci, strcpy(card->driver, "ALI5451"); strcpy(card->shortname, "ALI 5451"); - sprintf(card->longname, "%s at 0x%lx, irq %li", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, codec->port, codec->irq); snd_ali_printk("register card.\n"); From 304e3bb7a15d839cef64b24a1bcc1230b2ae6fa6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 May 2007 12:46:21 +0200 Subject: [PATCH 022/102] [ALSA] rme9652 - Fix the hw_pointer check The negative check in hw_pointer callback doesn't work because the value is unsigned. Cast to int in the comparison to fix this. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/rme9652/rme9652.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bd7dbd267ed1..2de27405a0bd 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -406,7 +406,7 @@ static snd_pcm_uframes_t rme9652_hw_pointer(struct snd_rme9652 *rme9652) } else if (!frag) return 0; offset -= rme9652->max_jitter; - if (offset < 0) + if ((int)offset < 0) offset += period_size * 2; } else { if (offset > period_size + rme9652->max_jitter) { From 3b194401a31b8df9a07a1850e3fe44fc7ba832a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Jun 2007 18:32:23 +0200 Subject: [PATCH 023/102] [ALSA] hda-codec - Fix Oops with AD1984 thinkpad model Fixed Oops with AD1984 thinkpad model. Also fixed the wrong init verbs for NID 0x03 and 0x04, which have apparently no mute bit. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4659fdeec3dc..c1216470595d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2883,8 +2883,8 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = { */ static struct hda_verb ad1884_init_verbs[] = { /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Port-A (HP) mixer */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -2960,7 +2960,7 @@ static int patch_ad1884(struct hda_codec *codec) * Lenovo Thinkpad T61/X61 */ static struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 4, + .num_items = 3, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, From 1919de0a9e939ef4f9453ef1630ca1345991e144 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Mon, 4 Jun 2007 23:11:23 +0200 Subject: [PATCH 024/102] [ALSA] More description on duplex streams with OSS emulation Add paragraph to the OSS document to clarify correct use of duplex streams. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/OSS-Emulation.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt index ec2a02541d5b..bfa0c9aacb4b 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -278,6 +278,21 @@ current mixer configuration by reading and writing the whole file image. +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + Unsupported Features ==================== From 538c49c41bf5a0a739ad2a5417ec25e060ee9d9e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Jun 2007 12:13:34 +0200 Subject: [PATCH 025/102] [ALSA] hda-codec - Fix AD1984 basic model Fix the amp direction of digital mic capture volume mixer, which resulted in -EINVAL. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c1216470595d..dff2e7917885 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2872,9 +2872,9 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = { HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, - HDA_OUTPUT), + HDA_INPUT), HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, - HDA_OUTPUT), + HDA_INPUT), { } /* end */ }; From 7d87de2db2213e6e9413532445b14c92dae42c85 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 5 Jun 2007 12:17:21 +0200 Subject: [PATCH 026/102] [ALSA] hda-codec - Add proper model for HP xw series Set the proper model for HP xw4550, xw4600, xw6600 and xw8600. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4776de93928b..c92c97c6cbe5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7592,8 +7592,12 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), From a361d84bfcd938208dea6c84aa19994b3d69e15d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 5 Jun 2007 12:30:55 +0200 Subject: [PATCH 027/102] [ALSA] hda-codec - Add support of ALC268 codec Added the support of new ALC268 codec chip. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 4 + sound/pci/hda/patch_realtek.c | 517 ++++++++++++++++++ 2 files changed, 521 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index ca3e6d5e6a53..7659e5071c41 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -822,6 +822,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) + ALC268 + 3stack 3-stack model + auto auto-config reading BIOS (default) + ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c92c97c6cbe5..49a0fa8ec679 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,13 @@ enum { ALC262_MODEL_LAST /* last tag */ }; +/* ALC268 models */ +enum { + ALC268_3ST, + ALC268_AUTO, + ALC268_MODEL_LAST /* last tag */ +}; + /* ALC861 models */ enum { ALC861_3ST, @@ -7803,6 +7810,515 @@ static int patch_alc262(struct hda_codec *codec) return 0; } +/* + * ALC268 channel source setting (2 channel) + */ +#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID +#define alc268_modes alc260_modes + +static hda_nid_t alc268_dac_nids[2] = { + /* front, hp */ + 0x02, 0x03 +}; + +static hda_nid_t alc268_adc_nids[2] = { + /* ADC0-1 */ + 0x08, 0x07 +}; + +static hda_nid_t alc268_adc_nids_alt[1] = { + /* ADC0 */ + 0x08 +}; + +static struct snd_kcontrol_new alc268_base_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_base_init_verbs[] = { + /* Unmute DAC0-1 and set vol = 0 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, + + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_volume_init_verbs[] = { + /* set output DAC */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* set PCBEEP vol = 0 */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))}, + + { } +}; + +#define alc268_mux_enum_info alc_mux_enum_info +#define alc268_mux_enum_get alc_mux_enum_get + +static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; + hda_nid_t nid = capture_mixers[adc_idx]; + unsigned int *cur_val = &spec->cur_mux[adc_idx]; + unsigned int i, idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx && !codec->in_resume) + return 0; + for (i = 0; i < imux->num_items; i++) { + unsigned int v = (i == idx) ? 0x7000 : 0x7080; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + v | (imux->items[i].index << 8)); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + idx ); + } + *cur_val = idx; + return 1; +} + +static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc268_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux alc268_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + }, +}; + +/* create input playback/capture controls for the given pin */ +static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, + const char *ctlname, int idx) +{ + char name[32]; + int err; + + sprintf(name, "%s Playback Volume", ctlname); + if (nid == 0x14) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x02, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (nid == 0x15) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x03, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else + return -1; + sprintf(name, "%s Playback Switch", ctlname); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + if (err < 0) + return err; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 2; /* only use one dac */ + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 2; + spec->multiout.dac_nids[1] = 3; + + nid = cfg->line_out_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Front", 0); + + nid = cfg->speaker_pins[0]; + if (nid == 0x1d) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Speaker Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + nid = cfg->hp_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Headphone", 0); + + nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; + if (nid == 0x16) { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + struct hda_input_mux *imux = &spec->private_imux; + int i, idx1; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch(cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + default: + continue; + } + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; + } + return 0; +} + +static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; + unsigned int dac_vol1, dac_vol2; + + if (speaker_nid) { + snd_hda_codec_write(codec, speaker_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } else { + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + } + + dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */ + if (line_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (line_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (hp_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (hp_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (line_nid != 0x16 || hp_nid != 0x16 || + spec->autocfg.line_out_pins[1] != 0x16 || + spec->autocfg.line_out_pins[2] != 0x16) + dac_vol1 = dac_vol2 = AMP_OUT_ZERO; + + snd_hda_codec_write(codec, 0x02, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1); + snd_hda_codec_write(codec, 0x03, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); +} + +/* pcm configuration: identiacal with ALC880 */ +#define alc268_pcm_analog_playback alc880_pcm_analog_playback +#define alc268_pcm_analog_capture alc880_pcm_analog_capture +#define alc268_pcm_digital_playback alc880_pcm_digital_playback + +/* + * BIOS auto configuration + */ +static int alc268_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc268_ignore[] = { 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc268_ignore); + if (err < 0) + return err; + if (!spec->autocfg.line_outs) + return 0; /* can't find valid BIOS pin config */ + + err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = 2; + + /* digital only support output */ + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; + spec->num_mux_defs = 1; + spec->input_mux = &spec->private_imux; + + return 1; +} + +#define alc268_auto_init_multi_out alc882_auto_init_multi_out +#define alc268_auto_init_hp_out alc882_auto_init_hp_out +#define alc268_auto_init_analog_input alc882_auto_init_analog_input + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc268_auto_init(struct hda_codec *codec) +{ + alc268_auto_init_multi_out(codec); + alc268_auto_init_hp_out(codec); + alc268_auto_init_mono_speaker_out(codec); + alc268_auto_init_analog_input(codec); +} + +/* + * configuration and preset + */ +static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC268_3ST] = "3stack", + [ALC268_AUTO] = "auto", +}; + +static struct snd_pci_quirk alc268_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), + {} +}; + +static struct alc_config_preset alc268_presets[] = { + [ALC268_3ST] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .init_verbs = { alc268_base_init_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + }, +}; + +static int patch_alc268(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST, + alc268_models, + alc268_cfg_tbl); + + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC268, " + "trying auto-probe from BIOS...\n"); + board_config = ALC268_AUTO; + } + + if (board_config == ALC268_AUTO) { + /* automatic parse from the BIOS config */ + err = alc268_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC268_3ST; + } + } + + if (board_config != ALC268_AUTO) + setup_preset(spec, &alc268_presets[board_config]); + + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_analog_playback = &alc268_pcm_analog_playback; + spec->stream_analog_capture = &alc268_pcm_analog_capture; + + spec->stream_name_digital = "ALC268 Digital"; + spec->stream_digital_playback = &alc268_pcm_digital_playback; + + if (board_config == ALC268_AUTO) { + if (!spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc268_adc_nids_alt; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids_alt); + spec->mixers[spec->num_mixers] = + alc268_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc268_adc_nids; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids); + spec->mixers[spec->num_mixers] = + alc268_capture_mixer; + spec->num_mixers++; + } + } + } + codec->patch_ops = alc_patch_ops; + if (board_config == ALC268_AUTO) + spec->init_hook = alc268_auto_init; + + return 0; +} + /* * ALC861 channel source setting (2/6 channel selection for 3-stack) */ @@ -10728,6 +11244,7 @@ static int patch_alc662(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, + { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, From 35739bb179386cf52c13d0779a42ecfe18dedf7e Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Wed, 6 Jun 2007 14:48:52 +0200 Subject: [PATCH 028/102] [ALSA] Yet another Uniwill laptop with ALC861 codec From: Dave Jones Rediffed version of the patch from .. http://bugzilla.kernel.org/show_bug.cgi?id=8016 that seems to be lingering for some time. Original patch by: Andy Shevchenko Signed-off-by: Dave Jones Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 49a0fa8ec679..f09fc6e72f4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9291,6 +9291,7 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), From 6f3bf657edf9c592cf9d92e07ab385bb0ede48c5 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 8 Jun 2007 11:39:35 +0200 Subject: [PATCH 029/102] [ALSA] hda-codec - Add quirk for another MSI laptop to alc883_cfg_tbl[] This hg commit adds the SSID of another MSI laptop to the alc883_cfg_tbl[] quirk list. Verified in #alsa/Freenode on Thu, Jun 7, 2007. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 f09fc6e72f4c..a31735d28352 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6388,6 +6388,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), From bc55197d5a19a58c63f826dbd326da70e97b88f8 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Fri, 8 Jun 2007 11:41:34 +0200 Subject: [PATCH 030/102] [ALSA] soc/sh: let SND_SOC_PCM_SH7760 depend on SH_DMABRG Let SND_SOC_PCM_SH7760 depend on SH_DMABRG rather than select'ing it. Gets rid of the Kbuild warning about SH_DMABRG being undefined on non-sh builds. Signed-off-by: Manuel Lauss Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/sh/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a332e51d6887..f03220d23e73 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -2,8 +2,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC - select SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support. From b959d1f83fc9cb707aef86ccb3f65e20f9572428 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jun 2007 12:25:25 +0200 Subject: [PATCH 031/102] [ALSA] hda-codec - Fix Thinkpad X61/T61 outputs Enable EAPD bit on Thinkpad X61/T61. The recent model requires it explicitly to make any sounds. In future, it should be toggled automatically on demand. Also, added a beep volume/switch. Some people like it although I don't understand why :) Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index dff2e7917885..760c897cde7c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2982,6 +2982,8 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3011,6 +3013,8 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* enable EAPD bit */ + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } /* end */ }; From 15cc4458c5c93bd5a616410815d5d165f0fe0900 Mon Sep 17 00:00:00 2001 From: Peer Chen Date: Fri, 8 Jun 2007 13:55:10 +0200 Subject: [PATCH 032/102] [ALSA] Add the MCP73/77 support to hda_intel driver Add the MCP73/MCP77 support to hda driver. The patch base on kernel 2.6.22-rc4 Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2fa281cbef91..74e5593a58e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1788,6 +1788,12 @@ static struct pci_device_id azx_ids[] = { { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ + { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); From 13d457094bc364e942884266036fd1b2ab74308b Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 11 Jun 2007 12:21:20 +0200 Subject: [PATCH 033/102] [ALSA] emu10k1 - EMU 1212 with 16 capture channels * adding 8 more 32-bit capture channels (total of 16) for emu1010 cards * adding some code comments and card details description Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/emu10k1.h | 16 +++++++ sound/pci/emu10k1/emu10k1_main.c | 52 +++++++++++++++++++++ sound/pci/emu10k1/emufx.c | 78 +++++++++++++++++++++++++++++++- sound/pci/emu10k1/emumixer.c | 16 +++++++ sound/pci/emu10k1/emupcm.c | 39 +++++++++++----- 5 files changed, 188 insertions(+), 13 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 23e45a4cf0e4..529d0a564367 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1120,6 +1120,16 @@ /************************************************************************************************/ /* EMU1010m HANA Destinations */ /************************************************************************************************/ +/* 32-bit destinations of signal in the Hana FPGA. Destinations are either + * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture + * - 16 x EMU_DST_ALICE2_EMU32_X. + */ +/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */ +/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture. + * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on + * setup of mixer control for each destination - see emumixer.c - + * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[] + */ #define EMU_DST_ALICE2_EMU32_0 0x000f /* 16 EMU32 channels to Alice2 +0 to +0xf */ #define EMU_DST_ALICE2_EMU32_1 0x0000 /* 16 EMU32 channels to Alice2 +0 to +0xf */ #define EMU_DST_ALICE2_EMU32_2 0x0001 /* 16 EMU32 channels to Alice2 +0 to +0xf */ @@ -1199,6 +1209,12 @@ /************************************************************************************************/ /* EMU1010m HANA Sources */ /************************************************************************************************/ +/* 32-bit sources of signal in the Hana FPGA. The sources are routed to + * destinations using mixer control for each destination - see emumixer.c + * Sources are either physical inputs of FPGA, + * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A + + * 16 x EMU_SRC_ALICE_EMU32B + */ #define EMU_SRC_SILENCE 0x0000 /* Silence */ #define EMU_SRC_DOCK_MIC_A1 0x0100 /* Audio Dock Mic A, 1st or 48kHz only */ #define EMU_SRC_DOCK_MIC_A2 0x0101 /* Audio Dock Mic A, 2nd or 96kHz */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 4a9b59ad8ab1..72367411ee6b 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -694,6 +694,37 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return 0; } +/* + * EMU-1010 - details found out from this driver, official MS Win drivers, + * testing the card: + * + * Audigy2 (aka Alice2): + * --------------------- + * * communication over PCI + * * conversion of 32-bit data coming over EMU32 links from HANA FPGA + * to 2 x 16-bit, using internal DSP instructions + * * slave mode, clock supplied by HANA + * * linked to HANA using: + * 32 x 32-bit serial EMU32 output channels + * 16 x EMU32 input channels + * (?) x I2S I/O channels (?) + * + * FPGA (aka HANA): + * --------------- + * * provides all (?) physical inputs and outputs of the card + * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) + * * provides clock signal for the card and Alice2 + * * two crystals - for 44.1kHz and 48kHz multiples + * * provides internal routing of signal sources to signal destinations + * * inputs/outputs to Alice2 - see above + * + * Current status of the driver: + * ---------------------------- + * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) + * * PCM device nb. 2: + * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops + * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops + */ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) { unsigned int i; @@ -850,6 +881,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1); + /* Pavel Hofman - setting defaults for 8 more capture channels + * Defaults only, users will set their own values anyways, let's + * just copy/paste. + */ + + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1); #endif #if 0 /* Original */ diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index c02012cccd8e..7206c0fa06f2 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1123,6 +1123,11 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF; } +/* + * Used for emu1010 - conversion from 32-bit capture inputs from HANA + * to 2 x 16-bit registers in audigy - their values are read via DMA. + * Conversion is performed by Audigy DSP instructions of FX8010. + */ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( struct snd_emu10k1_fx8010_code *icode, u32 *ptr, int tmp, int bit_shifter16, @@ -1193,7 +1198,11 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP); #if 1 - /* PCM front Playback Volume (independent from stereo mix) */ + /* PCM front Playback Volume (independent from stereo mix) + * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31) + * where gpr contains attenuation from corresponding mixer control + * (snd_emu10k1_init_stereo_control) + */ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100); @@ -1549,7 +1558,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu1010) { snd_printk("EMU inputs on\n"); - /* Capture 8 channels of S32_LE sound */ + /* Capture 16 (originally 8) channels of S32_LE sound */ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ @@ -1560,6 +1569,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); /* Right ADC in 1 of 2 */ gpr_map[gpr++] = 0x00000000; + /* Delaying by one sample: instead of copying the input + * value A_P16VIN to output A_FXBUS2 as in the first channel, + * we use an auxiliary register, delaying the value by one + * sample + */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; @@ -1583,6 +1597,66 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr_map[gpr++] = 0x00000000; snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + /* Pavel Hofman - we still have voices, A_FXBUS2s, and + * A_P16VINs available - + * let's add 8 more capture channels - total of 16 + */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x10)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x12)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x14)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x16)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x18)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1a)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1c)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1e)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), + A_C_00000000, A_C_00000000); #if 0 for (z = 4; z < 8; z++) { diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 4db6e1ca1665..7b2c1dcc5337 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -77,6 +77,10 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol, return 0; } +/* + * Items labels in enum mixer controls assigning source data to + * each destination + */ static char *emu1010_src_texts[] = { "Silence", "Dock Mic A", @@ -133,6 +137,9 @@ static char *emu1010_src_texts[] = { "DSP 31", }; +/* + * List of data sources available for each destination + */ static unsigned int emu1010_src_regs[] = { EMU_SRC_SILENCE,/* 0 */ EMU_SRC_DOCK_MIC_A1, /* 1 */ @@ -189,6 +196,10 @@ static unsigned int emu1010_src_regs[] = { EMU_SRC_ALICE_EMU32B+0xf, /* 52 */ }; +/* + * Data destinations - physical EMU outputs. + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_output_dst[] = { EMU_DST_DOCK_DAC1_LEFT1, /* 0 */ EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */ @@ -216,6 +227,11 @@ static unsigned int emu1010_output_dst[] = { EMU_DST_HANA_ADAT+7, /* 23 */ }; +/* + * Data destinations - HANA outputs going to Alice2 (audigy) for + * capture (EMU32 + I2S links) + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_input_dst[] = { EMU_DST_ALICE2_EMU32_0, EMU_DST_ALICE2_EMU32_1, diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index ab4f5df5241b..eda5cb373ded 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1233,24 +1233,26 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); if (emu->card_capabilities->emu1010) { - /* TODO + /* Nb. of channels has been increased to 16 */ + /* TODO * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 * rate_min = 44100, * rate_max = 192000, - * channels_min = 8, - * channels_max = 8, + * channels_min = 16, + * channels_max = 16, * Need to add mixer control to fix sample rate * - * There are 16 mono channels of 16bits each. + * There are 32 mono channels of 16bits each. * 24bit Audio uses 2x channels over 16bit * 96kHz uses 2x channels over 48kHz * 192kHz uses 4x channels over 48kHz - * So, for 48kHz 24bit, one has 8 channels - * for 96kHz 24bit, one has 4 channels - * for 192kHz 24bit, one has 2 channels + * So, for 48kHz 24bit, one has 16 channels + * for 96kHz 24bit, one has 8 channels + * for 192kHz 24bit, one has 4 channels + * */ #if 1 switch (emu->emu1010.internal_clock) { @@ -1258,13 +1260,15 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) /* For 44.1kHz */ runtime->hw.rates = SNDRV_PCM_RATE_44100; runtime->hw.rate_min = runtime->hw.rate_max = 44100; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; case 1: /* For 48kHz */ runtime->hw.rates = SNDRV_PCM_RATE_48000; runtime->hw.rate_min = runtime->hw.rate_max = 48000; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; }; #endif @@ -1282,7 +1286,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) #endif runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE; /* efx_voices_mask[0] is expected to be zero - * efx_voices_mask[1] is expected to have 16bits set + * efx_voices_mask[1] is expected to have 32bits set */ } else { runtime->hw.channels_min = runtime->hw.channels_max = 0; @@ -1787,11 +1791,24 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; - emu->efx_voices_mask[1] = 0xffff; + if (emu->card_capabilities->emu1010) + /* Pavel Hofman - 32 voices will be used for + * capture (write mode) - + * each bit = corresponding voice + */ + emu->efx_voices_mask[1] = 0xffffffff; + else + emu->efx_voices_mask[1] = 0xffff; } else { emu->efx_voices_mask[0] = 0xffff0000; emu->efx_voices_mask[1] = 0; } + /* For emu1010, the control has to set 32 upper bits (voices) + * out of the 64 bits (voices) to true for the 16-channels capture + * to work correctly. Correct A_FXWC2 initial value (0xffffffff) + * is already defined but the snd_emu10k1_pcm_efx_voices_mask + * control can override this register's value. + */ kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu); if (!kctl) return -ENOMEM; From 1c3973240b94a48766b7bfe743a4fcaad4eab3eb Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Mon, 11 Jun 2007 12:23:31 +0200 Subject: [PATCH 034/102] [ALSA] ali5451 warning fix sound/pci/ali5451/ali5451.c: In function 'snd_ali_prepare': sound/pci/ali5451/ali5451.c:716: warning: 'R2' may be used uninitialized in this function Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ali5451/ali5451.