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WSL2-Linux-Kernel
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WSL2-Linux-Kernel/sound/soc/soc-core.c

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/*
* soc-core.c -- ALSA SoC Audio Layer
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* with code, comments and ideas from :-
* Richard Purdie <richard@openedhand.com>
*
* 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.
*
* TODO:
* o Add hw rules to enforce rates, etc.
* o More testing with other codecs/machines.
* o Add more codecs and platforms to ensure good API coverage.
* o Support TDM on PCM and I2S
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
/* debug */
#define SOC_DEBUG 0
#if SOC_DEBUG
#define dbg(format, arg...) printk(format, ## arg)
#else
#define dbg(format, arg...)
#endif
static DEFINE_MUTEX(pcm_mutex);
static DEFINE_MUTEX(io_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
* It can be used to eliminate pops between different playback streams, e.g.
* between two audio tracks.
*/
static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
/*
* This function forces any delayed work to be queued and run.
*/
static int run_delayed_work(struct delayed_work *dwork)
{
int ret;
/* cancel any work waiting to be queued. */
ret = cancel_delayed_work(dwork);
/* if there was any work waiting then we run it now and
* wait for it's completion */
if (ret) {
schedule_delayed_work(dwork, 0);
flush_scheduled_work();
}
return ret;
}
#ifdef CONFIG_SND_SOC_AC97_BUS
/* unregister ac97 codec */
static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
{
if (codec->ac97->dev.bus)
device_unregister(&codec->ac97->dev);
return 0;
}
/* stop no dev release warning */
static void soc_ac97_device_release(struct device *dev){}
/* register ac97 codec to bus */
static int soc_ac97_dev_register(struct snd_soc_codec *codec)
{
int err;
codec->ac97->dev.bus = &ac97_bus_type;
codec->ac97->dev.parent = NULL;
codec->ac97->dev.release = soc_ac97_device_release;
snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
codec->card->number, 0, codec->name);
err = device_register(&codec->ac97->dev);
if (err < 0) {
snd_printk(KERN_ERR "Can't register ac97 bus\n");
codec->ac97->dev.bus = NULL;
return err;
}
return 0;
}
#endif
static inline const char *get_dai_name(int type)
{
switch (type) {
case SND_SOC_DAI_AC97_BUS:
case SND_SOC_DAI_AC97:
return "AC97";
case SND_SOC_DAI_I2S:
return "I2S";
case SND_SOC_DAI_PCM:
return "PCM";
}
return NULL;
}
/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
* startup for the cpu DAI, platform, machine and codec DAI.
*/
static int soc_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
mutex_lock(&pcm_mutex);
/* startup the audio subsystem */
if (cpu_dai->ops.startup) {
ret = cpu_dai->ops.startup(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open interface %s\n",
cpu_dai->name);
goto out;
}
}
if (platform->pcm_ops->open) {
ret = platform->pcm_ops->open(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
goto platform_err;
}
}
if (codec_dai->ops.startup) {
ret = codec_dai->ops.startup(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open codec %s\n",
codec_dai->name);
goto codec_dai_err;
}
}
if (machine->ops && machine->ops->startup) {
ret = machine->ops->startup(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
goto machine_err;
}
}
/* Check that the codec and cpu DAI's are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw.rate_min =
max(codec_dai->playback.rate_min,
cpu_dai->playback.rate_min);
runtime->hw.rate_max =
min(codec_dai->playback.rate_max,
cpu_dai->playback.rate_max);
runtime->hw.channels_min =
max(codec_dai->playback.channels_min,
cpu_dai->playback.channels_min);
runtime->hw.channels_max =
min(codec_dai->playback.channels_max,
cpu_dai->playback.channels_max);
runtime->hw.formats =
codec_dai->playback.formats & cpu_dai->playback.formats;
runtime->hw.rates =
codec_dai->playback.rates & cpu_dai->playback.rates;
} else {
runtime->hw.rate_min =
max(codec_dai->capture.rate_min,
cpu_dai->capture.rate_min);
runtime->hw.rate_max =
min(codec_dai->capture.rate_max,
cpu_dai->capture.rate_max);
runtime->hw.channels_min =
max(codec_dai->capture.channels_min,
cpu_dai->capture.channels_min);
runtime->hw.channels_max =
min(codec_dai->capture.channels_max,
cpu_dai->capture.channels_max);
runtime->hw.formats =
codec_dai->capture.formats & cpu_dai->capture.formats;
runtime->hw.rates =
codec_dai->capture.rates & cpu_dai->capture.rates;
}
snd_pcm_limit_hw_rates(runtime);
if (!runtime->hw.rates) {
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
codec_dai->name, cpu_dai->name);
goto machine_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
codec_dai->name, cpu_dai->name);
goto machine_err;
}
if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
codec_dai->name, cpu_dai->name);
goto machine_err;
}
dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
runtime->hw.channels_max);
dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
runtime->hw.rate_max);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->playback.active = codec_dai->playback.active = 1;
else
cpu_dai->capture.active = codec_dai->capture.active = 1;
cpu_dai->active = codec_dai->active = 1;
cpu_dai->runtime = runtime;
socdev->codec->active++;
mutex_unlock(&pcm_mutex);
return 0;
machine_err:
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
codec_dai_err:
if (platform->pcm_ops->close)
platform->pcm_ops->close(substream);
platform_err:
if (cpu_dai->ops.shutdown)
cpu_dai->ops.shutdown(substream);
out:
mutex_unlock(&pcm_mutex);
return ret;
}
/*
* Power down the audio subsystem pmdown_time msecs after close is called.
