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linux/sound/soc/soc-core.c
Peter Ujfalusi d11bb4a925 ASoC: core: Fix for the volume limiting when invert is in use 
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-11 09:34:11 +01:00

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/*
* soc-core.c -- ALSA SoC Audio Layer
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
* 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/debugfs.h>
#include <linux/platform_device.h>
#include <sound/ac97_codec.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>
static DEFINE_MUTEX(pcm_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
#ifdef CONFIG_DEBUG_FS
static struct dentry *debugfs_root;
#endif
static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(card_list);
static LIST_HEAD(dai_list);
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
static int snd_soc_register_card(struct snd_soc_card *card);
static int snd_soc_unregister_card(struct snd_soc_card *card);
/*
* 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;
}
/* codec register dump */
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
{
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) {
if (codec->readable_register && !codec->readable_register(i))
continue;
count += sprintf(buf + count, "%2x: ", i);
if (count >= PAGE_SIZE - 1)
break;
if (codec->display_register)
count += codec->display_register(codec, buf + count,
PAGE_SIZE - count, i);
else
count += snprintf(buf + count, PAGE_SIZE - count,
"%4x", codec->read(codec, i));
if (count >= PAGE_SIZE - 1)
break;
count += snprintf(buf + count, PAGE_SIZE - count, "\n");
if (count >= PAGE_SIZE - 1)
break;
}
/* Truncate count; min() would cause a warning */
if (count >= PAGE_SIZE)
count = PAGE_SIZE - 1;
return count;
}
static ssize_t codec_reg_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_device *devdata = dev_get_drvdata(dev);
return soc_codec_reg_show(devdata->card->codec, buf);
}
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
static ssize_t pmdown_time_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_device *socdev = dev_get_drvdata(dev);
struct snd_soc_card *card = socdev->card;
return sprintf(buf, "%ld\n", card->pmdown_time);
}
static ssize_t pmdown_time_set(struct device *dev,
struct device_attribute *attr,
const char *buf, size_t count)
{
struct snd_soc_device *socdev = dev_get_drvdata(dev);
struct snd_soc_card *card = socdev->card;
strict_strtol(buf, 10, &card->pmdown_time);
return count;
}
static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set);
#ifdef CONFIG_DEBUG_FS
static int codec_reg_open_file(struct inode *inode, struct file *file)
{
file->private_data = inode->i_private;
return 0;
}
static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
size_t count, loff_t *ppos)
{
ssize_t ret;
struct snd_soc_codec *codec = file->private_data;
char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
if (!buf)
return -ENOMEM;
ret = soc_codec_reg_show(codec, buf);
if (ret >= 0)
ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
kfree(buf);
return ret;
}
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;
char *start = buf;
unsigned long reg, value;
int step = 1;
struct snd_soc_codec *codec = file->private_data;
buf_size = min(count, (sizeof(buf)-1));
if (copy_from_user(buf, user_buf, buf_size))
return -EFAULT;
buf[buf_size] = 0;
if (codec->reg_cache_step)
step = codec->reg_cache_step;
while (*start == ' ')
start++;
reg = simple_strtoul(start, &start, 16);
if ((reg >= codec->reg_cache_size) || (reg % step))
return -EINVAL;
while (*start == ' ')
start++;
if (strict_strtoul(start, 16, &value))
return -EINVAL;
codec->write(codec, reg, value);
return buf_size;
}
static const struct file_operations codec_reg_fops = {
.open = codec_reg_open_file,
.read = codec_reg_read_file,
.write = codec_reg_write_file,
};
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
char codec_root[128];
if (codec->dev)
snprintf(codec_root, sizeof(codec_root),
"%s.%s", codec->name, dev_name(codec->dev));
else
snprintf(codec_root, sizeof(codec_root),
"%s", codec->name);
codec->debugfs_codec_root = debugfs_create_dir(codec_root,
debugfs_root);
if (!codec->debugfs_codec_root) {
printk(KERN_WARNING
"ASoC: Failed to create codec debugfs directory\n");
return;
}
codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
codec->debugfs_codec_root,
codec, &codec_reg_fops);
if (!codec->debugfs_reg)
printk(KERN_WARNING
"ASoC: Failed to create codec register debugfs file\n");
codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
codec->debugfs_codec_root,
&codec->pop_time);
if (!codec->debugfs_pop_time)
printk(KERN_WARNING
"Failed to create pop time debugfs file\n");
codec->debugfs_dapm = debugfs_create_dir("dapm",
codec->debugfs_codec_root);
if (!codec->debugfs_dapm)
printk(KERN_WARNING
"Failed to create DAPM debugfs directory\n");
snd_soc_dapm_debugfs_init(codec);
}
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
{
debugfs_remove_recursive(codec->debugfs_codec_root);
}
#else
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
}
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
{
}
#endif
#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 = codec->card->dev;
codec->ac97->dev.release = soc_ac97_device_release;
dev_set_name(&codec->ac97->dev, "%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 int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_card *card = socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret;
