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tmp_suning_uos_patched/sound/soc/soc-core.c
Liam Girdwood a71a468a50 [ALSA] ASoC: Add support for BCLK based on (Rate * Chn * Word Size) 
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
 1. BCLK = Rate * x
 2. BCLK = MCLK / x
 3. BCLK = Rate * Chn * Word Size.  (New)
Changes:-
 o Add support for RCW generation of BCLK
 o Update Documentation to include RCW.
 o Update DAI documentation for label = value DAI modes.
 o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:01:07 +01:00

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/*
* soc-core.c -- ALSA SoC Audio Layer
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
*
* Revision history
* 12th Aug 2005 Initial version.
* 25th Oct 2005 Working Codec, Interface and Platform registration.
*
* 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/driver.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
/* debug DAI capabilities matching */
#define SOC_DEBUG_DAI 0
#if SOC_DEBUG_DAI
#define dbgc(format, arg...) printk(format, ## arg)
#else
#define dbgc(format, arg...)
#endif
#define CODEC_CPU(codec, cpu) ((codec << 4) | cpu)
static DEFINE_MUTEX(pcm_mutex);
static DEFINE_MUTEX(io_mutex);
static struct workqueue_struct *soc_workq;
static struct work_struct soc_stream_work;
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
/* supported sample rates */
/* ATTENTION: these values depend on the definition in pcm.h! */
static const unsigned int rates[] = {
5512, 8000, 11025, 16000, 22050, 32000, 44100,
48000, 64000, 88200, 96000, 176400, 192000
};
/*
* 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)");
#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:
return "AC97";
case SND_SOC_DAI_I2S:
return "I2S";
case SND_SOC_DAI_PCM:
return "PCM";
}
return NULL;
}
/* get rate format from rate */
static inline int soc_get_rate_format(int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(rates); i++) {
if (rates[i] == rate)
return 1 << i;
}
return 0;
}
/* gets the audio system mclk/sysclk for the given parameters */
static unsigned inline int soc_get_mclk(struct snd_soc_pcm_runtime *rtd,
struct snd_soc_clock_info *info)
{
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_machine *machine = socdev->machine;
int i;
/* find the matching machine config and get it's mclk for the given
* sample rate and hardware format */
for(i = 0; i < machine->num_links; i++) {
if (machine->dai_link[i].cpu_dai == rtd->cpu_dai &&
machine->dai_link[i].config_sysclk)
return machine->dai_link[i].config_sysclk(rtd, info);
}
return 0;
}
/* changes a bitclk multiplier mask to a divider mask */
static u64 soc_bfs_rcw_to_div(u64 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
int i, j;
u64 bfs_ = 0;
int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
if (size <= 0)
return 0;
/* the minimum bit clock that has enough bandwidth */
min = size * rate * chn;
dbgc("rcw --> div min bclk %d with mclk %d\n", min, mclk);
for (i = 0; i < 64; i++) {
if ((bfs >> i) & 0x1) {
j = min * (i + 1);
bfs_ |= SND_SOC_FSBD(mclk/j);
dbgc("rcw --> div support mult %d\n",
SND_SOC_FSBD_REAL(1<<i));
}
}
return bfs_;
}
/* changes a bitclk divider mask to a multiplier mask */
static u64 soc_bfs_div_to_rcw(u64 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
int i, j;
u64 bfs_ = 0;
int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
if (size <= 0)
return 0;
/* the minimum bit clock that has enough bandwidth */
min = size * rate * chn;
dbgc("div to rcw min bclk %d with mclk %d\n", min, mclk);
for (i = 0; i < 64; i++) {
if ((bfs >> i) & 0x1) {
j = mclk / (i + 1);
if (j >= min) {
bfs_ |= SND_SOC_FSBW(j/min);
dbgc("div --> rcw support div %d\n",
SND_SOC_FSBW_REAL(1<<i));
}
}
}
return bfs_;
}
/* changes a constant bitclk to a multiplier mask */
static u64 soc_bfs_rate_to_rcw(u64 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
unsigned int bfs_ = rate * bfs;