* This is to ensure there are no pops or clicks in between any music tracks
* due to DAPM power cycling.
*/
static void close_delayed_work(struct work_struct *work)
{
struct snd_soc_device *socdev =
container_of(work, struct snd_soc_device, delayed_work.work);
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_dai *codec_dai;
int i;
mutex_lock(&pcm_mutex);
for (i = 0; i < codec->num_dai; i++) {
codec_dai = &codec->dai[i];
dbg("pop wq checking: %s status: %s waiting: %s\n",
codec_dai->playback.stream_name,
codec_dai->playback.active ? "active" : "inactive",
codec_dai->pop_wait ? "yes" : "no");
/* are we waiting on this codec DAI stream */
if (codec_dai->pop_wait == 1) {
/* Reduce power if no longer active */
if (codec->active == 0) {
dbg("pop wq D1 %s %s\n", codec->name,
codec_dai->playback.stream_name);
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_PREPARE);
}
codec_dai->pop_wait = 0;
snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_STOP);
/* Fall into standby if no longer active */
if (codec->active == 0) {
dbg("pop wq D3 %s %s\n", codec->name,
codec_dai->playback.stream_name);
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_STANDBY);
}
}
}
mutex_unlock(&pcm_mutex);
}
/*
* Called by ALSA when a PCM substream is closed. Private data can be
* freed here. The cpu DAI, codec DAI, machine and platform are also
* shutdown.
*/
static int soc_codec_close(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
mutex_lock(&pcm_mutex);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->playback.active = codec_dai->playback.active = 0;
else
cpu_dai->capture.active = codec_dai->capture.active = 0;
if (codec_dai->playback.active == 0 &&
codec_dai->capture.active == 0) {
cpu_dai->active = codec_dai->active = 0;
}
codec->active--;
if (cpu_dai->ops.shutdown)
cpu_dai->ops.shutdown(substream);
if (codec_dai->ops.shutdown)
codec_dai->ops.shutdown(substream);
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
if (platform->pcm_ops->close)
platform->pcm_ops->close(substream);
cpu_dai->runtime = NULL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* start delayed pop wq here for playback streams */
codec_dai->pop_wait = 1;
schedule_delayed_work(&socdev->delayed_work,
msecs_to_jiffies(pmdown_time));
} else {
/* capture streams can be powered down now */
snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_STOP);
if (codec->active == 0 && codec_dai->pop_wait == 0)
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_STANDBY);
}
mutex_unlock(&pcm_mutex);
return 0;
}
/*
* Called by ALSA when the PCM substream is prepared, can set format, sample
* rate, etc. This function is non atomic and can be called multiple times,
* it can refer to the runtime info.