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index e2c8666e06ad..05b4c8696941 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -731,8 +731,7 @@ static void snd_ali_detect_spdif_rate(struct snd_ali *codec) return; } - count = 0; - while (count++ <= 50000) { + for (count = 0; count <= 50000; count++) { snd_ali_delay(codec, 6); bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1)); R2 = bval & 0x1F; From febe3375ea690a6cf544c33fa0fea1a06ff451ee Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 12 Jun 2007 11:27:46 +0200 Subject: [PATCH 035/102] [ALSA] hda-codec - Add HP Pavillion quirk to Realtek code This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 a31735d28352..1950875b694c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6380,6 +6380,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), From a18519e1e4783628182743e6cca5f9dfa6a9e3bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jun 2007 12:58:23 +0200 Subject: [PATCH 036/102] [ALSA] hda-codec - Add model for Toshiba A135 Added model=lenovo for Toshiba A135 with ALC861VD codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 1950875b694c..d2cadb2038da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10006,6 +10006,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), {} }; From 8c87286fdcb67f796cc9477a9bc1a2f1e173d99f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2007 12:11:16 +0200 Subject: [PATCH 037/102] [ALSA] hda-codec - Fix ALC662 auto-configuration code The auto-configuration code didn't initialize the verbs correctly. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2cadb2038da..9c5c3cfa5ec7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11169,12 +11169,10 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - else if (err > 0) - /* hack - override the init verbs */ - spec->init_verbs[0] = alc662_auto_init_verbs; + spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; - return err; + return 1; } /* additional initialization for auto-configuration model */ @@ -11211,7 +11209,7 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) { alc_free(codec); return err; - } else if (err) { + } else if (!err) { printk(KERN_INFO "hda_codec: Cannot set up configuration " "from BIOS. Using base mode...\n"); From 2723f2a46823465f24b91cca0ace1e1389fe3507 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Tue, 19 Jun 2007 12:12:33 +0200 Subject: [PATCH 038/102] [ALSA] ASoC S3C24xx machine drivers - lm4857-h missing patch Somewhere when dealing with the s3c24xx patches and requested changes I managed to miss this file on re-submission. Its a header file required for compilation of neo1973-wm8573.c and should be a no brainer. Signed-off-by: Graeme Gregory Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/lm4857.h | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) create mode 100644 sound/soc/s3c24xx/lm4857.h diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h new file mode 100644 index 000000000000..0cf5b7011d6f --- /dev/null +++ b/sound/soc/s3c24xx/lm4857.h @@ -0,0 +1,32 @@ +/* + * lm4857.h -- ALSA Soc Audio Layer + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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. + * + * Revision history + * 18th Jun 2007 Initial version. + */ + +#ifndef LM4857_H_ +#define LM4857_H_ + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +#endif /*LM4857_H_*/ + From b69f3748400517d1023337b97456a0a1cdc74a79 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 19 Jun 2007 12:13:32 +0200 Subject: [PATCH 039/102] [ALSA] hda-codec - Add Fujitsu Siemens v3515 support to patch_conexant This patch adds support for the Fujitsu Seimens Amilo Pro v3515 notebook. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 bef214bcdddf..90ebc2c49889 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -802,6 +802,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_FUJITSU), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; From 76c08828709129bdce6c6a325e0342ba73f2618f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2007 12:17:42 +0200 Subject: [PATCH 040/102] [ALSA] hda-codec - Add model dell for Dell XPS M1210 Added a new model 'dell' for Dell XPS M1210 with STAC922x codec chip. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_sigmatel.c | 12 ++++++++++++ 2 files changed, 13 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7659e5071c41..a5a1ed20b3c0 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -927,6 +927,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF + dell Dell XPS M1210 intel-mac-v1 Intel Mac Type 1 intel-mac-v2 Intel Mac Type 2 intel-mac-v3 Intel Mac Type 3 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index addcb2a0ae66..36423ca693e7 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -59,6 +59,7 @@ enum { STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, + STAC_922X_DELL, STAC_INTEL_MAC_V1, STAC_INTEL_MAC_V2, STAC_INTEL_MAC_V3, @@ -586,10 +587,17 @@ static unsigned int intel_mac_v5_pin_configs[10] = { 0x400000fc, 0x400000fb, }; +static unsigned int stac922x_dell_pin_configs[10] = { + 0x0221121e, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221122f, 0x03451340, 0x40c003f2, + 0x50a003f3, 0x405003f4 +}; + static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, + [STAC_922X_DELL] = stac922x_dell_pin_configs, [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs, [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs, [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, @@ -608,6 +616,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", + [STAC_922X_DELL] = "dell", [STAC_INTEL_MAC_V1] = "intel-mac-v1", [STAC_INTEL_MAC_V2] = "intel-mac-v2", [STAC_INTEL_MAC_V3] = "intel-mac-v3", @@ -683,6 +692,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* Apple Mac Mini (early 2006) */ SND_PCI_QUIRK(0x8384, 0x7680, "Mac Mini", STAC_INTEL_MAC_V3), + /* Dell */ + SND_PCI_QUIRK(0x1028, 0x01d7, "Dell XPS M1210", STAC_922X_DELL), + {} /* terminator */ }; From 92a22beb756c53985e8ba98d39f2c8fc834557a4 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 19 Jun 2007 16:48:28 +0200 Subject: [PATCH 041/102] [ALSA] hda: add eapd support to additional idt codecs Added support for EAPD on the 927x, and 9227-8 IDT HDA codecs. Enabling EAPD powers the internal speaker amp otherwise there is no sound on systems with an internal amp. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 31 ++++++++++++++++++------------- 1 file changed, 18 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 36423ca693e7..270539a2bbc9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -844,6 +844,21 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) } } +static void stac92xx_enable_eapd(struct hda_codec *codec) +{ + /* Configure GPIO0 as output */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x00000001); + /* Configure GPIO0 as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); + /* Assert GPIO0 high */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, 0x00000001); + /* Enable GPIO0 */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, 0x00000001); +} + /* * Analog playback callbacks */ @@ -2205,7 +2220,8 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; - + stac92xx_enable_eapd(codec); + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2273,18 +2289,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - - /* Configure GPIO0 as EAPD output */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + stac92xx_enable_eapd(codec); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { From e25c05f1d5cf4e332db88b15e9daa1cabd6e17e0 Mon Sep 17 00:00:00 2001 From: Jonathan Phenix Date: Tue, 19 Jun 2007 18:31:28 +0200 Subject: [PATCH 042/102] [ALSA] hda-codec - Output MFG information for HDA devices This patch output information about the MFG (Modem Function Group) to the /proc/asound/card?/codec#? file. This information is important to identify if a HDA device has a modem or not. Signed-off-by: Jonathan Phenix Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_proc.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e313e685f161..ac15066fd300 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -250,6 +250,12 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + + if (codec->mfg) + snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg); + else + snd_iprintf(buffer, "No Modem Function Group found\n"); + if (! codec->afg) return; snd_iprintf(buffer, "Default PCM:\n"); From 0ac8551ea08e655fc6f35874803f091a17dacf90 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2007 15:46:13 +0200 Subject: [PATCH 043/102] [ALSA] hda-codec - Add AD1882 codec support Added AD1882 codec support. It has currently two models, 3stack and 6stack. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 4 + sound/pci/hda/patch_analog.c | 287 +++++++++++++++++- 2 files changed, 289 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index a5a1ed20b3c0..4d47f3f19631 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -874,6 +874,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1882 + 3stack 3-stack mode (default) + 6stack 6-stack mode + AD1884 N/A diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 760c897cde7c..4d7f8d11ad75 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,6 +1,6 @@ /* - * HD audio interface patch for AD1884, AD1981HD, AD1983, AD1984, AD1986A, - * AD1988 + * HD audio interface patch for AD1882, AD1884, AD1981HD, AD1983, AD1984, + * AD1986A, AD1988 * * Copyright (c) 2005-2007 Takashi Iwai * @@ -3113,10 +3113,293 @@ static int patch_ad1984(struct hda_codec *codec) } +/* + * AD1882 + * + * port-A - front hp-out + * port-B - front mic-in + * port-C - rear line-in, shared surr-out (3stack) + * port-D - rear line-out + * port-E - rear mic-in, shared clfe-out (3stack) + * port-F - rear surr-out (6stack) + * port-G - rear clfe-out (6stack) + */ + +static hda_nid_t ad1882_dac_nids[3] = { + 0x04, 0x03, 0x05 +}; + +static hda_nid_t ad1882_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1882_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1882_SPDIF_OUT 0x02 + +/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ +static struct hda_input_mux ad1882_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x1 }, + { "Mic", 0x4 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + { "Mix", 0x7 }, + }, +}; + +static struct snd_kcontrol_new ad1882_base_mixers[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_3stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_6stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb ad1882_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch4_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } /* end */ +}; + +static struct hda_channel_mode ad1882_modes[3] = { + { 2, ad1882_ch2_init }, + { 4, ad1882_ch4_init }, + { 6, ad1882_ch6_init }, +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1882_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-A (HP) mixer */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C (line-in) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C mixer - mute as input */ + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-E (mic-in) pin */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-E mixer - mute as input */ + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-F (surround) */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-G (CLFE) */ + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +/* models */ +enum { + AD1882_3STACK, + AD1882_6STACK, + AD1882_MODELS +}; + +static const char *ad1882_models[AD1986A_MODELS] = { + [AD1882_3STACK] = "3stack", + [AD1882_6STACK] = "6stack", +}; + + +static int patch_ad1882(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 6; + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = ad1882_dac_nids; + spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); + spec->adc_nids = ad1882_adc_nids; + spec->capsrc_nids = ad1882_capsrc_nids; + spec->input_mux = &ad1882_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1882_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1882_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, AD1882_MODELS, + ad1882_models, NULL); + switch (board_config) { + default: + case AD1882_3STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_3stack_mixers; + spec->channel_mode = ad1882_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + break; + case AD1882_6STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_6stack_mixers; + break; + } + return 0; +} + + /* * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, From bdec0c728256e1a3739820e244a5138bcfae5a91 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 25 Jun 2007 12:07:38 +0200 Subject: [PATCH 044/102] [ALSA] fix section mismatch Fix shared init/exit function attributes: WARNING: sound/built-in.o(.exit.text+0x4a1): Section mismatch: reference to .init.text: (between 'alsa_card_virmidi_exit' and 'alsa_card_serial_exit') WARNING: sound/built-in.o(.exit.text+0x4c1): Section mismatch: reference to .init.text: (between 'alsa_card_serial_exit' and 'ac97_bus_exit') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/drivers/serial-u16550.c | 2 +- sound/drivers/virmidi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 838a4277929d..d3e6a20edd38 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -998,7 +998,7 @@ static struct platform_driver snd_serial_driver = { }, }; -static void __init_or_module snd_serial_unregister_all(void) +static void snd_serial_unregister_all(void) { int i; diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 46f3d3486067..915c86773c21 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -145,7 +145,7 @@ static struct platform_driver snd_virmidi_driver = { }, }; -static void __init_or_module snd_virmidi_unregister_all(void) +static void snd_virmidi_unregister_all(void) { int i; From c12aad6efbee1d937438f1deabbef695add0628b Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 25 Jun 2007 12:08:01 +0200 Subject: [PATCH 045/102] [ALSA] more section mismatches Something about __init_or_module isn't working as expected (?). CONFIG_HOTPLUG=y CONFIG_MODULES=n Fix shared init/exit code helper: WARNING: sound/built-in.o(.exit.text+0x243): Section mismatch: reference to .init.text: (between 'alsa_card_mpu401_exit' and 'ac97_bus_exit') WARNING: sound/built-in.o(.exit.text+0x21b): Section mismatch: reference to .init.text: (between 'alsa_card_dummy_exit' and 'alsa_card_serial_exit') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/drivers/dummy.c | 2 +- sound/drivers/mpu401/mpu401.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a0f28f51fc7e..4360ae9de19c 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -659,7 +659,7 @@ static struct platform_driver snd_dummy_driver = { }, }; -static void __init_or_module snd_dummy_unregister_all(void) +static void snd_dummy_unregister_all(void) { int i; diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 1d563e515c17..67c6e9745418 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -228,7 +228,7 @@ static struct pnp_driver snd_mpu401_pnp_driver = { static struct pnp_driver snd_mpu401_pnp_driver; #endif -static void __init_or_module snd_mpu401_unregister_all(void) +static void snd_mpu401_unregister_all(void) { int i; From e65365de5ba280e058bd6b8b80c8790253268887 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jun 2007 12:09:32 +0200 Subject: [PATCH 046/102] [ALSA] Fix invalid schedule_timeout_interruptible() Fixed the invalid use of schedule_timeout_interruptible() without checking pending signals. Simply replaced with schedule_timeout(). Suggestions thanks to Jeff Garzik. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/core/seq/seq_instr.c | 6 +++--- sound/isa/ad1848/ad1848_lib.c | 4 ++-- sound/isa/sscape.c | 4 ++-- sound/isa/wavefront/wavefront_synth.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/via82xx.c | 4 ++-- sound/pci/via82xx_modem.c | 4 ++-- 7 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c index f30d171b6d96..5efe6523a589 100644 --- a/sound/core/seq/seq_instr.c +++ b/sound/core/seq/seq_instr.c @@ -109,7 +109,7 @@ void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr) spin_lock_irqsave(&list->lock, flags); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); @@ -199,7 +199,7 @@ int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list, instr = flist; flist = instr->next; while (instr->use) - schedule_timeout_interruptible(1); + schedule_timeout(1); if (snd_seq_instr_free(instr, atomic)<0) snd_printk(KERN_WARNING "instrument free problem\n"); instr = next; @@ -555,7 +555,7 @@ static int instr_free(struct snd_seq_kinstr_ops *ops, SNDRV_SEQ_INSTR_NOTIFY_REMOVE); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 8094282c2ae1..1bc2e3fd5721 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -245,7 +245,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) snd_printk(KERN_ERR "mce_down - auto calibration time out (2)\n"); return; } - time = schedule_timeout_interruptible(time); + time = schedule_timeout(time); spin_lock_irqsave(&chip->reg_lock, flags); } #if 0 @@ -258,7 +258,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); return; } - time = schedule_timeout_interruptible(time); + time = schedule_timeout(time); spin_lock_irqsave(&chip->reg_lock, flags); } spin_unlock_irqrestore(&chip->reg_lock, flags); diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 9ea417bcf3e5..cbad2a51cbaa 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -382,7 +382,7 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) unsigned long flags; unsigned char x; - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); @@ -409,7 +409,7 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) unsigned long flags; unsigned char x; - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 78020d832e04..bacc51c86587 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1780,7 +1780,7 @@ wavefront_should_cause_interrupt (snd_wavefront_t *dev, outb (val,port); spin_unlock_irq(&dev->irq_lock); while (1) { - if ((timeout = schedule_timeout_interruptible(timeout)) == 0) + if ((timeout = schedule_timeout(timeout)) == 0) return; if (dev->irq_ok) return; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 74e5593a58e1..c78ff901a572 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -538,7 +538,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (! chip->rirb.cmds) return chip->rirb.res; /* the last value */ - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_after_eq(timeout, jiffies)); if (chip->msi) { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 50c9f92cfd1b..6ea09df0c73a 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2098,7 +2098,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -2117,7 +2117,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 8cbf8eba4ae9..72425e73abae 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -983,7 +983,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -1001,7 +1001,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ From 6963f84c428ecd31780c97224004c2462b0664f7 Mon Sep 17 00:00:00 2001 From: Mike Crash Date: Mon, 25 Jun 2007 12:12:51 +0200 Subject: [PATCH 047/102] [ALSA] hda: support for S/PDIF out on ASUS M2V Added support for S/PDIF digital output on ASUS M2V motheboard - added new model '3stack-660-digout' and ALC660VD_3ST_DIG Signed-off-by: Mike Crash Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 17 ++++++++++++++++- 2 files changed, 17 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4d47f3f19631..b967dc850c60 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -863,6 +863,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-dig 3-jack with SPDIF OUT 6stack-dig 6-jack with SPDIF OUT 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c5c3cfa5ec7..4bff20f76b08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -122,6 +122,7 @@ enum { /* ALC861-VD models */ enum { ALC660VD_3ST, + ALC660VD_3ST_DIG, ALC861VD_3ST, ALC861VD_3ST_DIG, ALC861VD_6ST_DIG, @@ -9987,6 +9988,7 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re */ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", + [ALC660VD_3ST_DIG]= "3stack-660-digout", [ALC861VD_3ST] = "3stack", [ALC861VD_3ST_DIG] = "3stack-digout", [ALC861VD_6ST_DIG] = "6stack-digout", @@ -9998,7 +10000,7 @@ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), @@ -10023,6 +10025,19 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, }, + [ALC660VD_3ST_DIG] = { + .mixers = { alc861vd_3st_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + }, [ALC861VD_3ST] = { .mixers = { alc861vd_3st_mixer }, .init_verbs = { alc861vd_volume_init_verbs, From f8f794ab27dd3c6d9965ec5434775ead23d3d7a2 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Mon, 25 Jun 2007 12:14:25 +0200 Subject: [PATCH 048/102] [ALSA] HDA - Change default configuration for Fujitsu Si3515 This patch changes the default configuration for the Fujitsu Siemens Si3515 Laptop to the correct model=laptop. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 90ebc2c49889..0ffcf2001d28 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -802,7 +802,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; From 076c0e4fd80a43f64710fe9aabdc4c463b7f8621 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 26 Jun 2007 11:43:52 +0200 Subject: [PATCH 049/102] [ALSA] fix ensoniq section mismatch Fix ensoniq driver sections when HOTPLUG=n, but code is used during resume, so it cannot be discarded: WARNING: sound/pci/snd-ens1371.o(.text+0x5c0): Section mismatch: reference to .init.text: (between 'snd_ensoniq_chip_init' and 'snd_ensoniq_free') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ens1370.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 7c403965153b..21cb4268a59b 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1607,8 +1607,8 @@ struct es1371_quirk { unsigned char rev; /* revision */ }; -static int __devinit es1371_quirk_lookup(struct ensoniq *ensoniq, - struct es1371_quirk *list) +static int es1371_quirk_lookup(struct ensoniq *ensoniq, + struct es1371_quirk *list) { while (list->vid != (unsigned short)PCI_ANY_ID) { if (ensoniq->pci->vendor == list->vid && From 3c2b576d58b4542b1adc56cbfb92dbfa5eef89ee Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 27 Jun 2007 11:52:39 +0200 Subject: [PATCH 050/102] [ALSA] portman2x4 section mismatch FIx section mismatch when CONFIG_HOTPLUG=n: WARNING: sound/built-in.o(.exit.text+0x271): Section mismatch: reference to .init.text:snd_p ortman_unregister_all (between 'snd_portman_module_exit' and 'alsa_mpu401_uart_exit') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/drivers/portman2x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 497cafb57d9b..0eb9b5cebfcd 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -833,7 +833,7 @@ static struct platform_driver snd_portman_driver = { /********************************************************************* * module init stuff *********************************************************************/ -static void __init_or_module snd_portman_unregister_all(void) +static void snd_portman_unregister_all(void) { int i; From d10ad1a26da03fe089c02846a95fa5938c3ca3ce Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Mon, 2 Jul 2007 16:03:12 +0200 Subject: [PATCH 051/102] [ALSA] sound/pci/hda/patch_realtek.c: remove dead code This patch removes some dead code spotted by the Coverity checker. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bff20f76b08..5367d1377e11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11182,8 +11182,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - if (err < 0) - return err; spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; From df802952b723cdb08f2f8a3cba396c357290850b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jul 2007 19:18:00 +0200 Subject: [PATCH 052/102] [ALSA] hda-codec - Fix the error in DAC assignment of STAC codec The STAC codec support code has the automatic recognition of additional side-output pins, and this may result in an error when the assigned pin has no corresponding DAC connection. The patch fixes the problem -- it corrects the line_outs again without such additional pins. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 270539a2bbc9..e89180c6aaa2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1228,7 +1228,7 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) * and 9202/925x. For those, dac_nids[] must be hard-coded. */ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) + struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; int i, j, conn_len = 0; @@ -1253,6 +1253,13 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, } if (j == conn_len) { + if (spec->multiout.num_dacs > 0) { + /* we have already working output pins, + * so let's drop the broken ones again + */ + cfg->line_outs = spec->multiout.num_dacs; + break; + } /* error out, no available DAC found */ snd_printk(KERN_ERR "%s: No available DAC for pin 0x%x\n", From 87823090a8bd373f34bdb6f1d5cdc5f2f4da4905 Mon Sep 17 00:00:00 2001 From: Carlo Beccaria Date: Tue, 3 Jul 2007 08:04:25 +0200 Subject: [PATCH 053/102] [ALSA] usb-audio: add Roland SH-201 support Add quirks to detect the Roland SH-201. Signed-off-by: Carlo Beccaria Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 59f07e874e52..9274baf69901 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1239,6 +1239,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, /* TODO: add Edirol MD-P1 support */ +{ + /* Roland SH-201 */ + USB_DEVICE(0x0582, 0x00ad), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "SH-201", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { From c454fd4e888dc2b1423b6a65106a619e99a2deb4 Mon Sep 17 00:00:00 2001 From: Masakazu Mokuno Date: Tue, 3 Jul 2007 12:54:49 +0200 Subject: [PATCH 054/102] [ALSA] Add PS3 sound driver The patch adds the support for the sound feature of PS3 - the digital output of HDMI and SPDIF optical, and the analog output of AV multi. Signed-off-by: Masakazu Mokuno Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/ppc/Kconfig | 20 + sound/ppc/Makefile | 3 +- sound/ppc/snd_ps3.c | 1125 +++++++++++++++++++++++++++++++++++++++ sound/ppc/snd_ps3.h | 135 +++++ sound/ppc/snd_ps3_reg.h | 891 +++++++++++++++++++++++++++++++ 5 files changed, 2173 insertions(+), 1 deletion(-) create mode 100644 sound/ppc/snd_ps3.c create mode 100644 sound/ppc/snd_ps3.h create mode 100644 sound/ppc/snd_ps3_reg.h diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index a3fb1496e4dc..cacb0b136883 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -33,3 +33,23 @@ config SND_POWERMAC_AUTO_DRC option. endmenu + +menu "ALSA PowerPC devices" + depends on SND!=n && ( PPC64 || PPC32 ) + +config SND_PS3 + tristate "PS3 Audio support" + depends on SND && PS3_PS3AV + select SND_PCM + default m + help + Say Y here to include support for audio on the PS3 + + To compile this driver as a module, choose M here: the module + will be called snd_ps3. + +config SND_PS3_DEFAULT_START_DELAY + int "Startup delay time in ms" + depends on SND_PS3 + default "2000" +endmenu diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile index 4d95c652c8ca..eacee2d0675c 100644 --- a/sound/ppc/Makefile +++ b/sound/ppc/Makefile @@ -6,4 +6,5 @@ snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o # Toplevel Module Dependency -obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o +obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o +obj-$(CONFIG_SND_PS3) += snd_ps3.o diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c new file mode 100644 index 000000000000..1aa0b467599f --- /dev/null +++ b/sound/ppc/snd_ps3.c @@ -0,0 +1,1125 @@ +/* + * Audio support for PS3 + * Copyright (C) 2007 Sony Computer Entertainment Inc. + * All rights reserved. + * Copyright 2006, 2007 Sony Corporation + * + * 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 Licence. + * + * 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 + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "snd_ps3_reg.h" +#include "snd_ps3.h" + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("PS3 sound driver"); +MODULE_AUTHOR("Sony Computer Entertainment Inc."); + +/* module entries */ +static int __init snd_ps3_init(void); +static void __exit snd_ps3_exit(void); + +/* ALSA snd driver ops */ +static int snd_ps3_pcm_open(struct snd_pcm_substream *substream); +static int snd_ps3_pcm_close(struct snd_pcm_substream *substream); +static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream); +static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, + int cmd); +static snd_pcm_uframes_t snd_ps3_pcm_pointer(struct snd_pcm_substream + *substream); +static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream); + + +/* ps3_system_bus_driver entries */ +static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev); +static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev); + +/* address setup */ +static int snd_ps3_map_mmio(void); +static void snd_ps3_unmap_mmio(void); +static int snd_ps3_allocate_irq(void); +static void snd_ps3_free_irq(void); +static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start); + +/* interrupt handler */ +static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id); + + +/* set sampling rate/format */ +static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream); +/* take effect parameter change */ +static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card); +/* initialize avsetting and take it effect */ +static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card); +/* setup dma */ +static int snd_ps3_program_dma(struct snd_ps3_card_info *card, + enum snd_ps3_dma_filltype filltype); +static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card); + +static dma_addr_t v_to_bus(struct snd_ps3_card_info *, void *vaddr, int ch); + + +module_init(snd_ps3_init); +module_exit(snd_ps3_exit); + +/* + * global + */ +static struct snd_ps3_card_info the_card; + +static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY; + +module_param_named(start_delay, snd_ps3_start_delay, uint, 0644); +MODULE_PARM_DESC(start_delay, "time to insert silent data in milisec"); + +static int index = SNDRV_DEFAULT_IDX1; +static char *id = SNDRV_DEFAULT_STR1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for PS3 soundchip."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for PS3 soundchip."); + + +/* + * PS3 audio register access + */ +static inline u32 read_reg(unsigned int reg) +{ + return in_be32(the_card.mapped_mmio_vaddr + reg); +} +static inline void write_reg(unsigned int reg, u32 val) +{ + out_be32(the_card.mapped_mmio_vaddr + reg, val); +} +static inline void update_reg(unsigned int reg, u32 or_val) +{ + u32 newval = read_reg(reg) | or_val; + write_reg(reg, newval); +} +static inline void update_mask_reg(unsigned int reg, u32 mask, u32 or_val) +{ + u32 newval = (read_reg(reg) & mask) | or_val; + write_reg(reg, newval); +} + +/* + * ALSA defs + */ +const static struct snd_pcm_hardware snd_ps3_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_NONINTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE), + .rates = (SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000), + .rate_min = 44100, + .rate_max = 96000, + + .channels_min = 2, /* stereo only */ + .channels_max = 2, + + .buffer_bytes_max = PS3_AUDIO_FIFO_SIZE * 64, + + /* interrupt by four stages */ + .period_bytes_min = PS3_AUDIO_FIFO_STAGE_SIZE * 4, + .period_bytes_max = PS3_AUDIO_FIFO_STAGE_SIZE * 4, + + .periods_min = 16, + .periods_max = 32, /* buffer_size_max/ period_bytes_max */ + + .fifo_size = PS3_AUDIO_FIFO_SIZE +}; + +static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = +{ + .open = snd_ps3_pcm_open, + .close = snd_ps3_pcm_close, + .prepare = snd_ps3_pcm_prepare, + .ioctl = snd_pcm_lib_ioctl, + .trigger = snd_ps3_pcm_trigger, + .pointer = snd_ps3_pcm_pointer, + .hw_params = snd_ps3_pcm_hw_params, + .hw_free = snd_ps3_pcm_hw_free +}; + +static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card, + int count, int force_stop) +{ + int dma_ch, done, retries, stop_forced = 0; + uint32_t status; + + for (dma_ch = 0; dma_ch < 8; dma_ch ++) { + retries = count; + do { + status = read_reg(PS3_AUDIO_KICK(dma_ch)) & + PS3_AUDIO_KICK_STATUS_MASK; + switch (status) { + case PS3_AUDIO_KICK_STATUS_DONE: + case PS3_AUDIO_KICK_STATUS_NOTIFY: + case PS3_AUDIO_KICK_STATUS_CLEAR: + case PS3_AUDIO_KICK_STATUS_ERROR: + done = 1; + break; + default: + done = 0; + udelay(10); + } + } while (!done && --retries); + if (!retries && force_stop) { + pr_info("%s: DMA ch %d is not stopped.", + __func__, dma_ch); + /* last resort. force to stop dma. + * NOTE: this cause DMA done interrupts + */ + update_reg(PS3_AUDIO_CONFIG, PS3_AUDIO_CONFIG_CLEAR); + stop_forced = 1; + } + } + return stop_forced; +} + +/* + * wait for all dma is done. + * NOTE: caller should reset card->running before call. + * If not, the interrupt handler will re-start DMA, + * then DMA is never stopped. + */ +static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card) +{ + int stop_forced; + /* + * wait for the last dma is done + */ + + /* + * expected maximum DMA done time is 5.7ms + something (DMA itself). + * 5.7ms is from 16bit/sample 2ch 44.1Khz; the time next + * DMA kick event would occur. + */ + stop_forced = snd_ps3_verify_dma_stop(card, 700, 1); + + /* + * clear outstanding interrupts. + */ + update_reg(PS3_AUDIO_INTR_0, 0); + update_reg(PS3_AUDIO_AX_IS, 0); + + /* + *revert CLEAR bit since it will not reset automatically after DMA stop + */ + if (stop_forced) + update_mask_reg(PS3_AUDIO_CONFIG, ~PS3_AUDIO_CONFIG_CLEAR, 0); + /* ensure the hardware sees changes */ + wmb(); +} + +static void snd_ps3_kick_dma(struct snd_ps3_card_info *card) +{ + + update_reg(PS3_AUDIO_KICK(0), PS3_AUDIO_KICK_REQUEST); + /* ensure the hardware sees the change */ + wmb(); +} + +/* + * convert virtual addr to ioif bus addr. + */ +static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, + void * paddr, + int ch) +{ + return card->dma_start_bus_addr[ch] + + (paddr - card->dma_start_vaddr[ch]); +}; + + +/* + * increment ring buffer pointer. + * NOTE: caller must hold write spinlock + */ +static void snd_ps3_bump_buffer(struct snd_ps3_card_info *card, + enum snd_ps3_ch ch, size_t byte_count, + int stage) +{ + if (!stage) + card->dma_last_transfer_vaddr[ch] = + card->dma_next_transfer_vaddr[ch]; + card->dma_next_transfer_vaddr[ch] += byte_count; + if ((card->dma_start_vaddr[ch] + (card->dma_buffer_size / 2)) <= + card->dma_next_transfer_vaddr[ch]) { + card->dma_next_transfer_vaddr[ch] = card->dma_start_vaddr[ch]; + } +} +/* + * setup dmac to send data to audio and attenuate samples on the ring buffer + */ +static int snd_ps3_program_dma(struct snd_ps3_card_info *card, + enum snd_ps3_dma_filltype filltype) +{ + /* this dmac does not support over 4G */ + uint32_t dma_addr; + int fill_stages, dma_ch, stage; + enum snd_ps3_ch ch; + uint32_t ch0_kick_event = 0; /* initialize to mute gcc */ + void *start_vaddr; + unsigned long irqsave; + int silent = 0; + + switch (filltype) { + case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL: + silent = 1; + /* intentionally fall thru */ + case SND_PS3_DMA_FILLTYPE_FIRSTFILL: + ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS; + break; + + case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING: + silent = 1; + /* intentionally fall thru */ + case SND_PS3_DMA_FILLTYPE_RUNNING: + ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY; + break; + } + + snd_ps3_verify_dma_stop(card, 700, 0); + fill_stages = 4; + spin_lock_irqsave(&card->dma_lock, irqsave); + for (ch = 0; ch < 2; ch++) { + start_vaddr = card->dma_next_transfer_vaddr[0]; + for (stage = 0; stage < fill_stages; stage ++) { + dma_ch = stage * 2 + ch; + if (silent) + dma_addr = card->null_buffer_start_dma_addr; + else + dma_addr = + v_to_bus(card, + card->dma_next_transfer_vaddr[ch], + ch); + + write_reg(PS3_AUDIO_SOURCE(dma_ch), + (PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY | + dma_addr)); + + /* dst: fixed to 3wire#0 */ + if (ch == 0) + write_reg(PS3_AUDIO_DEST(dma_ch), + (PS3_AUDIO_DEST_TARGET_AUDIOFIFO | + PS3_AUDIO_AO_3W_LDATA(0))); + else + write_reg(PS3_AUDIO_DEST(dma_ch), + (PS3_AUDIO_DEST_TARGET_AUDIOFIFO | + PS3_AUDIO_AO_3W_RDATA(0))); + + /* count always 1 DMA block (1/2 stage = 128 bytes) */ + write_reg(PS3_AUDIO_DMASIZE(dma_ch), 0); + /* bump pointer if needed */ + if (!silent) + snd_ps3_bump_buffer(card, ch, + PS3_AUDIO_DMAC_BLOCK_SIZE, + stage); + + /* kick event */ + if (dma_ch == 0) + write_reg(PS3_AUDIO_KICK(dma_ch), + ch0_kick_event); + else + write_reg(PS3_AUDIO_KICK(dma_ch), + PS3_AUDIO_KICK_EVENT_AUDIO_DMA(dma_ch + - 1) | + PS3_AUDIO_KICK_REQUEST); + } + } + /* ensure the hardware sees the change */ + wmb(); + spin_unlock_irqrestore(&card->dma_lock, irqsave); + + return 0; +} + +/* + * audio mute on/off + * mute_on : 0 output enabled + * 1 mute + */ +static int snd_ps3_mute(int mute_on) +{ + return ps3av_audio_mute(mute_on); +} + +/* + * PCM operators + */ +static int snd_ps3_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + int pcm_index; + + pcm_index = substream->pcm->device; + /* to retrieve substream/runtime in interrupt handler */ + card->substream = substream; + + runtime->hw = snd_ps3_pcm_hw; + + card->start_delay = snd_ps3_start_delay; + + /* mute off */ + snd_ps3_mute(0); /* this function sleep */ + + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + PS3_AUDIO_FIFO_STAGE_SIZE * 4 * 2); + return 0; +}; + +static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + size_t size; + + /* alloc transport buffer */ + size = params_buffer_bytes(hw_params); + snd_pcm_lib_malloc_pages(substream, size); + return 0; +}; + +static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, + unsigned int delay_ms) +{ + int ret; + int rate ; + + rate = substream->runtime->rate; + ret = snd_pcm_format_size(substream->runtime->format, + rate * delay_ms / 1000) + * substream->runtime->channels; + + pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", + __func__, + delay_ms, + rate, + snd_pcm_format_size(substream->runtime->format, rate), + rate * delay_ms / 1000, + ret); + + return ret; +}; + +static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + unsigned long irqsave; + + if (!snd_ps3_set_avsetting(substream)) { + /* some parameter changed */ + write_reg(PS3_AUDIO_AX_IE, + PS3_AUDIO_AX_IE_ASOBEIE(0) | + PS3_AUDIO_AX_IE_ASOBUIE(0)); + /* + * let SPDIF device re-lock with SPDIF signal, + * start with some silence + */ + card->silent = snd_ps3_delay_to_bytes(substream, + card->start_delay) / + (PS3_AUDIO_FIFO_STAGE_SIZE * 4); /* every 4 times */ + } + + /* restart ring buffer pointer */ + spin_lock_irqsave(&card->dma_lock, irqsave); + { + card->dma_buffer_size = runtime->dma_bytes; + + card->dma_last_transfer_vaddr[SND_PS3_CH_L] = + card->dma_next_transfer_vaddr[SND_PS3_CH_L] = + card->dma_start_vaddr[SND_PS3_CH_L] = + runtime->dma_area; + card->dma_start_bus_addr[SND_PS3_CH_L] = runtime->dma_addr; + + card->dma_last_transfer_vaddr[SND_PS3_CH_R] = + card->dma_next_transfer_vaddr[SND_PS3_CH_R] = + card->dma_start_vaddr[SND_PS3_CH_R] = + runtime->dma_area + (runtime->dma_bytes / 2); + card->dma_start_bus_addr[SND_PS3_CH_R] = + runtime->dma_addr + (runtime->dma_bytes / 2); + + pr_debug("%s: vaddr=%p bus=%#lx\n", __func__, + card->dma_start_vaddr[SND_PS3_CH_L], + card->dma_start_bus_addr[SND_PS3_CH_L]); + + } + spin_unlock_irqrestore(&card->dma_lock, irqsave); + + /* ensure the hardware sees the change */ + mb(); + + return 0; +}; + +static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* clear outstanding interrupts */ + update_reg(PS3_AUDIO_AX_IS, 0); + + spin_lock(&card->dma_lock); + { + card->running = 1; + } + spin_unlock(&card->dma_lock); + + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + while (read_reg(PS3_AUDIO_KICK(7)) & + PS3_AUDIO_KICK_STATUS_MASK) { + udelay(1); + } + snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); + snd_ps3_kick_dma(card); + break; + + case SNDRV_PCM_TRIGGER_STOP: + spin_lock(&card->dma_lock); + { + card->running = 0; + } + spin_unlock(&card->dma_lock); + snd_ps3_wait_for_dma_stop(card); + break; + default: + break; + + } + + return ret; +}; + +/* + * report current pointer + */ +static snd_pcm_uframes_t snd_ps3_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + size_t bytes; + snd_pcm_uframes_t ret; + + spin_lock(&card->dma_lock); + { + bytes = (size_t)(card->dma_last_transfer_vaddr[SND_PS3_CH_L] - + card->dma_start_vaddr[SND_PS3_CH_L]); + } + spin_unlock(&card->dma_lock); + + ret = bytes_to_frames(substream->runtime, bytes * 2); + + return ret; +}; + +static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream) +{ + int ret; + ret = snd_pcm_lib_free_pages(substream); + return ret; +}; + +static int snd_ps3_pcm_close(struct snd_pcm_substream *substream) +{ + /* mute on */ + snd_ps3_mute(1); + return 0; +}; + +static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card) +{ + /* + * avsetting driver seems to never change the followings + * so, init them here once + */ + + /* no dma interrupt needed */ + write_reg(PS3_AUDIO_INTR_EN_0, 0); + + /* use every 4 buffer empty interrupt */ + update_mask_reg(PS3_AUDIO_AX_IC, + PS3_AUDIO_AX_IC_AASOIMD_MASK, + PS3_AUDIO_AX_IC_AASOIMD_EVERY4); + + /* enable 3wire clocks */ + update_mask_reg(PS3_AUDIO_AO_3WMCTRL, + ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED | + PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED), + 0); + update_reg(PS3_AUDIO_AO_3WMCTRL, + PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT); +} + +/* + * av setting + * NOTE: calling this function may generate audio interrupt. + */ +static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) +{ + int ret, retries, i; + pr_debug("%s: start\n", __func__); + + ret = ps3av_set_audio_mode(card->avs.avs_audio_ch, + card->avs.avs_audio_rate, + card->avs.avs_audio_width, + card->avs.avs_audio_format, + card->avs.avs_audio_source); + /* + * Reset the following unwanted settings: + */ + + /* disable all 3wire buffers */ + update_mask_reg(PS3_AUDIO_AO_3WMCTRL, + ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(1) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), + 0); + wmb(); /* ensure the hardware sees the change */ + /* wait for actually stopped */ + retries = 1000; + while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & + (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(1) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(2) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) && + --retries) { + udelay(1); + } + + /* reset buffer pointer */ + for (i = 0; i < 4; i++) { + update_reg(PS3_AUDIO_AO_3WCTRL(i), + PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET); + udelay(10); + } + wmb(); /* ensure the hardware actually start resetting */ + + /* enable 3wire#0 buffer */ + update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0)); + + + /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */ + update_mask_reg(PS3_AUDIO_AO_3WCTRL(0), + ~PS3_AUDIO_AO_3WCTRL_ASODF, + PS3_AUDIO_AO_3WCTRL_ASODF_LSB); + update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0), + ~PS3_AUDIO_AO_SPDCTRL_SPODF, + PS3_AUDIO_AO_SPDCTRL_SPODF_LSB); + /* ensure all the setting above is written back to register */ + wmb(); + /* avsetting driver altered AX_IE, caller must reset it if you want */ + pr_debug("%s: end\n", __func__); + return ret; +} + +static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) +{ + int ret; + pr_debug("%s: start\n", __func__); + card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2; + card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; + card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; + card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; + card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; + + ret = snd_ps3_change_avsetting(card); + + snd_ps3_audio_fixup(card); + + /* to start to generate SPDIF signal, fill data */ + snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + pr_debug("%s: end\n", __func__); + return ret; +} + +/* + * set sampling rate according to the substream + */ +static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) +{ + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + struct snd_ps3_avsetting_info avs; + + avs = card->avs; + + pr_debug("%s: called freq=%d width=%d\n", __func__, + substream->runtime->rate, + snd_pcm_format_width(substream->runtime->format)); + + pr_debug("%s: before freq=%d width=%d\n", __func__, + card->avs.avs_audio_rate, card->avs.avs_audio_width); + + /* sample rate */ + switch (substream->runtime->rate) { + case 44100: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K; + break; + case 48000: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; + break; + case 88200: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K; + break; + case 96000: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K; + break; + default: + pr_info("%s: invalid rate %d\n", __func__, + substream->runtime->rate); + return 1; + } + + /* width */ + switch (snd_pcm_format_width(substream->runtime->format)) { + case 16: + avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; + break; + case 24: + avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24; + break; + default: + pr_info("%s: invalid width %d\n", __func__, + snd_pcm_format_width(substream->runtime->format)); + return 1; + } + + if ((card->avs.avs_audio_width != avs.avs_audio_width) || + (card->avs.avs_audio_rate != avs.avs_audio_rate)) { + card->avs = avs; + snd_ps3_change_avsetting(card); + + pr_debug("%s: after freq=%d width=%d\n", __func__, + card->avs.avs_audio_rate, card->avs.avs_audio_width); + + return 0; + } else + return 1; +} + + + +static int snd_ps3_map_mmio(void) +{ + the_card.mapped_mmio_vaddr = + ioremap(the_card.ps3_dev->m_region->bus_addr, + the_card.ps3_dev->m_region->len); + + if (!the_card.mapped_mmio_vaddr) { + pr_info("%s: ioremap 0 failed p=%#lx l=%#lx \n", + __func__, the_card.ps3_dev->m_region->lpar_addr, + the_card.ps3_dev->m_region->len); + return -ENXIO; + } + + return 0; +}; + +static void snd_ps3_unmap_mmio(void) +{ + iounmap(the_card.mapped_mmio_vaddr); + the_card.mapped_mmio_vaddr = NULL; +} + +static int snd_ps3_allocate_irq(void) +{ + int ret; + u64 lpar_addr, lpar_size; + u64 __iomem *mapped; + + /* FIXME: move this to device_init (H/W probe) */ + + /* get irq outlet */ + ret = lv1_gpu_device_map(1, &lpar_addr, &lpar_size); + if (ret) { + pr_info("%s: device map 1 failed %d\n", __func__, + ret); + return -ENXIO; + } + + mapped = ioremap(lpar_addr, lpar_size); + if (!mapped) { + pr_info("%s: ioremap 1 failed \n", __func__); + return -ENXIO; + } + + the_card.audio_irq_outlet = in_be64(mapped); + + iounmap(mapped); + ret = lv1_gpu_device_unmap(1); + if (ret) + pr_info("%s: unmap 1 failed\n", __func__); + + /* irq */ + ret = ps3_irq_plug_setup(PS3_BINDING_CPU_ANY, + the_card.audio_irq_outlet, + &the_card.irq_no); + if (ret) { + pr_info("%s:ps3_alloc_irq failed (%d)\n", __func__, ret); + return ret; + } + + ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED, + SND_PS3_DRIVER_NAME, &the_card); + if (ret) { + pr_info("%s: request_irq failed (%d)\n", __func__, ret); + goto cleanup_irq; + } + + return 0; + + cleanup_irq: + ps3_irq_plug_destroy(the_card.irq_no); + return ret; +}; + +static void snd_ps3_free_irq(void) +{ + free_irq(the_card.irq_no, &the_card); + ps3_irq_plug_destroy(the_card.irq_no); +} + +static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) +{ + uint64_t val; + int ret; + + val = (ioaddr_start & (0x0fUL << 32)) >> (32 - 20) | + (0x03UL << 24) | + (0x0fUL << 12) | + (PS3_AUDIO_IOID); + + ret = lv1_gpu_attribute(0x100, 0x007, val, 0, 0); + if (ret) + pr_info("%s: gpu_attribute failed %d\n", __func__, + ret); +} + +static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) +{ + int ret; + u64 lpar_addr, lpar_size; + + BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1)); + BUG_ON(dev->match_id != PS3_MATCH_ID_SOUND); + + the_card.ps3_dev = dev; + + ret = ps3_open_hv_device(dev); + + if (ret) + return -ENXIO; + + /* setup MMIO */ + ret = lv1_gpu_device_map(2, &lpar_addr, &lpar_size); + if (ret) { + pr_info("%s: device map 2 failed %d\n", __func__, ret); + goto clean_open; + } + ps3_mmio_region_init(dev, dev->m_region, lpar_addr, lpar_size, + PAGE_SHIFT); + + ret = snd_ps3_map_mmio(); + if (ret) + goto clean_dev_map; + + /* setup DMA area */ + ps3_dma_region_init(dev, dev->d_region, + PAGE_SHIFT, /* use system page size */ + 0, /* dma type; not used */ + NULL, + _ALIGN_UP(SND_PS3_DMA_REGION_SIZE, PAGE_SIZE)); + dev->d_region->ioid = PS3_AUDIO_IOID; + + ret = ps3_dma_region_create(dev->d_region); + if (ret) { + pr_info("%s: region_create\n", __func__); + goto clean_mmio; + } + + snd_ps3_audio_set_base_addr(dev->d_region->bus_addr); + + /* CONFIG_SND_PS3_DEFAULT_START_DELAY */ + the_card.start_delay = snd_ps3_start_delay; + + /* irq */ + if (snd_ps3_allocate_irq()) { + ret = -ENXIO; + goto clean_dma_region; + } + + /* create card instance */ + the_card.card = snd_card_new(index, id, THIS_MODULE, 0); + if (!the_card.card) { + ret = -ENXIO; + goto clean_irq; + } + + strcpy(the_card.card->driver, "PS3"); + strcpy(the_card.card->shortname, "PS3"); + strcpy(the_card.card->longname, "PS3 sound"); + /* create PCM devices instance */ + /* NOTE:this driver works assuming pcm:substream = 1:1 */ + ret = snd_pcm_new(the_card.card, + "SPDIF", + 0, /* instance index, will be stored pcm.device*/ + 1, /* output substream */ + 0, /* input substream */ + &(the_card.pcm)); + if (ret) + goto clean_card; + + the_card.pcm->private_data = &the_card; + strcpy(the_card.pcm->name, "SPDIF"); + + /* set pcm ops */ + snd_pcm_set_ops(the_card.pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_ps3_pcm_spdif_ops); + + the_card.pcm->info_flags = SNDRV_PCM_INFO_NONINTERLEAVED; + /* pre-alloc PCM DMA buffer*/ + ret = snd_pcm_lib_preallocate_pages_for_all(the_card.pcm, + SNDRV_DMA_TYPE_DEV, + &dev->core, + SND_PS3_PCM_PREALLOC_SIZE, + SND_PS3_PCM_PREALLOC_SIZE); + if (ret < 0) { + pr_info("%s: prealloc failed\n", __func__); + goto clean_card; + } + + /* + * allocate null buffer + * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2 + * PAGE_SIZE is enogh + */ + if (!