if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
machine->symmetric_rates) {
dev_dbg(card->dev, "Symmetry forces %dHz rate\n",
machine->rate);
ret = snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
machine->rate,
machine->rate);
if (ret < 0) {
dev_err(card->dev,
"Unable to apply rate symmetry constraint: %d\n", ret);
return ret;
}
}
return 0;
}
/*
* 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_soc_card *card = socdev->card;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = card->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, cpu_dai);
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, codec_dai);
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;
if (codec_dai->playback.rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
runtime->hw.rates |= cpu_dai->playback.rates;
if (cpu_dai->playback.rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
runtime->hw.rates |= codec_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;
if (codec_dai->capture.rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
runtime->hw.rates |= cpu_dai->capture.rates;
if (cpu_dai->capture.rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
runtime->hw.rates |= codec_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 config_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
codec_dai->name, cpu_dai->name);
goto config_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 config_err;
}
/* Symmetry only applies if we've already got an active stream. */
if (cpu_dai->active || codec_dai->active) {
ret = soc_pcm_apply_symmetry(substream);
if (ret != 0)
goto config_err;
}
pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
runtime->hw.channels_max);
pr_debug("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++;
} else {
cpu_dai->capture.active++;
codec_dai->capture.active++;
}
cpu_dai->active++;
codec_dai->active++;
card->codec->active++;
mutex_unlock(&pcm_mutex);
return 0;
config_err:
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
machine_err:
if (codec_dai->ops->shutdown)
codec_dai->ops->shutdown(substream, codec_dai);
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, cpu_dai);
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_card *card = container_of(work, struct snd_soc_card,
delayed_work.work);
struct snd_soc_codec *codec = card->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];
pr_debug("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) {
codec_dai->pop_wait = 0;
snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_STOP);
}
}
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_card *card = socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = card->codec;
mutex_lock(&pcm_mutex);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback.active--;
codec_dai->playback.active--;
} else {
cpu_dai->capture.active--;
codec_dai->capture.active--;
}
cpu_dai->active--;
codec_dai->active--;
codec->active--;
/* Muting the DAC suppresses artifacts caused during digital
* shutdown, for example from stopping clocks.
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_soc_dai_digital_mute(codec_dai, 1);
if (cpu_dai->ops->shutdown)
cpu_dai->ops->shutdown(substream, cpu_dai);
if (codec_dai->ops->shutdown)
codec_dai->ops->shutdown(substream, codec_dai);
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
if (platform->pcm_ops->close)
platform->pcm_ops->close(substream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* start delayed pop wq here for playback streams */
codec_dai->pop_wait = 1;
schedule_delayed_work(&card->delayed_work,
msecs_to_jiffies(card->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);
}
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_card *card = socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = card->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, codec_dai);
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, cpu_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: cpu DAI prepare error\n");
goto out;
}
}
/* cancel any delayed stream shutdown that is pending */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
codec_dai->pop_wait) {
codec_dai->pop_wait = 0;
cancel_delayed_work(&card->delayed_work);
}
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_card *card = socdev->card;
struct snd_soc_platform *platform = card->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, codec_dai);
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, cpu_dai);
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;
}
}
machine->rate = params_rate(params);
out:
mutex_unlock(&pcm_mutex);
return ret;
platform_err:
if (cpu_dai->ops->hw_free)
cpu_dai->ops->hw_free(substream, cpu_dai);
interface_err:
if (codec_dai->ops->hw_free)
codec_dai->ops->hw_free(substream, codec_dai);
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_card *card = socdev->card;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = card->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, codec_dai);
if (cpu_dai->ops->hw_free)
cpu_dai->ops->hw_free(substream, cpu_dai);
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_card *card= socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = card->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, codec_dai);
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, cpu_dai);
if (ret < 0)
return ret;
}
return 0;
}
/*
* soc level wrapper for pointer callback
* If cpu_dai, codec_dai, platform driver has the delay callback, than
* the runtime->delay will be updated accordingly.
