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Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kerne…
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…l/git/broonie/sound-2.6 into topic/asoc
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Takashi Iwai committed May 12, 2009
2 parents ddc4097 + 13e2c86 commit dd4a416
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Showing 9 changed files with 283 additions and 40 deletions.
2 changes: 1 addition & 1 deletion sound/arm/pxa2xx-ac97-lib.c
Original file line number Diff line number Diff line change
Expand Up @@ -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;
Expand Down
8 changes: 8 additions & 0 deletions sound/soc/atmel/Kconfig
Original file line number Diff line number Diff line change
Expand Up @@ -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.
1 change: 1 addition & 0 deletions sound/soc/atmel/Makefile
Original file line number Diff line number Diff line change
Expand Up @@ -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
203 changes: 203 additions & 0 deletions sound/soc/atmel/snd-soc-afeb9260.c
Original file line number Diff line number Diff line change
@@ -0,0 +1,203 @@
/*
* 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");

52 changes: 19 additions & 33 deletions sound/soc/codecs/twl4030.c
Original file line number Diff line number Diff line change
Expand Up @@ -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[] =
Expand Down Expand Up @@ -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),
Expand Down
4 changes: 2 additions & 2 deletions sound/soc/pxa/Kconfig
Original file line number Diff line number Diff line change
Expand Up @@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800
Toshiba e800 PDA

config SND_PXA2XX_SOC_EM_X270
tristate "SoC Audio support for CompuLab EM-x270"
tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
depends on SND_PXA2XX_SOC && MACH_EM_X270
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
Say Y if you want to add support for SoC audio on
CompuLab EM-x270.
CompuLab EM-x270, eXeda and CM-X300 machines.

config SND_PXA2XX_SOC_PALM27X
bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
Expand Down
9 changes: 5 additions & 4 deletions sound/soc/pxa/em-x270.c
Original file line number Diff line number Diff line change
@@ -1,7 +1,7 @@
/*
* em-x270.c -- SoC audio for EM-X270
* SoC audio driver for EM-X270, eXeda and CM-X300
*
* Copyright 2007 CompuLab, Ltd.
* Copyright 2007, 2009 CompuLab, Ltd.
*
* Author: Mike Rapoport <mike@compulab.co.il>
*
Expand Down Expand Up @@ -68,7 +68,8 @@ static int __init em_x270_init(void)
{
int ret;

if (!machine_is_em_x270())
if (!(machine_is_em_x270() || machine_is_exeda()
|| machine_is_cm_x300()))
return -ENODEV;

em_x270_snd_device = platform_device_alloc("soc-audio", -1);
Expand All @@ -95,5 +96,5 @@ module_exit(em_x270_exit);

/* Module information */
MODULE_AUTHOR("Mike Rapoport");
MODULE_DESCRIPTION("ALSA SoC EM-X270");
MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
MODULE_LICENSE("GPL");
1 change: 1 addition & 0 deletions sound/soc/pxa/pxa2xx-i2s.c
Original file line number Diff line number Diff line change
Expand Up @@ -329,6 +329,7 @@ struct snd_soc_dai pxa_i2s_dai = {
.rates = PXA2XX_I2S_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = &pxa_i2s_dai_ops,
.symmetric_rates = 1,
};

EXPORT_SYMBOL_GPL(pxa_i2s_dai);
Expand Down
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