int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
if (size <= 0)
return 0;
/* the minimum bit clock that has enough bandwidth */
min = size * rate * chn;
dbgc("rate --> rcw min bclk %d with mclk %d\n", min, mclk);
if (bfs_ < min)
return 0;
else {
bfs_ = SND_SOC_FSBW(bfs_/min);
dbgc("rate --> rcw support div %d\n", SND_SOC_FSBW_REAL(bfs_));
return bfs_;
}
}
/* changes a bitclk multiplier mask to a divider mask */
static u64 soc_bfs_rate_to_div(u64 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
unsigned int bfs_ = rate * bfs;
int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
if (size <= 0)
return 0;
/* the minimum bit clock that has enough bandwidth */
min = size * rate * chn;
dbgc("rate --> div min bclk %d with mclk %d\n", min, mclk);
if (bfs_ < min)
return 0;
else {
bfs_ = SND_SOC_FSBW(mclk/bfs_);
dbgc("rate --> div support div %d\n", SND_SOC_FSBD_REAL(bfs_));
return bfs_;
}
}
/* Matches codec DAI and SoC CPU DAI hardware parameters */
static int soc_hw_match_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai_mode *codec_dai_mode = NULL;
struct snd_soc_dai_mode *cpu_dai_mode = NULL;
struct snd_soc_clock_info clk_info;
unsigned int fs, mclk, rate = params_rate(params),
chn, j, k, cpu_bclk, codec_bclk, pcmrate;
u16 fmt = 0;
u64 codec_bfs, cpu_bfs;
dbg("asoc: match version %s\n", SND_SOC_VERSION);
clk_info.rate = rate;
pcmrate = soc_get_rate_format(rate);
/* try and find a match from the codec and cpu DAI capabilities */
for (j = 0; j < rtd->codec_dai->caps.num_modes; j++) {
for (k = 0; k < rtd->cpu_dai->caps.num_modes; k++) {
codec_dai_mode = &rtd->codec_dai->caps.mode[j];
cpu_dai_mode = &rtd->cpu_dai->caps.mode[k];
if (!(codec_dai_mode->pcmrate & cpu_dai_mode->pcmrate &
pcmrate)) {
dbgc("asoc: DAI[%d:%d] failed to match rate\n", j, k);
continue;
}
fmt = codec_dai_mode->fmt & cpu_dai_mode->fmt;
if (!(fmt & SND_SOC_DAIFMT_FORMAT_MASK)) {
dbgc("asoc: DAI[%d:%d] failed to match format\n", j, k);
continue;
}
if (!(fmt & SND_SOC_DAIFMT_CLOCK_MASK)) {
dbgc("asoc: DAI[%d:%d] failed to match clock masters\n",
j, k);
continue;
}
if (!(fmt & SND_SOC_DAIFMT_INV_MASK)) {
dbgc("asoc: DAI[%d:%d] failed to match invert\n", j, k);
continue;
}
if (!(codec_dai_mode->pcmfmt & cpu_dai_mode->pcmfmt)) {
dbgc("asoc: DAI[%d:%d] failed to match pcm format\n", j, k);
continue;
}
if (!(codec_dai_mode->pcmdir & cpu_dai_mode->pcmdir)) {
dbgc("asoc: DAI[%d:%d] failed to match direction\n", j, k);
continue;
}
/* todo - still need to add tdm selection */
rtd->cpu_dai->dai_runtime.fmt =
rtd->codec_dai->dai_runtime.fmt =
1 << (ffs(fmt & SND_SOC_DAIFMT_FORMAT_MASK) -1) |
1 << (ffs(fmt & SND_SOC_DAIFMT_CLOCK_MASK) - 1) |
1 << (ffs(fmt & SND_SOC_DAIFMT_INV_MASK) - 1);
clk_info.bclk_master =
rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK;
/* make sure the ratio between rate and master
* clock is acceptable*/
fs = (cpu_dai_mode->fs & codec_dai_mode->fs);
if (fs == 0) {
dbgc("asoc: DAI[%d:%d] failed to match FS\n", j, k);
continue;
}
clk_info.fs = rtd->cpu_dai->dai_runtime.fs =
rtd->codec_dai->dai_runtime.fs = fs;
/* calculate audio system clocking using slowest clocks possible*/
mclk = soc_get_mclk(rtd, &clk_info);
if (mclk == 0) {
dbgc("asoc: DAI[%d:%d] configuration not clockable\n", j, k);
dbgc("asoc: rate %d fs %d master %x\n", rate, fs,
clk_info.bclk_master);
continue;
}
/* calculate word size (per channel) and frame size */
rtd->codec_dai->dai_runtime.pcmfmt =
rtd->cpu_dai->dai_runtime.pcmfmt =
1 << params_format(params);
chn = params_channels(params);
/* i2s always has left and right */
if (params_channels(params) == 1 &&
rtd->cpu_dai->dai_runtime.fmt & (SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_LEFT_J))
chn <<= 1;
/* Calculate bfs - the ratio between bitclock and the sample rate
* We must take into consideration the dividers and multipliers
* used in the codec and cpu DAI modes. We always choose the
* lowest possible clocks to reduce power.