*/
static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_card *card = socdev->card;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t offset = 0;
snd_pcm_sframes_t delay = 0;
if (platform->pcm_ops->pointer)
offset = platform->pcm_ops->pointer(substream);
if (cpu_dai->ops->delay)
delay += cpu_dai->ops->delay(substream, cpu_dai);
if (codec_dai->ops->delay)
delay += codec_dai->ops->delay(substream, codec_dai);
if (platform->delay)
delay += platform->delay(substream, codec_dai);
runtime->delay = delay;
return offset;
}
/* 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,
.pointer = soc_pcm_pointer,
};
#ifdef CONFIG_PM
/* powers down audio subsystem for suspend */
static int soc_suspend(struct device *dev)
{
struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = card->codec;
int i;
/* If the initialization of this soc device failed, there is no codec
* associated with it. Just bail out in this case.
*/
if (!codec)
return 0;
/* 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 < card->num_links; i++) {
struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (dai->ops->digital_mute && dai->playback.active)
dai->ops->digital_mute(dai, 1);
}
/* suspend all pcms */
for (i = 0; i < card->num_links; i++) {
if (card->dai_link[i].ignore_suspend)
continue;
snd_pcm_suspend_all(card->dai_link[i].pcm);
}
if (card->suspend_pre)
card->suspend_pre(pdev, PMSG_SUSPEND);
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (cpu_dai->suspend && !cpu_dai->ac97_control)
cpu_dai->suspend(cpu_dai);
if (platform->suspend)
platform->suspend(&card->dai_link[i]);
}
/* close any waiting streams and save state */
run_delayed_work(&card->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 (card->dai_link[i].ignore_suspend)
continue;
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 there are paths active then the CODEC will be held with
* bias _ON and should not be suspended. */
if (codec_dev->suspend) {
switch (codec->bias_level) {
case SND_SOC_BIAS_STANDBY:
case SND_SOC_BIAS_OFF:
codec_dev->suspend(pdev, PMSG_SUSPEND);
break;
default:
dev_dbg(socdev->dev, "CODEC is on over suspend\n");
break;
}
}
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (cpu_dai->suspend && cpu_dai->ac97_control)
cpu_dai->suspend(cpu_dai);
}
if (card->suspend_post)
card->suspend_post(pdev, PMSG_SUSPEND);
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_card *card = container_of(work,
struct snd_soc_card,
deferred_resume_work);
struct snd_soc_device *socdev = card->socdev;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = card->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_dbg(socdev->dev, "starting resume work\n");
/* Bring us up into D2 so that DAPM starts enabling things */
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D2);
if (card->resume_pre)
card->resume_pre(pdev);
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (cpu_dai->resume && cpu_dai->ac97_control)
cpu_dai->resume(cpu_dai);
}
/* If the CODEC was idle over suspend then it will have been
* left with bias OFF or STANDBY and suspended so we must now
* resume. Otherwise the suspend was suppressed.
*/
if (codec_dev->resume) {
switch (codec->bias_level) {
case SND_SOC_BIAS_STANDBY:
case SND_SOC_BIAS_OFF:
codec_dev->resume(pdev);
break;
default:
dev_dbg(socdev->dev, "CODEC was on over suspend\n");
break;
}
}
for (i = 0; i < codec->num_dai; i++) {
char *stream = codec->dai[i].playback.stream_name;
if (card->dai_link[i].ignore_suspend)
continue;
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 < card->num_links; i++) {
struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (dai->ops->digital_mute && dai->playback.active)
dai->ops->digital_mute(dai, 0);
}
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (card->dai_link[i].ignore_suspend)
continue;
if (cpu_dai->resume && !cpu_dai->ac97_control)
cpu_dai->resume(cpu_dai);
if (platform->resume)
platform->resume(&card->dai_link[i]);
}
if (card->resume_post)
card->resume_post(pdev);
dev_dbg(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 device *dev)
{
struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
/* If the initialization of this soc device failed, there is no codec
* associated with it. Just bail out in this case.
*/
if (!card->codec)
return 0;
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
* problem and may take a substantial amount of time to resume
* due to I/O costs and anti-pop so handle them out of line.