*/
static int soc_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
mutex_lock(&pcm_mutex);
if (machine->ops && machine->ops->prepare) {
ret = machine->ops->prepare(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: machine prepare error\n");
goto out;
}
}
if (platform->pcm_ops->prepare) {
ret = platform->pcm_ops->prepare(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: platform prepare error\n");
goto out;
}
}
if (codec_dai->ops.prepare) {
ret = codec_dai->ops.prepare(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: codec DAI prepare error\n");
goto out;
}
}
if (cpu_dai->ops.prepare) {
ret = cpu_dai->ops.prepare(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: cpu DAI prepare error\n");
goto out;
}
}
/* we only want to start a DAPM playback stream if we are not waiting
* on an existing one stopping */
if (codec_dai->pop_wait) {
/* we are waiting for the delayed work to start */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
snd_soc_dapm_stream_event(socdev->codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
else {
codec_dai->pop_wait = 0;
cancel_delayed_work(&socdev->delayed_work);
snd_soc_dai_digital_mute(codec_dai, 0);
}
} else {
/* no delayed work - do we need to power up codec */
if (codec->bias_level != SND_SOC_BIAS_ON) {
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_PREPARE);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_START);
else
snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
snd_soc_dai_digital_mute(codec_dai, 0);
} else {
/* codec already powered - power on widgets */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_START);
else
snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
snd_soc_dai_digital_mute(codec_dai, 0);
}
}
out:
mutex_unlock(&pcm_mutex);
return ret;
}
/*
* Called by ALSA when the hardware params are set by application. This
* function can also be called multiple times and can allocate buffers
* (using snd_pcm_lib_* ). It's non-atomic.
*/
static int soc_pcm_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_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
mutex_lock(&pcm_mutex);
if (machine->ops && machine->ops->hw_params) {
ret = machine->ops->hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: machine hw_params failed\n");
goto out;
}
}
if (codec_dai->ops.hw_params) {
ret = codec_dai->ops.hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: can't set codec %s hw params\n",
codec_dai->name);
goto codec_err;
}
}
if (cpu_dai->ops.hw_params) {
ret = cpu_dai->ops.hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: interface %s hw params failed\n",
cpu_dai->name);
goto interface_err;
}
}
if (platform->pcm_ops->hw_params) {
ret = platform->pcm_ops->hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: platform %s hw params failed\n",
platform->name);
goto platform_err;
}
}
out:
mutex_unlock(&pcm_mutex);
return ret;
platform_err:
if (cpu_dai->ops.hw_free)
cpu_dai->ops.hw_free(substream);
interface_err:
if (codec_dai->ops.hw_free)
codec_dai->ops.hw_free(substream);
codec_err:
if (machine->ops && machine->ops->hw_free)
machine->ops->hw_free(substream);
mutex_unlock(&pcm_mutex);
return ret;
}
/*
* Free's resources allocated by hw_params, can be called multiple times
*/
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
mutex_lock(&pcm_mutex);
/* apply codec digital mute */
if (!codec->active)
snd_soc_dai_digital_mute(codec_dai, 1);
/* free any machine hw params */
if (machine->ops && machine->ops->hw_free)
machine->ops->hw_free(substream);
/* free any DMA resources */
if (platform->pcm_ops->hw_free)
platform->pcm_ops->hw_free(substream);
/* now free hw params for the DAI's */
if (codec_dai->ops.hw_free)
codec_dai->ops.hw_free(substream);
if (cpu_dai->ops.hw_free)
cpu_dai->ops.hw_free(substream);
mutex_unlock(&pcm_mutex);
return 0;
}
static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret;
if (codec_dai->ops.trigger) {
ret = codec_dai->ops.trigger(substream, cmd);
if (ret < 0)
return ret;
}
if (platform->pcm_ops->trigger) {
ret = platform->pcm_ops->trigger(substream, cmd);
if (ret < 0)
return ret;
}
if (cpu_dai->ops.trigger) {
ret = cpu_dai->ops.trigger(substream, cmd);
if (ret < 0)
return ret;
}
return 0;
}
/* ASoC PCM operations */
static struct snd_pcm_ops soc_pcm_ops = {
.open = soc_pcm_open,
.close = soc_codec_close,
.hw_params = soc_pcm_hw_params,
.hw_free = soc_pcm_hw_free,
.prepare = soc_pcm_prepare,
.trigger = soc_pcm_trigger,
};
#ifdef CONFIG_PM
/* powers down audio subsystem for suspend */
static int soc_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = socdev->codec;
int i;
/* Due to the resume being scheduled into a workqueue we could
* suspend before that's finished - wait for it to complete.