*/
switch (CODEC_CPU(codec_dai_mode->flags, cpu_dai_mode->flags)) {
case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_DIV):
/* cpu & codec bfs dividers */
rtd->cpu_dai->dai_runtime.bfs =
rtd->codec_dai->dai_runtime.bfs =
1 << (fls(codec_dai_mode->bfs & cpu_dai_mode->bfs) - 1);
break;
case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RCW):
/* normalise bfs codec divider & cpu rcw mult */
codec_bfs = soc_bfs_div_to_rcw(codec_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
rtd->cpu_dai->dai_runtime.bfs =
1 << (ffs(codec_bfs & cpu_dai_mode->bfs) - 1);
cpu_bfs = soc_bfs_rcw_to_div(cpu_dai_mode->bfs, rate, mclk,
rtd->codec_dai->dai_runtime.pcmfmt, chn);
rtd->codec_dai->dai_runtime.bfs =
1 << (fls(codec_dai_mode->bfs & cpu_bfs) - 1);
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_DIV):
/* normalise bfs codec rcw mult & cpu divider */
codec_bfs = soc_bfs_rcw_to_div(codec_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
rtd->cpu_dai->dai_runtime.bfs =
1 << (fls(codec_bfs & cpu_dai_mode->bfs) -1);
cpu_bfs = soc_bfs_div_to_rcw(cpu_dai_mode->bfs, rate, mclk,
rtd->codec_dai->dai_runtime.pcmfmt, chn);
rtd->codec_dai->dai_runtime.bfs =
1 << (ffs(codec_dai_mode->bfs & cpu_bfs) -1);
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RCW):
/* codec & cpu bfs rate rcw multipliers */
rtd->cpu_dai->dai_runtime.bfs =
rtd->codec_dai->dai_runtime.bfs =
1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1);
break;
case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RATE):
/* normalise cpu bfs rate const multiplier & codec div */
cpu_bfs = soc_bfs_rate_to_div(cpu_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
if(codec_dai_mode->bfs & cpu_bfs) {
rtd->codec_dai->dai_runtime.bfs = cpu_bfs;
rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs;
} else
rtd->cpu_dai->dai_runtime.bfs = 0;
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RATE):
/* normalise cpu bfs rate const multiplier & codec rcw mult */
cpu_bfs = soc_bfs_rate_to_rcw(cpu_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
if(codec_dai_mode->bfs & cpu_bfs) {
rtd->codec_dai->dai_runtime.bfs = cpu_bfs;
rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs;
} else
rtd->cpu_dai->dai_runtime.bfs = 0;
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RCW):
/* normalise cpu bfs rate rcw multiplier & codec const mult */
codec_bfs = soc_bfs_rate_to_rcw(codec_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
if(cpu_dai_mode->bfs & codec_bfs) {
rtd->cpu_dai->dai_runtime.bfs = codec_bfs;
rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs;
} else
rtd->cpu_dai->dai_runtime.bfs = 0;
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_DIV):
/* normalise cpu bfs div & codec const mult */
codec_bfs = soc_bfs_rate_to_div(codec_dai_mode->bfs, rate,
mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
if(codec_dai_mode->bfs & codec_bfs) {
rtd->cpu_dai->dai_runtime.bfs = codec_bfs;
rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs;
} else
rtd->cpu_dai->dai_runtime.bfs = 0;
break;
case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RATE):
/* cpu & codec constant mult */
if(codec_dai_mode->bfs == cpu_dai_mode->bfs)
rtd->cpu_dai->dai_runtime.bfs =
rtd->codec_dai->dai_runtime.bfs =
codec_dai_mode->bfs;
else