*/
if (cpu_dai->ac97_control) {
dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
soc_resume_deferred(&card->deferred_resume_work);
} else {
dev_dbg(socdev->dev, "Scheduling resume work\n");
if (!schedule_work(&card->deferred_resume_work))
dev_err(socdev->dev, "resume work item may be lost\n");
}
return 0;
}
#else
#define soc_suspend NULL
#define soc_resume NULL
#endif
static struct snd_soc_dai_ops null_dai_ops = {
};
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct platform_device *pdev = container_of(card->dev,
struct platform_device,
dev);
struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct snd_soc_dai *dai;
int i, found, ret, ac97;
if (card->instantiated)
return;
found = 0;
list_for_each_entry(platform, &platform_list, list)
if (card->platform == platform) {
found = 1;
break;
}
if (!found) {
dev_dbg(card->dev, "Platform %s not registered\n",
card->platform->name);
return;
}
ac97 = 0;
for (i = 0; i < card->num_links; i++) {
found = 0;
list_for_each_entry(dai, &dai_list, list)
if (card->dai_link[i].cpu_dai == dai) {
found = 1;
break;
}
if (!found) {
dev_dbg(card->dev, "DAI %s not registered\n",
card->dai_link[i].cpu_dai->name);
return;
}
if (card->dai_link[i].cpu_dai->ac97_control)
ac97 = 1;
}
for (i = 0; i < card->num_links; i++) {
if (!card->dai_link[i].codec_dai->ops)
card->dai_link[i].codec_dai->ops = &null_dai_ops;
}
/* If we have AC97 in the system then don't wait for the
* codec. This will need revisiting if we have to handle
* systems with mixed AC97 and non-AC97 parts. Only check for
* DAIs currently; we can't do this per link since some AC97
* codecs have non-AC97 DAIs.
*/
if (!ac97)
for (i = 0; i < card->num_links; i++) {
found = 0;
list_for_each_entry(dai, &dai_list, list)
if (card->dai_link[i].codec_dai == dai) {
found = 1;
break;
}
if (!found) {
dev_dbg(card->dev, "DAI %s not registered\n",
card->dai_link[i].codec_dai->name);
return;
}
}
/* Note that we do not current check for codec components */
dev_dbg(card->dev, "All components present, instantiating\n");
/* Found everything, bring it up */
card->pmdown_time = pmdown_time;
if (card->probe) {
ret = card->probe(pdev);
if (ret < 0)
return;
}
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->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;
}
codec = card->codec;
if (platform->probe) {
ret = platform->probe(pdev);
if (ret < 0)
goto platform_err;
}
/* DAPM stream work */
INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
#ifdef CONFIG_PM
/* deferred resume work */
INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
#endif
for (i = 0; i < card->num_links; i++) {
if (card->dai_link[i].init) {
ret = card->dai_link[i].init(codec);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to init %s\n",
card->dai_link[i].stream_name);
continue;
}
}
if (card->dai_link[i].codec_dai->ac97_control)
ac97 = 1;
}
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
"%s", card->name);
snprintf(codec->card->longname, sizeof(codec->card->longname),
"%s (%s)", card->name, codec->name);
/* Make sure all DAPM widgets are instantiated */
snd_soc_dapm_new_widgets(codec);
ret = snd_card_register(codec->card);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
codec->name);
goto card_err;
}
mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
/* Only instantiate AC97 if not already done by the adaptor
* for the generic AC97 subsystem.
*/
if (ac97 && strcmp(codec->name, "AC97") != 0) {
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 card_err;
}
}
#endif
ret = snd_soc_dapm_sys_add(card->socdev->dev);
if (ret < 0)
printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time);
if (ret < 0)
printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n");
ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg);
if (ret < 0)
printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
soc_init_codec_debugfs(codec);
mutex_unlock(&codec->mutex);
card->instantiated = 1;
return;
card_err:
if (platform->remove)
platform->remove(pdev);
platform_err:
if (codec_dev->remove)
codec_dev->remove(pdev);
cpu_dai_err:
for (i--; i >= 0; i--) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
if (card->remove)
card->remove(pdev);
}
/*
* Attempt to initialise any uninitalised cards. Must be called with
* client_mutex.