*/
snd_power_lock(codec->card);
snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
snd_power_unlock(codec->card);
/* we're going to block userspace touching us until resume completes */
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
/* mute any active DAC's */
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
if (dai->dai_ops.digital_mute && dai->playback.active)
dai->dai_ops.digital_mute(dai, 1);
}
/* suspend all pcms */
for (i = 0; i < machine->num_links; i++)
snd_pcm_suspend_all(machine->dai_link[i].pcm);
if (machine->suspend_pre)
machine->suspend_pre(pdev, state);
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
cpu_dai->suspend(pdev, cpu_dai);
if (platform->suspend)
platform->suspend(pdev, cpu_dai);
}
/* close any waiting streams and save state */
run_delayed_work(&socdev->delayed_work);
codec->suspend_bias_level = codec->bias_level;
for (i = 0; i < codec->num_dai; i++) {
char *stream = codec->dai[i].playback.stream_name;
if (stream != NULL)
snd_soc_dapm_stream_event(codec, stream,
SND_SOC_DAPM_STREAM_SUSPEND);
stream = codec->dai[i].capture.stream_name;
if (stream != NULL)
snd_soc_dapm_stream_event(codec, stream,
SND_SOC_DAPM_STREAM_SUSPEND);
}
if (codec_dev->suspend)
codec_dev->suspend(pdev, state);
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
cpu_dai->suspend(pdev, cpu_dai);
}
if (machine->suspend_post)
machine->suspend_post(pdev, state);
return 0;
}
/* deferred resume work, so resume can complete before we finished
* setting our codec back up, which can be very slow on I2C
*/
static void soc_resume_deferred(struct work_struct *work)
{
struct snd_soc_device *socdev = container_of(work,
struct snd_soc_device,
deferred_resume_work);
struct snd_soc_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = socdev->codec;
struct platform_device *pdev = to_platform_device(socdev->dev);
int i;
/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
* so userspace apps are blocked from touching us
*/
dev_info(socdev->dev, "starting resume work\n");
if (machine->resume_pre)
machine->resume_pre(pdev);
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
cpu_dai->resume(pdev, cpu_dai);
}
if (codec_dev->resume)
codec_dev->resume(pdev);
for (i = 0; i < codec->num_dai; i++) {
char *stream = codec->dai[i].playback.stream_name;
if (stream != NULL)
snd_soc_dapm_stream_event(codec, stream,
SND_SOC_DAPM_STREAM_RESUME);
stream = codec->dai[i].capture.stream_name;
if (stream != NULL)
snd_soc_dapm_stream_event(codec, stream,
SND_SOC_DAPM_STREAM_RESUME);
}
/* unmute any active DACs */
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
if (dai->dai_ops.digital_mute && dai->playback.active)
dai->dai_ops.digital_mute(dai, 0);
}
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
cpu_dai->resume(pdev, cpu_dai);
if (platform->resume)
platform->resume(pdev, cpu_dai);
}
if (machine->resume_post)
machine->resume_post(pdev);
dev_info(socdev->dev, "resume work completed\n");
/* userspace can access us now we are back as we were before */
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
}
/* powers up audio subsystem after a suspend */
static int soc_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
dev_info(socdev->dev, "scheduling resume work\n");
if (!schedule_work(&socdev->deferred_resume_work))
dev_err(socdev->dev, "work item may be lost\n");
return 0;
}
#else
#define soc_suspend NULL
#define soc_resume NULL
#endif
/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
int ret = 0, i;
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
if (machine->probe) {
ret = machine->probe(pdev);
if (ret < 0)
return ret;
}
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->probe) {
ret = cpu_dai->probe(pdev, cpu_dai);
if (ret < 0)
goto cpu_dai_err;
}
}
if (codec_dev->probe) {
ret = codec_dev->probe(pdev);
if (ret < 0)
goto cpu_dai_err;
}
if (platform->probe) {
ret = platform->probe(pdev);
if (ret < 0)
goto platform_err;
}
/* DAPM stream work */
INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
#ifdef CONFIG_PM
/* deferred resume work */
INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
#endif
return 0;
platform_err:
if (codec_dev->remove)
codec_dev->remove(pdev);
cpu_dai_err:
for (i--; i >= 0; i--) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
if (machine->remove)
machine->remove(pdev);
return ret;
}
/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
int i;
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
run_delayed_work(&socdev->delayed_work);
if (platform->remove)
platform->remove(pdev);
if (codec_dev->remove)
codec_dev->remove(pdev);
for (i = 0; i < machine->num_links; i++) {
struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