rtd->cpu_dai->dai_runtime.bfs =
rtd->codec_dai->dai_runtime.bfs = 0;
break;
}
/* make sure the bit clock speed is acceptable */
if (!rtd->cpu_dai->dai_runtime.bfs ||
!rtd->codec_dai->dai_runtime.bfs) {
dbgc("asoc: DAI[%d:%d] failed to match BFS\n", j, k);
dbgc("asoc: cpu_dai %llu codec %llu\n",
rtd->cpu_dai->dai_runtime.bfs,
rtd->codec_dai->dai_runtime.bfs);
dbgc("asoc: mclk %d hwfmt %x\n", mclk, fmt);
continue;
}
goto found;
}
}
printk(KERN_ERR "asoc: no matching DAI found between codec and CPU\n");
return -EINVAL;
found:
/* we have matching DAI's, so complete the runtime info */
rtd->codec_dai->dai_runtime.pcmrate =
rtd->cpu_dai->dai_runtime.pcmrate =
soc_get_rate_format(rate);
rtd->codec_dai->dai_runtime.priv = codec_dai_mode->priv;
rtd->cpu_dai->dai_runtime.priv = cpu_dai_mode->priv;
rtd->codec_dai->dai_runtime.flags = codec_dai_mode->flags;
rtd->cpu_dai->dai_runtime.flags = cpu_dai_mode->flags;
/* for debug atm */
dbg("asoc: DAI[%d:%d] Match OK\n", j, k);
if (rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) {
codec_bclk = (rtd->codec_dai->dai_runtime.fs * params_rate(params)) /
SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
dbg("asoc: codec fs %d mclk %d bfs div %d bclk %d\n",
rtd->codec_dai->dai_runtime.fs, mclk,
SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk);
} else if(rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) {
codec_bclk = params_rate(params) * rtd->codec_dai->dai_runtime.bfs;
dbg("asoc: codec fs %d mclk %d bfs rate mult %llu bclk %d\n",
rtd->codec_dai->dai_runtime.fs, mclk,
rtd->codec_dai->dai_runtime.bfs, codec_bclk);
} else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) {
codec_bclk = params_rate(params) * params_channels(params) *
snd_pcm_format_physical_width(rtd->codec_dai->dai_runtime.pcmfmt) *
SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs);
dbg("asoc: codec fs %d mclk %d bfs rcw mult %d bclk %d\n",
rtd->codec_dai->dai_runtime.fs, mclk,
SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk);
} else
codec_bclk = 0;
if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) {
cpu_bclk = (rtd->cpu_dai->dai_runtime.fs * params_rate(params)) /
SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs);
dbg("asoc: cpu fs %d mclk %d bfs div %d bclk %d\n",
rtd->cpu_dai->dai_runtime.fs, mclk,
SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk);
} else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) {
cpu_bclk = params_rate(params) * rtd->cpu_dai->dai_runtime.bfs;
dbg("asoc: cpu fs %d mclk %d bfs rate mult %llu bclk %d\n",
rtd->cpu_dai->dai_runtime.fs, mclk,
rtd->cpu_dai->dai_runtime.bfs, cpu_bclk);
} else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) {
cpu_bclk = params_rate(params) * params_channels(params) *
snd_pcm_format_physical_width(rtd->cpu_dai->dai_runtime.pcmfmt) *
SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs);
dbg("asoc: cpu fs %d mclk %d bfs mult rcw %d bclk %d\n",
rtd->cpu_dai->dai_runtime.fs, mclk,
SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk);
} else
cpu_bclk = 0;
/*
* Check we have matching bitclocks. If we don't then it means the
* sysclock returned by either the codec or cpu DAI (selected by the
* machine sysclock function) is wrong compared with the supported DAI
* modes for the codec or cpu DAI.