*/
static void snd_soc_instantiate_cards(void)
{
struct snd_soc_card *card;
list_for_each_entry(card, &card_list, list)
snd_soc_instantiate_card(card);
}
/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
int ret = 0;
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
/* Bodge while we push things out of socdev */
card->socdev = socdev;
/* Bodge while we unpick instantiation */
card->dev = &pdev->dev;
ret = snd_soc_register_card(card);
if (ret != 0) {
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
return 0;
}
/* 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_card *card = socdev->card;
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
if (card->instantiated) {
run_delayed_work(&card->delayed_work);
if (platform->remove)
platform->remove(pdev);
if (codec_dev->remove)
codec_dev->remove(pdev);
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
if (card->remove)
card->remove(pdev);
}
snd_soc_unregister_card(card);
return 0;
}
static int soc_poweroff(struct device *dev)
{
struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
if (!card->instantiated)
return 0;
/* Flush out pmdown_time work - we actually do want to run it
* now, we're shutting down so no imminent restart. */
run_delayed_work(&card->delayed_work);
snd_soc_dapm_shutdown(socdev);
return 0;
}
static const struct dev_pm_ops soc_pm_ops = {
.suspend = soc_suspend,
.resume = soc_resume,
.poweroff = soc_poweroff,
};
/* ASoC platform driver */
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
.owner = THIS_MODULE,
.pm = &soc_pm_ops,
},
.probe = soc_probe,
.remove = soc_remove,
};
/* 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_card *card = socdev->card;
struct snd_soc_codec *codec = card->codec;
struct snd_soc_platform *platform = card->platform;
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 = card->codec;
/* check client and interface hw capabilities */
snprintf(new_name, sizeof(new_name), "%s %s-%d",
dai_link->stream_name, codec_dai->name, 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 = platform->pcm_ops->mmap;
soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
soc_pcm_ops.copy = platform->pcm_ops->copy;
soc_pcm_ops.silence = platform->pcm_ops->silence;
soc_pcm_ops.ack = platform->pcm_ops->ack;
soc_pcm_ops.page = 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 = 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 = platform->pcm_free;
printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
cpu_dai->name);
return ret;
}
/**
* snd_soc_codec_volatile_register: Report if a register is volatile.
*
* @codec: CODEC to query.
* @reg: Register to query.
*
* Boolean function indiciating if a CODEC register is volatile.
*/
int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
{
if (codec->volatile_register)
return codec->volatile_register(reg);
else
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register);
/**
* 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;
codec->dev = &codec->ac97->dev;
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 int mask, unsigned int value)
{
int change;
unsigned int old, new;
old = snd_soc_read(codec, reg);
new = (old & ~mask) | value;
change = old != new;
if (change)
snd_soc_write(codec, reg, new);
return change;
}
EXPORT_SYMBOL_GPL(snd_soc_update_bits);
/**
* snd_soc_update_bits_locked - update codec register bits
* @codec: audio codec
* @reg: codec register
* @mask: register mask
* @value: new value
*
* Writes new register value, and takes the codec mutex.
*
* Returns 1 for change else 0.
*/
int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
unsigned short reg, unsigned int mask,
unsigned int value)
{
int change;
mutex_lock(&codec->mutex);
change = snd_soc_update_bits(codec, reg, mask, value);
mutex_unlock(&codec->mutex);
return change;
}
EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked);
/**
* 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 int mask, unsigned int value)
{
int change;
unsigned int old, new;
old = snd_soc_read(codec, reg);
new = (old & ~mask) | value;
change = old != new;
return change;
}
EXPORT_SYMBOL_GPL(snd_soc_test_bits);
/**
* snd_soc_new_pcms - create new sound card and pcms
* @socdev: the SoC audio device
* @idx: ALSA card index
* @xid: card identification
*
* 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_card *card = socdev->card;
struct snd_soc_codec *codec = card->codec;
int ret, i;
mutex_lock(&codec->mutex);
/* register a sound card */
ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
if (ret < 0) {
printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
codec->name);
mutex_unlock(&codec->mutex);
return ret;
}
codec->socdev = socdev;
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 < card->num_links; i++) {
ret = soc_new_pcm(socdev, &card->dai_link[i], i);
if (ret < 0) {
printk(KERN_ERR "asoc: can't create pcm %s\n",
card->dai_link[i].stream_name);
mutex_unlock(&codec->mutex);
return ret;
}
/* Check for codec->ac97 to handle the ac97.c fun */
if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) {
snd_ac97_dev_add_pdata(codec->ac97,
card->dai_link[i].cpu_dai->ac97_pdata);
}
}
mutex_unlock(&codec->mutex);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
/**
* 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->card->codec;
#ifdef CONFIG_SND_SOC_AC97_BUS
struct snd_soc_dai *codec_dai;
int i;
#endif
mutex_lock(&codec->mutex);
soc_cleanup_codec_debugfs(codec);
#ifdef CONFIG_SND_SOC_AC97_BUS
for (i = 0; i < codec->num_dai; i++) {
codec_dai = &codec->dai[i];
if (codec_dai->ac97_control && codec->ac97 &&
strcmp(codec->name, "AC97") != 0) {
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
* @long_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_add_controls - add an array of controls to a codec.
