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Merge branch 'for-2.6.40' into for-2.6.41

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This commit is contained in:
Mark Brown 2011-05-30 10:54:18 +08:00
parent 74ab24af4f ea02c63d57
commit d21685ec25
45 changed files with 1305 additions and 253 deletions
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6
MAINTAINERS
View file

@ -5840,7 +5840,7 @@ F: include/sound/
F: sound/
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
M: Liam Girdwood <lrg@slimlogic.co.uk>
M: Liam Girdwood <lrg@ti.com>
M: Mark Brown <broonie@opensource.wolfsonmicro.com>
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
@ -6093,7 +6093,7 @@ F: drivers/mmc/host/tifm_sd.c
F: include/linux/tifm.h
TI TWL4030 SERIES SOC CODEC DRIVER
M: Peter Ujfalusi <peter.ujfalusi@nokia.com>
M: Peter Ujfalusi <peter.ujfalusi@ti.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
F: sound/soc/codecs/twl4030*
@ -6736,7 +6736,7 @@ F: drivers/scsi/vmw_pvscsi.c
F: drivers/scsi/vmw_pvscsi.h
VOLTAGE AND CURRENT REGULATOR FRAMEWORK
M: Liam Girdwood <lrg@slimlogic.co.uk>
M: Liam Girdwood <lrg@ti.com>
M: Mark Brown <broonie@opensource.wolfsonmicro.com>
W: http://opensource.wolfsonmicro.com/node/15
W: http://www.slimlogic.co.uk/?p=48

26
include/sound/ak4641.h Normal file
View file

@ -0,0 +1,26 @@
/*
* AK4641 ALSA SoC Codec driver
*
* Copyright 2009 Philipp Zabel
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __AK4641_H
#define __AK4641_H
/**
* struct ak4641_platform_data - platform specific AK4641 configuration
* @gpio_power: GPIO to control external power to AK4641
* @gpio_npdn: GPIO connected to AK4641 nPDN pin
*
* Both GPIO parameters are optional.
*/
struct ak4641_platform_data {
int gpio_power;
int gpio_npdn;
};
#endif /* __AK4641_H */

2
include/sound/soc.h
View file

@ -702,6 +702,8 @@ struct snd_soc_aux_dev {
/* SoC card */
struct snd_soc_card {
const char *name;
const char *long_name;
const char *driver_name;
struct device *dev;
struct snd_card *snd_card;
struct module *owner;

2
include/sound/tlv320dac33-plat.h
View file

@ -1,7 +1,7 @@
/*
* Platform header for Texas Instruments TLV320DAC33 codec driver
*
* Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*

2
include/sound/tpa6130a2-plat.h
View file

@ -3,7 +3,7 @@
*
* Copyright (C) Nokia Corporation
*
* Written by Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License

2
sound/soc/atmel/sam9g20_wm8731.c
View file

@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
"at91sam9g20ek_wm8731 "
": at91sam9g20ek_wm8731_init() called\n");
ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_MCLK,
MCLK_RATE, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret);

4
sound/soc/codecs/Kconfig
View file

@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
select SND_SOC_ALC5623 if I2C
@ -139,6 +140,9 @@ config SND_SOC_AK4104
config SND_SOC_AK4535
tristate
config SND_SOC_AK4641
tristate
config SND_SOC_AK4642
tristate

2
sound/soc/codecs/Makefile
View file

@ -7,6 +7,7 @@ snd-soc-ad73311-objs := ad73311.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o