*/
if (cpu_bclk != codec_bclk && cpu_bclk){
printk(KERN_ERR
"asoc: codec and cpu bitclocks differ, audio may be wrong speed\n"
);
printk(KERN_ERR "asoc: codec %d != cpu %d\n", codec_bclk, cpu_bclk);
}
switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
dbg("asoc: DAI codec BCLK master, LRC master\n");
break;
case SND_SOC_DAIFMT_CBS_CFM:
dbg("asoc: DAI codec BCLK slave, LRC master\n");
break;
case SND_SOC_DAIFMT_CBM_CFS:
dbg("asoc: DAI codec BCLK master, LRC slave\n");
break;
case SND_SOC_DAIFMT_CBS_CFS:
dbg("asoc: DAI codec BCLK slave, LRC slave\n");
break;
}
dbg("asoc: mode %x, invert %x\n",
rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK,
rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK);
dbg("asoc: audio rate %d chn %d fmt %x\n", params_rate(params),
params_channels(params), params_format(params));
return 0;
}
static inline u32 get_rates(struct snd_soc_dai_mode *modes, int nmodes)
{
int i;
u32 rates = 0;
for(i = 0; i < nmodes; i++)
rates |= modes[i].pcmrate;
return rates;
}
static inline u64 get_formats(struct snd_soc_dai_mode *modes, int nmodes)
{
int i;
u64 formats = 0;
for(i = 0; i < nmodes; i++)
formats |= modes[i].pcmfmt;
return formats;
}
/*
* 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_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec_dai *codec_dai = rtd->codec_dai;
struct snd_soc_cpu_dai *cpu_dai = rtd->cpu_dai;
int ret = 0;
mutex_lock(&pcm_mutex);
/* startup the audio subsystem */
if (rtd->cpu_dai->ops.startup) {
ret = rtd->cpu_dai->ops.startup(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open interface %s\n",
rtd->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 (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;
}
}
if (rtd->codec_dai->ops.startup) {
ret = rtd->codec_dai->ops.startup(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open codec %s\n",
rtd->codec_dai->name);
goto codec_dai_err;
}
}
/* create runtime params from DMA, codec and cpu DAI */
if (runtime->hw.rates)
runtime->hw.rates &=
get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) &
get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
else
runtime->hw.rates =
get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) &
get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
if (runtime->hw.formats)
runtime->hw.formats &=
get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) &
get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
else
runtime->hw.formats =
get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) &
get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
/* Check that the codec and cpu DAI's are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw.rate_min =
max(rtd->codec_dai->playback.rate_min,
rtd->cpu_dai->playback.rate_min);
runtime->hw.rate_max =
min(rtd->codec_dai->playback.rate_max,
rtd->cpu_dai->playback.rate_max);
runtime->hw.channels_min =
max(rtd->codec_dai->playback.channels_min,
rtd->cpu_dai->playback.channels_min);
runtime->hw.channels_max =
min(rtd->codec_dai->playback.channels_max,
rtd->cpu_dai->playback.channels_max);
} else {
runtime->hw.rate_min =
max(rtd->codec_dai->capture.rate_min,
rtd->cpu_dai->capture.rate_min);
runtime->hw.rate_max =
min(rtd->codec_dai->capture.rate_max,
rtd->cpu_dai->capture.rate_max);
runtime->hw.channels_min =
max(rtd->codec_dai->capture.channels_min,
rtd->cpu_dai->capture.channels_min);
runtime->hw.channels_max =
min(rtd->codec_dai->capture.channels_max,
rtd->cpu_dai->capture.channels_max);
}
snd_pcm_limit_hw_rates(runtime);
if (!runtime->hw.rates) {
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
rtd->codec_dai->name, rtd->cpu_dai->name);
goto codec_dai_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
rtd->codec_dai->name, rtd->cpu_dai->name);
goto codec_dai_err;
}
if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
rtd->codec_dai->name, rtd->cpu_dai->name);
goto codec_dai_err;
}
dbg("asoc: %s <-> %s info:\n", rtd->codec_dai->name, rtd->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)
rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 1;
else
rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 1;
rtd->cpu_dai->active = rtd->codec_dai->active = 1;
rtd->cpu_dai->runtime = runtime;
socdev->codec->active++;
mutex_unlock(&pcm_mutex);
return 0;
codec_dai_err:
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
machine_err:
if (platform->pcm_ops->close)
platform->pcm_ops->close(substream);
platform_err:
if (rtd->cpu_dai->ops.shutdown)
rtd->cpu_dai->ops.shutdown(substream);
out:
mutex_unlock(&pcm_mutex);
return ret;
}
/*
* Power down the audio subsytem 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(void *data)
{
struct snd_soc_device *socdev = data;
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_codec_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) {
codec_dai->pop_wait = 0;
snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_STOP);
/* power down the codec power domain if no longer active */
if (codec->active == 0) {
dbg("pop wq D3 %s %s\n", codec->name,
codec_dai->playback.stream_name);
if (codec->dapm_event)
codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
}
}
}
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_machine *machine = socdev->machine;
struct snd_soc_platform *platform = socdev->platform;