if (machine->remove)
machine->remove(pdev);
return 0;
}
/* ASoC platform driver */
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
.owner = THIS_MODULE,
},
.probe = soc_probe,
.remove = soc_remove,
.suspend = soc_suspend,
.resume = soc_resume,
};
/* create a new pcm */
static int soc_new_pcm(struct snd_soc_device *socdev,
struct snd_soc_dai_link *dai_link, int num)
{
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_dai *codec_dai = dai_link->codec_dai;
struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
struct snd_soc_pcm_runtime *rtd;
struct snd_pcm *pcm;
char new_name[64];
int ret = 0, playback = 0, capture = 0;
rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
if (rtd == NULL)
return -ENOMEM;
rtd->dai = dai_link;
rtd->socdev = socdev;
codec_dai->codec = socdev->codec;
/* check client and interface hw capabilities */
sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
get_dai_name(cpu_dai->type), num);
if (codec_dai->playback.channels_min)
playback = 1;
if (codec_dai->capture.channels_min)
capture = 1;
ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
capture, &pcm);
if (ret < 0) {
printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
codec->name);
kfree(rtd);
return ret;
}
dai_link->pcm = pcm;
pcm->private_data = rtd;
soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
soc_pcm_ops.page = socdev->platform->pcm_ops->page;
if (playback)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
if (capture)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
if (ret < 0) {
printk(KERN_ERR "asoc: platform pcm constructor failed\n");
kfree(rtd);
return ret;
}
pcm->private_free = socdev->platform->pcm_free;
printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
cpu_dai->name);
return ret;
}
/* codec register dump */
static ssize_t codec_reg_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_device *devdata = dev_get_drvdata(dev);
struct snd_soc_codec *codec = devdata->codec;
int i, step = 1, count = 0;
if (!codec->reg_cache_size)
return 0;
if (codec->reg_cache_step)
step = codec->reg_cache_step;
count += sprintf(buf, "%s registers\n", codec->name);
for (i = 0; i < codec->reg_cache_size; i += step)
count += sprintf(buf + count, "%2x: %4x\n", i,
codec->read(codec, i));
return count;
}
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
/**
* snd_soc_new_ac97_codec - initailise AC97 device
* @codec: audio codec
* @ops: AC97 bus operations
* @num: AC97 codec number
*
* Initialises AC97 codec resources for use by ad-hoc devices only.
*/
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num)
{
mutex_lock(&codec->mutex);
codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
if (codec->ac97 == NULL) {
mutex_unlock(&codec->mutex);
return -ENOMEM;
}
codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
if (codec->ac97->bus == NULL) {
kfree(codec->ac97);
codec->ac97 = NULL;
mutex_unlock(&codec->mutex);
return -ENOMEM;
}
codec->ac97->bus->ops = ops;
codec->ac97->num = num;
mutex_unlock(&codec->mutex);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
/**
* snd_soc_free_ac97_codec - free AC97 codec device
* @codec: audio codec
*
* Frees AC97 codec device resources.
*/
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
{
mutex_lock(&codec->mutex);
kfree(codec->ac97->bus);
kfree(codec->ac97);
codec->ac97 = NULL;
mutex_unlock(&codec->mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
/**
* snd_soc_update_bits - update codec register bits
* @codec: audio codec
* @reg: codec register
* @mask: register mask
* @value: new value
*
* Writes new register value.
*
* Returns 1 for change else 0.
*/
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value)
{
int change;
unsigned short old, new;
mutex_lock(&io_mutex);
old = snd_soc_read(codec, reg);
new = (old & ~mask) | value;
change = old != new;
if (change)
snd_soc_write(codec, reg, new);
mutex_unlock(&io_mutex);
return change;
}
EXPORT_SYMBOL_GPL(snd_soc_update_bits);
/**
* snd_soc_test_bits - test register for change
* @codec: audio codec
* @reg: codec register
* @mask: register mask
* @value: new value
*
* Tests a register with a new value and checks if the new value is
* different from the old value.
*
* Returns 1 for change else 0.
*/
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value)
{
int change;
unsigned short old, new;
mutex_lock(&io_mutex);
old = snd_soc_read(codec, reg);
new = (old & ~mask) | value;
change = old != new;
mutex_unlock(&io_mutex);
return change;
}
EXPORT_SYMBOL_GPL(snd_soc_test_bits);
/**
* snd_soc_new_pcms - create new sound card and pcms
* @socdev: the SoC audio device
*
* Create a new sound card based upon the codec and interface pcms.
*
* Returns 0 for success, else error.