(the_card.null_buffer_start_vaddr = + dma_alloc_coherent(&the_card.ps3_dev->core, + PAGE_SIZE, + &the_card.null_buffer_start_dma_addr, + GFP_KERNEL))) { + pr_info("%s: nullbuffer alloc failed\n", __func__); + goto clean_preallocate; + } + pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__, + the_card.null_buffer_start_vaddr, + the_card.null_buffer_start_dma_addr); + /* set default sample rate/word width */ + snd_ps3_init_avsetting(&the_card); + + /* register the card */ + ret = snd_card_register(the_card.card); + if (ret < 0) + goto clean_dma_map; + + pr_info("%s started. start_delay=%dms\n", + the_card.card->longname, the_card.start_delay); + return 0; + +clean_dma_map: + dma_free_coherent(&the_card.ps3_dev->core, + PAGE_SIZE, + the_card.null_buffer_start_vaddr, + the_card.null_buffer_start_dma_addr); +clean_preallocate: + snd_pcm_lib_preallocate_free_for_all(the_card.pcm); +clean_card: + snd_card_free(the_card.card); +clean_irq: + snd_ps3_free_irq(); +clean_dma_region: + ps3_dma_region_free(dev->d_region); +clean_mmio: + snd_ps3_unmap_mmio(); +clean_dev_map: + lv1_gpu_device_unmap(2); +clean_open: + ps3_close_hv_device(dev); + /* + * there is no destructor function to pcm. + * midlayer automatically releases if the card removed + */ + return ret; +}; /* snd_ps3_probe */ + +/* called when module removal */ +static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev) +{ + int ret; + pr_info("%s:start id=%d\n", __func__, dev->match_id); + if (dev->match_id != PS3_MATCH_ID_SOUND) + return -ENXIO; + + /* + * ctl and preallocate buffer will be freed in + * snd_card_free + */ + ret = snd_card_free(the_card.card); + if (ret) + pr_info("%s: ctl freecard=%d\n", __func__, ret); + + dma_free_coherent(&dev->core, + PAGE_SIZE, + the_card.null_buffer_start_vaddr, + the_card.null_buffer_start_dma_addr); + + ps3_dma_region_free(dev->d_region); + + snd_ps3_free_irq(); + snd_ps3_unmap_mmio(); + + lv1_gpu_device_unmap(2); + ps3_close_hv_device(dev); + pr_info("%s:end id=%d\n", __func__, dev->match_id); + return 0; +} /* snd_ps3_remove */ + +static struct ps3_system_bus_driver snd_ps3_bus_driver_info = { + .match_id = PS3_MATCH_ID_SOUND, + .probe = snd_ps3_driver_probe, + .remove = snd_ps3_driver_remove, + .shutdown = snd_ps3_driver_remove, + .core = { + .name = SND_PS3_DRIVER_NAME, + .owner = THIS_MODULE, + }, +}; + + +/* + * Interrupt handler + */ +static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) +{ + + uint32_t port_intr; + int underflow_occured = 0; + struct snd_ps3_card_info *card = dev_id; + + if (!card->running) { + update_reg(PS3_AUDIO_AX_IS, 0); + update_reg(PS3_AUDIO_INTR_0, 0); + return IRQ_HANDLED; + } + + port_intr = read_reg(PS3_AUDIO_AX_IS); + /* + *serial buffer empty detected (every 4 times), + *program next dma and kick it + */ + if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) { + write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0)); + if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { + write_reg(PS3_AUDIO_AX_IS, port_intr); + underflow_occured = 1; + } + if (card->silent) { + /* we are still in silent time */ + snd_ps3_program_dma(card, + (underflow_occured) ? + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : + SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); + snd_ps3_kick_dma(card); + card->silent --; + } else { + snd_ps3_program_dma(card, + (underflow_occured) ? + SND_PS3_DMA_FILLTYPE_FIRSTFILL : + SND_PS3_DMA_FILLTYPE_RUNNING); + snd_ps3_kick_dma(card); + snd_pcm_period_elapsed(card->substream); + } + } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { + write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0)); + /* + * serial out underflow, but buffer empty not detected. + * in this case, fill fifo with 0 to recover. After + * filling dummy data, serial automatically start to + * consume them and then will generate normal buffer + * empty interrupts. + * If both buffer underflow and buffer empty are occured, + * it is better to do nomal data transfer than empty one + */ + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + } + /* clear interrupt cause */ + return IRQ_HANDLED; +}; + +/* + * module/subsystem initialize/terminate + */ +static int __init snd_ps3_init(void) +{ + int ret; + + if (!firmware_has_feature(FW_FEATURE_PS3_LV1)) + return -ENXIO; + + memset(&the_card, 0, sizeof(the_card)); + spin_lock_init(&the_card.dma_lock); + + /* register systembus DRIVER, this calls our probe() func */ + ret = ps3_system_bus_driver_register(&snd_ps3_bus_driver_info); + + return ret; +} + +static void __exit snd_ps3_exit(void) +{ + ps3_system_bus_driver_unregister(&snd_ps3_bus_driver_info); +} + +MODULE_ALIAS(PS3_MODULE_ALIAS_SOUND); diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h new file mode 100644 index 000000000000..4b7e6fbbe500 --- /dev/null +++ b/sound/ppc/snd_ps3.h @@ -0,0 +1,135 @@ +/* + * Audio support for PS3 + * Copyright (C) 2007 Sony Computer Entertainment Inc. + * All rights reserved. + * Copyright 2006, 2007 Sony Corporation + * + * 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 Licence. + * + * 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 + */ + +#if !defined(_SND_PS3_H_) +#define _SND_PS3_H_ + +#include + +#define SND_PS3_DRIVER_NAME "snd_ps3" + +enum snd_ps3_out_channel { + SND_PS3_OUT_SPDIF_0, + SND_PS3_OUT_SPDIF_1, + SND_PS3_OUT_SERIAL_0, + SND_PS3_OUT_DEVS +}; + +enum snd_ps3_dma_filltype { + SND_PS3_DMA_FILLTYPE_FIRSTFILL, + SND_PS3_DMA_FILLTYPE_RUNNING, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL, + SND_PS3_DMA_FILLTYPE_SILENT_RUNNING +}; + +enum snd_ps3_ch { + SND_PS3_CH_L = 0, + SND_PS3_CH_R = 1, + SND_PS3_CH_MAX = 2 +}; + +struct snd_ps3_avsetting_info { + uint32_t avs_audio_ch; /* fixed */ + uint32_t avs_audio_rate; + uint32_t avs_audio_width; + uint32_t avs_audio_format; /* fixed */ + uint32_t avs_audio_source; /* fixed */ +}; +/* + * PS3 audio 'card' instance + * there should be only ONE hardware. + */ +struct snd_ps3_card_info { + struct ps3_system_bus_device *ps3_dev; + struct snd_card *card; + + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + + /* hvc info */ + u64 audio_lpar_addr; + u64 audio_lpar_size; + + /* registers */ + void __iomem *mapped_mmio_vaddr; + + /* irq */ + u64 audio_irq_outlet; + unsigned int irq_no; + + /* remember avsetting */ + struct snd_ps3_avsetting_info avs; + + /* dma buffer management */ + spinlock_t dma_lock; + /* dma_lock start */ + void * dma_start_vaddr[2]; /* 0 for L, 1 for R */ + dma_addr_t dma_start_bus_addr[2]; + size_t dma_buffer_size; + void * dma_last_transfer_vaddr[2]; + void * dma_next_transfer_vaddr[2]; + int silent; + /* dma_lock end */ + + int running; + + /* null buffer */ + void *null_buffer_start_vaddr; + dma_addr_t null_buffer_start_dma_addr; + + /* start delay */ + unsigned int start_delay; + +}; + + +/* PS3 audio DMAC block size in bytes */ +#define PS3_AUDIO_DMAC_BLOCK_SIZE (128) +/* one stage (stereo) of audio FIFO in bytes */ +#define PS3_AUDIO_FIFO_STAGE_SIZE (256) +/* how many stages the fifo have */ +#define PS3_AUDIO_FIFO_STAGE_COUNT (8) +/* fifo size 128 bytes * 8 stages * stereo (2ch) */ +#define PS3_AUDIO_FIFO_SIZE \ + (PS3_AUDIO_FIFO_STAGE_SIZE * PS3_AUDIO_FIFO_STAGE_COUNT) + +/* PS3 audio DMAC max block count in one dma shot = 128 (0x80) blocks*/ +#define PS3_AUDIO_DMAC_MAX_BLOCKS (PS3_AUDIO_DMASIZE_BLOCKS_MASK + 1) + +#define PS3_AUDIO_NORMAL_DMA_START_CH (0) +#define PS3_AUDIO_NORMAL_DMA_COUNT (8) +#define PS3_AUDIO_NULL_DMA_START_CH \ + (PS3_AUDIO_NORMAL_DMA_START_CH + PS3_AUDIO_NORMAL_DMA_COUNT) +#define PS3_AUDIO_NULL_DMA_COUNT (2) + +#define SND_PS3_MAX_VOL (0x0F) +#define SND_PS3_MIN_VOL (0x00) +#define SND_PS3_MIN_ATT SND_PS3_MIN_VOL +#define SND_PS3_MAX_ATT SND_PS3_MAX_VOL + +#define SND_PS3_PCM_PREALLOC_SIZE \ + (PS3_AUDIO_DMAC_BLOCK_SIZE * PS3_AUDIO_DMAC_MAX_BLOCKS * 4) + +#define SND_PS3_DMA_REGION_SIZE \ + (SND_PS3_PCM_PREALLOC_SIZE + PAGE_SIZE) + +#define PS3_AUDIO_IOID (1UL) + +#endif /* _SND_PS3_H_ */ diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h new file mode 100644 index 000000000000..03fdee4aaaf2 --- /dev/null +++ b/sound/ppc/snd_ps3_reg.h @@ -0,0 +1,891 @@ +/* + * Audio support for PS3 + * Copyright (C) 2007 Sony Computer Entertainment Inc. + * Copyright 2006, 2007 Sony 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 + * 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 + */ + +/* + * interrupt / configure registers + */ + +#define PS3_AUDIO_INTR_0 (0x00000100) +#define PS3_AUDIO_INTR_EN_0 (0x00000140) +#define PS3_AUDIO_CONFIG (0x00000200) + +/* + * DMAC registers + * n:0..9 + */ +#define PS3_AUDIO_DMAC_REGBASE(x) (0x0000210 + 0x20 * (x)) + +#define PS3_AUDIO_KICK(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x00) +#define PS3_AUDIO_SOURCE(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x04) +#define PS3_AUDIO_DEST(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x08) +#define PS3_AUDIO_DMASIZE(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x0C) + +/* + * mute control + */ +#define PS3_AUDIO_AX_MCTRL (0x00004000) +#define PS3_AUDIO_AX_ISBP (0x00004004) +#define PS3_AUDIO_AX_AOBP (0x00004008) +#define PS3_AUDIO_AX_IC (0x00004010) +#define PS3_AUDIO_AX_IE (0x00004014) +#define PS3_AUDIO_AX_IS (0x00004018) + +/* + * three wire serial + * n:0..3 + */ +#define PS3_AUDIO_AO_MCTRL (0x00006000) +#define PS3_AUDIO_AO_3WMCTRL (0x00006004) + +#define PS3_AUDIO_AO_3WCTRL(n) (0x00006200 + 0x200 * (n)) + +/* + * S/PDIF + * n:0..1 + * x:0..11 + * y:0..5 + */ +#define PS3_AUDIO_AO_SPD_REGBASE(n) (0x00007200 + 0x200 * (n)) + +#define PS3_AUDIO_AO_SPDCTRL(n) \ + (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x00) +#define PS3_AUDIO_AO_SPDUB(n, x) \ + (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x04 + 0x04 * (x)) +#define PS3_AUDIO_AO_SPDCS(n, y) \ + (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x34 + 0x04 * (y)) + + +/* + PS3_AUDIO_INTR_0 register tells an interrupt handler which audio + DMA channel triggered the interrupt. The interrupt status for a channel + can be cleared by writing a '1' to the corresponding bit. A new interrupt + cannot be generated until the previous interrupt has been cleared. + + Note that the status reported by PS3_AUDIO_INTR_0 is independent of the + value of PS3_AUDIO_INTR_EN_0. + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ +#define PS3_AUDIO_INTR_0_CHAN(n) (1 << ((n) * 2)) +#define PS3_AUDIO_INTR_0_CHAN9 PS3_AUDIO_INTR_0_CHAN(9) +#define PS3_AUDIO_INTR_0_CHAN8 PS3_AUDIO_INTR_0_CHAN(8) +#define PS3_AUDIO_INTR_0_CHAN7 PS3_AUDIO_INTR_0_CHAN(7) +#define PS3_AUDIO_INTR_0_CHAN6 PS3_AUDIO_INTR_0_CHAN(6) +#define PS3_AUDIO_INTR_0_CHAN5 PS3_AUDIO_INTR_0_CHAN(5) +#define PS3_AUDIO_INTR_0_CHAN4 PS3_AUDIO_INTR_0_CHAN(4) +#define PS3_AUDIO_INTR_0_CHAN3 PS3_AUDIO_INTR_0_CHAN(3) +#define PS3_AUDIO_INTR_0_CHAN2 PS3_AUDIO_INTR_0_CHAN(2) +#define PS3_AUDIO_INTR_0_CHAN1 PS3_AUDIO_INTR_0_CHAN(1) +#define PS3_AUDIO_INTR_0_CHAN0 PS3_AUDIO_INTR_0_CHAN(0) + +/* + The PS3_AUDIO_INTR_EN_0 register specifies which DMA channels can generate + an interrupt to the PU. Each bit of PS3_AUDIO_INTR_EN_0 is ANDed with the + corresponding bit in PS3_AUDIO_INTR_0. The resulting bits are OR'd together + to generate the Audio interrupt. + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_EN_0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + + Bit assignments are same as PS3_AUDIO_INTR_0 +*/ + +/* + PS3_AUDIO_CONFIG + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 C|0 0 0 0 0 0 0 0| CONFIG + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + +*/ + +/* The CLEAR field cancels all pending transfers, and stops any running DMA + transfers. Any interrupts associated with the canceled transfers + will occur as if the transfer had finished. + Since this bit is designed to recover from DMA related issues + which are caused by unpredictable situations, it is prefered to wait + for normal DMA transfer end without using this bit. +*/ +#define PS3_AUDIO_CONFIG_CLEAR (1 << 8) /* RWIVF */ + +/* + PS3_AUDIO_AX_MCTRL: Audio Port Mute Control Register + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|A|A|0 0 0 0 0 0 0|S|S|A|A|A|A| AX_MCTRL + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + +/* 3 Wire Audio Serial Output Channel Mutes (0..3) */ +#define PS3_AUDIO_AX_MCTRL_ASOMT(n) (1 << (3 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_ASO3MT (1 << 0) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_ASO2MT (1 << 1) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_ASO1MT (1 << 2) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_ASO0MT (1 << 3) /* RWIVF */ + +/* S/PDIF mutes (0,1)*/ +#define PS3_AUDIO_AX_MCTRL_SPOMT(n) (1 << (5 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_SPO1MT (1 << 4) /* RWIVF */ +#define PS3_AUDIO_AX_MCTRL_SPO0MT (1 << 5) /* RWIVF */ + +/* All 3 Wire Serial Outputs Mute */ +#define PS3_AUDIO_AX_MCTRL_AASOMT (1 << 13) /* RWIVF */ + +/* All S/PDIF Mute */ +#define PS3_AUDIO_AX_MCTRL_ASPOMT (1 << 14) /* RWIVF */ + +/* All Audio Outputs Mute */ +#define PS3_AUDIO_AX_MCTRL_AAOMT (1 << 15) /* RWIVF */ + +/* + S/PDIF Outputs Buffer Read/Write Pointer Register + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B|0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B| AX_ISBP + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + +*/ +/* + S/PDIF Output Channel Read Buffer Numbers + Buffer number is value of field. + Indicates current read access buffer ID from Audio Data + Transfer controller of S/PDIF Output +*/ + +#define PS3_AUDIO_AX_ISBP_SPOBRN_MASK(n) (0x7 << 4 * (1 - (n))) /* R-IUF */ +#define PS3_AUDIO_AX_ISBP_SPO1BRN_MASK (0x7 << 0) /* R-IUF */ +#define PS3_AUDIO_AX_ISBP_SPO0BRN_MASK (0x7 << 4) /* R-IUF */ + +/* +S/PDIF Output Channel Buffer Write Numbers +Indicates current write access buffer ID from bus master. +*/ +#define PS3_AUDIO_AX_ISBP_SPOBWN_MASK(n) (0x7 << 4 * (5 - (n))) /* R-IUF */ +#define PS3_AUDIO_AX_ISBP_SPO1BWN_MASK (0x7 << 16) /* R-IUF */ +#define PS3_AUDIO_AX_ISBP_SPO0BWN_MASK (0x7 << 20) /* R-IUF */ + +/* + 3 Wire Audio Serial Outputs Buffer Read/Write + Pointer Register + Buffer number is value of field + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B|0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B| AX_AOBP + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + +/* +3 Wire Audio Serial Output Channel Buffer Read Numbers +Indicates current read access buffer Id from Audio Data Transfer +Controller of 3 Wire Audio Serial Output Channels +*/ +#define PS3_AUDIO_AX_AOBP_ASOBRN_MASK(n) (0x7 << 4 * (3 - (n))) /* R-IUF */ + +#define PS3_AUDIO_AX_AOBP_ASO3BRN_MASK (0x7 << 0) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO2BRN_MASK (0x7 << 4) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO1BRN_MASK (0x7 << 8) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO0BRN_MASK (0x7 << 12) /* R-IUF */ + +/* +3 Wire Audio Serial Output Channel Buffer Write Numbers +Indicates current write access buffer ID from bus master. +*/ +#define PS3_AUDIO_AX_AOBP_ASOBWN_MASK(n) (0x7 << 4 * (7 - (n))) /* R-IUF */ + +#define PS3_AUDIO_AX_AOBP_ASO3BWN_MASK (0x7 << 16) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO2BWN_MASK (0x7 << 20) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO1BWN_MASK (0x7 << 24) /* R-IUF */ +#define PS3_AUDIO_AX_AOBP_ASO0BWN_MASK (0x7 << 28) /* R-IUF */ + + + +/* +Audio Port Interrupt Condition Register +For the fields in this register, the following values apply: +0 = Interrupt is generated every interrupt event. +1 = Interrupt is generated every 2 interrupt events. +2 = Interrupt is generated every 4 interrupt events. +3 = Reserved + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0|0 0|SPO|0 0|SPO|0 0|AAS|0 0 0 0 0 0 0 0 0 0 0 0| AX_IC + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ +/* +All 3-Wire Audio Serial Outputs Interrupt Mode +Configures the Interrupt and Signal Notification +condition of all 3-wire Audio Serial Outputs. +*/ +#define PS3_AUDIO_AX_IC_AASOIMD_MASK (0x3 << 12) /* RWIVF */ +#define PS3_AUDIO_AX_IC_AASOIMD_EVERY1 (0x0 << 12) /* RWI-V */ +#define PS3_AUDIO_AX_IC_AASOIMD_EVERY2 (0x1 << 12) /* RW--V */ +#define PS3_AUDIO_AX_IC_AASOIMD_EVERY4 (0x2 << 12) /* RW--V */ + +/* +S/PDIF Output Channel Interrupt Modes +Configures the Interrupt and signal Notification +conditions of S/PDIF output channels. +*/ +#define PS3_AUDIO_AX_IC_SPO1IMD_MASK (0x3 << 16) /* RWIVF */ +#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY1 (0x0 << 16) /* RWI-V */ +#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY2 (0x1 << 16) /* RW--V */ +#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY4 (0x2 << 16) /* RW--V */ + +#define PS3_AUDIO_AX_IC_SPO0IMD_MASK (0x3 << 20) /* RWIVF */ +#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY1 (0x0 << 20) /* RWI-V */ +#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY2 (0x1 << 20) /* RW--V */ +#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY4 (0x2 << 20) /* RW--V */ + +/* +Audio Port interrupt Enable Register +Configures whether to enable or disable each Interrupt Generation. + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IE + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + +*/ + +/* +3 Wire Audio Serial Output Channel Buffer Underflow +Interrupt Enables +Select enable/disable of Buffer Underflow Interrupts for +3-Wire Audio Serial Output Channels +DISABLED=Interrupt generation disabled. +*/ +#define PS3_AUDIO_AX_IE_ASOBUIE(n) (1 << (3 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO3BUIE (1 << 0) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO2BUIE (1 << 1) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO1BUIE (1 << 2) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO0BUIE (1 << 3) /* RWIVF */ + +/* S/PDIF Output Channel Buffer Underflow Interrupt Enables */ + +#define PS3_AUDIO_AX_IE_SPOBUIE(n) (1 << (7 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO1BUIE (1 << 6) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO0BUIE (1 << 7) /* RWIVF */ + +/* S/PDIF Output Channel One Block Transfer Completion Interrupt Enables */ + +#define PS3_AUDIO_AX_IE_SPOBTCIE(n) (1 << (11 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO1BTCIE (1 << 10) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO0BTCIE (1 << 11) /* RWIVF */ + +/* 3-Wire Audio Serial Output Channel Buffer Empty Interrupt Enables */ + +#define PS3_AUDIO_AX_IE_ASOBEIE(n) (1 << (19 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO3BEIE (1 << 16) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO2BEIE (1 << 17) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO1BEIE (1 << 18) /* RWIVF */ +#define PS3_AUDIO_AX_IE_ASO0BEIE (1 << 19) /* RWIVF */ + +/* S/PDIF Output Channel Buffer Empty Interrupt Enables */ + +#define PS3_AUDIO_AX_IE_SPOBEIE(n) (1 << (23 - (n))) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO1BEIE (1 << 22) /* RWIVF */ +#define PS3_AUDIO_AX_IE_SPO0BEIE (1 << 23) /* RWIVF */ + +/* +Audio Port Interrupt Status Register +Indicates Interrupt status, which interrupt has occured, and can clear +each interrupt in this register. +Writing 1b to a field containing 1b clears field and de-asserts interrupt. +Writing 0b to a field has no effect. +Field vaules are the following: +0 - Interrupt hasn't occured. +1 - Interrupt has occured. + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IS + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + + Bit assignment are same as AX_IE +*/ + +/* +Audio Output Master Control Register +Configures Master Clock and other master Audio Output Settings + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0|SCKSE|0|SCKSE| MR0 | MR1 |MCL|MCL|0 0 0 0|0 0 0 0 0 0 0 0| AO_MCTRL + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + +/* +MCLK Output Control +Controls mclko[1] output. +0 - Disable output (fixed at High) +1 - Output clock produced by clock selected +with scksel1 by mr1 +2 - Reserved +3 - Reserved +*/ + +#define PS3_AUDIO_AO_MCTRL_MCLKC1_MASK (0x3 << 12) /* RWIVF */ +#define PS3_AUDIO_AO_MCTRL_MCLKC1_DISABLED (0x0 << 12) /* RWI-V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC1_ENABLED (0x1 << 12) /* RW--V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD2 (0x2 << 12) /* RW--V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD3 (0x3 << 12) /* RW--V */ + +/* +MCLK Output Control +Controls mclko[0] output. +0 - Disable output (fixed at High) +1 - Output clock produced by clock selected +with SCKSEL0 by MR0 +2 - Reserved +3 - Reserved +*/ +#define PS3_AUDIO_AO_MCTRL_MCLKC0_MASK (0x3 << 14) /* RWIVF */ +#define PS3_AUDIO_AO_MCTRL_MCLKC0_DISABLED (0x0 << 14) /* RWI-V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC0_ENABLED (0x1 << 14) /* RW--V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD2 (0x2 << 14) /* RW--V */ +#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD3 (0x3 << 14) /* RW--V */ +/* +Master Clock Rate 1 +Sets the divide ration of Master Clock1 (clock output from +mclko[1] for the input clock selected by scksel1. +*/ +#define PS3_AUDIO_AO_MCTRL_MR1_MASK (0xf << 16) +#define PS3_AUDIO_AO_MCTRL_MR1_DEFAULT (0x0 << 16) /* RWI-V */ +/* +Master Clock Rate 0 +Sets the divide ratio of Master Clock0 (clock output from +mclko[0] for the input clock selected by scksel0). +*/ +#define PS3_AUDIO_AO_MCTRL_MR0_MASK (0xf << 20) /* RWIVF */ +#define PS3_AUDIO_AO_MCTRL_MR0_DEFAULT (0x0 << 20) /* RWI-V */ +/* +System Clock Select 0/1 +Selects the system clock to be used as Master Clock 0/1 +Input the system clock that is appropriate for the sampling +rate. +*/ +#define PS3_AUDIO_AO_MCTRL_SCKSEL1_MASK (0x7 << 24) /* RWIVF */ +#define PS3_AUDIO_AO_MCTRL_SCKSEL1_DEFAULT (0x2 << 24) /* RWI-V */ + +#define PS3_AUDIO_AO_MCTRL_SCKSEL0_MASK (0x7 << 28) /* RWIVF */ +#define PS3_AUDIO_AO_MCTRL_SCKSEL0_DEFAULT (0x2 << 28) /* RWI-V */ + + +/* +3-Wire Audio Output Master Control Register +Configures clock, 3-Wire Audio Serial Output Enable, and +other 3-Wire Audio Serial Output Master Settings + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |A|A|A|A|0 0 0|A| ASOSR |0 0 0 0|A|A|A|A|A|A|0|1|0 0 0 0 0 0 0 0| AO_3WMCTRL + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + + +/* +LRCKO Polarity +0 - Reserved +1 - default +*/ +#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK (1 << 8) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT (1 << 8) /* RW--V */ + +/* LRCK Output Disable */ + +#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD (1 << 10) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_ENABLED (0 << 10) /* RW--V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED (1 << 10) /* RWI-V */ + +/* Bit Clock Output Disable */ + +#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD (1 << 11) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_ENABLED (0 << 11) /* RW--V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED (1 << 11) /* RWI-V */ + +/* +3-Wire Audio Serial Output Channel 0-3 Operational +Status. Each bit becomes 1 after each 3-Wire Audio +Serial Output Channel N is in action by setting 1 to +asoen. +Each bit becomes 0 after each 3-Wire Audio Serial Output +Channel N is out of action by setting 0 to asoen. +*/ +#define PS3_AUDIO_AO_3WMCTRL_ASORUN(n) (1 << (15 - (n))) /* R-IVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(n) (0 << (15 - (n))) /* R-I-V */ +#define PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(n) (1 << (15 - (n))) /* R---V */ +#define PS3_AUDIO_AO_3WMCTRL_ASORUN0 \ + PS3_AUDIO_AO_3WMCTRL_ASORUN(0) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_STOPPED \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(0) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_RUNNING \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(0) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN1 \ + PS3_AUDIO_AO_3WMCTRL_ASORUN(1) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_STOPPED \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(1) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_RUNNING \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(1) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN2 \ + PS3_AUDIO_AO_3WMCTRL_ASORUN(2) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_STOPPED \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(2) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_RUNNING \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(2) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN3 \ + PS3_AUDIO_AO_3WMCTRL_ASORUN(3) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_STOPPED \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(3) +#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_RUNNING \ + PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(3) + +/* +Sampling Rate +Specifies the divide ratio of the bit clock (clock output +from bclko) used by the 3-wire Audio Output Clock, whcih +is applied to the master clock selected by mcksel. +Data output is synchronized with this clock. +*/ +#define PS3_AUDIO_AO_3WMCTRL_ASOSR_MASK (0xf << 20) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV2 (0x1 << 20) /* RWI-V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV4 (0x2 << 20) /* RW--V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV8 (0x4 << 20) /* RW--V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV12 (0x6 << 20) /* RW--V */ + +/* +Master Clock Select +0 - Master Clock 0 +1 - Master Clock 1 +*/ +#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL (1 << 24) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK0 (0 << 24) /* RWI-V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK1 (1 << 24) /* RW--V */ + +/* +Enables and disables 4ch 3-Wire Audio Serial Output +operation. Each Bit from 0 to 3 corresponds to an +output channel, which means that each output channel +can be enabled or disabled individually. When +multiple channels are enabled at the same time, output +operations are performed in synchronization. +Bit 0 - Output Channel 0 (SDOUT[0]) +Bit 1 - Output Channel 