664
sound/soc/codecs/ak4641.c Normal file
View file

@ -0,0 +1,664 @@
/*
* ak4641.c -- AK4641 ALSA Soc Audio driver
*
* Copyright (C) 2008 Harald Welte <laforge@gnufiish.org>
* Copyright (C) 2011 Dmitry Artamonow <mad_soft@inbox.ru>
*
* Based on ak4535.c by Richard Purdie
*
* 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/init.h>
#include <linux/delay.h>
#include <linux/gpio.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <sound/ak4641.h>
#include "ak4641.h"
/* codec private data */
struct ak4641_priv {
struct snd_soc_codec *codec;
unsigned int sysclk;
int deemph;
int playback_fs;
};
/*
* ak4641 register cache
*/
static const u8 ak4641_reg[AK4641_CACHEREGNUM] = {
0x00, 0x80, 0x00, 0x80,
0x02, 0x00, 0x11, 0x05,
0x00, 0x00, 0x36, 0x10,
0x00, 0x00, 0x57, 0x00,
0x88, 0x88, 0x08, 0x08
};
static const int deemph_settings[] = {44100, 0, 48000, 32000};
static int ak4641_set_deemph(struct snd_soc_codec *codec)
{
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int i, best = 0;
for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) {
/* if deemphasis is on, select the nearest available rate */
if (ak4641->deemph && deemph_settings[i] != 0 &&
abs(deemph_settings[i] - ak4641->playback_fs) <
abs(deemph_settings[best] - ak4641->playback_fs))
best = i;
if (!ak4641->deemph && deemph_settings[i] == 0)
best = i;
}
dev_dbg(codec->dev, "Set deemphasis %d\n", best);
return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best);
}
static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int deemph = ucontrol->value.enumerated.item[0];
if (deemph > 1)
return -EINVAL;
ak4641->deemph = deemph;
return ak4641_set_deemph(codec);
}
static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = ak4641->deemph;
return 0;
};
static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"};
static const char *ak4641_hp_out[] = {"Stereo", "Mono"};
static const char *ak4641_mic_select[] = {"Internal", "External"};
static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"};
static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0);
static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0);
static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0);
static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0);
static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0);
static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0);
static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0);
static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
static const struct soc_enum ak4641_mono_out_enum =
SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out);
static const struct soc_enum ak4641_hp_out_enum =
SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out);
static const struct soc_enum ak4641_mic_select_enum =
SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select);
static const struct soc_enum ak4641_mic_or_dac_enum =
SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac);
static const struct snd_kcontrol_new ak4641_snd_controls[] = {
SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1,
mono_gain_tlv),
SOC_ENUM("Headphone Output", ak4641_hp_out_enum),
SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
ak4641_get_deemph, ak4641_put_deemph),
SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv),
SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0),
SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0),
SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0),
SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0),
SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv),
SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0),
SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0),
SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv),
SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT,
AK4641_RATT, 0, 255, 1, master_tlv),
SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv),
SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0),
SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv),
};
/* Mono 1 Mixer */
static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0,
mic_mono_sidetone_tlv),
SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0),
SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0),
};
/* Stereo Mixer */
static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0,
mic_stereo_sidetone_tlv),
SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0),
SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0),
SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0),
};
/* Input Mixer */
static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = {
SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0),
SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0),
};
/* Mic mux */
static const struct snd_kcontrol_new ak4641_mic_mux_control =
SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum);
/* Input mux */
static const struct snd_kcontrol_new ak4641_input_mux_control =
SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum);
/* mono 2 switch */
static const struct snd_kcontrol_new ak4641_mono2_control =
SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0);
/* ak4641 dapm widgets */
static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_stereo_mixer_controls[0],
ARRAY_SIZE(ak4641_stereo_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_mono1_mixer_controls[0],
ARRAY_SIZE(ak4641_mono1_mixer_controls)),
SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_input_mixer_controls[0],
ARRAY_SIZE(ak4641_input_mixer_controls)),
SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0,
&ak4641_mic_mux_control),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
&ak4641_input_mux_control),
SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
&ak4641_mono2_control),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_OUTPUT("MOUT1"),
SND_SOC_DAPM_OUTPUT("MOUT2"),
SND_SOC_DAPM_OUTPUT("MICOUT"),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0),
SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0),
SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0),
SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0),
SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0),
SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0),
SND_SOC_DAPM_INPUT("MICIN"),
SND_SOC_DAPM_INPUT("MICEXT"),
SND_SOC_DAPM_INPUT("AUX"),
SND_SOC_DAPM_INPUT("AIN"),
};
static const struct snd_soc_dapm_route ak4641_audio_map[] = {
/* Stereo Mixer */
{"Stereo Mixer", "Playback Switch", "DAC"},
{"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"},
{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
/* Mono 1 Mixer */
{"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"},
{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
/* Mic */
{"Mic", NULL, "AIN"},
{"Mic Mux", "Internal", "Mic Int Bias"},
{"Mic Mux", "External", "Mic Ext Bias"},
{"Mic Int Bias", NULL, "MICIN"},
{"Mic Ext Bias", NULL, "MICEXT"},
{"MICOUT", NULL, "Mic Mux"},
/* Input Mux */
{"Input Mux", "Microphone", "Mic"},
{"Input Mux", "Voice DAC", "Voice DAC"},
/* Line Out */
{"LOUT", NULL, "Line Out"},
{"ROUT", NULL, "Line Out"},
{"Line Out", NULL, "Stereo Mixer"},
/* Mono 1 Out */
{"MOUT1", NULL, "Mono Out"},
{"Mono Out", NULL, "Mono1 Mixer"},
/* Mono 2 Out */
{"MOUT2", NULL, "Mono 2 Enable"},
{"Mono 2 Enable", "Switch", "Mono Out 2"},
{"Mono Out 2", NULL, "Stereo Mixer"},
{"Voice ADC", NULL, "Mono 2 Enable"},
/* Aux In */
{"AUX In", NULL, "AUX"},
/* ADC */
{"ADC", NULL, "Input Mixer"},
{"Input Mixer", "Mic Capture Switch", "Mic"},
{"Input Mixer", "Aux Capture Switch", "AUX In"},
};
static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
ak4641->sysclk = freq;
return 0;
}
static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int rate = params_rate(params), fs = 256;
u8 mode2;
if (rate)
fs = ak4641->sysclk / rate;
else
return -EINVAL;
/* set fs */
switch (fs) {
case 1024:
mode2 = (0x2 << 5);
break;
case 512:
mode2 = (0x1 << 5);
break;
case 256:
mode2 = (0x0 << 5);
break;
default:
dev_err(codec->dev, "Error: unsupported fs=%d\n", fs);
return -EINVAL;
}
snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2);
/* Update de-emphasis filter for the new rate */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ak4641->playback_fs = rate;
ak4641_set_deemph(codec);
};
return 0;
}
static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 btif;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
btif = (0x3 << 5);
break;
case SND_SOC_DAIFMT_LEFT_J:
btif = (0x2 << 5);
break;
case SND_SOC_DAIFMT_DSP_A: /* MSB after FRM */
btif = (0x0 << 5);
break;
case SND_SOC_DAIFMT_DSP_B: /* MSB during FRM */
btif = (0x1 << 5);
break;
default:
return -EINVAL;
}
return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif);
}
static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 mode1 = 0;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
mode1 = 0x02;
break;
case SND_SOC_DAIFMT_LEFT_J:
mode1 = 0x01;
break;
default:
return -EINVAL;
}
return snd_soc_write(codec, AK4641_MODE1, mode1);
}
static int ak4641_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0);
}
static int ak4641_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct ak4641_platform_data *pdata = codec->dev->platform_data;
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
/* unmute */
snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0);
break;
case SND_SOC_BIAS_PREPARE:
/* mute */
snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 1);
mdelay(1);
if (pdata && gpio_is_valid(pdata->gpio_npdn))
gpio_set_value(pdata->gpio_npdn, 1);
mdelay(1);
ret = snd_soc_cache_sync(codec);
if (ret) {
dev_err(codec->dev,
"Failed to sync cache: %d\n", ret);
return ret;
}
}
snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80);
snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0);
break;
case SND_SOC_BIAS_OFF:
snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0);
if (pdata && gpio_is_valid(pdata->gpio_npdn))
gpio_set_value(pdata->gpio_npdn, 0);
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 0);
codec->cache_sync = 1;
break;
}
codec->dapm.bias_level = level;
return 0;
}
#define AK4641_RATES (SNDRV_PCM_RATE_8000_48000)
#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000)
#define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
static struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
.hw_params = ak4641_i2s_hw_params,
.set_fmt = ak4641_i2s_set_dai_fmt,
.digital_mute = ak4641_mute,
.set_sysclk = ak4641_set_dai_sysclk,
};
static struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
.hw_params = NULL, /* rates are controlled by BT chip */
.set_fmt = ak4641_pcm_set_dai_fmt,
.digital_mute = ak4641_mute,
.set_sysclk = ak4641_set_dai_sysclk,
};
struct snd_soc_dai_driver ak4641_dai[] = {
{
.name = "ak4641-hifi",
.id = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = AK4641_RATES,
.formats = AK4641_FORMATS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = AK4641_RATES,
.formats = AK4641_FORMATS,
},
.ops = &ak4641_i2s_dai_ops,
.symmetric_rates = 1,
},
{
.name = "ak4641-voice",
.id = 1,
.playback = {
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
.rates = AK4641_RATES_BT,
.formats = AK4641_FORMATS,
},
.capture = {
.stream_name = "Voice Capture",
.channels_min = 1,
.channels_max = 1,
.rates = AK4641_RATES_BT,
.formats = AK4641_FORMATS,
},
.ops = &ak4641_pcm_dai_ops,
.symmetric_rates = 1,
},
};
static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int ak4641_resume(struct snd_soc_codec *codec)
{
ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
static int ak4641_probe(struct snd_soc_codec *codec)
{
struct ak4641_platform_data *pdata = codec->dev->platform_data;
int ret;
if (pdata) {
if (gpio_is_valid(pdata->gpio_power)) {
ret = gpio_request_one(pdata->gpio_power,
GPIOF_OUT_INIT_LOW, "ak4641 power");
if (ret)
goto err_out;
}
if (gpio_is_valid(pdata->gpio_npdn)) {
ret = gpio_request_one(pdata->gpio_npdn,
GPIOF_OUT_INIT_LOW, "ak4641 npdn");
if (ret)
goto err_gpio;
udelay(1); /* > 150 ns */
gpio_set_value(pdata->gpio_npdn, 1);
}
}
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
goto err_register;
}
/* power on device */
ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
err_register:
if (pdata) {
if (gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 0);
if (gpio_is_valid(pdata->gpio_npdn))
gpio_free(pdata->gpio_npdn);
}
err_gpio:
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_free(pdata->gpio_power);
err_out:
return ret;
}
static int ak4641_remove(struct snd_soc_codec *codec)
{
struct ak4641_platform_data *pdata = codec->dev->platform_data;
ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
if (pdata) {
if (gpio_is_valid(pdata->gpio_power)) {
gpio_set_value(pdata->gpio_power, 0);
gpio_free(pdata->gpio_power);
}
if (gpio_is_valid(pdata->gpio_npdn))
gpio_free(pdata->gpio_npdn);
}
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
.probe = ak4641_probe,
.remove = ak4641_remove,
.suspend = ak4641_suspend,
.resume = ak4641_resume,
.controls = ak4641_snd_controls,
.num_controls = ARRAY_SIZE(ak4641_snd_controls),
.dapm_widgets = ak4641_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4641_dapm_widgets),
.dapm_routes = ak4641_audio_map,
.num_dapm_routes = ARRAY_SIZE(ak4641_audio_map),
.set_bias_level = ak4641_set_bias_level,
.reg_cache_size = ARRAY_SIZE(ak4641_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = ak4641_reg,
.reg_cache_step = 1,
};
static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ak4641_priv *ak4641;
int ret;
ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL);
if (!ak4641)
return -ENOMEM;
i2c_set_clientdata(i2c, ak4641);
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
ak4641_dai, ARRAY_SIZE(ak4641_dai));
if (ret < 0)
kfree(ak4641);
return ret;
}
static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
{
snd_soc_unregister_codec(&i2c->dev);
kfree(i2c_get_clientdata(i2c));
return 0;
}
static const struct i2c_device_id ak4641_i2c_id[] = {
{ "ak4641", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id);
static struct i2c_driver ak4641_i2c_driver = {
.driver = {
.name = "ak4641",
.owner = THIS_MODULE,
},
.probe = ak4641_i2c_probe,
.remove = __devexit_p(ak4641_i2c_remove),
.id_table = ak4641_i2c_id,
};
static int __init ak4641_modinit(void)
{
int ret;
ret = i2c_add_driver(&ak4641_i2c_driver);
if (ret != 0)
pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
return ret;
}
module_init(ak4641_modinit);
static void __exit ak4641_exit(void)
{
i2c_del_driver(&ak4641_i2c_driver);
}
module_exit(ak4641_exit);
MODULE_DESCRIPTION("SoC AK4641 driver");
MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
MODULE_LICENSE("GPL");