*/
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
{
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_machine *machine = socdev->machine;
int ret = 0, i;
mutex_lock(&codec->mutex);
/* register a sound card */
codec->card = snd_card_new(idx, xid, codec->owner, 0);
if (!codec->card) {
printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
codec->name);
mutex_unlock(&codec->mutex);
return -ENODEV;
}
codec->card->dev = socdev->dev;
codec->card->private_data = codec;
strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
/* create the pcms */
for (i = 0; i < machine->num_links; i++) {
ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
if (ret < 0) {
printk(KERN_ERR "asoc: can't create pcm %s\n",
machine->dai_link[i].stream_name);
mutex_unlock(&codec->mutex);
return ret;
}
}
mutex_unlock(&codec->mutex);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
/**
* snd_soc_register_card - register sound card
* @socdev: the SoC audio device
*
* Register a SoC sound card. Also registers an AC97 device if the
* codec is AC97 for ad hoc devices.
*
* Returns 0 for success, else error.
*/
int snd_soc_register_card(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_machine *machine = socdev->machine;
int ret = 0, i, ac97 = 0, err = 0;
for (i = 0; i < machine->num_links; i++) {
if (socdev->machine->dai_link[i].init) {
err = socdev->machine->dai_link[i].init(codec);
if (err < 0) {
printk(KERN_ERR "asoc: failed to init %s\n",
socdev->machine->dai_link[i].stream_name);
continue;
}
}
if (socdev->machine->dai_link[i].codec_dai->type ==
SND_SOC_DAI_AC97_BUS)
ac97 = 1;
}
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
"%s", machine->name);
snprintf(codec->card->longname, sizeof(codec->card->longname),
"%s (%s)", machine->name, codec->name);
ret = snd_card_register(codec->card);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
codec->name);
goto out;
}
mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
if (ac97) {
ret = soc_ac97_dev_register(codec);
if (ret < 0) {
printk(KERN_ERR "asoc: AC97 device register failed\n");
snd_card_free(codec->card);
mutex_unlock(&codec->mutex);
goto out;
}
}
#endif
err = snd_soc_dapm_sys_add(socdev->dev);
if (err < 0)
printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
err = device_create_file(socdev->dev, &dev_attr_codec_reg);
if (err < 0)
printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
mutex_unlock(&codec->mutex);
out:
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_card);
/**
* snd_soc_free_pcms - free sound card and pcms
* @socdev: the SoC audio device
*
* Frees sound card and pcms associated with the socdev.
* Also unregister the codec if it is an AC97 device.
*/
void snd_soc_free_pcms(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
#ifdef CONFIG_SND_SOC_AC97_BUS
struct snd_soc_dai *codec_dai;
int i;
#endif
mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
for (i = 0; i < codec->num_dai; i++) {
codec_dai = &codec->dai[i];
if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
soc_ac97_dev_unregister(codec);
goto free_card;
}
}
free_card:
#endif
if (codec->card)
snd_card_free(codec->card);
device_remove_file(socdev->dev, &dev_attr_codec_reg);
mutex_unlock(&codec->mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
/**
* snd_soc_set_runtime_hwparams - set the runtime hardware parameters
* @substream: the pcm substream
* @hw: the hardware parameters
*
* Sets the substream runtime hardware parameters.
*/
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
const struct snd_pcm_hardware *hw)
{
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw.info = hw->info;
runtime->hw.formats = hw->formats;
runtime->hw.period_bytes_min = hw->period_bytes_min;
runtime->hw.period_bytes_max = hw->period_bytes_max;
runtime->hw.periods_min = hw->periods_min;
runtime->hw.periods_max = hw->periods_max;
runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
runtime->hw.fifo_size = hw->fifo_size;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
/**
* snd_soc_cnew - create new control
* @_template: control template
* @data: control private data
* @lnng_name: control long name
*
* Create a new mixer control from a template control.
*
* Returns 0 for success, else error.
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, char *long_name)
{
struct snd_kcontrol_new template;
memcpy(&template, _template, sizeof(template));
if (long_name)
template.name = long_name;
template.index = 0;
return snd_ctl_new1(&template, data);
}
EXPORT_SYMBOL_GPL(snd_soc_cnew);
/**
* snd_soc_info_enum_double - enumerated double mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a double enumerated
* mixer control.
*
* Returns 0 for success.
*/
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
uinfo->value.enumerated.items = e->mask;
if (uinfo->value.enumerated.item > e->mask - 1)
uinfo->value.enumerated.item = e->mask - 1;
strcpy(uinfo->value.enumerated.name,
e->texts[uinfo->value.enumerated.item]);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
/**
* snd_soc_get_enum_double - enumerated double mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a double enumerated mixer.
*
* Returns 0 for success.
