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Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623', 'asoc/topic/alc5632', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next

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This commit is contained in:
Mark Brown 2014-12-08 13:11:50 +00:00
parent 4aad340f1b e48d73c697 2849bde56a 5c9dc0898f e9c7f34a7e de82bf6c05
commit 1f491557d2
8 changed files with 33 additions and 220 deletions
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9
sound/soc/atmel/Kconfig
View file

@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731
help
Say Y if you want to add support for audio SoC on an
at91sam9x5 based board that is using WM8731 codec.
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
select SND_ATMEL_SOC_PDC
select SND_ATMEL_SOC_SSC
select SND_SOC_TLV320AIC23_I2C
help
Say Y here to support sound on AFEB9260 board.

1
sound/soc/atmel/Makefile
View file

@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o

5
sound/soc/atmel/atmel_ssc_dai.c
View file

@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
* transmit and receive, so if a value has already
* been set, it must match this value.
*/
if (ssc_p->cmr_div == 0)
if (ssc_p->dir_mask !=
(SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE))
ssc_p->cmr_div = div;
else if (ssc_p->cmr_div == 0)
ssc_p->cmr_div = div;
else
if (div != ssc_p->cmr_div)

151
sound/soc/atmel/snd-soc-afeb9260.c
View file

@ -1,151 +0,0 @@
/*
* 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 <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->codec_dai;
int 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 afeb9260_audio_map[] = {
{"Headphone Jack", NULL, "LHPOUT"},
{"Headphone Jack", NULL, "RHPOUT"},
{"LLINEIN", NULL, "Line In"},
{"RLINEIN", NULL, "Line In"},
{"MICIN", NULL, "Mic Jack"},
};
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link afeb9260_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
.codec_name = "tlv320aic23-codec.0-001a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &afeb9260_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_machine_afeb9260 = {
.name = "AFEB9260",
.owner = THIS_MODULE,
.dai_link = &afeb9260_dai,
.num_links = 1,
.dapm_widgets = tlv320aic23_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
.dapm_routes = afeb9260_audio_map,
.num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
};
static struct platform_device *afeb9260_snd_device;
static int __init afeb9260_soc_init(void)
{
int err;
struct device *dev;
if (!(machine_is_afeb9260()))
return -ENODEV;
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, &snd_soc_machine_afeb9260);
err = platform_device_add(afeb9260_snd_device);
if (err)
goto err1;
dev = &afeb9260_snd_device->dev;
return 0;
err1:
platform_device_put(afeb9260_snd_device);
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");

13
sound/soc/codecs/ak4671.c
View file

@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
.ops = &ak4671_dai_ops,
};
static int ak4671_probe(struct snd_soc_codec *codec)
{
return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
}
static int ak4671_remove(struct snd_soc_codec *codec)
{
ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
.probe = ak4671_probe,
.remove = ak4671_remove,
.set_bias_level = ak4671_set_bias_level,
.controls = ak4671_snd_controls,
.num_controls = ARRAY_SIZE(ak4671_snd_controls),

22
sound/soc/codecs/alc5623.c
View file

@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec)
{
struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
regcache_cache_only(alc5623->regmap, true);
return 0;
@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec)
return ret;
}
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge alc5623 caps */
if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
codec->dapm.bias_level = SND_SOC_BIAS_ON;
alc5623_set_bias_level(codec, codec->dapm.bias_level);
}
return 0;
}
@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec)
alc5623_reset(codec);
/* power on device */
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (alc5623->add_ctrl) {
snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
alc5623->add_ctrl);
@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec)
return 0;
}
/* power down chip */
static int alc5623_remove(struct snd_soc_codec *codec)
{
alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
.probe = alc5623_probe,
.remove = alc5623_remove,
.suspend = alc5623_suspend,
.resume = alc5623_resume,
.set_bias_level = alc5623_set_bias_level,
.suspend_bias_off = true,
};
static const struct regmap_config alc5623_regmap = {

22
sound/soc/codecs/alc5632.c
View file

@ -1038,23 +1038,15 @@ static struct snd_soc_dai_driver alc5632_dai = {
};
#ifdef CONFIG_PM
static int alc5632_suspend(struct snd_soc_codec *codec)
{
alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int alc5632_resume(struct snd_soc_codec *codec)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
regcache_sync(alc5632->regmap);
alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
#else
#define alc5632_suspend NULL
#define alc5632_resume NULL
#endif
@ -1062,9 +1054,6 @@ static int alc5632_probe(struct snd_soc_codec *codec)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
/* power on device */
alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
switch (alc5632->id) {
case 0x5c:
snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls,
@ -1077,19 +1066,12 @@ static int alc5632_probe(struct snd_soc_codec *codec)
return 0;
}
/* power down chip */
static int alc5632_remove(struct snd_soc_codec *codec)
{
alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
.probe = alc5632_probe,
.remove = alc5632_remove,
.suspend = alc5632_suspend,
.resume = alc5632_resume,
.set_bias_level = alc5632_set_bias_level,
.suspend_bias_off = true,
.controls = alc5632_snd_controls,
.num_controls = ARRAY_SIZE(alc5632_snd_controls),
.dapm_widgets = alc5632_dapm_widgets,

30
sound/soc/codecs/arizona.c
View file

@ -61,6 +61,11 @@
#define ARIZONA_FLL_MIN_OUTDIV 2
#define ARIZONA_FLL_MAX_OUTDIV 7
#define ARIZONA_FMT_DSP_MODE_A 0
#define ARIZONA_FMT_DSP_MODE_B 1
#define ARIZONA_FMT_I2S_MODE 2
#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3
#define arizona_fll_err(_fll, fmt, ...) \
dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_warn(_fll, fmt, ...) \
@ -648,7 +653,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum,
EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum);
static const char * const arizona_in_dmic_osr_text[] = {
"1.536MHz", "3.072MHz", "6.144MHz",
"1.536MHz", "3.072MHz", "6.144MHz", "768kHz",
};
const struct soc_enum arizona_in_dmic_osr[] = {
@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
mode = 0;
mode = ARIZONA_FMT_DSP_MODE_A;
break;
case SND_SOC_DAIFMT_DSP_B:
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK)
!= SND_SOC_DAIFMT_CBM_CFM) {
arizona_aif_err(dai, "DSP_B not valid in slave mode\n");
return -EINVAL;
}
mode = ARIZONA_FMT_DSP_MODE_B;
break;
case SND_SOC_DAIFMT_I2S:
mode = 2;
mode = ARIZONA_FMT_I2S_MODE;
break;
case SND_SOC_DAIFMT_LEFT_J:
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK)
!= SND_SOC_DAIFMT_CBM_CFM) {
arizona_aif_err(dai, "LEFT_J not valid in slave mode\n");
return -EINVAL;
}
mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE;
break;
default:
arizona_aif_err(dai, "Unsupported DAI format %d\n",
@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
/* Force multiple of 2 channels for I2S mode */
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) {
val &= ARIZONA_AIF1_FMT_MASK;
if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) {
arizona_aif_dbg(dai, "Forcing stereo mode\n");
bclk_target /= channels;
bclk_target *= channels + 1;