47
sound/soc/codecs/ak4641.h Normal file
View file

@ -0,0 +1,47 @@
/*
* ak4641.h -- AK4641 SoC Audio driver
*
* Copyright 2008 Harald Welte <laforge@gnufiish.org>
*
* Based on ak4535.h
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _AK4641_H
#define _AK4641_H
/* AK4641 register space */
#define AK4641_PM1 0x00
#define AK4641_PM2 0x01
#define AK4641_SIG1 0x02
#define AK4641_SIG2 0x03
#define AK4641_MODE1 0x04
#define AK4641_MODE2 0x05
#define AK4641_DAC 0x06
#define AK4641_MIC 0x07
#define AK4641_TIMER 0x08
#define AK4641_ALC1 0x09
#define AK4641_ALC2 0x0a
#define AK4641_PGA 0x0b
#define AK4641_LATT 0x0c
#define AK4641_RATT 0x0d
#define AK4641_VOL 0x0e
#define AK4641_STATUS 0x0f
#define AK4641_EQLO 0x10
#define AK4641_EQMID 0x11
#define AK4641_EQHI 0x12
#define AK4641_BTIF 0x13
#define AK4641_CACHEREGNUM 0x14
#define AK4641_DAI_HIFI 0
#define AK4641_DAI_VOICE 1
#endif

