Merge branch 'for-2.6.31' of... (dd4a4164) · Commits · e / devices / android_kernel_sony_msm8994

sound/arm/pxa2xx-ac97-lib.c

+1 −1

Original line number	Diff line number	Diff line
		@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
		switch (resetgpio_action) {
		case RESETGPIO_NORMAL_ALTFUNC:
		if (reset_gpio == 113)
		mode = 113 \| GPIO_OUT \| GPIO_DFLT_LOW;
		mode = 113 \| GPIO_ALT_FN_2_OUT;
		if (reset_gpio == 95)
		mode = 95 \| GPIO_ALT_FN_1_OUT;
		break;

sound/soc/atmel/Kconfig

+8 −0

Original line number	Diff line number	Diff line
		@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
		and FRAME signals on the PlayPaq. Unless you want to play
		with the AT32 as the SSC master, you probably want to say N here,
		as this will give you better sound quality.

		config SND_AT91_SOC_AFEB9260
		tristate "SoC Audio support for AFEB9260 board"
		depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
		select SND_ATMEL_SOC_SSC
		select SND_SOC_TLV320AIC23
		help
		Say Y here to support sound on AFEB9260 board.

sound/soc/atmel/Makefile

+1 −0

Original line number	Diff line number	Diff line
		@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o

		obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
		obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
		obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o

sound/soc/atmel/snd-soc-afeb9260.c

0 → 100644

+203 −0

Original line number	Diff line number	Diff line
		/*
		* afeb9260.c -- SoC audio for AFEB9260
		*
		* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
		*
		* This program is free software; you can redistribute it and/or
		* modify it under the terms of the GNU General Public License
		* version 2 as published by the Free Software Foundation.
		*
		* This program is distributed in the hope that it will be useful, but
		* WITHOUT ANY WARRANTY; without even the implied warranty of
		* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
		* General Public License for more details.
		*
		* 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., 51 Franklin St, Fifth Floor, Boston, MA
		* 02110-1301 USA
		*
		*/

		#include <linux/module.h>
		#include <linux/moduleparam.h>
		#include <linux/kernel.h>
		#include <linux/clk.h>
		#include <linux/platform_device.h>

		#include <linux/atmel-ssc.h>
		#include <sound/core.h>
		#include <sound/pcm.h>
		#include <sound/pcm_params.h>
		#include <sound/soc.h>
		#include <sound/soc-dapm.h>

		#include <asm/mach-types.h>
		#include <mach/hardware.h>
		#include <linux/gpio.h>

		#include "../codecs/tlv320aic23.h"
		#include "atmel-pcm.h"
		#include "atmel_ssc_dai.h"

		#define CODEC_CLOCK 12000000

		static int afeb9260_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->dai->codec_dai;
		struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
		int err;

		/* Set codec DAI configuration */
		err = snd_soc_dai_set_fmt(codec_dai,
		SND_SOC_DAIFMT_I2S\|
		SND_SOC_DAIFMT_NB_IF \|
		SND_SOC_DAIFMT_CBM_CFM);
		if (err < 0) {
		printk(KERN_ERR "can't set codec DAI configuration\n");
		return err;
		}

		/* Set cpu DAI configuration */
		err = snd_soc_dai_set_fmt(cpu_dai,
		SND_SOC_DAIFMT_I2S \|
		SND_SOC_DAIFMT_NB_IF \|
		SND_SOC_DAIFMT_CBM_CFM);
		if (err < 0) {
		printk(KERN_ERR "can't set cpu DAI configuration\n");
		return err;
		}

		/* Set the codec system clock for DAC and ADC */
		err =
		snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);

		if (err < 0) {
		printk(KERN_ERR "can't set codec system clock\n");
		return err;
		}

		return err;
		}

		static struct snd_soc_ops afeb9260_ops = {
		.hw_params = afeb9260_hw_params,
		};

		static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
		SND_SOC_DAPM_HP("Headphone Jack", NULL),
		SND_SOC_DAPM_LINE("Line In", NULL),
		SND_SOC_DAPM_MIC("Mic Jack", NULL),
		};

		static const struct snd_soc_dapm_route audio_map[] = {
		{"Headphone Jack", NULL, "LHPOUT"},
		{"Headphone Jack", NULL, "RHPOUT"},

		{"LLINEIN", NULL, "Line In"},
		{"RLINEIN", NULL, "Line In"},

		{"MICIN", NULL, "Mic Jack"},
		};

		static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
		{

		/* Add afeb9260 specific widgets */
		snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
		ARRAY_SIZE(tlv320aic23_dapm_widgets));