*/
int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned short val, bitmask;
for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
;
val = snd_soc_read(codec, e->reg);
ucontrol->value.enumerated.item[0]
= (val >> e->shift_l) & (bitmask - 1);
if (e->shift_l != e->shift_r)
ucontrol->value.enumerated.item[1] =
(val >> e->shift_r) & (bitmask - 1);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
/**
* snd_soc_put_enum_double - enumerated double mixer put callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a double enumerated mixer.
*
* Returns 0 for success.
*/
int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned short val;
unsigned short mask, bitmask;
for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
;
if (ucontrol->value.enumerated.item[0] > e->mask - 1)
return -EINVAL;
val = ucontrol->value.enumerated.item[0] << e->shift_l;
mask = (bitmask - 1) << e->shift_l;
if (e->shift_l != e->shift_r) {
if (ucontrol->value.enumerated.item[1] > e->mask - 1)
return -EINVAL;
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
mask |= (bitmask - 1) << e->shift_r;
}
return snd_soc_update_bits(codec, e->reg, mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
/**
* snd_soc_info_enum_ext - external enumerated single mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about an external enumerated
* single mixer.
*
* Returns 0 for success.
*/
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = e->mask;
if (uinfo->value.enumerated.item > e->mask - 1)
uinfo->value.enumerated.item = e->mask - 1;
strcpy(uinfo->value.enumerated.name,
e->texts[uinfo->value.enumerated.item]);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
/**
* snd_soc_info_volsw_ext - external single mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a single external mixer control.
*
* Returns 0 for success.
*/
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int max = kcontrol->private_value;
if (max == 1)
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = max;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
/**
* snd_soc_info_volsw - single mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a single mixer control.
*
* Returns 0 for success.
*/
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int max = (kcontrol->private_value >> 16) & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
if (max == 1)
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = shift == rshift ? 1 : 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = max;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
/**
* snd_soc_get_volsw - single mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a single mixer control.
*
* Returns 0 for success.
*/
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
int max = (kcontrol->private_value >> 16) & 0xff;
int mask = (1 << fls(max)) - 1;
int invert = (kcontrol->private_value >> 24) & 0x01;
ucontrol->value.integer.value[0] =
(snd_soc_read(codec, reg) >> shift) & mask;
if (shift != rshift)
ucontrol->value.integer.value[1] =
(snd_soc_read(codec, reg) >> rshift) & mask;
if (invert) {
ucontrol->value.integer.value[0] =
max - ucontrol->value.integer.value[0];
if (shift != rshift)
ucontrol->value.integer.value[1] =
max - ucontrol->value.integer.value[1];
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
/**
* snd_soc_put_volsw - single mixer put callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a single mixer control.
*
* Returns 0 for success.
*/
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
int max = (kcontrol->private_value >> 16) & 0xff;
int mask = (1 << fls(max)) - 1;
int invert = (kcontrol->private_value >> 24) & 0x01;
unsigned short val, val2, val_mask;
val = (ucontrol->value.integer.value[0] & mask);
if (invert)
val = max - val;
val_mask = mask << shift;
val = val << shift;
if (shift != rshift) {
val2 = (ucontrol->value.integer.value[1] & mask);
if (invert)
val2 = max - val2;
val_mask |= mask << rshift;
val |= val2 << rshift;
}
return snd_soc_update_bits(codec, reg, val_mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
/**
* snd_soc_info_volsw_2r - double mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a double mixer control that
* spans 2 codec registers.
*
* Returns 0 for success.
*/
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int max = (kcontrol->private_value >> 12) & 0xff;
if (max == 1)
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = max;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
/**
* snd_soc_get_volsw_2r - double mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a double mixer control that spans 2 registers.
*
* Returns 0 for success.
*/
int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int reg2 = (kcontrol->private_value >> 24) & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int max = (kcontrol->private_value >> 12) & 0xff;
int mask = (1<<fls(max))-1;
int invert = (kcontrol->private_value >> 20) & 0x01;
ucontrol->value.integer.value[0] =
(snd_soc_read(codec, reg) >> shift) & mask;
ucontrol->value.integer.value[1] =
(snd_soc_read(codec, reg2) >> shift) & mask;
if (invert) {
ucontrol->value.integer.value[0] =
max - ucontrol->value.integer.value[0];
ucontrol->value.integer.value[1] =
max - ucontrol->value.integer.value[1];
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
/**
* snd_soc_put_volsw_2r - double mixer set callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a double mixer control that spans 2 registers.
*
* Returns 0 for success.