26
sound/soc/codecs/dmic.c
View file

@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = {
},
};
static struct snd_soc_codec_driver soc_dmic = {};
static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_INPUT("DMic"),
};
static const struct snd_soc_dapm_route intercon[] = {
{"DMIC AIF", NULL, "DMic"},
};
static int dmic_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
ARRAY_SIZE(dmic_dapm_widgets));
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(dapm);
return 0;
}
static struct snd_soc_codec_driver soc_dmic = {
.probe = dmic_probe,
};
static int __devinit dmic_dev_probe(struct platform_device *pdev)
{

62
sound/soc/codecs/max98088.c
View file

@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =
ARRAY_SIZE(max98088_exmode_texts),
max98088_exmode_texts,
max98088_exmode_values);
static const struct snd_kcontrol_new max98088_exmode_controls =
SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum);
static const char *max98088_ex_thresh[] = { /* volts PP */
"0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"};
@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0),
SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0),
SOC_ENUM("EX Limiter Mode", max98088_exmode_enum),
SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum),
SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum),
@ -808,10 +807,10 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
/* Left speaker mixer switch */
static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0),
@ -836,10 +835,10 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = {
/* Left headphone mixer switch */
static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = {
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0),
@ -864,10 +863,10 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = {
/* Left earpiece/receiver mixer switch */
static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = {
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0),
@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = {
SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),
SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0,
&max98088_exmode_controls),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("SPKL"),
@ -1568,6 +1564,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
int reg;
if (mute)
reg = M98088_DAI_MUTE;
else
reg = 0;
snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY,
M98088_DAI_MUTE_MASK, reg);
return 0;
}
static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
int reg;
if (mute)
reg = M98088_DAI_MUTE;
else
reg = 0;
snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY,
M98088_DAI_MUTE_MASK, reg);
return 0;
}
static void max98088_sync_cache(struct snd_soc_codec *codec)
{
u16 *reg_cache = codec->reg_cache;
@ -1629,12 +1655,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops = {
.set_sysclk = max98088_dai_set_sysclk,
.set_fmt = max98088_dai1_set_fmt,
.hw_params = max98088_dai1_hw_params,
.digital_mute = max98088_dai1_digital_mute,
};
static struct snd_soc_dai_ops max98088_dai2_ops = {
.set_sysclk = max98088_dai_set_sysclk,
.set_fmt = max98088_dai2_set_fmt,
.hw_params = max98088_dai2_hw_params,
.digital_mute = max98088_dai2_digital_mute,
};
static struct snd_soc_dai_driver max98088_dai[] = {

13
sound/soc/codecs/max98088.h
View file

@ -133,6 +133,19 @@
#define M98088_REC_LINEMODE (1<<7)
#define M98088_REC_LINEMODE_MASK (1<<7)
/* M98088_REG_2D_MIX_SPK_CNTL */
#define M98088_MIX_SPKR_GAIN_MASK (3<<2)
#define M98088_MIX_SPKR_GAIN_SHIFT 2
#define M98088_MIX_SPKL_GAIN_MASK (3<<0)
#define M98088_MIX_SPKL_GAIN_SHIFT 0
/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */
#define M98088_DAI_MUTE (1<<7)
#define M98088_DAI_MUTE_MASK (1<<7)
#define M98088_DAI_VOICE_GAIN_MASK (3<<4)
#define M98088_DAI_ATTENUATION_MASK (0xF<<0)
#define M98088_DAI_ATTENUATION_SHIFT 0
/* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */
#define M98088_MICPRE_MASK (3<<5)
#define M98088_MICPRE_SHIFT 5

16
sound/soc/codecs/max98095.c
View file

@ -1870,16 +1870,14 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
BUG_ON(channel > 1);
cdata = &max98095->dai[channel];
if (!pdata || !max98095->eq_textcnt)
return 0;
if (sel >= pdata->eq_cfgcnt)
return -EINVAL;
cdata = &max98095->dai[channel];
cdata->eq_sel = sel;
if (!pdata || !max98095->eq_textcnt)
return 0;
fs = cdata->rate;
/* Find the selected configuration with nearest sample rate */
@ -2018,16 +2016,14 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
BUG_ON(channel > 1);
cdata = &max98095->dai[channel];
if (!pdata || !max98095->bq_textcnt)
return 0;
if (sel >= pdata->bq_cfgcnt)
return -EINVAL;
cdata = &max98095->dai[channel];
cdata->bq_sel = sel;
if (!pdata || !max98095->bq_textcnt)
return 0;
fs = cdata->rate;
/* Find the selected configuration with nearest sample rate */

8
sound/soc/codecs/spdif_transciever.c
View file

@ -21,7 +21,7 @@
#include <sound/pcm.h>
#include <sound/initval.h>
MODULE_LICENSE("GPL");
#define DRV_NAME "spdif-dit"
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = {
.probe = spdif_dit_probe,
.remove = spdif_dit_remove,
.driver = {
.name = "spdif-dit",
.name = DRV_NAME,
.owner = THIS_MODULE,
},
};
@ -74,3 +74,7 @@ static void __exit dit_exit(void)
module_init(dit_modinit);
module_exit(dit_exit);
MODULE_AUTHOR("Steve Chen <schen@mvista.com>");
MODULE_DESCRIPTION("SPDIF dummy codec driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:" DRV_NAME);

2
sound/soc/codecs/ssm2602.c
View file

@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0),
SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),

3
sound/soc/codecs/tlv320aic3x.c
View file

@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;

4
sound/soc/codecs/tlv320dac33.c
View file

@ -1,7 +1,7 @@
/*
* ALSA SoC Texas Instruments TLV320DAC33 codec driver
*
* Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*
@ -1658,5 +1658,5 @@ module_exit(dac33_module_exit);
MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver");
MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>");
MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
MODULE_LICENSE("GPL");

2
sound/soc/codecs/tlv320dac33.h
View file

@ -1,7 +1,7 @@
/*
* ALSA SoC Texas Instruments TLV320DAC33 codec driver
*
* Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*