		/* Set up afeb9260 specific audio path audio_map */
		snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));

		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
		snd_soc_dapm_enable_pin(codec, "Line In");
		snd_soc_dapm_enable_pin(codec, "Mic Jack");

		snd_soc_dapm_sync(codec);

		return 0;
		}

		/* Digital audio interface glue - connects codec <--> CPU */
		static struct snd_soc_dai_link afeb9260_dai = {
		.name = "TLV320AIC23",
		.stream_name = "AIC23",
		.cpu_dai = &atmel_ssc_dai[0],
		.codec_dai = &tlv320aic23_dai,
		.init = afeb9260_tlv320aic23_init,
		.ops = &afeb9260_ops,
		};

		/* Audio machine driver */
		static struct snd_soc_card snd_soc_machine_afeb9260 = {
		.name = "AFEB9260",
		.platform = &atmel_soc_platform,
		.dai_link = &afeb9260_dai,
		.num_links = 1,
		};

		/* Audio subsystem */
		static struct snd_soc_device afeb9260_snd_devdata = {
		.card = &snd_soc_machine_afeb9260,
		.codec_dev = &soc_codec_dev_tlv320aic23,
		};

		static struct platform_device *afeb9260_snd_device;

		static int __init afeb9260_soc_init(void)
		{
		int err;
		struct device *dev;
		struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
		struct ssc_device *ssc = NULL;

		if (!(machine_is_afeb9260()))
		return -ENODEV;

		ssc = ssc_request(0);
		if (IS_ERR(ssc)) {
		printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
		err = PTR_ERR(ssc);
		ssc = NULL;
		goto err_ssc;
		}
		ssc_p->ssc = ssc;

		afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
		if (!afeb9260_snd_device) {
		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
		return -ENOMEM;
		}

		platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
		afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
		err = platform_device_add(afeb9260_snd_device);
		if (err)
		goto err1;

		dev = &afeb9260_snd_device->dev;

		return 0;
		err1:
		platform_device_del(afeb9260_snd_device);
		platform_device_put(afeb9260_snd_device);
		err_ssc:
		return err;

		}

		static void __exit afeb9260_soc_exit(void)
		{
		platform_device_unregister(afeb9260_snd_device);
		}

		module_init(afeb9260_soc_init);
		module_exit(afeb9260_soc_exit);

		MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
		MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
		MODULE_LICENSE("GPL");

sound/soc/codecs/twl4030.c

+19 −33

Original line number	Diff line number	Diff line
		@@ -422,36 +422,18 @@ static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
		SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);

		/* Left analog microphone selection */
		static const char *twl4030_analoglmic_texts[] =
		{"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};

		static const unsigned int twl4030_analoglmic_values[] =
		{0x0, 0x1, 0x2, 0x4, 0x8};

		static const struct soc_enum twl4030_analoglmic_enum =
		SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
		ARRAY_SIZE(twl4030_analoglmic_texts),
		twl4030_analoglmic_texts,
		twl4030_analoglmic_values);

		static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
		SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
		static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
		SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
		SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
		SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
		SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
		};

		/* Right analog microphone selection */
		static const char *twl4030_analogrmic_texts[] =
		{"Off", "Sub mic", "AUXR"};

		static const unsigned int twl4030_analogrmic_values[] =
		{0x0, 0x1, 0x4};

		static const struct soc_enum twl4030_analogrmic_enum =
		SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
		ARRAY_SIZE(twl4030_analogrmic_texts),
		twl4030_analogrmic_texts,
		twl4030_analogrmic_values);

		static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
		SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
		static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
		SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
		SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0),
		};

		/* TX1 L/R Analog/Digital microphone selection */
		static const char *twl4030_micpathtx1_texts[] =
		@@ -1138,11 +1120,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
		SND_SOC_DAPM_POST_PMU\|SND_SOC_DAPM_POST_PMD\|
		SND_SOC_DAPM_POST_REG),

		/* Analog input muxes with switch for the capture amplifiers */
		SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
		SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
		/* Analog input mixers for the capture amplifiers */
		SND_SOC_DAPM_MIXER("Analog Left Capture Route",
		TWL4030_REG_ANAMICL, 4, 0,
		&twl4030_dapm_analoglmic_controls[0],
		ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
		SND_SOC_DAPM_MIXER("Analog Right Capture Route",
		TWL4030_REG_ANAMICR, 4, 0,
		&twl4030_dapm_analogrmic_controls[0],
		ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),

		SND_SOC_DAPM_PGA("ADC Physical Left",
		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),