Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next

This commit is contained in:
Mark Brown 2013-06-17 17:20:25 +01:00
commit 3843e87ebd
7 changed files with 4 additions and 408 deletions

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@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X
Say Y if you want to add support for SoC audio on
Palm T|X, T5, E2 or LifeDrive handheld computer.
config SND_SOC_SAARB
tristate "SoC Audio support for Marvell Saarb"
depends on SND_PXA2XX_SOC && MACH_SAARB
select MFD_88PM860X
select SND_PXA_SOC_SSP
select SND_SOC_88PM860X
help
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
config SND_SOC_TAVOREVB3
tristate "SoC Audio support for Marvell Tavor EVB3"
depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
select MFD_88PM860X
select SND_PXA_SOC_SSP
select SND_SOC_88PM860X
help
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
config SND_PXA910_SOC
tristate "SoC Audio for Marvell PXA910 chip"
depends on ARCH_MMP && SND

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@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-saarb-objs := saarb.o
snd-soc-tavorevb3-objs := tavorevb3.o
snd-soc-zylonite-objs := zylonite.o
snd-soc-hx4700-objs := hx4700.o
snd-soc-magician-objs := magician.o
@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o

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@ -147,7 +147,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
vma->vm_end - vma->vm_start, vma->vm_page_prot);
}
struct snd_pcm_ops mmp_pcm_ops = {
static struct snd_pcm_ops mmp_pcm_ops = {
.open = mmp_pcm_open,
.close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
@ -208,7 +208,7 @@ static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
return 0;
}
int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm_substream *substream;
struct snd_pcm *pcm = rtd->pcm;
@ -229,7 +229,7 @@ err:
return ret;
}
struct snd_soc_platform_driver mmp_soc_platform = {
static struct snd_soc_platform_driver mmp_soc_platform = {
.ops = &mmp_pcm_ops,
.pcm_new = mmp_pcm_new,
.pcm_free = mmp_pcm_free_dma_buffers,

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@ -388,7 +388,7 @@ static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
.set_fmt = mmp_sspa_set_dai_fmt,
};
struct snd_soc_dai_driver mmp_sspa_dai = {
static struct snd_soc_dai_driver mmp_sspa_dai = {
.probe = mmp_sspa_probe,
.playback = {
.channels_min = 1,

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@ -1,190 +0,0 @@
/*
* saarb.c -- SoC audio for saarb
*
* Copyright (C) 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include "../codecs/88pm860x-codec.h"
#include "pxa-ssp.h"
static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
static struct platform_device *saarb_snd_device;
static struct snd_soc_jack hs_jack, mic_jack;
static struct snd_soc_jack_pin hs_jack_pins[] = {
{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
};
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
};
/* saarb machine dapm widgets */
static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
SND_SOC_DAPM_SPK("Ext Speaker", NULL),
SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
};
/* saarb machine audio map */
static const struct snd_soc_dapm_route saarb_audio_map[] = {
{"Headset Stereophone", NULL, "HS1"},
{"Headset Stereophone", NULL, "HS2"},
{"Ext Speaker", NULL, "LSP"},
{"Ext Speaker", NULL, "LSN"},
{"Lineout Out 1", NULL, "LINEOUT1"},
{"Lineout Out 2", NULL, "LINEOUT2"},
{"MIC1P", NULL, "Mic1 Bias"},
{"MIC1N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Ext Mic 1"},
{"MIC2P", NULL, "Mic1 Bias"},
{"MIC2N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Headset Mic 2"},
{"MIC3P", NULL, "Mic3 Bias"},
{"MIC3N", NULL, "Mic3 Bias"},
{"Mic3 Bias", NULL, "Ext Mic 3"},
};
static int saarb_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int width = snd_pcm_format_physical_width(params_format(params));
int ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
return ret;
}
static struct snd_soc_ops saarb_i2s_ops = {
.hw_params = saarb_i2s_hw_params,
};
static struct snd_soc_dai_link saarb_dai[] = {
{
.name = "88PM860x I2S",
.stream_name = "I2S Audio",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "88pm860x-i2s",
.platform_name = "pxa-pcm-audio",
.codec_name = "88pm860x-codec",
.init = saarb_pm860x_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &saarb_i2s_ops,
},
};
static struct snd_soc_card snd_soc_card_saarb = {
.name = "Saarb",
.owner = THIS_MODULE,
.dai_link = saarb_dai,
.num_links = ARRAY_SIZE(saarb_dai),
.dapm_widgets = saarb_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets),
.dapm_routes = saarb_audio_map,
.num_dapm_routes = ARRAY_SIZE(saarb_audio_map),
};
static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
/* connected pins */
snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
&hs_jack);
snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
&mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
/* headphone, microphone detection & headset short detection */
pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
return 0;
}
static int __init saarb_init(void)
{
int ret;
if (!machine_is_saarb())
return -ENODEV;
saarb_snd_device = platform_device_alloc("soc-audio", -1);
if (!saarb_snd_device)
return -ENOMEM;
platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
ret = platform_device_add(saarb_snd_device);
if (ret)
platform_device_put(saarb_snd_device);
return ret;
}
static void __exit saarb_exit(void)
{
platform_device_unregister(saarb_snd_device);
}
module_init(saarb_init);
module_exit(saarb_exit);
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
MODULE_LICENSE("GPL");