4
sound/soc/codecs/tpa6130a2.c
View file

@ -3,7 +3,7 @@
*
* Copyright (C) Nokia Corporation
*
* Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -495,7 +495,7 @@ static void __exit tpa6130a2_exit(void)
i2c_del_driver(&tpa6130a2_i2c_driver);
}
MODULE_AUTHOR("Peter Ujfalusi");
MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver");
MODULE_LICENSE("GPL");

2
sound/soc/codecs/tpa6130a2.h
View file

@ -3,7 +3,7 @@
*
* Copyright (C) Nokia Corporation
*
* Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License

6
sound/soc/codecs/twl6040.c
View file

@ -960,9 +960,9 @@ static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0);
/*
* AFMGAIN volume control:
* from 18 to 24 dB in 6 dB steps
* from -18 to 24 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0);
static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0);
/*
* HSGAIN volume control:
@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
/* AFM gains */
SOC_DOUBLE_TLV("Aux FM Volume",
TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv),
TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume",

2
sound/soc/codecs/wm1250-ev1.c
View file

@ -22,7 +22,7 @@ SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DAC", "wm1250-ev1 Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_INPUT("WM1250 Input"),
SND_SOC_DAPM_INPUT("WM1250 Output"),
SND_SOC_DAPM_OUTPUT("WM1250 Output"),
};
static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {

2
sound/soc/codecs/wm8731.c
View file

@ -198,7 +198,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
{
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
return wm8731->sysclk_type == WM8731_SYSCLK_MCLK;
return wm8731->sysclk_type == WM8731_SYSCLK_XTAL;
}
static const struct snd_soc_dapm_route wm8731_intercon[] = {

3
sound/soc/codecs/wm8903.c
View file

@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct snd_soc_codec *codec = widget->codec;
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
u16 reg;

1
sound/soc/codecs/wm8915.c
View file

@ -19,7 +19,6 @@
#include <linux/gcd.h>
#include <linux/gpio.h>
#include <linux/i2c.h>
#include <linux/delay.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
#include <linux/workqueue.h>

29
sound/soc/codecs/wm8958-dsp2.c
View file

@ -99,7 +99,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
len = fw->size - len;
while (len) {
if (len < 12) {
dev_err(codec->dev, "%s short data block of %d\n",
dev_err(codec->dev, "%s short data block of %zd\n",
name, len);
goto err;
}
@ -107,7 +107,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
memcpy(&data32, data + 4, sizeof(data32));
block_len = be32_to_cpu(data32);
if (block_len + 8 > len) {
dev_err(codec->dev, "%d byte block longer than file\n",
dev_err(codec->dev, "%zd byte block longer than file\n",
block_len);
goto err;
}
@ -141,7 +141,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
case WM_FW_BLOCK_I:
case WM_FW_BLOCK_A:
case WM_FW_BLOCK_C:
dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name,
dev_dbg(codec->dev, "%s: %zd bytes of %x@%x\n", name,
block_len, (data32 >> 24) & 0xff,
data32 & 0xffffff);
@ -362,6 +362,10 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start)
path, wm8994->dsp_active, start, pwr_reg, reg);
if (start && ena) {
/* If the DSP is already running then noop */
if (reg & WM8958_DSP2_ENA)
return;
/* If either AIFnCLK is not yet enabled postpone */
if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
& WM8994_AIF1CLK_ENA_MASK) &&
@ -508,6 +512,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0])
return 0;
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@ -628,6 +635,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0])
return 0;
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@ -689,6 +699,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (hpf < 3) {
if (wm8994->hpf1_ena[hpf % 3] ==
ucontrol->value.integer.value[0])
return 0;
} else {
if (wm8994->hpf2_ena[hpf % 3] ==
ucontrol->value.integer.value[0])
return 0;
}
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@ -782,6 +802,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0])
return 0;
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;

3
sound/soc/codecs/wm8993.c
View file

@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
static int class_w_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct snd_soc_codec *codec = widget->codec;
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
int ret;

31
sound/soc/codecs/wm8994.c
View file

@ -877,7 +877,8 @@ static const char *hp_mux_text[] = {
static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec = w->codec;
int ret;
@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec = w->codec;
int ret;
@ -2416,8 +2418,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = codec->control_data;
int i, ret;
switch (control->type) {
case WM8994:
snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
break;
case WM8958:
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, 0);
break;
}
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
sizeof(struct wm8994_fll_config));
@ -2435,6 +2448,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
static int wm8994_resume(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = codec->control_data;
int i, ret;
unsigned int val, mask;
@ -2473,6 +2487,19 @@ static int wm8994_resume(struct snd_soc_codec *codec)
i + 1, ret);
}
switch (control->type) {
case WM8994:
if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
snd_soc_update_bits(codec, WM8994_MICBIAS,
WM8994_MICD_ENA, WM8994_MICD_ENA);
break;
case WM8958:
if (wm8994->jack_cb)
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, WM8958_MICD_ENA);
break;
}
return 0;
}
#else

4
sound/soc/codecs/wm8995.c
View file

@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source,
static int wm8995_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *w;
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec;
int ret;
w = snd_kcontrol_chip(kcontrol);
codec = w->codec;
ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
wm8995_update_class_w(codec);

32
sound/soc/codecs/wm_hubs.c
View file

@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = {
SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
inmix_sw_tlv),
@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
{ "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" },
{ "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" },
{ "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" },
{ "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" },
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" },
@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
{ "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
{ "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" },
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" },
@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME,
WM8993_IN2_VU, WM8993_IN2_VU);
snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT,
WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT,
WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME,
WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC);
WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC,
WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC);
snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC);
snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME,
WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC);
WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU,
WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU);
snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU);

2
sound/soc/davinci/davinci-mcasp.c
View file

@ -909,6 +909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->sram_size = pdata->sram_size_playback;
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
mem->start);
@ -925,6 +926,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->sram_size = pdata->sram_size_capture;
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
mem->start);