1 (SDOUT[1]) +Bit 2 - Output Channel 2 (SDOUT[2]) +Bit 3 - Output Channel 3 (SDOUT[3]) +*/ +#define PS3_AUDIO_AO_3WMCTRL_ASOEN(n) (1 << (31 - (n))) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(n) (0 << (31 - (n))) /* RWI-V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(n) (1 << (31 - (n))) /* RW--V */ + +#define PS3_AUDIO_AO_3WMCTRL_ASOEN0 \ + PS3_AUDIO_AO_3WMCTRL_ASOEN(0) /* RWIVF */ +#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_DISABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(0) /* RWI-V */ +#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_ENABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(0) /* RW--V */ +#define PS3_AUDIO_A1_3WMCTRL_ASOEN0 \ + PS3_AUDIO_AO_3WMCTRL_ASOEN(1) /* RWIVF */ +#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_DISABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(1) /* RWI-V */ +#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_ENABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(1) /* RW--V */ +#define PS3_AUDIO_A2_3WMCTRL_ASOEN0 \ + PS3_AUDIO_AO_3WMCTRL_ASOEN(2) /* RWIVF */ +#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_DISABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(2) /* RWI-V */ +#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_ENABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(2) /* RW--V */ +#define PS3_AUDIO_A3_3WMCTRL_ASOEN0 \ + PS3_AUDIO_AO_3WMCTRL_ASOEN(3) /* RWIVF */ +#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_DISABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(3) /* RWI-V */ +#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_ENABLED \ + PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(3) /* RW--V */ + +/* +3-Wire Audio Serial output Channel 0-3 Control Register +Configures settings for 3-Wire Serial Audio Output Channel 0-3 + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|0 0 0 0|A|0|ASO|0 0 0|0|0|0|0|0| AO_3WCTRL + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + +*/ +/* +Data Bit Mode +Specifies the number of data bits +0 - 16 bits +1 - reserved +2 - 20 bits +3 - 24 bits +*/ +#define PS3_AUDIO_AO_3WCTRL_ASODB_MASK (0x3 << 8) /* RWIVF */ +#define PS3_AUDIO_AO_3WCTRL_ASODB_16BIT (0x0 << 8) /* RWI-V */ +#define PS3_AUDIO_AO_3WCTRL_ASODB_RESVD (0x1 << 8) /* RWI-V */ +#define PS3_AUDIO_AO_3WCTRL_ASODB_20BIT (0x2 << 8) /* RW--V */ +#define PS3_AUDIO_AO_3WCTRL_ASODB_24BIT (0x3 << 8) /* RW--V */ +/* +Data Format Mode +Specifies the data format where (LSB side or MSB) the data(in 20 bit +or 24 bit resolution mode) is put in a 32 bit field. +0 - Data put on LSB side +1 - Data put on MSB side +*/ +#define PS3_AUDIO_AO_3WCTRL_ASODF (1 << 11) /* RWIVF */ +#define PS3_AUDIO_AO_3WCTRL_ASODF_LSB (0 << 11) /* RWI-V */ +#define PS3_AUDIO_AO_3WCTRL_ASODF_MSB (1 << 11) /* RW--V */ +/* +Buffer Reset +Performs buffer reset. Writing 1 to this bit initializes the +corresponding 3-Wire Audio Output buffers(both L and R). +*/ +#define PS3_AUDIO_AO_3WCTRL_ASOBRST (1 << 16) /* CWIVF */ +#define PS3_AUDIO_AO_3WCTRL_ASOBRST_IDLE (0 << 16) /* -WI-V */ +#define PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET (1 << 16) /* -W--T */ + +/* +S/PDIF Audio Output Channel 0/1 Control Register +Configures settings for S/PDIF Audio Output Channel 0/1. + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |S|0 0 0|S|0 0|S| SPOSR |0 0|SPO|0 0 0 0|S|0|SPO|0 0 0 0 0 0 0|S| AO_SPDCTRL + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ +/* +Buffer reset. Writing 1 to this bit initializes the +corresponding S/PDIF output buffer pointer. +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPOBRST (1 << 0) /* CWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_IDLE (0 << 0) /* -WI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_RESET (1 << 0) /* -W--T */ + +/* +Data Bit Mode +Specifies number of data bits +0 - 16 bits +1 - Reserved +2 - 20 bits +3 - 24 bits +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPODB_MASK (0x3 << 8) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPODB_16BIT (0x0 << 8) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPODB_RESVD (0x1 << 8) /* RW--V */ +#define PS3_AUDIO_AO_SPDCTRL_SPODB_20BIT (0x2 << 8) /* RW--V */ +#define PS3_AUDIO_AO_SPDCTRL_SPODB_24BIT (0x3 << 8) /* RW--V */ +/* +Data format Mode +Specifies the data format, where (LSB side or MSB) +the data(in 20 or 24 bit resolution) is put in the +32 bit field. +0 - LSB Side +1 - MSB Side +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPODF (1 << 11) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPODF_LSB (0 << 11) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPODF_MSB (1 << 11) /* RW--V */ +/* +Source Select +Specifies the source of the S/PDIF output. When 0, output +operation is controlled by 3wen[0] of AO_3WMCTRL register. +The SR must have the same setting as the a0_3wmctrl reg. +0 - 3-Wire Audio OUT Ch0 Buffer +1 - S/PDIF buffer +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPOSS_MASK (0x3 << 16) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSS_3WEN (0x0 << 16) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSS_SPDIF (0x1 << 16) /* RW--V */ +/* +Sampling Rate +Specifies the divide ratio of the bit clock (clock output +from bclko) used by the S/PDIF Output Clock, which +is applied to the master clock selected by mcksel. +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPOSR (0xf << 20) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV2 (0x1 << 20) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV4 (0x2 << 20) /* RW--V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV8 (0x4 << 20) /* RW--V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV12 (0x6 << 20) /* RW--V */ +/* +Master Clock Select +0 - Master Clock 0 +1 - Master Clock 1 +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL (1 << 24) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK0 (0 << 24) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK1 (1 << 24) /* RW--V */ + +/* +S/PDIF Output Channel Operational Status +This bit becomes 1 after S/PDIF Output Channel is in +action by setting 1 to spoen. This bit becomes 0 +after S/PDIF Output Channel is out of action by setting +0 to spoen. +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPORUN (1 << 27) /* R-IVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPORUN_STOPPED (0 << 27) /* R-I-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPORUN_RUNNING (1 << 27) /* R---V */ + +/* +S/PDIF Audio Output Channel Output Enable +Enables and disables output operation. This bit is used +only when sposs = 1 +*/ +#define PS3_AUDIO_AO_SPDCTRL_SPOEN (1 << 31) /* RWIVF */ +#define PS3_AUDIO_AO_SPDCTRL_SPOEN_DISABLED (0 << 31) /* RWI-V */ +#define PS3_AUDIO_AO_SPDCTRL_SPOEN_ENABLED (1 << 31) /* RW--V */ + +/* +S/PDIF Audio Output Channel Channel Status +Setting Registers. +Configures channel status bit settings for each block +(192 bits). +Output is performed from the MSB(AO_SPDCS0 register bit 31). +The same value is added for subframes within the same frame. + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + | SPOCS | AO_SPDCS + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + +S/PDIF Audio Output Channel User Bit Setting +Configures user bit settings for each block (384 bits). +Output is performed from the MSB(ao_spdub0 register bit 31). + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + | SPOUB | AO_SPDUB + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ +/***************************************************************************** + * + * DMAC register + * + *****************************************************************************/ +/* +The PS3_AUDIO_KICK register is used to initiate a DMA transfer and monitor +its status + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0|STATU|0 0 0| EVENT |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|R| KICK + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ +/* +The REQUEST field is written to ACTIVE to initiate a DMA request when EVENT +occurs. +It will return to the DONE state when the request is completed. +The registers for a DMA channel should only be written if REQUEST is IDLE. +*/ + +#define PS3_AUDIO_KICK_REQUEST (1 << 0) /* RWIVF */ +#define PS3_AUDIO_KICK_REQUEST_IDLE (0 << 0) /* RWI-V */ +#define PS3_AUDIO_KICK_REQUEST_ACTIVE (1 << 0) /* -W--T */ + +/* + *The EVENT field is used to set the event in which + *the DMA request becomes active. + */ +#define PS3_AUDIO_KICK_EVENT_MASK (0x1f << 16) /* RWIVF */ +#define PS3_AUDIO_KICK_EVENT_ALWAYS (0x00 << 16) /* RWI-V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY (0x01 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_UNDERFLOW (0x02 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_EMPTY (0x03 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_UNDERFLOW (0x04 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_EMPTY (0x05 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_UNDERFLOW (0x06 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_EMPTY (0x07 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_UNDERFLOW (0x08 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF0_BLOCKTRANSFERCOMPLETE \ + (0x09 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF0_UNDERFLOW (0x0A << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF0_EMPTY (0x0B << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF1_BLOCKTRANSFERCOMPLETE \ + (0x0C << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF1_UNDERFLOW (0x0D << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_SPDIF1_EMPTY (0x0E << 16) /* RW--V */ + +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA(n) \ + ((0x13 + (n)) << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA0 (0x13 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA1 (0x14 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA2 (0x15 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA3 (0x16 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA4 (0x17 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA5 (0x18 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA6 (0x19 << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA7 (0x1A << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA8 (0x1B << 16) /* RW--V */ +#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA9 (0x1C << 16) /* RW--V */ + +/* +The STATUS field can be used to monitor the progress of a DMA request. +DONE indicates the previous request has completed. +EVENT indicates that the DMA engine is waiting for the EVENT to occur. +PENDING indicates that the DMA engine has not started processing this +request, but the EVENT has occured. +DMA indicates that the data transfer is in progress. +NOTIFY indicates that the notifier signalling end of transfer is being written. +CLEAR indicated that the previous transfer was cleared. +ERROR indicates the previous transfer requested an unsupported +source/destination combination. +*/ + +#define PS3_AUDIO_KICK_STATUS_MASK (0x7 << 24) /* R-IVF */ +#define PS3_AUDIO_KICK_STATUS_DONE (0x0 << 24) /* R-I-V */ +#define PS3_AUDIO_KICK_STATUS_EVENT (0x1 << 24) /* R---V */ +#define PS3_AUDIO_KICK_STATUS_PENDING (0x2 << 24) /* R---V */ +#define PS3_AUDIO_KICK_STATUS_DMA (0x3 << 24) /* R---V */ +#define PS3_AUDIO_KICK_STATUS_NOTIFY (0x4 << 24) /* R---V */ +#define PS3_AUDIO_KICK_STATUS_CLEAR (0x5 << 24) /* R---V */ +#define PS3_AUDIO_KICK_STATUS_ERROR (0x6 << 24) /* R---V */ + +/* +The PS3_AUDIO_SOURCE register specifies the source address for transfers. + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + | START |0 0 0 0 0|TAR| SOURCE + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + +/* +The Audio DMA engine uses 128-byte transfers, thus the address must be aligned +to a 128 byte boundary. The low seven bits are assumed to be 0. +*/ + +#define PS3_AUDIO_SOURCE_START_MASK (0x01FFFFFF << 7) /* RWIUF */ + +/* +The TARGET field specifies the memory space containing the source address. +*/ + +#define PS3_AUDIO_SOURCE_TARGET_MASK (3 << 0) /* RWIVF */ +#define PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY (2 << 0) /* RW--V */ + +/* +The PS3_AUDIO_DEST register specifies the destination address for transfers. + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + | START |0 0 0 0 0|TAR| DEST + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + +/* +The Audio DMA engine uses 128-byte transfers, thus the address must be aligned +to a 128 byte boundary. The low seven bits are assumed to be 0. +*/ + +#define PS3_AUDIO_DEST_START_MASK (0x01FFFFFF << 7) /* RWIUF */ + +/* +The TARGET field specifies the memory space containing the destination address +AUDIOFIFO = Audio WriteData FIFO, +*/ + +#define PS3_AUDIO_DEST_TARGET_MASK (3 << 0) /* RWIVF */ +#define PS3_AUDIO_DEST_TARGET_AUDIOFIFO (1 << 0) /* RW--V */ + +/* +PS3_AUDIO_DMASIZE specifies the number of 128-byte blocks + 1 to transfer. +So a value of 0 means 128-bytes will get transfered. + + + 31 24 23 16 15 8 7 0 + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ + |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0| BLOCKS | DMASIZE + +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+ +*/ + + +#define PS3_AUDIO_DMASIZE_BLOCKS_MASK (0x7f << 0) /* RWIUF */ + +/* + * source/destination address for internal fifos + */ +#define PS3_AUDIO_AO_3W_LDATA(n) (0x1000 + (0x100 * (n))) +#define PS3_AUDIO_AO_3W_RDATA(n) (0x1080 + (0x100 * (n))) + +#define PS3_AUDIO_AO_SPD_DATA(n) (0x2000 + (0x400 * (n))) + + +/* + * field attiribute + * + * Read + * ' ' = Other Information + * '-' = Field is part of a write-only register + * 'C' = Value read is always the same, constant value line follows (C) + * 'R' = Value is read + * + * Write + * ' ' = Other Information + * '-' = Must not be written (D), value ignored when written (R,A,F) + * 'W' = Can be written + * + * Internal State + * ' ' = Other Information + * '-' = No internal state + * 'X' = Internal state, initial value is unknown + * 'I' = Internal state, initial value is known and follows (I) + * + * Declaration/Size + * ' ' = Other Information + * '-' = Does Not Apply + * 'V' = Type is void + * 'U' = Type is unsigned integer + * 'S' = Type is signed integer + * 'F' = Type is IEEE floating point + * '1' = Byte size (008) + * '2' = Short size (016) + * '3' = Three byte size (024) + * '4' = Word size (032) + * '8' = Double size (064) + * + * Define Indicator + * ' ' = Other Information + * 'D' = Device + * 'M' = Memory + * 'R' = Register + * 'A' = Array of Registers + * 'F' = Field + * 'V' = Value + * 'T' = Task + */ + From 95093a23b9090025f8a6be0ac93859210fbba678 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Tue, 3 Jul 2007 20:07:39 +0200 Subject: [PATCH 055/102] [ALSA] usb-audio - Add quirk for Roland Juno-G Added a quirk for Roland Juno-G. It's simply a copy of Fantom-X with the IDs changed. From: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 9274baf69901..8fca24e2635c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1103,6 +1103,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x582, 0x00a6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "Juno-G", + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + } +}, { /* * This quirk is for the "Advanced" modes of the Edirol UA-25. * If the switch is not in an advanced setting, the UA-25 has From 8481da5a8d009d9bbac3d1483b579940e6dd9d59 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Tue, 3 Jul 2007 20:05:19 +0200 Subject: [PATCH 056/102] [ALSA] hda-codec - Add support for HP Spartan Added the model for HP Spartan with Conexant codec chip. From: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 0ffcf2001d28..4d8e8af5c819 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -801,6 +801,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), From 8f41b56645f03e6f50407211d6f0f0627ae95e4e Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Tue, 3 Jul 2007 20:02:49 +0200 Subject: [PATCH 057/102] [ALSA] hda-codec - Add support for HP Nettle Added the model for HP Nettle mobo with ALC883. From: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 5367d1377e11..a4876a31311e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6410,6 +6410,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; From 4700418cfc045296ee453342dc2fb142dc752aed Mon Sep 17 00:00:00 2001 From: Paul Vojta Date: Wed, 4 Jul 2007 10:34:22 +0200 Subject: [PATCH 058/102] [ALSA] opl3sa2 - Add Neomagic MagicWave 3D ISA PnP ID Add Neomagic MagicWave 3D to list of supported devices for opl3sa2 driver. Signed-off-by: Paul Vojta Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/opl3sa2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 4f6800b43b0e..e70db32991d9 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -164,6 +164,8 @@ static struct pnp_card_device_id snd_opl3sa2_pnpids[] = { { .id = "YMH0801", .devs = { { "YMH0021" } } }, /* NeoMagic MagicWave 3DX */ { .id = "NMX2200", .devs = { { "YMH2210" } } }, + /* NeoMagic MagicWave 3D */ + { .id = "NMX2200", .devs = { { "NMX2210" } } }, /* --- */ { .id = "" } /* end */ }; From 4210861964145617cf27b2b9c45860bff3e8290c Mon Sep 17 00:00:00 2001 From: Paul Vojta Date: Wed, 4 Jul 2007 10:35:29 +0200 Subject: [PATCH 059/102] [ALSA] nm256 - Add mention of opl3sa2 to a diagnostic message Adds mention of opl3sa2 driver to a diagnostic message for NeoMagic nm256 driver. Signed-off-by: Paul Vojta Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/nm256/nm256.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 03b3a4792f73..c7621bd770a6 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1533,7 +1533,8 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, printk(KERN_ERR " force the driver to load by " "passing in the module parameter\n"); printk(KERN_ERR " force_ac97=1\n"); - printk(KERN_ERR " or try sb16 or cs423x drivers instead.\n"); + printk(KERN_ERR " or try sb16, opl3sa2, or " + "cs423x drivers instead.\n"); err = -ENXIO; goto __error; } From 0a8c5da3c066a1e92a66a493e6fe04d5ce7a206d Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Wed, 4 Jul 2007 15:17:38 +0200 Subject: [PATCH 060/102] [ALSA] hda-codec - Fix 5.1 output in LG LW20 Implemented LG LW20 5.1 support based on ALC880 threestack modes. It was previously set to use one DAC and 2 jack modes, giving 2-channel output only. Analog CD playing support is still missing. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4876a31311e..8deca0f7d7ce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1900,17 +1900,38 @@ static struct hda_input_mux alc880_lg_lw_capture_source = { }, }; +#define alc880_lg_lw_modes alc880_threestack_modes + static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, { } /* end */ }; static struct hda_verb alc880_lg_lw_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ + /* set capture source to mic-in */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1920,7 +1941,6 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* HP-out */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* mic-in to input */ @@ -2864,11 +2884,11 @@ static struct alc_config_preset alc880_presets[] = { .mixers = { alc880_lg_lw_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_lg_lw_init_verbs }, - .num_dacs = 1, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), .dac_nids = alc880_dac_nids, .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, + .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes), + .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, .unsol_event = alc880_lg_lw_unsol_event, .init_hook = alc880_lg_lw_automute, From 83c34218a867085f7108438408c8323b46dbfc4a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 5 Jul 2007 11:43:05 +0200 Subject: [PATCH 061/102] [ALSA] hda-codec - Some fixes for Realtek codec supports - Added BenQ T31-16 AMD Turion64x2 (ALC262) support - Added models for ASUS machines with ALC861 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8deca0f7d7ce..2d35b4cfb2b8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,6 +94,7 @@ enum { ALC262_HP_BPC_D7000_WF, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, + ALC262_BENQ_T31, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -6888,7 +6889,16 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; - +static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { } /* end */ +}; #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -7220,6 +7230,15 @@ static struct hda_verb alc262_EAPD_verbs[] = { {} }; +static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3050}, + {} +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -7641,6 +7660,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), + SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -7745,6 +7765,17 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + }, + [ALC262_BENQ_T31] = { + .mixers = { alc262_benq_t31_mixer }, + .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_hippo_unsol_event, }, }; @@ -9311,7 +9342,9 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), + SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), + SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), From 067b5a844c7b9703b0bdcd8cf003a621734c3bae Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 5 Jul 2007 12:35:15 +0200 Subject: [PATCH 062/102] [ALSA] hda-codec - Fix ALC662 auto mode Fixed ALC662 auto mode, wrong NIDs were chosen. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d35b4cfb2b8..acaf8c39eea6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11013,7 +11013,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, for (i = 0; i < cfg->line_outs; i++) { if (!spec->multiout.dac_nids[i]) continue; - nid = alc880_idx_to_dac(i); + nid = alc880_idx_to_mixer(i); if (i == 2) { /* Center/LFE */ err = add_control(spec, ALC_CTL_WIDGET_VOL, From 1624cb9a35d4276c8ea0d939d580a6945ac72ac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Jul 2007 13:10:51 +0200 Subject: [PATCH 063/102] [ALSA] hda-codec - Fix the recording selection on VAIO laptop Fix the mismatch of init verb and capture mux with VAIO STAC9872. Also, changed the item names to be more intuitive. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e89180c6aaa2..f20b4faf6ccb 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2332,8 +2332,8 @@ static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ - { "Line", 0x1 }, - { "Mic", 0x2 }, + { "Mic Jack", 0x1 }, + { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; @@ -2344,7 +2344,7 @@ static struct hda_verb vaio_init[] = { {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ @@ -2360,7 +2360,7 @@ static struct hda_verb vaio_ar_init[] = { {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ /* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ /* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ From cd1e3b40429c172b29c04a9c777be2ca1ac012e0 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 6 Jul 2007 12:10:01 +0200 Subject: [PATCH 064/102] [ALSA] hda-codec - Fix HP Nettle 5.1 output Fixed HP Nettle 6ch/8ch channel routing based on ALC883 sixstack and swapping rear and center/LFE in verbs and mixer settings. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 65 ++++++++++++++++++++++++++++++++++- 1 file changed, 64 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index acaf8c39eea6..33c6fafd358f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -172,6 +172,7 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, + ALC888_HP_NETTLE, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6024,6 +6025,46 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6155,6 +6196,14 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_hp_nettle_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ + { } +}; + /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { @@ -6397,6 +6446,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", + [ALC888_HP_NETTLE] = "hp-nettle", [ALC883_AUTO] = "auto", }; @@ -6431,7 +6481,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6616,6 +6666,19 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, }, + [ALC888_HP_NETTLE] = { + .mixers = { alc888_hp_nettle_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_hp_nettle_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, }; From 8341de60c76c400eaa453c91810ba6995d2fdb57 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 6 Jul 2007 12:10:45 +0200 Subject: [PATCH 065/102] [ALSA] hda-codec - Add HP Lucknow 5.1 support Added HP Lucknow (ALC888) 5.1 surround support based on ALC883 6-channel threestack with changes in channel routing. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 81 +++++++++++++++++++++++++++++++++++ 1 file changed, 81 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33c6fafd358f..993b503a0538 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -173,6 +173,7 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC888_HP_NETTLE, + ALC888_HP_LUCKNOW, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6065,6 +6066,44 @@ static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_hp_lucknow_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6204,6 +6243,34 @@ static struct hda_verb alc888_hp_nettle_verbs[] = { { } }; +static struct hda_verb alc888_hp_lucknow_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + { } +}; + +static struct hda_verb alc888_hp_lucknow_2ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } +}; + +static struct hda_verb alc888_hp_lucknow_6ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } +}; + +static struct hda_channel_mode alc888_hp_lucknow_modes[2] = { + { 2, alc888_hp_lucknow_2ch_init }, + { 6, alc888_hp_lucknow_6ch_init }, +}; + /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { @@ -6447,6 +6514,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC888_HP_NETTLE] = "hp-nettle", + [ALC888_HP_LUCKNOW] = "hp-lucknow", [ALC883_AUTO] = "auto", }; @@ -6482,6 +6550,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6679,6 +6748,18 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC888_HP_LUCKNOW] = { + .mixers = { alc888_hp_lucknow_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_hp_lucknow_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc888_hp_lucknow_modes), + .channel_mode = alc888_hp_lucknow_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, }; From cdb0e9a7af3ad34f3f879fda9251039c4cad3403 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 6 Jul 2007 12:11:22 +0200 Subject: [PATCH 066/102] [ALSA] Update Claudio's CREDITS information Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- CREDITS | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/CREDITS b/CREDITS index 79fd13dbb8e4..10c214dc95e7 100644 --- a/CREDITS +++ b/CREDITS @@ -2212,13 +2212,13 @@ S: 2300 Copenhagen S S: Denmark N: Claudio S. Matsuoka -E: claudio@conectiva.com -E: claudio@helllabs.org +E: cmatsuoka@gmail.com +E: claudio@mandriva.com W: http://helllabs.org/~claudio -D: V4L, OV511 driver hacks +D: V4L, OV511 and HDA-codec hacks S: Conectiva S.A. -S: R. Tocantins 89 -S: 80050-430 Curitiba PR +S: Souza Naves 1250 +S: 80050-040 Curitiba PR S: Brazil N: Heinz Mauelshagen From 0f40502e0b5300a37c4cfc7bf93f84c2702dbbe9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 12:24:11 +0200 Subject: [PATCH 067/102] [ALSA] hda-codec - Add missing model parameter strings for ALC262 Added the missing model option values for ALC262. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 993b503a0538..2f074ba57480 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7778,7 +7778,8 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", - [ALC262_BENQ_ED8] = "sony-assamd", + [ALC262_BENQ_T31] = "benq-t31", + [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_AUTO] = "auto", }; From 940d244851f8a204b5aedae678d9ab96c20b2fec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 12:24:55 +0200 Subject: [PATCH 068/102] [ALSA] Add missing model entries for HD-audio codecs Added missing model entries for HD-audio codecs in the module option list. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index b967dc850c60..85596ede7e11 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -809,6 +809,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hp-3013 HP machines (3013-variant) fujitsu Fujitsu S7020 acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V basic fixed pin assignment (old default model) auto auto-config reading BIOS (default) @@ -817,8 +819,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hp-bpc HP xw4400/6400/8400/9400 laptops hp-bpc-d7000 HP BPC D7000 benq Benq ED8 + benq-t31 Benq T31 hippo Hippo (ATI) with jack detection, Sony UX-90s hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) @@ -826,6 +830,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack model auto auto-config reading BIOS (default) + ALC662 + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 6stack-dig 6-stack with SPDIF + lenovo-101e Lenovo laptop + auto auto-config reading BIOS (default) + ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O @@ -842,9 +854,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) medion Medion Laptops + medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + hp-nettle HP Nettle + hp-lucknow HP Lucknow auto auto-config reading BIOS (default) ALC861/660 @@ -865,6 +883,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-660 3-jack (for ALC660VD) 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 + dallas Dallas laptops auto auto-config reading BIOS (default) CMI9880 From 1a7fa543cb5d8185d293a03842b33136d8dcc04c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 12:27:25 +0200 Subject: [PATCH 069/102] [ALSA] Fix substream to check in PCM drain The check of a substream in snd_pcm_drain() might not be always correct since runtime can point a different substream (although the PCM state of each linked substream should be same, in theory). This patch fixes it. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a96733a5beb8..59b29cd482ae 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1487,7 +1487,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); /* pre-start/stop - all running streams are changed to DRAINING state */ From 47a2327eac01fa944ab935fc5bb03c3032b2ec37 Mon Sep 17 00:00:00 2001 From: "Robert P. J. Day" Date: Fri, 6 Jul 2007 13:27:16 +0200 Subject: [PATCH 070/102] [ALSA] Remove unreferenced header file include/sound/wavefront_fx.h Signed-off-by: Robert P. J. Day Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/wavefront_fx.h | 9 --------- 1 file changed, 9 deletions(-) delete mode 100644 include/sound/wavefront_fx.h diff --git a/include/sound/wavefront_fx.h b/include/sound/wavefront_fx.h deleted file mode 100644 index cec92b141796..000000000000 --- a/include/sound/wavefront_fx.h +++ /dev/null @@ -1,9 +0,0 @@ -#ifndef __SOUND_WAVEFRONT_FX_H -#define __SOUND_WAVEFRONT_FX_H - -extern int snd_wavefront_fx_detect (snd_wavefront_t *); -extern void snd_wavefront_fx_ioctl (snd_synth_t *sdev, - unsigned int cmd, - unsigned long arg); - -#endif __SOUND_WAVEFRONT_FX_H From a4eed138add1018846d17e813560b0c7c0ae8e01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 18:17:04 +0200 Subject: [PATCH 071/102] [ALSA] hda-codec - Fix number of pin widgets with STAC codecs Use ARRAY_SIZE() to calculate the number of pin widgets for each codec chip. This also fixes a typo in STAC9205 code that gives a wrong number (14 instead of 12), which resulted in fatal communication errors. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f20b4faf6ccb..0dccba231e8b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1958,7 +1958,7 @@ static int patch_stac9200(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, @@ -2008,7 +2008,7 @@ static int patch_stac925x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS, stac925x_models, @@ -2080,7 +2080,7 @@ static int patch_stac922x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 10; + spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, @@ -2181,7 +2181,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, stac927x_models, @@ -2266,7 +2266,7 @@ static int patch_stac9205(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS, stac9205_models, From 43bbb6ccc67775b432c31d5fd2e1050329ffdcbf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 20:22:05 +0200 Subject: [PATCH 072/102] [ALSA] hda-intel - Show the last command in warning messages Show the last issued command in warning messages when any communication error with CORB/RIRB occurs. Also, a bit code clean-up for composing the command value in *_send_cmd(). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 45 ++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c78ff901a572..92bc8b3fa2a0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -341,6 +341,9 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + + /* for debugging */ + unsigned int last_cmd; /* last issued command (to sync) */ }; /* driver types */ @@ -466,18 +469,10 @@ static void azx_free_cmd_io(struct azx *chip) } /* send a command */ -static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int para) +static int azx_corb_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; unsigned int wp; - u32 val; - - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; /* add command to corb */ wp = azx_readb(chip, CORBWP); @@ -543,7 +538,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) if (chip->msi) { snd_printk(KERN_WARNING "hda_intel: No response from codec, " - "disabling MSI...\n"); + "disabling MSI: last cmd=0x%08x\n", chip->last_cmd); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -555,13 +550,15 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) if (!chip->polling_mode) { snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " - "switching to polling mode...\n"); + "switching to polling mode: last cmd=0x%08x\n", + chip->last_cmd); chip->polling_mode = 1; goto again; } snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode...\n"); + "switching to single_cmd mode: last cmd=0x%08x\n", + chip->last_cmd); chip->rirb.rp = azx_readb(chip, RIRBWP); chip->rirb.cmds = 0; /* switch to single_cmd mode */ @@ -581,20 +578,11 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) */ /* send a command */ -static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, - unsigned int para) +static int azx_single_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; - u32 val; int timeout = 50; - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; - while (timeout--) { /* check ICB busy bit */ if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) { @@ -639,10 +627,19 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, unsigned int para) { struct azx *chip = codec->bus->private_data; + u32 val; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= para; + chip->last_cmd = val; + if (chip->single_cmd) - return azx_single_send_cmd(codec, nid, direct, verb, para); + return azx_single_send_cmd(codec, val); else - return azx_corb_send_cmd(codec, nid, direct, verb, para); + return azx_corb_send_cmd(codec, val); } /* get a response */ From d20b9b20e5f3bb82a6f38583154cb2f163d465d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jul 2007 10:36:22 +0200 Subject: [PATCH 073/102] [ALSA] aica - fix behaviour in poor resource conditions Fix behaviour in poor resource conditions. With this new patch, tidied with indent and with SNDRV_PCM_TRIGGER_STOP being used. Signed-off by: Adrian McMenamin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/sh/aica.c | 82 ++++++++++++++++++++++--------------------------- sound/sh/aica.h | 1 + 2 files changed, 37 insertions(+), 46 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 97bb86a58622..739786529ca5 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -64,12 +64,6 @@ module_param(enable, bool, 0644); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); /* Use workqueue */ - -static struct spu_work_holder { - struct work_struct spu_dma_work; - void *sspointer; -} spu_working; - static struct workqueue_struct *aica_queue; /* Simple platform device */ @@ -100,9 +94,9 @@ static void spu_write_wait(void) break; /* To ensure hardware failure doesn't wedge kernel */ time_count++; - if (time_count > 0x10000) - { - snd_printk("WARNING: G2 FIFO appears to be blocked.\n"); + if (time_count > 0x10000) { + snd_printk + ("WARNING: G2 FIFO appears to be blocked.\n"); break; } } @@ -226,11 +220,11 @@ static int aica_dma_transfer(int channels, int buffer_size, runtime = substream->runtime; for (q = 0; q < channels; q++) { err = dma_xfer(AICA_DMA_CHANNEL, - (unsigned long)(runtime->dma_area + - (AICA_BUFFER_SIZE * q) / - channels + - AICA_PERIOD_SIZE * - period_offset), + (unsigned long) (runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + AICA_PERIOD_SIZE * period_offset, buffer_size / channels, AICA_DMA_MODE); @@ -244,26 +238,25 @@ static int aica_dma_transfer(int channels, int buffer_size, static void startup_aica(struct snd_card_aica *dreamcastcard) { spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, - dreamcastcard->channel, - sizeof(struct aica_channel)); + dreamcastcard->channel, sizeof(struct aica_channel)); aica_chn_start(); } static void run_spu_dma(struct work_struct *work) { int buffer_size; - struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_card_aica *dreamcastcard; - struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); - substream = holder-> sspointer; - dreamcastcard = substream->pcm->private_data; - runtime = substream->runtime; + dreamcastcard = + container_of(work, struct snd_card_aica, spu_dma_work); + runtime = dreamcastcard->substream->runtime; if (unlikely(dreamcastcard->dma_check == 0)) { - buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + buffer_size = + frames_to_bytes(runtime, runtime->buffer_size); if (runtime->channels > 1) dreamcastcard->channel->flags |= 0x01; - aica_dma_transfer(runtime->channels, buffer_size, substream); + aica_dma_transfer(runtime->channels, buffer_size, + dreamcastcard->substream); startup_aica(dreamcastcard); dreamcastcard->clicks = buffer_size / (AICA_PERIOD_SIZE * runtime->channels); @@ -271,13 +264,11 @@ static void run_spu_dma(struct work_struct *work) } else { aica_dma_transfer(runtime->channels, AICA_PERIOD_SIZE * runtime->channels, - substream); + dreamcastcard->substream); snd_pcm_period_elapsed(dreamcastcard->substream); dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) - { dreamcastcard->clicks %= AICA_PERIOD_NUMBER; - } mod_timer(&dreamcastcard->timer, jiffies + 1); } } @@ -289,7 +280,7 @@ static void aica_period_elapsed(unsigned long timer_var) struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream; struct snd_card_aica *dreamcastcard; - substream = (struct snd_pcm_substream *)timer_var; + substream = (struct snd_pcm_substream *) timer_var; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -307,27 +298,24 @@ static void aica_period_elapsed(unsigned long timer_var) dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(spu_working.spu_dma_work)); + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); } static void spu_begin_dma(struct snd_pcm_substream *substream) { - /* Must be atomic */ struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; - /* Use queue to do the heavy lifting */ - spu_working.sspointer = substream; - INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); - queue_work(aica_queue, &(spu_working.spu_dma_work)); + /*get the queue to do the work */ + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); return; } init_timer(&(dreamcastcard->timer)); - dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.data = (unsigned long) substream; dreamcastcard->timer.function = aica_period_elapsed; dreamcastcard->timer.expires = jiffies + 4; add_timer(&(dreamcastcard->timer)); @@ -366,7 +354,9 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; - del_timer(&dreamcastcard->timer); + flush_workqueue(aica_queue); + if (dreamcastcard->timer.data) + del_timer(&dreamcastcard->timer); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -385,7 +375,8 @@ static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream { /* Allocate a DMA buffer using ALSA built-ins */ return - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); } static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream @@ -402,15 +393,11 @@ static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); aica_chn_halt(); break; default: @@ -444,7 +431,8 @@ static int __init snd_aicapcmchip(struct snd_card_aica int err; /* AICA has no capture ability */ err = - snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, + &pcm); if (unlikely(err < 0)) return err; pcm->private_data = dreamcastcard; @@ -524,9 +512,7 @@ static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; dreamcastcard->master_volume = ucontrol->value.integer.value[0]; spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, - dreamcastcard->channel, - sizeof(struct aica_channel)); - + dreamcastcard->channel, sizeof(struct aica_channel)); return 1; } @@ -610,6 +596,8 @@ static int __init snd_aica_probe(struct platform_device *devptr) strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strcpy(dreamcastcard->card->longname, "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Prepare to use the queue */ + INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) @@ -663,8 +651,10 @@ static int __init aica_init(void) static void __exit aica_exit(void) { - /* Destroy the aica kernel thread */ - destroy_workqueue(aica_queue); + /* Destroy the aica kernel thread * + * being extra cautious to check if it exists*/ + if (likely(aica_queue)) + destroy_workqueue(aica_queue); platform_device_unregister(pd); platform_driver_unregister(&snd_aica_driver); /* Kill any sound still playing and reset ARM7 to safe state */ diff --git a/sound/sh/aica.h b/sound/sh/aica.h index 0603b5b0376b..8c11e3d10a50 100644 --- a/sound/sh/aica.h +++ b/sound/sh/aica.h @@ -69,6 +69,7 @@ struct aica_channel { }; struct snd_card_aica { + struct work_struct spu_dma_work; struct snd_card *card; struct aica_channel *channel; struct snd_pcm_substream *substream; From f167684a6454183d41588982ed26e4c6ee6feb48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jul 2007 10:39:44 +0200 Subject: [PATCH 074/102] [ALSA] usbaudio - Add quirk for Roland EXR series Added a quirk for Roland EXR series. Reported by Andre Coetzee. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 8fca24e2635c..f88a72ccd122 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1060,7 +1060,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_STANDARD_INTERFACE } }, - /* TODO: add Roland EXR support */ +{ + USB_DEVICE(0x0582, 0x0060), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "EXR Series", + .ifnum = 0, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, { /* has ID 0x0067 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0065), From c34532da91c208bd5567419984ea6a4bf60dad60 Mon Sep 17 00:00:00 2001 From: George Shapovalov Date: Mon, 9 Jul 2007 10:42:35 +0200 Subject: [PATCH 075/102] [ALSA] usb-audio: another Logitech camera/microphone ID match From: George Shapovalov Another case of ALSA ticket #3040: we need an explicit USB ID match for another Logitech camera/microphone device. Signed-off-by: Daniel Drake Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f88a72ccd122..06392914b9f8 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -52,6 +52,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08ae, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | From 341d4eb0cffdcd5fbdbd696dc26671f91e7c418c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jul 2007 17:53:18 +0200 Subject: [PATCH 076/102] [ALSA] hda-codec - Fix Toshiba A100 with ALC861 One of the newly added model entries causes a regression on Toshiba A100. It works fine with model=auto. So, let's deselect the entry again. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f074ba57480..804276685324 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9491,7 +9491,10 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), + /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) + * Any other models that need this preset? + */ + /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), From f95474ec05106a2af46220fbf8624adc17fcb3e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jul 2007 00:47:43 +0200 Subject: [PATCH 077/102] [ALSA] hda-codec - Fix default pin config of Abit AW9D-MAX Fix the default pin configuration of Abit AW9D-MAX. The board has a broken BIOS that doesn't set the correct pin default configs, which screws up the auto-configuration of snd-hda-intel driver. The patch enables the override of default pin config values. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 57 +++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 804276685324..d7eb22f5ca38 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -723,6 +723,38 @@ static void alc_subsystem_id(struct hda_codec *codec, } } +/* + * Fix-up pin default configurations + */ + +struct alc_pincfg { + hda_nid_t nid; + u32 val; +}; + +static void alc_fix_pincfg(struct hda_codec *codec, + const struct snd_pci_quirk *quirk, + const struct alc_pincfg **pinfix) +{ + const struct alc_pincfg *cfg; + + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (!quirk) + return; + + cfg = pinfix[quirk->value]; + for (; cfg->nid; cfg++) { + int i; + u32 val = cfg->val; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, cfg->nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + val & 0xff); + val >>= 8; + } + } +} + /* * ALC880 3-stack model * @@ -5409,6 +5441,29 @@ static struct alc_config_preset alc882_presets[] = { }; +/* + * Pin config fixes + */ +enum { + PINFIX_ABIT_AW9D_MAX +}; + +static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } +}; + +static const struct alc_pincfg *alc882_pin_fixes[] = { + [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +}; + +static struct snd_pci_quirk alc882_pinfix_tbl[] = { + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + {} +}; + /* * BIOS auto configuration */ @@ -5532,6 +5587,8 @@ static int patch_alc882(struct hda_codec *codec) } } + alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); From 706568967c7d99a5975b2a21aacd791dc838ea8c Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 10 Jul 2007 11:34:32 +0200 Subject: [PATCH 078/102] [ALSA] fix SND_CS5530=y, ISA=n compilation This patch fixes the following compile error with CONFIG_SND_CS5530=y, CONFIG_ISA=n: <-- snip --> ... LD .tmp_vmlinux1 sound/built-in.o: In function 'snd_sb16_capture_trigger': sb16_main.c:(.text+0x212ad): undefined reference to 'snd_sbdsp_command' sb16_main.c:(.text+0x212cc): undefined reference to 'snd_sbdsp_command' sb16_main.c:(.text+0x212f7): undefined reference to 'snd_sbdsp_command' sound/built-in.o: In function 'snd_sb16_playback_trigger': sb16_main.c:(.text+0x2136d): undefined reference to 'snd_sbdsp_command' sb16_main.c:(.text+0x2138c): undefined reference to 'snd_sbdsp_command' ... <-- snip --> Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90ddabae2e9..ea5084abe60f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -1,8 +1,5 @@ # ALSA ISA drivers -menu "ISA devices" - depends on SND!=n && ISA && ISA_DMA_API - config SND_AD1848_LIB tristate select SND_PCM @@ -24,6 +21,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +menu "ISA devices" + depends on SND!=n && ISA && ISA_DMA_API + config SND_ADLIB tristate "AdLib FM card" depends on SND From cfc24ec89b488eeb5dfe4189761205fca38eb58e Mon Sep 17 00:00:00 2001 From: Ritesh Raj Sarraf Date: Tue, 10 Jul 2007 13:03:57 +0200 Subject: [PATCH 079/102] [ALSA] USB ID for intergrated quickcam on dell xps m1210 This patch adds Logitech QuickCam USB ID for Dell's XPS M1210 notebooks. Signed-off-by: Ritesh Raj Sarraf Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 06392914b9f8..32de6c3f4ddd 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -61,6 +61,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08c6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | From 2dcd522f89d17936d08665c22ab0ee415b8c5d56 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 10 Jul 2007 17:04:57 +0200 Subject: [PATCH 080/102] [ALSA] HDA-Intel - Add support for MSI K9AGM2-FIH motherboard This patch adds support for the MSI K9AGM2-FIH on-board audio. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_atihdmi.c | 1 + sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index f8eb4c90f801..72d3ab9751ac 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -172,6 +172,7 @@ static int patch_atihdmi(struct hda_codec *codec) */ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, + { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R600 HDMI", .patch = patch_atihdmi }, {} /* terminator */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d7eb22f5ca38..b145667de17c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6586,6 +6586,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), From be38114a49853e441e694d690e631937de5fc524 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Wed, 11 Jul 2007 10:31:08 +0200 Subject: [PATCH 081/102] [ALSA] hda-codec - Add VIA HDA to si3054 Add VIA HDA identification to use the HDA-based Motorola modem of the Clevo m540 laptop. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_si3054.