* Convienience function to add a list of controls. Many codecs were
* duplicating this code.
*
* @codec: codec to add controls to
* @controls: array of controls to add
* @num_controls: number of elements in the array
*
* Return 0 for success, else error.
*/
int snd_soc_add_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls)
{
struct snd_card *card = codec->card;
int err, i;
for (i = 0; i < num_controls; i++) {
const struct snd_kcontrol_new *control = &controls[i];
err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
if (err < 0) {
dev_err(codec->dev, "%s: Failed to add %s\n",
codec->name, control->name);
return err;
}
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_add_controls);
/**
* 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->max;
if (uinfo->value.enumerated.item > e->max - 1)
uinfo->value.enumerated.item = e->max - 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
* @ucontrol: 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 int val, bitmask;
for (bitmask = 1; bitmask < e->max; 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
* @ucontrol: 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 int val;
unsigned int mask, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
;
if (ucontrol->value.enumerated.item[0] > e->max - 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->max - 1)
return -EINVAL;
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
mask |= (bitmask - 1) << e->shift_r;
}
return snd_soc_update_bits_locked(codec, e->reg, mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
/**
* snd_soc_get_value_enum_double - semi enumerated double mixer get callback
* @kcontrol: mixer control
* @ucontrol: control element information
*
* Callback to get the value of a double semi enumerated mixer.
*
* Semi enumerated mixer: the enumerated items are referred as values. Can be
* used for handling bitfield coded enumeration for example.
*
* Returns 0 for success.
*/
int snd_soc_get_value_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 int reg_val, val, mux;
reg_val = snd_soc_read(codec, e->reg);
val = (reg_val >> e->shift_l) & e->mask;
for (mux = 0; mux < e->max; mux++) {
if (val == e->values[mux])
break;
}
ucontrol->value.enumerated.item[0] = mux;
if (e->shift_l != e->shift_r) {
val = (reg_val >> e->shift_r) & e->mask;
for (mux = 0; mux < e->max; mux++) {
if (val == e->values[mux])
break;
}
ucontrol->value.enumerated.item[1] = mux;
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);
/**
* snd_soc_put_value_enum_double - semi enumerated double mixer put callback
* @kcontrol: mixer control
* @ucontrol: control element information
*
* Callback to set the value of a double semi enumerated mixer.
*
* Semi enumerated mixer: the enumerated items are referred as values. Can be
* used for handling bitfield coded enumeration for example.
*
* Returns 0 for success.
*/
int snd_soc_put_value_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 int val;
unsigned int mask;
if (ucontrol->value.enumerated.item[0] > e->max - 1)
return -EINVAL;
val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
if (ucontrol->value.enumerated.item[1] > e->max - 1)
return -EINVAL;
val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
mask |= e->mask << e->shift_r;
}
return snd_soc_update_bits_locked(codec, e->reg, mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_value_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->max;
if (uinfo->value.enumerated.item > e->max - 1)
uinfo->value.enumerated.item = e->max - 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 && !strstr(kcontrol->id.name, " Volume"))
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)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int platform_max;
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
if (!mc->platform_max)
mc->platform_max = mc->max;
platform_max = mc->platform_max;
if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
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 = platform_max;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
/**
* snd_soc_get_volsw - single mixer get callback
* @kcontrol: mixer control
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
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
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned int 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_locked(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)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int platform_max;
if (!mc->platform_max)
mc->platform_max = mc->max;
platform_max = mc->platform_max;
if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
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 = platform_max;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
/**
* snd_soc_get_volsw_2r - double mixer get callback
* @kcontrol: mixer control
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
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
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
int err;
unsigned int 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_locked(codec, reg, val_mask, val);
if (err < 0)
return err;
err = snd_soc_update_bits_locked(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)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int platform_max;
int min = mc->min;
if (!mc->platform_max)
mc->platform_max = mc->max;
platform_max = mc->platform_max;
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = platform_max - min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
/**
* snd_soc_get_volsw_s8 - signed mixer get callback
* @kcontrol: mixer control
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
int min = mc->min;
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
* @ucontrol: 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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
int min = mc->min;
unsigned int val;
val = (ucontrol->value.integer.value[0]+min) & 0xff;
val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
return snd_soc_update_bits_locked(codec, reg, 0xffff, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
/**
* snd_soc_limit_volume - Set new limit to an existing volume control.