6
sound/soc/omap/omap-mcbsp.c
View file

@ -4,7 +4,7 @@
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jhnikula@gmail.com>
* Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
* 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
* 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
*/
if (cpu_is_omap343x() || cpu_is_omap44xx()) {
if (cpu_is_omap34xx() || cpu_is_omap44xx()) {
/*
* Rule for the buffer size. We should not allow
* smaller buffer than the FIFO size to avoid underruns
@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
default:
return -EINVAL;
}
if (cpu_is_omap343x()) {
if (cpu_is_omap34xx()) {
dma_data->set_threshold = omap_mcbsp_set_threshold;
/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (omap_mcbsp_get_dma_op_mode(bus_id) ==

2
sound/soc/omap/omap-mcbsp.h
View file

@ -4,7 +4,7 @@
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jhnikula@gmail.com>
* Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License

7
sound/soc/omap/omap-pcm.c
View file

@ -4,7 +4,7 @@
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jhnikula@gmail.com>
* Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = 32,
@ -195,7 +196,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
if ((cpu_is_omap1510()))
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
else
else if (!substream->runtime->no_period_wakeup)
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
if (!(cpu_class_is_omap1())) {

2
sound/soc/omap/omap-pcm.h
View file

@ -4,7 +4,7 @@
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jhnikula@gmail.com>
* Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License

2
sound/soc/omap/rx51.c
View file

@ -3,7 +3,7 @@
*
* Copyright (C) 2008 - 2009 Nokia Corporation
*
* Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com>
* Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
* Eduardo Valentin <eduardo.valentin@nokia.com>
* Jarkko Nikula <jhnikula@gmail.com>
*

9
sound/soc/pxa/Kconfig
View file

@ -155,6 +155,15 @@ config SND_SOC_RAUMFELD
help
Say Y if you want to add support for SoC audio on Raumfeld devices
config SND_PXA2XX_SOC_HX4700
tristate "SoC Audio support for HP iPAQ hx4700"
depends on SND_PXA2XX_SOC && MACH_H4700
select SND_PXA2XX_SOC_I2S
select SND_SOC_AK4641
help
Say Y if you want to add support for SoC audio on the
HP iPAQ hx4700.
config SND_PXA2XX_SOC_MAGICIAN
tristate "SoC Audio support for HTC Magician"
depends on SND_PXA2XX_SOC && MACH_MAGICIAN

2
sound/soc/pxa/Makefile
View file

@ -22,6 +22,7 @@ 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
snd-soc-mioa701-objs := mioa701_wm9713.o
snd-soc-z2-objs := z2.o
@ -37,6 +38,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
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

255
sound/soc/pxa/hx4700.c Normal file
View file

@ -0,0 +1,255 @@
/*
* SoC audio for HP iPAQ hx4700
*
* Copyright (c) 2009 Philipp Zabel
*
* 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.
*
*/
#include <linux/module.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/delay.h>
#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <mach/hx4700.h>
#include <asm/mach-types.h>
#include "pxa2xx-i2s.h"
#include "../codecs/ak4641.h"
static struct snd_soc_jack hs_jack;
/* Headphones jack detection DAPM pin */
static struct snd_soc_jack_pin hs_jack_pin[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "Speaker",
/* disable speaker when hp jack is inserted */
.mask = SND_JACK_HEADPHONE,
.invert = 1,
},
};
/* Headphones jack detection GPIO */
static struct snd_soc_jack_gpio hs_jack_gpio = {
.gpio = GPIO75_HX4700_EARPHONE_nDET,
.invert = true,
.name = "hp-gpio",
.report = SND_JACK_HEADPHONE,
.debounce_time = 200,
};
/*
* iPAQ hx4700 uses I2S for capture and playback.
*/
static int hx4700_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 ret = 0;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the I2S system clock as output */
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
/* inform codec driver about clock freq *
* (PXA I2S always uses divider 256) */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params),
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops hx4700_ops = {
.hw_params = hx4700_hw_params,
};
static int hx4700_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
static int hx4700_hp_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
/* hx4700 machine dapm widgets */
static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power),
SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power),
SND_SOC_DAPM_MIC("Built-in Microphone", NULL),
};
/* hx4700 machine audio_map */
static const struct snd_soc_dapm_route hx4700_audio_map[] = {
/* Headphone connected to LOUT, ROUT */
{"Headphone Jack", NULL, "LOUT"},
{"Headphone Jack", NULL, "ROUT"},
/* Speaker connected to MOUT2 */
{"Speaker", NULL, "MOUT2"},
/* Microphone connected to MICIN */
{"MICIN", NULL, "Built-in Microphone"},
{"AIN", NULL, "MICOUT"},
};
/*
* Logic for a ak4641 as connected on a HP iPAQ hx4700
*/
static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* NC codec pins */
/* FIXME: is anything connected here? */
snd_soc_dapm_nc_pin(dapm, "MOUT1");
snd_soc_dapm_nc_pin(dapm, "MICEXT");
snd_soc_dapm_nc_pin(dapm, "AUX");
/* Jack detection API stuff */
err = snd_soc_jack_new(codec, "Headphone Jack",
SND_JACK_HEADPHONE, &hs_jack);
if (err)
return err;
err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin),
hs_jack_pin);
if (err)
return err;
err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio);
return err;
}
/* hx4700 digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link hx4700_dai = {
.name = "ak4641",
.stream_name = "AK4641",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "ak4641-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "ak4641.0-0012",
.init = hx4700_ak4641_init,
.ops = &hx4700_ops,
};
/* hx4700 audio machine driver */
static struct snd_soc_card snd_soc_card_hx4700 = {
.name = "iPAQ hx4700",
.dai_link = &hx4700_dai,
.num_links = 1,
.dapm_widgets = hx4700_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets),
.dapm_routes = hx4700_audio_map,
.num_dapm_routes = ARRAY_SIZE(hx4700_audio_map),
};
static struct gpio hx4700_audio_gpios[] = {
{ GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" },
{ GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" },
};
static int __devinit hx4700_audio_probe(struct platform_device *pdev)
{
int ret;
if (!machine_is_h4700())
return -ENODEV;
ret = gpio_request_array(hx4700_audio_gpios,
ARRAY_SIZE(hx4700_audio_gpios));
if (ret)
return ret;
snd_soc_card_hx4700.dev = &pdev->dev;
ret = snd_soc_register_card(&snd_soc_card_hx4700);
if (ret)
return ret;
return 0;
}
static int __devexit hx4700_audio_remove(struct platform_device *pdev)
{
snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
snd_soc_unregister_card(&snd_soc_card_hx4700);
gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios));
return 0;
}
static struct platform_driver hx4700_audio_driver = {
.driver = {
.name = "hx4700-audio",
.owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
},
.probe = hx4700_audio_probe,
.remove = __devexit_p(hx4700_audio_remove),
};
static int __init hx4700_modinit(void)
{
return platform_driver_register(&hx4700_audio_driver);
}
module_init(hx4700_modinit);
static void __exit hx4700_modexit(void)
{
platform_driver_unregister(&hx4700_audio_driver);
}
module_exit(hx4700_modexit);
MODULE_AUTHOR("Philipp Zabel");
MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:hx4700-audio");