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43f537ef40bf..5dd9f61b7cba 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -304,6 +304,8 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, + /* VIA HDA on Clevo m540 */ + { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, /* Asus A8J Modem (SM56) */ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, {} From f8c78b82b9e47d933629e7060e4a83a277b15ffe Mon Sep 17 00:00:00 2001 From: Thibault Le Meur Date: Thu, 12 Jul 2007 11:26:35 +0200 Subject: [PATCH 082/102] [ALSA] usb-audio - Fix audiophile-USB quirk for little-endian Audiophile-usb fix (corrects little-endianness in 16bit modes, resets interfaces at device initialization, and updates the documentation). Signed-off-by: Thibault Le Meur Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/Audiophile-Usb.txt | 226 +++++++++++++------- sound/usb/usbaudio.c | 9 +- 2 files changed, 157 insertions(+), 78 deletions(-) diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index e40cce83327c..5b7a5487d505 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.4 ======================================================== Thibault Le Meur @@ -6,8 +6,17 @@ This document is a guide to using the M-Audio Audiophile USB (tm) device with ALSA and JACK. +History +======= +* v1.4 - Thibault Le Meur (2007-07-11) + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal + - Modifying document structure + + 1 - Audiophile USB Specs and correct usage ========================================== + This part is a reminder of important facts about the functions and limitations of the device. @@ -25,18 +34,18 @@ The device has 4 audio interfaces, and 2 MIDI ports: The internal DAC/ADC has the following characteristics: * sample depth of 16 or 24 bits * sample rate from 8kHz to 96kHz -* Two ports can't use different sample depths at the same time. Moreover, the -Audiophile USB documentation gives the following Warning: "Please exit any -audio application running before switching between bit depths" +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be activated at the same time depending on the audio mode selected: - * 16-bit/48kHz ==> 4 channels in/4 channels out + * 16-bit/48kHz ==> 4 channels in + 4 channels out - Ai+Ao+Di+Do - * 24-bit/48kHz ==> 4 channels in/2 channels out, - or 2 channels in/4 channels out + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do - * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only) + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) - Ai or Ao or Di or Do Important facts about the Digital interface: @@ -52,44 +61,53 @@ source is connected synchronization error (for instance sound played at an odd sample rate) -2 - Audiophile USB support in ALSA -================================== +2 - Audiophile USB MIDI support in ALSA +======================================= -2.1 - MIDI ports ----------------- -The Audiophile USB MIDI ports will be automatically supported once the +The Audiophile USB MIDI ports will be automatically supported once the following modules have been loaded: * snd-usb-audio * snd-seq-midi No additional setting is required. -2.2 - Audio ports ------------------ + +3 - Audiophile USB Audio support in ALSA +======================================== Audio functions of the Audiophile USB device are handled by the snd-usb-audio module. This module can work in a default mode (without any device-specific parameter), or in an "advanced" mode with the device-specific parameter called "device_setup". -2.2.1 - Default Alsa driver mode +3.1 - Default Alsa driver mode +------------------------------ -The default behavior of the snd-usb-audio driver is to parse the device -capabilities at startup and enable all functions inside the device (including -all ports at any supported sample rates and sample depths). This approach -has the advantage to let the driver easily switch from sample rates/depths -automatically according to the need of the application claiming the device. +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. -In this case the Audiophile ports are mapped to alsa pcm devices in the -following way (I suppose the device's index is 1): +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): * hw:1,0 is Ao in playback and Di in capture * hw:1,1 is Do in playback and Ai in capture * hw:1,2 is Do in AC3/DTS passthrough mode -You must note as well that the device uses Big Endian byte encoding so that -supported audio format are S16_BE for 16-bit depth modes and S24_3BE for -24-bits depth mode. One exception is the hw:1,2 port which is Little Endian -compliant and thus uses S16_LE. +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. One exception is the hw:1,2 port which is reported +to be Little Endian compliant (supposedly supporting S16_LE) but processes +in fact only S16_BE streams. Examples: * playing a S24_3BE encoded raw file to the Ao port @@ -99,21 +117,23 @@ Examples: * playing a S16_BE encoded raw file to the Do port % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw -If you're happy with the default Alsa driver setup and don't experience any +If you're happy with the default Alsa driver mode and don't experience any issue with this mode, then you can skip the following chapter. -2.2.2 - Advanced module setup +3.2 - Advanced module setup +--------------------------- Due to the hardware constraints described above, the device initialization made by the Alsa driver in default mode may result in a corrupted state of the device. For instance, a particularly annoying issue is that the sound captured -from the Ai port sounds distorted (as if boosted with an excessive high volume -gain). +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). For people having this problem, the snd-usb-audio module has a new module -parameter called "device_setup". +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) -2.2.2.1 - Initializing the working mode of the Audiophile USB +3.2.1 - Initializing the working mode of the Audiophile USB As far as the Audiophile USB device is concerned, this value let the user specify: @@ -121,33 +141,57 @@ specify: * the sample rate * whether the Di port is used or not -Here is a list of supported device_setup values for this device: - * device_setup=0x00 (or omitted) - - Alsa driver default mode - - maintains backward compatibility with setups that do not use this - parameter by not introducing any change - - results sometimes in corrupted sound as described earlier +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + * device_setup=0x01 - 16bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x11 - 16bits 48kHz mode with Di enabled - Ai,Ao,Di,Do can be used at the same time - hw:1,0 is available in capture mode - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + * device_setup=0x09 - 24bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x19 - 24bits 48kHz mode with Di enabled - 3 ports from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in capture mode and an active digital source must be connected to Di - hw:1,2 is not available + * device_setup=0x0D or 0x10 - 24bits 96kHz mode - Di is enabled by default for this mode but does not need to be connected @@ -155,34 +199,61 @@ Here is a list of supported device_setup values for this device: - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in captured mode - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +This mode is untested, I have no AC3 compliant device to test it. I uses: + * device_setup=0x03 - 16bits 48kHz mode with only the Do port enabled - AC3 with DTS passthru (not tested) - Caution with this setup the Do port is mapped to the pcm device hw:1,0 -2.2.2.2 - Setting and switching configurations with the device_setup parameter +3.2.2 - How to use the device_setup parameter +---------------------------------------------- The parameter can be given: + * By manually probing the device (as root): # modprobe -r snd-usb-audio # modprobe snd-usb-audio index=1 device_setup=0x09 + * Or while configuring the modules options in your modules configuration file - For Fedora distributions, edit the /etc/modprobe.conf file: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 -IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: -------------------------------------------- - * You may need to _first_ initialize the module with the correct device_setup - parameter and _only_after_ turn on the Audiophile USB device - * This is especially true when switching the sample depth: +CAUTION when initializaing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead in a misconfiguration of the device. In this case + turn off the device, unproble the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: - first turn off the device - de-register the snd-usb-audio module (modprobe -r) - change the device_setup parameter by changing the device_setup option in /etc/modprobe.conf - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. -2.2.2.3 - Audiophile USB's device_setup structure +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure If you want to understand the device_setup magic numbers for the Audiophile USB, you need some very basic understanding of binary computation. However, @@ -228,12 +299,12 @@ Caution: - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll only be able to use one at the same time -2.2.3 - USB implementation details for this device +3.2.3.2 - USB implementation details for this device You may safely skip this section if you're not interested in driver -development. +hacking. -This section describes some internal aspects of the device and summarize the +This section describes some internal aspects of the device and summarizes the data I got by usb-snooping the windows and Linux drivers. The M-Audio Audiophile USB has 7 USB Interfaces: @@ -293,43 +364,45 @@ parse_audio_endpoints function uses a quirk called "audiophile_skip_setting_quirk" in order to prevent AltSettings not corresponding to device_setup from being registered in the driver. -3 - Audiophile USB and Jack support +4 - Audiophile USB and Jack support =================================== This section deals with support of the Audiophile USB device in Jack. -The main issue regarding this support is that the device is Big Endian -compliant. -3.1 - Using the plug alsa plugin --------------------------------- +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels -Jack doesn't directly support big endian devices. Thus, one way to have support -for this device with Alsa is to use the Alsa "plug" converter. +4.1 - Direct support in Jackd +----------------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +extacly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. For instance here is one way to run Jack with 2 playback channels on Ao and 2 capture channels from Ai: % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 - However you may see the following warning message: "You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer." -3.2 - Patching alsa to use direct pcm device --------------------------------------------- -A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. -However it has not been included in the CVS tree. - -You can find it at the following URL: -http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687& -atid=425939 - -After having applied the patch you can run jackd with the following command -line: - % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 - -3.2 - Getting 2 input and/or output interfaces in Jack +4.3 - Getting 2 input and/or output interfaces in Jack ------------------------------------------------------ As you can see, starting the Jack server this way will only enable 1 stereo @@ -339,6 +412,7 @@ This is due to the following restrictions: * Jack can only open one capture device and one playback device at a time * The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 (and optionally hw:1,2) + If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to combine the Alsa devices into one logical "complex" device. @@ -348,13 +422,11 @@ It is related to another device (ice1712) but can be adapted to suit the Audiophile USB. Enabling multiple Audiophile USB interfaces for Jackd will certainly require: -* patching Jack with the previously mentioned "Big Endian" patch -* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page) -* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) * define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc file * start jackd with this device -I had no success in testing this for now, but this may be due to my OS -configuration. If you have any success with this kind of setup, please -drop me an email. +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8ebc1adb5ed9..834b0aff5ec1 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2350,7 +2350,9 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * return 1; break; case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - return 1; + if (device_setup[chip->index] == 0x00 || + fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3) + return 1; } return 0; } @@ -3251,6 +3253,11 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { + /* Reset ALL ifaces to 0 altsetting. + * Call it for every possible altsetting of every interface. + */ + usb_set_interface(chip->dev, iface, 0); + if (device_setup[chip->index] & AUDIOPHILE_SET) { if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS) && altno != 6) From ef2ec0dd65bc542f9ab45ea2fac6920cb1afa13b Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Thu, 12 Jul 2007 11:27:37 +0200 Subject: [PATCH 083/102] [ALSA] hda-codec - Add LG LW20 si3054 modem id Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_si3054.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 5dd9f61b7cba..6d2ecc38905c 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -308,6 +308,8 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, /* Asus A8J Modem (SM56) */ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, + /* LG LW20 modem */ + { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 }, {} }; From 82fb159aa3d187aac0fce076739d7d8884e7a0af Mon Sep 17 00:00:00 2001 From: Matt Reimer Date: Thu, 12 Jul 2007 12:27:24 +0200 Subject: [PATCH 084/102] [ALSA] Make s3c24xx_i2s_set_clkdiv() change the correct bits Make s3c24xx_i2s_set_clkdiv() change the correct bits. Signed-off-by: Matt Reimer Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 8ca314dc8891..39f02462e07d 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, DBG("Entered %s\n", __FUNCTION__); switch (div_id) { - case S3C24XX_DIV_MCLK: + case S3C24XX_DIV_BCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; - case S3C24XX_DIV_BCLK: + case S3C24XX_DIV_MCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; From 1b290a510c12be91b6a2f7d6a07ef61b8ee9f06a Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 12 Jul 2007 15:17:34 +0200 Subject: [PATCH 085/102] [ALSA] hda: stac9202 mixer fix Added support for line out controls for STAC9202 cards, and fixed issue where master mixer control was being created twice for headphone and speaker outs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0dccba231e8b..be53e9685bc0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -334,8 +334,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -1401,7 +1399,15 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2", From cac19c3ba123369b7c1199e20bb01fb265004ff5 Mon Sep 17 00:00:00 2001 From: Thibault Le Meur Date: Fri, 13 Jul 2007 11:50:23 +0200 Subject: [PATCH 086/102] [ALSA] usb-audio - Fix AC3 with M-Audio Audiophile USB Fixed AC3 interface in device_setup=0x00 mode thanks to Hakan Lennestal and updated documentation Signed-off-by: Thibault Le Meur Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/Audiophile-Usb.txt | 22 +++++++++++++++------ sound/usb/usbaudio.c | 13 +++++++++++- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 5b7a5487d505..2ad5e6306c44 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.4 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 ======================================================== Thibault Le Meur @@ -12,6 +12,8 @@ History - Added Low Endianness nature of 16bits-modes found by Hakan Lennestal - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info 1 - Audiophile USB Specs and correct usage @@ -105,9 +107,12 @@ way (I suppose the device's index is 1): In this mode, the device uses Big Endian byte-encoding so that supported audio format are S16_BE for 16-bit depth modes and S24_3BE for -24-bits depth mode. One exception is the hw:1,2 port which is reported -to be Little Endian compliant (supposedly supporting S16_LE) but processes -in fact only S16_BE streams. +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. Examples: * playing a S24_3BE encoded raw file to the Ao port @@ -116,6 +121,8 @@ Examples: % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw * playing a S16_BE encoded raw file to the Do port % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw If you're happy with the default Alsa driver mode and don't experience any issue with this mode, then you can skip the following chapter. @@ -205,13 +212,16 @@ mode" above for an aplay command example) 3.2.1.3 - AC3 w/ DTS passthru mode -This mode is untested, I have no AC3 compliant device to test it. I uses: +Thanks to Hakan Lennestal, I now have a report saying that this mode works. * device_setup=0x03 - 16bits 48kHz mode with only the Do port enabled - - AC3 with DTS passthru (not tested) + - AC3 with DTS passthru - Caution with this setup the Do port is mapped to the pcm device hw:1,0 +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + 3.2.2 - How to use the device_setup parameter ---------------------------------------------- diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 834b0aff5ec1..7bd5852fcc0d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2532,7 +2532,18 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * * but we give normal PCM format to get the existing * apps working... */ - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + switch (chip->usb_id) { + + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + if (device_setup[chip->index] == 0x00 && + fp->altsetting == 6) + pcm_format = SNDRV_PCM_FORMAT_S16_BE; + else + pcm_format = SNDRV_PCM_FORMAT_S16_LE; + break; + default: + pcm_format = SNDRV_PCM_FORMAT_S16_LE; + } } else { pcm_format = parse_audio_format_i_type(chip, fp, format, fmt); if (pcm_format < 0) From e4f41da9e5f03e67ebe6067aad40c6a79443c3ae Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 11:51:06 +0200 Subject: [PATCH 087/102] [ALSA] hda-codec - Add LG LW20 line-in capture source Add line-in to the list of LG LW20 capture sources. Also fix the LG LW pin assignment list comment. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b145667de17c..e8782c27a459 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1921,17 +1921,18 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) * Pin assignment: * Speaker-out: 0x14 * Mic-In: 0x18 - * Built-in Mic-In: 0x19 (?) - * HP-Out: 0x1b + * Built-in Mic-In: 0x19 + * Line-In: 0x1b + * HP-Out: 0x1a * SPDIF-Out: 0x1e */ -/* seems analog CD is not working */ static struct hda_input_mux alc880_lg_lw_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, + { "Line In", 0x2 }, }, }; From e87f97a3e842a816f30f087d5bfac68ef2afaef2 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 11:51:32 +0200 Subject: [PATCH 088/102] [ALSA] hda-codec - Add quirk for HP Samba This patch fixes 5.1 surround output for the HP Samba using the same quirk as the HP Lucknow. If HP machines are uniform in their wiring, we should rename ALC888_HP_NETTLE to ALC888_6ST_HP and ALC888_HP_LUCKNOW to ALC888_3ST_HP for generic HP 6stack and 3stack configurations. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 e8782c27a459..fa259a98ea74 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6610,6 +6610,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_HP_LUCKNOW), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; From 4723c022e1960abb5608b37ef61443da7d6296bb Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 14:36:19 +0200 Subject: [PATCH 089/102] [ALSA] hda-codec - Rename HP model-specific quirks Rename ALC888_HP_NETTLE and ALC888_HP_LUCKNOW models to the more generic names ALC888_6ST_HP and ALC888_3ST_HP since HP seems to be consistent in the wiring of their 3stack and 6stack ALC888-based systems. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 4 +- sound/pci/hda/patch_realtek.c | 48 +++++++++---------- 2 files changed, 26 insertions(+), 26 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 85596ede7e11..5bb45cd5f0e6 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -861,8 +861,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 - hp-nettle HP Nettle - hp-lucknow HP Lucknow + 6stack-hp HP machines with 6stack (Nettle boards) + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa259a98ea74..d427c32301ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -172,8 +172,8 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, - ALC888_HP_NETTLE, - ALC888_HP_LUCKNOW, + ALC888_6ST_HP, + ALC888_3ST_HP, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6084,7 +6084,7 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { +static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -6124,7 +6124,7 @@ static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_hp_lucknow_mixer[] = { +static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -6293,7 +6293,7 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_hp_nettle_verbs[] = { +static struct hda_verb alc888_6st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ @@ -6301,14 +6301,14 @@ static struct hda_verb alc888_hp_nettle_verbs[] = { { } }; -static struct hda_verb alc888_hp_lucknow_verbs[] = { +static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; -static struct hda_verb alc888_hp_lucknow_2ch_init[] = { +static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -6316,7 +6316,7 @@ static struct hda_verb alc888_hp_lucknow_2ch_init[] = { { } }; -static struct hda_verb alc888_hp_lucknow_6ch_init[] = { +static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -6324,9 +6324,9 @@ static struct hda_verb alc888_hp_lucknow_6ch_init[] = { { } }; -static struct hda_channel_mode alc888_hp_lucknow_modes[2] = { - { 2, alc888_hp_lucknow_2ch_init }, - { 6, alc888_hp_lucknow_6ch_init }, +static struct hda_channel_mode alc888_3st_hp_modes[2] = { + { 2, alc888_3st_hp_2ch_init }, + { 6, alc888_3st_hp_6ch_init }, }; /* toggle front-jack and RCA according to the hp-jack state */ @@ -6571,8 +6571,8 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", - [ALC888_HP_NETTLE] = "hp-nettle", - [ALC888_HP_LUCKNOW] = "hp-lucknow", + [ALC888_6ST_HP] = "6stack-hp", + [ALC888_3ST_HP] = "3stack-hp", [ALC883_AUTO] = "auto", }; @@ -6608,9 +6608,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), - SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), - SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_HP_LUCKNOW), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6795,9 +6795,9 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, }, - [ALC888_HP_NETTLE] = { - .mixers = { alc888_hp_nettle_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_hp_nettle_verbs }, + [ALC888_6ST_HP] = { + .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -6808,15 +6808,15 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, - [ALC888_HP_LUCKNOW] = { - .mixers = { alc888_hp_lucknow_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_hp_lucknow_verbs }, + [ALC888_3ST_HP] = { + .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), .adc_nids = alc883_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc888_hp_lucknow_modes), - .channel_mode = alc888_hp_lucknow_modes, + .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), + .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, From ef64adbbba1b215e8b523b50ef9d568b4bf2e1e3 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Sat, 14 Jul 2007 00:26:16 +0200 Subject: [PATCH 090/102] [ALSA] hda-codec - Add quirks for HP dx2200/dx2250 HP dx2200 and dx2250 use Micro-Star International (MSI) motherboards (models MS-7254 and MS-7297 respectively) with an ALC862 codec in threestack configuration. Adding this quirk allows correct 5.1 sound output in these systems. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d427c32301ff..28fa99da1916 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9560,6 +9560,8 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), {} }; From c28054d4b31d78272f65c0d11db0796f50fb9569 Mon Sep 17 00:00:00 2001 From: Yoann Padioleau Date: Sat, 14 Jul 2007 00:33:15 +0200 Subject: [PATCH 091/102] [ALSA] snd-aoa-codec-onyx: fix typo Parse error in ifdef or bad use of macro. Signed-off-by: Yoann Padioleau Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/aoa/codecs/snd-aoa-codec-onyx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c index ded516717940..028852374f21 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -661,7 +661,7 @@ static struct transfer_info onyx_transfers[] = { .tag = 2, }, #ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE -Once alsa gets supports for this kind of thing we can add it... + /* Once alsa gets supports for this kind of thing we can add it... */ { /* digital compressed output */ .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE, @@ -713,7 +713,7 @@ static int onyx_prepare(struct codec_info_item *cii, if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) { /* mute and lock analog output */ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); - if (onyx_write_register(onyx + if (onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT)) goto out_unlock; From 7c64ec343a7d7b2d73b19e6bb8d9fd15f0462890 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 16 Jul 2007 11:01:30 +0200 Subject: [PATCH 092/102] [ALSA] timer: check for incorrect device state in non-debug compiles, too Convert the snd_assert()s to simple if()s to prevent crashes when one of the timer instance ioctls is called before the file is bound to a timer device. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/timer.c | 27 ++++++++++++++++++--------- 1 file changed, 18 insertions(+), 9 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 67520b3c0042..f2bbacedd567 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1549,9 +1549,11 @@ static int snd_timer_user_info(struct file *file, int err = 0; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; t = tu->timeri->timer; - snd_assert(t != NULL, return -ENXIO); + if (!t) + return -EBADFD; info = kzalloc(sizeof(*info), GFP_KERNEL); if (! info) @@ -1579,9 +1581,11 @@ static int snd_timer_user_params(struct file *file, int err; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; t = tu->timeri->timer; - snd_assert(t != NULL, return -ENXIO); + if (!t) + return -EBADFD; if (copy_from_user(¶ms, _params, sizeof(params))) return -EFAULT; if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) { @@ -1675,7 +1679,8 @@ static int snd_timer_user_status(struct file *file, struct snd_timer_status status; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; memset(&status, 0, sizeof(status)); status.tstamp = tu->tstamp; status.resolution = snd_timer_resolution(tu->timeri); @@ -1695,7 +1700,8 @@ static int snd_timer_user_start(struct file *file) struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; snd_timer_stop(tu->timeri); tu->timeri->lost = 0; tu->last_resolution = 0; @@ -1708,7 +1714,8 @@ static int snd_timer_user_stop(struct file *file) struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0; } @@ -1718,7 +1725,8 @@ static int snd_timer_user_continue(struct file *file) struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; tu->timeri->lost = 0; return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0; } @@ -1729,7 +1737,8 @@ static int snd_timer_user_pause(struct file *file) struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0; } From 87d483630a4bd5af2beb2155c7ae8c408729a1a6 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 17 Jul 2007 11:52:24 +0200 Subject: [PATCH 093/102] [ALSA] hda: Enable SPDIF in/out on some stac9205 boards Added support for some boards with SPDIF in/out, and cleaned up the GPIO enable function. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 95 ++++++++++++++++++++++------------ 1 file changed, 61 insertions(+), 34 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index be53e9685bc0..9b1d433178bf 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -44,6 +44,7 @@ enum { enum { STAC_9205_REF, + STAC_M43xx, STAC_9205_MODELS }; @@ -218,7 +219,6 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, - }; static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, @@ -773,7 +773,8 @@ static unsigned int ref9205_pin_configs[12] = { }; static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { - ref9205_pin_configs, + [STAC_REF] = ref9205_pin_configs, + [STAC_M43xx] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -784,6 +785,10 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8, + "Dell Precision", STAC_M43xx), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff, + "Dell Precision", STAC_M43xx), {} /* terminator */ }; @@ -813,48 +818,56 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) return 0; } +static void stac92xx_set_config_reg(struct hda_codec *codec, + hda_nid_t pin_nid, unsigned int pin_config) +{ + int i; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + i = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", + pin_nid, i); +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - unsigned int pin_cfg; - if (! spec->pin_nids || ! spec->pin_configs) - return; + if (!spec->pin_configs) + return; - for (i = 0; i < spec->num_pins; i++) { - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - spec->pin_configs[i] & 0x000000ff); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (spec->pin_configs[i] & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (spec->pin_configs[i] & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - spec->pin_configs[i] >> 24); - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); - } + for (i = 0; i < spec->num_pins; i++) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); } -static void stac92xx_enable_eapd(struct hda_codec *codec) +static void stac92xx_enable_gpio_mask(struct hda_codec *codec, + int gpio_mask, int gpio_data) { - /* Configure GPIO0 as output */ + /* Configure GPIOx as output */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ + AC_VERB_SET_GPIO_DIRECTION, gpio_mask); + /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ + /* Assert GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ + AC_VERB_SET_GPIO_DATA, gpio_data); + /* Enable GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + AC_VERB_SET_GPIO_MASK, gpio_mask); } /* @@ -2233,7 +2246,8 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + /* GPIO0 High = Enable EAPD */ + stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { @@ -2265,7 +2279,7 @@ static int patch_stac927x(struct hda_codec *codec) static int patch_stac9205(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, gpio_mask, gpio_data; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2302,8 +2316,21 @@ static int patch_stac9205(struct hda_codec *codec) spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + + if (spec->board_config == STAC_M43xx) { + /* Enable SPDIF in/out */ + stac92xx_set_config_reg(codec, 0x1f, 0x01441030); + stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + gpio_mask = 0x00000007; /* GPIO0-2 */ + /* GPIO0 High = EAPD, GPIO1 Low = DRM, + * GPIO2 High = Headphone Mute + */ + gpio_data = 0x00000005; + } else + gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + + stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) { From 89f157d9e6bf08b65f93dcb1dca1de037079885f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jul 2007 11:23:21 +0200 Subject: [PATCH 094/102] [ALSA] cs46xx - Fix PM resume Fixed PM resume of cs46xx devices. It now restores properly the DSP image and kick-off the DSP. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/cs46xx.h | 4 + include/sound/cs46xx_dsp_spos.h | 2 + sound/pci/cs46xx/cs46xx_lib.c | 77 ++++++++++++--- sound/pci/cs46xx/cs46xx_lib.h | 3 + sound/pci/cs46xx/dsp_spos.c | 170 +++++++++++++++++++++++--------- 5 files changed, 197 insertions(+), 59 deletions(-) diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 685928e6f65a..353910ce9755 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1723,6 +1723,10 @@ struct snd_cs46xx { struct snd_cs46xx_pcm *playback_pcm; unsigned int play_ctl; #endif + +#ifdef CONFIG_PM + u32 *saved_regs; +#endif }; int snd_cs46xx_create(struct snd_card *card, diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index da934def31e9..d9da9e59cf37 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -107,6 +107,7 @@ struct dsp_scb_descriptor { char scb_name[DSP_MAX_SCB_NAME]; u32 address; int index; + u32 *data; struct dsp_scb_descriptor * sub_list_ptr; struct dsp_scb_descriptor * next_scb_ptr; @@ -127,6 +128,7 @@ struct dsp_task_descriptor { int size; u32 address; int index; + u32 *data; }; struct dsp_pcm_channel_descriptor { diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index bef1f6d1859c..71d7aab9d869 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2897,6 +2897,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) } #endif +#ifdef CONFIG_PM + kfree(chip->saved_regs); +#endif + pci_disable_device(chip->pci); kfree(chip); return 0; @@ -3140,6 +3144,23 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) /* * start and load DSP */ + +static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) +{ + unsigned int tmp; + + snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); + + tmp = snd_cs46xx_peek(chip, BA1_PFIE); + tmp &= ~0x0000f03f; + snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ + + tmp = snd_cs46xx_peek(chip, BA1_CIE); + tmp &= ~0x0000003f; + tmp |= 0x00000001; + snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ +} + int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) { unsigned int tmp; @@ -3214,19 +3235,7 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) snd_cs46xx_proc_start(chip); - /* - * Enable interrupts on the part. - */ - snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); - - tmp = snd_cs46xx_peek(chip, BA1_PFIE); - tmp &= ~0x0000f03f; - snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ - - tmp = snd_cs46xx_peek(chip, BA1_CIE); - tmp &= ~0x0000003f; - tmp |= 0x00000001; - snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ + cs46xx_enable_stream_irqs(chip); #ifndef CONFIG_SND_CS46XX_NEW_DSP /* set the attenuation to 0dB */ @@ -3665,11 +3674,19 @@ static struct cs_card_type __devinitdata cards[] = { * APM support */ #ifdef CONFIG_PM +static unsigned int saved_regs[] = { + BA0_ACOSV, + BA0_ASER_FADDR, + BA0_ASER_MASTER, + BA1_PVOL, + BA1_CVOL, +}; + int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; @@ -3680,6 +3697,10 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* save some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]); + amp_saved = chip->amplifier; /* turn off amp */ chip->amplifier_ctrl(chip, -chip->amplifier); @@ -3698,7 +3719,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3716,6 +3737,16 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_cs46xx_chip_init(chip); + snd_cs46xx_reset(chip); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + cs46xx_dsp_resume(chip); + /* restore some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]); +#else + snd_cs46xx_download_image(chip); +#endif + #if 0 snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, chip->ac97_general_purpose); @@ -3730,6 +3761,13 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* reset playback/capture */ + snd_cs46xx_set_play_sample_rate(chip, 8000); + snd_cs46xx_set_capture_sample_rate(chip, 8000); + snd_cs46xx_proc_start(chip); + + cs46xx_enable_stream_irqs(chip); + if (amp_saved) chip->amplifier_ctrl(chip, 1); /* turn amp on */ else @@ -3896,6 +3934,15 @@ int __devinit snd_cs46xx_create(struct snd_card *card, snd_cs46xx_proc_init(card, chip); +#ifdef CONFIG_PM + chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * + ARRAY_SIZE(saved_regs), GFP_KERNEL); + if (!chip->saved_regs) { + snd_cs46xx_free(chip); + return -ENOMEM; + } +#endif + chip->active_ctrl(chip, -1); /* disable CLKRUN */ snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index f75750c2bd24..20dcd72f06c1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -86,6 +86,9 @@ static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip); void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip); int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module); +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip); +#endif struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type); #ifdef CONFIG_PROC_FS diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 336e77e2600c..590b35d91df2 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -306,13 +306,59 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) mutex_unlock(&chip->spos_mutex); } +static int dsp_load_parameter(struct snd_cs46xx *chip, + struct dsp_segment_desc *parameter) +{ + u32 doffset, dsize; + + if (!parameter) { + snd_printdd("dsp_spos: module got no parameter segment\n"); + return 0; + } + + doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); + dsize = parameter->size * 4; + + snd_printdd("dsp_spos: " + "downloading parameter data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { + snd_printk(KERN_ERR "dsp_spos: " + "failed to download parameter data to DSP\n"); + return -EINVAL; + } + return 0; +} + +static int dsp_load_sample(struct snd_cs46xx *chip, + struct dsp_segment_desc *sample) +{ + u32 doffset, dsize; + + if (!sample) { + snd_printdd("dsp_spos: module got no sample segment\n"); + return 0; + } + + doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); + dsize = sample->size * 4; + + snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + + if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { + snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); + return -EINVAL; + } + return 0; +} + int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); - struct dsp_segment_desc * parameter = get_segment_desc (module,SEGTYPE_SP_PARAMETER); - struct dsp_segment_desc * sample = get_segment_desc (module,SEGTYPE_SP_SAMPLE); u32 doffset, dsize; + int err; if (ins->nmodules == DSP_MAX_MODULES - 1) { snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n"); @@ -326,49 +372,20 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE); } - if (parameter == NULL) { - snd_printdd("dsp_spos: module got no parameter segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current parameter data may be overwriten!\n"); - } - - doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); - dsize = parameter->size * 4; - - snd_printdd("dsp_spos: downloading parameter data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to download parameter data to DSP\n"); - return -EINVAL; - } - } + err = dsp_load_parameter(chip, get_segment_desc(module, + SEGTYPE_SP_PARAMETER)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing sample area\n"); snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE); } - if (sample == NULL) { - snd_printdd("dsp_spos: module got no sample segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current sample data may be overwriten\n"); - } - - doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); - dsize = sample->size * 4; - - snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); - return -EINVAL; - } - } - + err = dsp_load_sample(chip, get_segment_desc(module, + SEGTYPE_SP_SAMPLE)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing code area\n"); @@ -986,7 +1003,10 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return NULL; } - strcpy(ins->tasks[ins->ntask].task_name,name); + if (name) + strcpy(ins->tasks[ins->ntask].task_name, name); + else + strcpy(ins->tasks[ins->ntask].task_name, "(NULL)"); ins->tasks[ins->ntask].address = dest; ins->tasks[ins->ntask].size = size; @@ -995,7 +1015,8 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) desc = (ins->tasks + ins->ntask); ins->ntask++; - add_symbol (chip,name,dest,SYMBOL_PARAMETER); + if (name) + add_symbol (chip,name,dest,SYMBOL_PARAMETER); return desc; } @@ -1006,6 +1027,7 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 desc = _map_scb (chip,name,dest); if (desc) { + desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); @@ -1023,6 +1045,7 @@ cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_da desc = _map_task_tree (chip,name,dest,size); if (desc) { + desc->data = task_data; _dsp_create_task_tree(chip,task_data,dest,size); } else { snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n"); @@ -1320,8 +1343,10 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) 0x0000ffff }; - /* dirty hack ... */ - _dsp_create_task_tree (chip,(u32 *)&mix2_ostream_spb,WRITE_BACK_SPB,2); + if (!cs46xx_dsp_create_task_tree(chip, NULL, + (u32 *)&mix2_ostream_spb, + WRITE_BACK_SPB, 2)) + goto _fail_end; } /* input sample converter */ @@ -1622,7 +1647,6 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, return 0; } - static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; @@ -1894,3 +1918,61 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) return 0; } + +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip) +{ + struct dsp_spos_instance * ins = chip->dsp_spos_instance; + int i, err; + + /* clear parameter, sample and code areas */ + snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, + DSP_PARAMETER_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, + DSP_SAMPLE_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE); + + for (i = 0; i < ins->nmodules; i++) { + struct dsp_module_desc *module = &ins->modules[i]; + struct dsp_segment_desc *seg; + u32 doffset, dsize; + + seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER); + err = dsp_load_parameter(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE); + err = dsp_load_sample(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM); + if (!seg) + continue; + + doffset = seg->offset * 4 + module->load_address * 4 + + DSP_CODE_BYTE_OFFSET; + dsize = seg->size * 4; + err = snd_cs46xx_download(chip, + ins->code.data + module->load_address, + doffset, dsize); + if (err < 0) + return err; + } + + for (i = 0; i < ins->ntask; i++) { + struct dsp_task_descriptor *t = &ins->tasks[i]; + _dsp_create_task_tree(chip, t->data, t->address, t->size); + } + + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + _dsp_create_scb(chip, s->data, s->address); + } + + return 0; +} +#endif From 3663d845e557989d09e856c1e9e708e80a976dd4 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 14 Jul 2007 02:18:26 +0100 Subject: [PATCH 095/102] [ALSA] snd-emu10k1: Initial support for E-Mu 1616 and 1616m. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/emu10k1_main.c | 45 +++++++++++++++++++++++++------- 1 file changed, 36 insertions(+), 9 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 72367411ee6b..6135fc99fa5e 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -51,9 +51,13 @@ #define HANA_FILENAME "emu/hana.fw" #define DOCK_FILENAME "emu/audio_dock.fw" +#define EMU1010B_FILENAME "emu/emu1010b.fw" +#define MICRO_DOCK_FILENAME "emu/micro_dock.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010B_FILENAME); +MODULE_FIRMWARE(MICRO_DOCK_FILENAME); /************************************************************************* @@ -660,10 +664,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return err; } snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); +#if 0 if (fw_entry->size != 0x133a4) { snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); return -EINVAL; } +#endif /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -758,7 +764,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x1f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -766,19 +772,26 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x1f) == 0x15) { /* FPGA failed to return to programming mode */ return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); - if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if (reg != 0x55) { + if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -995,16 +1008,23 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } } + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if (reg != 0x55) { + if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); return 0; @@ -1282,6 +1302,13 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spi_dac = 1, .i2c_adc = 1, .spk71 = 1} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1 , + .emu1010 = 2} , {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", From d9e8a552d486eff3281e58754d126500782dcd74 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 14 Jul 2007 10:24:49 +0100 Subject: [PATCH 096/102] [ALSA] snd-emu10k1:Enable E-Mu 1616m notebook firmware loading. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/emu10k1_main.c | 36 +++++++++++++++++++++----------- 1 file changed, 24 insertions(+), 12 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6135fc99fa5e..404ae1be0a4b 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -53,11 +53,13 @@ #define DOCK_FILENAME "emu/audio_dock.fw" #define EMU1010B_FILENAME "emu/emu1010b.fw" #define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); MODULE_FIRMWARE(EMU1010B_FILENAME); MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); /************************************************************************* @@ -764,7 +766,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if ((reg & 0x1f) == 0x15) { + if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -772,8 +774,9 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if ((reg & 0x1f) == 0x15) { + if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ + snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); @@ -787,11 +790,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); return err; } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if ((reg & 0x1f) != 0x15) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -1016,6 +1024,10 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { return err; } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } } snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); @@ -1024,7 +1036,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if ((reg & 0x1f) != 0x15) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); return 0; @@ -1299,9 +1311,8 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .ca_cardbus_chip = 1, - .spi_dac = 1, - .i2c_adc = 1, - .spk71 = 1} , + .spk71 = 1 , + .emu1010 = 3} , {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", .id = "EMU1010", @@ -1742,12 +1753,13 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->fx8010.extout_mask = extout_mask; emu->enable_ir = enable_ir; + if (emu->card_capabilities->ca_cardbus_chip) { + if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + goto error; + } if (emu->card_capabilities->ecard) { if ((err = snd_emu10k1_ecard_init(emu)) < 0) goto error; - } else if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) - goto error; } else if (emu->card_capabilities->emu1010) { if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { snd_emu10k1_free(emu); @@ -1893,10 +1905,10 @@ void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu) void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) { + if (emu->card_capabilities->ca_cardbus_chip) + snd_emu10k1_cardbus_init(emu); if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); - else if (emu->card_capabilities->ca_cardbus_chip) - snd_emu10k1_cardbus_init(emu); else if (emu->card_capabilities->emu1010) snd_emu10k1_emu1010_init(emu); else From 8632649b1c991aab883a8538e493c33e362d077a Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Fri, 18 May 2007 20:35:50 +0100 Subject: [PATCH 097/102] [ALSA] snd-ca0106: Add support for X-Fi Extreme Audio. This card is just a normal SB Live 24bit, but under a different marketing name. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/ca0106/ca0106_main.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 9fd7b8a5b75e..fcab8fb97e38 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -168,6 +168,25 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); #include "ca0106.h" static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ /* AudigyLS[SB0310] */ { .serial = 0x10021102, .name = "AudigyLS [SB0310]", From c5d9f1cdbf067b5500886c5171159e99657341ce Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Thu, 19 Jul 2007 23:18:32 +0200 Subject: [PATCH 098/102] [ALSA] hda-codec - Add quirk for Asus P5LD2 Without the proper model setting, the Asus P5LD2 rear outputs remain completely silent unlike other systems where the front speakers usually work. This patch adds the P5LD2 to the quirk table. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- 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 28fa99da1916..5825d9958f34 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5348,6 +5348,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), {} From c54728d8aa59283ece86cb745e5085ece8b4eedb Mon Sep 17 00:00:00 2001 From: Nicola Fagnani Date: Thu, 19 Jul 2007 23:28:52 +0200 Subject: [PATCH 099/102] [ALSA] hda-intel: support for iMac 24'' released on 09/2006 Basic audio support for the iMac 24'' model released on 09/2006, including headphone jack detection with automatic speaker muting. This iMac uses the Realtek ALC885 codec, not a Sigmatel one as in other models. Functionality has been tested for internal speakers, headphone and microphone. Signed-off-by: Nicola Fagnani Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- .../sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 74 ++++++++++++++++++- 2 files changed, 74 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 5bb45cd5f0e6..241e26c4ff92 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 macpro MacPro support + imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5825d9958f34..9a47eec5a27b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -153,6 +153,7 @@ enum { ALC882_TARGA, ALC882_ASUS_A7J, ALC885_MACPRO, + ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -5118,6 +5119,60 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* iMac 24 mixer. */ +static struct snd_kcontrol_new alc885_imac24_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), + { } /* end */ +}; + +/* iMac 24 init verbs. */ +static struct hda_verb alc885_imac24_init_verbs[] = { + /* Internal speakers: output 0 (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Internal speakers: output 0 (0x0c) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Headphone: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Front Mic: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +/* Processes unsolicited events. */ +static void alc885_imac24_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_imac24_automute(codec); +} + static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -5338,6 +5393,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", + [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5410,6 +5466,19 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc885_imac24_unsol_event, + .init_hook = alc885_imac24_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, alc882_capture_mixer }, @@ -5582,6 +5651,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; + case 0x106b1000: /* iMac 24 */ + board_config = ALC885_IMAC24; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5608,7 +5680,7 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO) { + if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { alc882_gpio_mute(codec, 0, 0); alc882_gpio_mute(codec, 1, 0); } From aafc03398baefaac380b34a1ee6356e7eed8e663 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 May 2007 18:01:06 +0200 Subject: [PATCH 100/102] [ALSA] hda-codec - Fix input with STAC92xx The recent fix for STAC92xx surround outputs broke the input pin setting for shared line-in and mic jacks. This patch fixes the breakage. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9b1d433178bf..3f25de72966b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2266,13 +2266,6 @@ static int patch_stac927x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; - /* Fix Mux capture level; max to 2 */ - snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, - (0 << AC_AMPCAP_OFFSET_SHIFT) | - (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (0 << AC_AMPCAP_MUTE_SHIFT)); - return 0; } From 1f6268e982c7d35adee34d8f92a17644c7b3f78a Mon Sep 17 00:00:00 2001 From: "Jaroslav Kysela perex@suse.cz" Date: Fri, 20 Jul 2007 10:42:56 +0200 Subject: [PATCH 101/102] [ALSA] remove duplicate Logitech Quickcam USB ID in usbquirks.h Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 32de6c3f4ddd..5a2f518c6629 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -70,15 +70,6 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x0850, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL -}, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | From 53555eb758388fff3982875129ca96080caed147 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 20 Jul 2007 11:13:35 +0200 Subject: [PATCH 102/102] [ALSA] version 1.0.14 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 8e5b2f0f5946..6bbcfefd2c38 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by alsa/ksync script. */ #define CONFIG_SND_VERSION "1.0.14" -#define CONFIG_SND_DATE " (Thu May 31 09:03:25 2007 UTC)" +#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)"