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@ -1,189 +0,0 @@
/*
* tavorevb3.c -- SoC audio for Tavor EVB3
*
* Copyright (C) 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include "../codecs/88pm860x-codec.h"
#include "pxa-ssp.h"
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
static struct platform_device *evb3_snd_device;
static struct snd_soc_jack hs_jack, mic_jack;
static struct snd_soc_jack_pin hs_jack_pins[] = {
{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
};
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
};
/* tavorevb3 machine dapm widgets */
static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
SND_SOC_DAPM_SPK("Ext Speaker", NULL),
SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
};
/* tavorevb3 machine audio map */
static const struct snd_soc_dapm_route evb3_audio_map[] = {
{"Headset Stereophone", NULL, "HS1"},
{"Headset Stereophone", NULL, "HS2"},
{"Ext Speaker", NULL, "LSP"},
{"Ext Speaker", NULL, "LSN"},
{"Lineout Out 1", NULL, "LINEOUT1"},
{"Lineout Out 2", NULL, "LINEOUT2"},
{"MIC1P", NULL, "Mic1 Bias"},
{"MIC1N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Ext Mic 1"},
{"MIC2P", NULL, "Mic1 Bias"},
{"MIC2N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Headset Mic 2"},
{"MIC3P", NULL, "Mic3 Bias"},
{"MIC3N", NULL, "Mic3 Bias"},
{"Mic3 Bias", NULL, "Ext Mic 3"},
};
static int evb3_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int width = snd_pcm_format_physical_width(params_format(params));
int ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
return ret;
}
static struct snd_soc_ops evb3_i2s_ops = {
.hw_params = evb3_i2s_hw_params,
};
static struct snd_soc_dai_link evb3_dai[] = {
{
.name = "88PM860x I2S",
.stream_name = "I2S Audio",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "88pm860x-i2s",
.platform_name = "pxa-pcm-audio",
.codec_name = "88pm860x-codec",
.init = evb3_pm860x_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &evb3_i2s_ops,
},
};
static struct snd_soc_card snd_soc_card_evb3 = {
.name = "Tavor EVB3",
.owner = THIS_MODULE,
.dai_link = evb3_dai,
.num_links = ARRAY_SIZE(evb3_dai),
.dapm_widgets = evb3_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets),
.dapm_routes = evb3_audio_map,
.num_dapm_routes = ARRAY_SIZE(evb3_audio_map),
};
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
/* connected pins */
snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
&hs_jack);
snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
&mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
/* headphone, microphone detection & headset short detection */
pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
return 0;
}
static int __init tavorevb3_init(void)
{
int ret;
if (!machine_is_tavorevb3())
return -ENODEV;
evb3_snd_device = platform_device_alloc("soc-audio", -1);
if (!evb3_snd_device)
return -ENOMEM;
platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
ret = platform_device_add(evb3_snd_device);
if (ret)
platform_device_put(evb3_snd_device);
return ret;
}
static void __exit tavorevb3_exit(void)
{
platform_device_unregister(evb3_snd_device);
}
module_init(tavorevb3_init);
module_exit(tavorevb3_exit);
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
MODULE_LICENSE("GPL");

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@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = {
.resume_pre = &zylonite_resume_pre,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
.owner = THIS_MODULE,
};
static struct platform_device *zylonite_snd_ac97_device;