*
* @codec: where to look for the control
* @name: Name of the control
* @max: new maximum limit
*
* Return 0 for success, else error.
*/
int snd_soc_limit_volume(struct snd_soc_codec *codec,
const char *name, int max)
{
struct snd_card *card = codec->card;
struct snd_kcontrol *kctl;
struct soc_mixer_control *mc;
int found = 0;
int ret = -EINVAL;
/* Sanity check for name and max */
if (unlikely(!name || max <= 0))
return -EINVAL;
list_for_each_entry(kctl, &card->controls, list) {
if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) {
found = 1;
break;
}
}
if (found) {
mc = (struct soc_mixer_control *)kctl->private_value;
if (max <= mc->max) {
mc->platform_max = max;
ret = 0;
}
}
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
/**
* 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->ops && dai->ops->set_sysclk)
return 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
* @div_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->ops && dai->ops->set_clkdiv)
return 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
* @source: DAI specific source for the PLL
* @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, int source,
unsigned int freq_in, unsigned int freq_out)
{
if (dai->ops && dai->ops->set_pll)
return dai->ops->set_pll(dai, pll_id, source,
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
* @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->ops && dai->ops->set_fmt)
return 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
* @tx_mask: bitmask representing active TX slots.
* @rx_mask: bitmask representing active RX slots.
* @slots: Number of slots in use.
* @slot_width: Width in bits for each slot.
*
* 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 tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
if (dai->ops && dai->ops->set_tdm_slot)
return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask,
slots, slot_width);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
/**
* snd_soc_dai_set_channel_map - configure DAI audio channel map
* @dai: DAI
* @tx_num: how many TX channels
* @tx_slot: pointer to an array which imply the TX slot number channel
* 0~num-1 uses
* @rx_num: how many RX channels
* @rx_slot: pointer to an array which imply the RX slot number channel
* 0~num-1 uses
*
* configure the relationship between channel number and TDM slot number.
*/
int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
unsigned int tx_num, unsigned int *tx_slot,
unsigned int rx_num, unsigned int *rx_slot)
{
if (dai->ops && dai->ops->set_channel_map)
return dai->ops->set_channel_map(dai, tx_num, tx_slot,
rx_num, rx_slot);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
/**
* 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->ops && dai->ops->set_tristate)
return 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->ops && dai->ops->digital_mute)
return dai->ops->digital_mute(dai, mute);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
/**
* snd_soc_register_card - Register a card with the ASoC core
*
* @card: Card to register
*
* Note that currently this is an internal only function: it will be
* exposed to machine drivers after further backporting of ASoC v2
* registration APIs.
*/
static int snd_soc_register_card(struct snd_soc_card *card)
{
if (!card->name || !card->dev)
return -EINVAL;
INIT_LIST_HEAD(&card->list);
card->instantiated = 0;
mutex_lock(&client_mutex);
list_add(&card->list, &card_list);
snd_soc_instantiate_cards();
mutex_unlock(&client_mutex);
dev_dbg(card->dev, "Registered card '%s'\n", card->name);
return 0;
}
/**
* snd_soc_unregister_card - Unregister a card with the ASoC core
*
* @card: Card to unregister
*
* Note that currently this is an internal only function: it will be
* exposed to machine drivers after further backporting of ASoC v2
* registration APIs.