*/
int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int reg2 = (kcontrol->private_value >> 24) & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int max = (kcontrol->private_value >> 12) & 0xff;
int mask = (1 << fls(max)) - 1;
int invert = (kcontrol->private_value >> 20) & 0x01;
int err;
unsigned short val, val2, val_mask;
val_mask = mask << shift;
val = (ucontrol->value.integer.value[0] & mask);
val2 = (ucontrol->value.integer.value[1] & mask);
if (invert) {
val = max - val;
val2 = max - val2;
}
val = val << shift;
val2 = val2 << shift;
err = snd_soc_update_bits(codec, reg, val_mask, val);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
return err;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
/**
* snd_soc_info_volsw_s8 - signed mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a signed mixer control.
*
* Returns 0 for success.
*/
int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int max = (signed char)((kcontrol->private_value >> 16) & 0xff);
int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = max-min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
/**
* snd_soc_get_volsw_s8 - signed mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a signed mixer control.
*
* Returns 0 for success.
*/
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
int val = snd_soc_read(codec, reg);
ucontrol->value.integer.value[0] =
((signed char)(val & 0xff))-min;
ucontrol->value.integer.value[1] =
((signed char)((val >> 8) & 0xff))-min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
/**
* snd_soc_put_volsw_sgn - signed mixer put callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a signed mixer control.
*
* Returns 0 for success.
*/
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
unsigned short val;
val = (ucontrol->value.integer.value[0]+min) & 0xff;
val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
return snd_soc_update_bits(codec, reg, 0xffff, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
/**
* snd_soc_dai_set_sysclk - configure DAI system or master clock.
* @dai: DAI
* @clk_id: DAI specific clock ID
* @freq: new clock frequency in Hz
* @dir: new clock direction - input/output.
*
* Configures the DAI master (MCLK) or system (SYSCLK) clocking.
*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
if (dai->dai_ops.set_sysclk)
return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
/**
* snd_soc_dai_set_clkdiv - configure DAI clock dividers.
* @dai: DAI
* @clk_id: DAI specific clock divider ID
* @div: new clock divisor.
*
* Configures the clock dividers. This is used to derive the best DAI bit and
* frame clocks from the system or master clock. It's best to set the DAI bit
* and frame clocks as low as possible to save system power.
*/
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div)
{
if (dai->dai_ops.set_clkdiv)
return dai->dai_ops.set_clkdiv(dai, div_id, div);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
/**
* snd_soc_dai_set_pll - configure DAI PLL.
* @dai: DAI
* @pll_id: DAI specific PLL ID
* @freq_in: PLL input clock frequency in Hz
* @freq_out: requested PLL output clock frequency in Hz
*
* Configures and enables PLL to generate output clock based on input clock.
*/
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
if (dai->dai_ops.set_pll)
return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
/**
* snd_soc_dai_set_fmt - configure DAI hardware audio format.
* @dai: DAI
* @clk_id: DAI specific clock ID
* @fmt: SND_SOC_DAIFMT_ format value.
*
* Configures the DAI hardware format and clocking.
*/
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
if (dai->dai_ops.set_fmt)
return dai->dai_ops.set_fmt(dai, fmt);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
/**
* snd_soc_dai_set_tdm_slot - configure DAI TDM.
* @dai: DAI
* @mask: DAI specific mask representing used slots.
* @slots: Number of slots in use.
*
* Configures a DAI for TDM operation. Both mask and slots are codec and DAI
* specific.
*/
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int mask, int slots)
{
if (dai->dai_ops.set_sysclk)
return dai->dai_ops.set_tdm_slot(dai, mask, slots);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
/**
* snd_soc_dai_set_tristate - configure DAI system or master clock.
* @dai: DAI
* @tristate: tristate enable
*
* Tristates the DAI so that others can use it.
*/
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
{
if (dai->dai_ops.set_sysclk)
return dai->dai_ops.set_tristate(dai, tristate);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
/**
* snd_soc_dai_digital_mute - configure DAI system or master clock.
* @dai: DAI
* @mute: mute enable
*
* Mutes the DAI DAC.
*/
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
{
if (dai->dai_ops.digital_mute)
return dai->dai_ops.digital_mute(dai, mute);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
static int __devinit snd_soc_init(void)
{
printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
return platform_driver_register(&soc_driver);
}
static void snd_soc_exit(void)
{
platform_driver_unregister(&soc_driver);
}
module_init(snd_soc_init);
module_exit(snd_soc_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:soc-audio");
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