94
sound/soc/pxa/raumfeld.c
View file

@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = {
.hw_params = raumfeld_cs4270_hw_params,
};
static int raumfeld_line_suspend(struct snd_soc_card *card)
static int raumfeld_analog_suspend(struct snd_soc_card *card)
{
raumfeld_enable_audio(false);
return 0;
}
static int raumfeld_line_resume(struct snd_soc_card *card)
static int raumfeld_analog_resume(struct snd_soc_card *card)
{
raumfeld_enable_audio(true);
return 0;
@ -225,32 +225,53 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
.hw_params = raumfeld_ak4104_hw_params,
};
static struct snd_soc_dai_link raumfeld_dai[] = {
{
.name = "ak4104",
.stream_name = "Playback",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "ak4104-hifi",
.platform_name = "pxa-pcm-audio",
.ops = &raumfeld_ak4104_ops,
.codec_name = "ak4104-codec.0",
},
{
.name = "CS4270",
.stream_name = "CS4270",
.cpu_dai_name = "pxa-ssp-dai.0",
.platform_name = "pxa-pcm-audio",
.codec_dai_name = "cs4270-hifi",
.codec_name = "cs4270-codec.0-0048",
.ops = &raumfeld_cs4270_ops,
},};
#define DAI_LINK_CS4270 \
{ \
.name = "CS4270", \
.stream_name = "CS4270", \
.cpu_dai_name = "pxa-ssp-dai.0", \
.platform_name = "pxa-pcm-audio", \
.codec_dai_name = "cs4270-hifi", \
.codec_name = "cs4270-codec.0-0048", \
.ops = &raumfeld_cs4270_ops, \
}
static struct snd_soc_card snd_soc_raumfeld = {
.name = "Raumfeld",
.dai_link = raumfeld_dai,
.suspend_post = raumfeld_line_suspend,
.resume_pre = raumfeld_line_resume,
.num_links = ARRAY_SIZE(raumfeld_dai),
#define DAI_LINK_AK4104 \
{ \
.name = "ak4104", \
.stream_name = "Playback", \
.cpu_dai_name = "pxa-ssp-dai.1", \
.codec_dai_name = "ak4104-hifi", \
.platform_name = "pxa-pcm-audio", \
.ops = &raumfeld_ak4104_ops, \
.codec_name = "spi0.0", \
}
static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] =
{
DAI_LINK_CS4270,
DAI_LINK_AK4104,
};
static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] =
{
DAI_LINK_CS4270,
};
static struct snd_soc_card snd_soc_raumfeld_connector = {
.name = "Raumfeld Connector",
.dai_link = snd_soc_raumfeld_connector_dai,
.num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai),
.suspend_post = raumfeld_analog_suspend,
.resume_pre = raumfeld_analog_resume,
};
static struct snd_soc_card snd_soc_raumfeld_speaker = {
.name = "Raumfeld Speaker",
.dai_link = snd_soc_raumfeld_speaker_dai,
.num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai),
.suspend_post = raumfeld_analog_suspend,
.resume_pre = raumfeld_analog_resume,
};
static struct platform_device *raumfeld_audio_device;
@ -271,22 +292,25 @@ static int __init raumfeld_audio_init(void)
set_max9485_clk(MAX9485_MCLK_FREQ_122880);
/* Register LINE and SPDIF */
/* Register analog device */
raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
if (!raumfeld_audio_device)
return -ENOMEM;
platform_set_drvdata(raumfeld_audio_device,
&snd_soc_raumfeld);
ret = platform_device_add(raumfeld_audio_device);
/* no S/PDIF on Speakers */
if (machine_is_raumfeld_speaker())
platform_set_drvdata(raumfeld_audio_device,
&snd_soc_raumfeld_speaker);
if (machine_is_raumfeld_connector())
platform_set_drvdata(raumfeld_audio_device,
&snd_soc_raumfeld_connector);
ret = platform_device_add(raumfeld_audio_device);
if (ret < 0)
return ret;
raumfeld_enable_audio(true);
return ret;
return 0;
}
static void __exit raumfeld_audio_exit(void)