*/
static int snd_soc_unregister_card(struct snd_soc_card *card)
{
mutex_lock(&client_mutex);
list_del(&card->list);
mutex_unlock(&client_mutex);
dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
return 0;
}
/**
* snd_soc_register_dai - Register a DAI with the ASoC core
*
* @dai: DAI to register
*/
int snd_soc_register_dai(struct snd_soc_dai *dai)
{
if (!dai->name)
return -EINVAL;
/* The device should become mandatory over time */
if (!dai->dev)
printk(KERN_WARNING "No device for DAI %s\n", dai->name);
if (!dai->ops)
dai->ops = &null_dai_ops;
INIT_LIST_HEAD(&dai->list);
mutex_lock(&client_mutex);
list_add(&dai->list, &dai_list);
snd_soc_instantiate_cards();
mutex_unlock(&client_mutex);
pr_debug("Registered DAI '%s'\n", dai->name);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dai);
/**
* snd_soc_unregister_dai - Unregister a DAI from the ASoC core
*
* @dai: DAI to unregister
*/
void snd_soc_unregister_dai(struct snd_soc_dai *dai)
{
mutex_lock(&client_mutex);
list_del(&dai->list);
mutex_unlock(&client_mutex);
pr_debug("Unregistered DAI '%s'\n", dai->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
/**
* snd_soc_register_dais - Register multiple DAIs with the ASoC core
*
* @dai: Array of DAIs to register
* @count: Number of DAIs
*/
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
{
int i, ret;
for (i = 0; i < count; i++) {
ret = snd_soc_register_dai(&dai[i]);
if (ret != 0)
goto err;
}
return 0;
err:
for (i--; i >= 0; i--)
snd_soc_unregister_dai(&dai[i]);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dais);
/**
* snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
*
* @dai: Array of DAIs to unregister
* @count: Number of DAIs
*/
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
{
int i;
for (i = 0; i < count; i++)
snd_soc_unregister_dai(&dai[i]);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
/**
* snd_soc_register_platform - Register a platform with the ASoC core
*
* @platform: platform to register
*/
int snd_soc_register_platform(struct snd_soc_platform *platform)
{
if (!platform->name)
return -EINVAL;
INIT_LIST_HEAD(&platform->list);
mutex_lock(&client_mutex);
list_add(&platform->list, &platform_list);
snd_soc_instantiate_cards();
mutex_unlock(&client_mutex);
pr_debug("Registered platform '%s'\n", platform->name);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_platform);
/**
* snd_soc_unregister_platform - Unregister a platform from the ASoC core
*
* @platform: platform to unregister
*/
void snd_soc_unregister_platform(struct snd_soc_platform *platform)
{
mutex_lock(&client_mutex);
list_del(&platform->list);
mutex_unlock(&client_mutex);
pr_debug("Unregistered platform '%s'\n", platform->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
static u64 codec_format_map[] = {
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
};
/* Fix up the DAI formats for endianness: codecs don't actually see
* the endianness of the data but we're using the CPU format
* definitions which do need to include endianness so we ensure that
* codec DAIs always have both big and little endian variants set.
*/
static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
{
int i;
for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
if (stream->formats & codec_format_map[i])
stream->formats |= codec_format_map[i];
}
/**
* snd_soc_register_codec - Register a codec with the ASoC core
*
* @codec: codec to register
*/
int snd_soc_register_codec(struct snd_soc_codec *codec)
{
int i;
if (!codec->name)
return -EINVAL;
/* The device should become mandatory over time */
if (!codec->dev)
printk(KERN_WARNING "No device for codec %s\n", codec->name);
INIT_LIST_HEAD(&codec->list);
for (i = 0; i < codec->num_dai; i++) {
fixup_codec_formats(&codec->dai[i].playback);
fixup_codec_formats(&codec->dai[i].capture);
}
mutex_lock(&client_mutex);
list_add(&codec->list, &codec_list);
snd_soc_instantiate_cards();
mutex_unlock(&client_mutex);
pr_debug("Registered codec '%s'\n", codec->name);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_codec);
/**
* snd_soc_unregister_codec - Unregister a codec from the ASoC core
*
* @codec: codec to unregister
*/
void snd_soc_unregister_codec(struct snd_soc_codec *codec)
{
mutex_lock(&client_mutex);
list_del(&codec->list);
mutex_unlock(&client_mutex);
pr_debug("Unregistered codec '%s'\n", codec->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
static int __init snd_soc_init(void)
{
#ifdef CONFIG_DEBUG_FS
debugfs_root = debugfs_create_dir("asoc", NULL);
if (IS_ERR(debugfs_root) || !debugfs_root) {
printk(KERN_WARNING
"ASoC: Failed to create debugfs directory\n");
debugfs_root = NULL;
}
#endif
return platform_driver_register(&soc_driver);
}
static void __exit snd_soc_exit(void)
{
#ifdef CONFIG_DEBUG_FS
debugfs_remove_recursive(debugfs_root);
#endif
platform_driver_unregister(&soc_driver);
}
module_init(snd_soc_init);
module_exit(snd_soc_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:soc-audio");
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