struct snd_soc_codec *codec = socdev->codec;
mutex_lock(&pcm_mutex);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 0;
else
rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 0;
if (rtd->codec_dai->playback.active == 0 &&
rtd->codec_dai->capture.active == 0) {
rtd->cpu_dai->active = rtd->codec_dai->active = 0;
}
codec->active--;
if (rtd->cpu_dai->ops.shutdown)
rtd->cpu_dai->ops.shutdown(substream);
if (rtd->codec_dai->ops.shutdown)
rtd->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);
rtd->cpu_dai->runtime = NULL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* start delayed pop wq here for playback streams */
rtd->codec_dai->pop_wait = 1;
queue_delayed_work(soc_workq, &soc_stream_work,
msecs_to_jiffies(pmdown_time));
} else {
/* capture streams can be powered down now */
snd_soc_dapm_stream_event(codec, rtd->codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_STOP);
if (codec->active == 0 && rtd->codec_dai->pop_wait == 0){
if (codec->dapm_event)
codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
}
}
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_platform *platform = socdev->platform;
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
mutex_lock(&pcm_mutex);
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 (rtd->codec_dai->ops.prepare) {
ret = rtd->codec_dai->ops.prepare(substream);
if (ret < 0) {
printk(KERN_ERR "asoc: codec DAI prepare error\n");
goto out;
}
}
if (rtd->cpu_dai->ops.prepare)
ret = rtd->cpu_dai->ops.prepare(substream);
/* we only want to start a DAPM playback stream if we are not waiting
* on an existing one stopping */
if (rtd->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(codec,
rtd->codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
else {
rtd->codec_dai->pop_wait = 0;
cancel_delayed_work(&soc_stream_work);
if (rtd->codec_dai->digital_mute)
rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0);
}
} else {
/* no delayed work - do we need to power up codec */
if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
if (codec->dapm_event)
codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_soc_dapm_stream_event(codec,
rtd->codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_START);
else
snd_soc_dapm_stream_event(codec,
rtd->codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
if (codec->dapm_event)
codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
if (rtd->codec_dai->digital_mute)
rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0);
} else {
/* codec already powered - power on widgets */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_soc_dapm_stream_event(codec,
rtd->codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_START);
else
snd_soc_dapm_stream_event(codec,
rtd->codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
if (rtd->codec_dai->digital_mute)
rtd->codec_dai->digital_mute(codec, rtd->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_platform *platform = socdev->platform;
struct snd_soc_machine *machine = socdev->machine;
int ret = 0;
mutex_lock(&pcm_mutex);
/* we don't need to match any AC97 params */
if (rtd->cpu_dai->type != SND_SOC_DAI_AC97) {
ret = soc_hw_match_params(substream, params);
if (ret < 0)
goto out;
} else {
struct snd_soc_clock_info clk_info;
clk_info.rate = params_rate(params);
ret = soc_get_mclk(rtd, &clk_info);
if (ret < 0)
goto out;
}
if (rtd->codec_dai->ops.hw_params) {
ret = rtd->codec_dai->ops.hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: can't set codec %s hw params\n",
rtd->codec_dai->name);
goto out;
}
}
if (rtd->cpu_dai->ops.hw_params) {
ret = rtd->cpu_dai->ops.hw_params(substream, params);
if (ret < 0) {
printk(KERN_ERR "asoc: can't set interface %s hw params\n",
rtd->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: can't set platform %s hw params\n",
platform->name);
goto platform_err;
}
}
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 machine_err;
}
}
out:
mutex_unlock(&pcm_mutex);
return ret;
machine_err:
if (platform->pcm_ops->hw_free)
platform->pcm_ops->hw_free(substream);
platform_err:
if (rtd->cpu_dai->ops.hw_free)
rtd->cpu_dai->ops.hw_free(substream);
interface_err:
if (rtd->codec_dai->ops.hw_free)
rtd->codec_dai->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_platform *platform = socdev->platform;
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_machine *machine = socdev->machine;
mutex_lock(&pcm_mutex);
/* apply codec digital mute */
if (!codec->active && rtd->codec_dai->digital_mute)
rtd->codec_dai->digital_mute(codec, rtd->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 (rtd->codec_dai->ops.hw_free)
rtd->codec_dai->ops.hw_free(substream);
if (rtd->cpu_dai->ops.hw_free)
rtd->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_platform *platform = socdev->platform;
int ret;
if (rtd->codec_dai->ops.trigger) {
ret = rtd->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 (rtd->cpu_dai->ops.trigger) {
ret = rtd->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;
/* mute any active DAC's */
for(i = 0; i < machine->num_links; i++) {
struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
if (dai->digital_mute && dai->playback.active)
dai->digital_mute(codec, dai, 1);
}
if (machine->suspend_pre)
machine->suspend_pre(pdev, state);
for(i = 0; i < machine->num_links; i++) {