140
sound/soc/soc-cache.c
View file

@ -20,25 +20,15 @@
#include <trace/events/asoc.h>
#if defined(CONFIG_SPI_MASTER)
static int do_spi_write(void *control_data, const void *msg,
int len)
#ifdef CONFIG_SPI_MASTER
static int do_spi_write(void *control, const char *data, int len)
{
struct spi_device *spi = control_data;
struct spi_transfer t;
struct spi_message m;
struct spi_device *spi = control;
int ret;
if (len <= 0)
return 0;
spi_message_init(&m);
memset(&t, 0, sizeof t);
t.tx_buf = msg;
t.len = len;
spi_message_add_tail(&t, &m);
spi_sync(spi, &m);
ret = spi_write(spi, data, len);
if (ret < 0)
return ret;
return len;
}
@ -101,29 +91,13 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[2];
u16 data;
data[0] = (reg << 4) | ((value >> 8) & 0x000f);
data[1] = value & 0x00ff;
data = cpu_to_be16((reg << 12) | (value & 0xffffff));
return do_hw_write(codec, reg, value, data, 2);
return do_hw_write(codec, reg, value, &data, 2);
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_4_12_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[2];
msg[0] = data[1];
msg[1] = data[0];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_4_12_spi_write NULL
#endif
static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@ -140,21 +114,6 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
return do_hw_write(codec, reg, value, &data, 2);
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_7_9_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[2];
msg[0] = data[0];
msg[1] = data[1];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_7_9_spi_write NULL
#endif
static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@ -173,21 +132,6 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
return do_hw_read(codec, reg);
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_8_8_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[2];
msg[0] = data[0];
msg[1] = data[1];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_8_8_spi_write NULL
#endif
static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@ -206,22 +150,6 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
return do_hw_read(codec, reg);
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_8_16_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[3];
msg[0] = data[0];
msg[1] = data[1];
msg[2] = data[2];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_8_16_spi_write NULL
#endif
#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
static unsigned int do_i2c_read(struct snd_soc_codec *codec,
void *reg, int reglen,
@ -318,27 +246,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
memcpy(data, &rval, sizeof(rval));
data[2] = value;
reg &= 0xff;
return do_hw_write(codec, reg, value, data, 3);
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_16_8_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[3];
msg[0] = data[0];
msg[1] = data[1];
msg[2] = data[2];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_16_8_spi_write NULL
#endif
#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
unsigned int r)
@ -373,23 +284,6 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
return do_hw_write(codec, reg, value, data, sizeof(data));
}
#if defined(CONFIG_SPI_MASTER)
static int snd_soc_16_16_spi_write(void *control_data, const char *data,
int len)
{
u8 msg[4];
msg[0] = data[0];
msg[1] = data[1];
msg[2] = data[2];
msg[3] = data[3];
return do_spi_write(control_data, msg, len);
}
#else
#define snd_soc_16_16_spi_write NULL
#endif
/* Primitive bulk write support for soc-cache. The data pointed to by
* `data' needs to already be in the form the hardware expects
* including any leading register specific data. Any data written
@ -419,7 +313,7 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r
#endif
#if defined(CONFIG_SPI_MASTER)
case SND_SOC_SPI:
ret = do_spi_write(codec->control_data, data, len);
ret = spi_write(codec->control_data, data, len);
break;
#endif
default:
@ -438,43 +332,36 @@ static struct {
int addr_bits;
int data_bits;
int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
int (*spi_write)(void *, const char *, int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
{
.addr_bits = 4, .data_bits = 12,
.write = snd_soc_4_12_write, .read = snd_soc_4_12_read,
.spi_write = snd_soc_4_12_spi_write,
},
{
.addr_bits = 7, .data_bits = 9,
.write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
.spi_write = snd_soc_7_9_spi_write,
},
{
.addr_bits = 8, .data_bits = 8,
.write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
.i2c_read = snd_soc_8_8_read_i2c,
.spi_write = snd_soc_8_8_spi_write,
},
{
.addr_bits = 8, .data_bits = 16,
.write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
.i2c_read = snd_soc_8_16_read_i2c,
.spi_write = snd_soc_8_16_spi_write,
},
{
.addr_bits = 16, .data_bits = 8,
.write = snd_soc_16_8_write, .read = snd_soc_16_8_read,
.i2c_read = snd_soc_16_8_read_i2c,
.spi_write = snd_soc_16_8_spi_write,
},
{
.addr_bits = 16, .data_bits = 16,
.write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
.i2c_read = snd_soc_16_16_read_i2c,
.spi_write = snd_soc_16_16_spi_write,
},
};
@ -535,8 +422,9 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
break;
case SND_SOC_SPI:
if (io_types[i].spi_write)
codec->hw_write = io_types[i].spi_write;
#ifdef CONFIG_SPI_MASTER
codec->hw_write = do_spi_write;
#endif
codec->control_data = container_of(codec->dev,
struct spi_device,

16
sound/soc/soc-core.c
View file

@ -242,7 +242,7 @@ static ssize_t codec_reg_write_file(struct file *file,
const char __user *user_buf, size_t count, loff_t *ppos)
{
char buf[32];
int buf_size;
size_t buf_size;
char *start = buf;
unsigned long reg, value;
int step = 1;
@ -1307,10 +1307,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
/* no, then find CPU DAI from registered DAIs*/
list_for_each_entry(cpu_dai, &dai_list, list) {
if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) {
if (!try_module_get(cpu_dai->dev->driver->owner))
return -ENODEV;
rtd->cpu_dai = cpu_dai;
goto find_codec;
}
@ -1623,11 +1619,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
/* probe the cpu_dai */
if (!cpu_dai->probed) {
if (!try_module_get(cpu_dai->dev->driver->owner))
return -ENODEV;
if (cpu_dai->driver->probe) {
ret = cpu_dai->driver->probe(cpu_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n",
cpu_dai->name);
module_put(cpu_dai->dev->driver->owner);
return ret;
}
}
@ -1927,9 +1927,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
card->num_dapm_routes);
snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
"%s", card->driver_name ? card->driver_name : card->name);
if (card->late_probe) {
ret = card->late_probe(card);

7
sound/soc/soc-dapm.c
View file

@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
}
static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *kcontrolw,
const struct snd_kcontrol_new *kcontrol_new,
struct snd_kcontrol **kcontrol)
{
@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
*kcontrol = NULL;
list_for_each_entry(w, &dapm->card->widgets, list) {
if (w == kcontrolw || w->dapm != kcontrolw->dapm)
continue;
for (i = 0; i < w->num_kcontrols; i++) {
if (&w->kcontrol_news[i] == kcontrol_new) {
if (w->kcontrols)
@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
return -EINVAL;
}
shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0],
shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0],
&kcontrol);
if (kcontrol) {
wlist = kcontrol->private_data;
@ -1110,7 +1113,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
trace_snd_soc_dapm_start(card);
list_for_each_entry(d, &card->dapm_list, list)
if (d->n_widgets)
if (d->n_widgets || d->codec == NULL)
d->dev_power = 0;
/* Check which widgets we need to power and store them in

2
sound/soc/tegra/tegra_i2s.c
View file

@ -114,7 +114,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
debugfs_remove(i2s->debug);
}
#else
static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
{
}