struct snd_soc_cpu_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 */
flush_workqueue(soc_workq);
codec->suspend_dapm_state = codec->dapm_state;
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_cpu_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;
}
/* powers up audio subsystem after a suspend */
static int soc_resume(struct platform_device *pdev)
{
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;
if (machine->resume_pre)
machine->resume_pre(pdev);
for(i = 0; i < machine->num_links; i++) {
struct snd_soc_cpu_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 DAC's */
for(i = 0; i < machine->num_links; i++) {
struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
if (dai->digital_mute && dai->playback.active)
dai->digital_mute(codec, dai, 0);
}
for(i = 0; i < machine->num_links; i++) {
struct snd_soc_cpu_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);
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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->probe) {
ret = cpu_dai->probe(pdev);
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 */
soc_workq = create_workqueue("kdapm");
if (soc_workq == NULL)
goto work_err;
INIT_WORK(&soc_stream_work, close_delayed_work, socdev);
return 0;
work_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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev);
}
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;
if (soc_workq)
destroy_workqueue(soc_workq);
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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev);
}
if (machine->remove)
machine->remove(pdev);
return 0;
}
/* ASoC platform driver */
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
},
.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_codec_dai *codec_dai = dai_link->codec_dai;
struct snd_soc_cpu_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->cpu_dai = cpu_dai;
rtd->codec_dai = codec_dai;
rtd->socdev = socdev;
/* 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;
}
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_get_rate - get int sample rate
* @hwpcmrate: the hardware pcm rate
*
* Returns the audio rate integaer value, else 0.
*/
int snd_soc_get_rate(int hwpcmrate)
{
int rate = ffs(hwpcmrate) - 1;
if (rate > ARRAY_SIZE(rates))
return 0;
return rates[rate];
}
EXPORT_SYMBOL_GPL(snd_soc_get_rate);
/**
* 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;
mutex_lock(&codec->mutex);
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].cpu_dai->type == SND_SOC_DAI_AC97)
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 codec %s\n",
codec->name);
goto out;
}
#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);
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 entries\n");
out:
mutex_unlock(&codec->mutex);
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;
mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
if (codec->ac97)
soc_ac97_dev_unregister(codec);
#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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
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 mask = kcontrol->private_value;
uinfo->type =
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = mask;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
/**
* snd_soc_info_bool_ext - external single boolean mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to provide information about a single boolean external mixer control.
*
* Returns 0 for success.
*/
int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_bool_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 mask = (kcontrol->private_value >> 16) & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
uinfo->type =
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = shift == rshift ? 1 : 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = mask;
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 mask = (kcontrol->private_value >> 16) & 0xff;
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] =
mask - ucontrol->value.integer.value[0];
if (shift != rshift)
ucontrol->value.integer.value[1] =
mask - 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 mask = (kcontrol->private_value >> 16) & 0xff;
int invert = (kcontrol->private_value >> 24) & 0x01;
int err;
unsigned short val, val2, val_mask;
val = (ucontrol->value.integer.value[0] & mask);
if (invert)
val = mask - val;
val_mask = mask << shift;
val = val << shift;
if (shift != rshift) {
val2 = (ucontrol->value.integer.value[1] & mask);
if (invert)
val2 = mask - val2;
val_mask |= mask << rshift;
val |= val2 << rshift;
}
err = snd_soc_update_bits(codec, reg, val_mask, val);
return err;
}
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 mask = (kcontrol->private_value >> 12) & 0xff;
uinfo->type =
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = mask;
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 mask = (kcontrol->private_value >> 12) & 0xff;
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] =
mask - ucontrol->value.integer.value[0];
ucontrol->value.integer.value[1] =
mask - 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 mask = (kcontrol->private_value >> 12) & 0xff;
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 = mask - val;
val2 = mask - val2;
}
val = val << shift;
val2 = val2 << shift;
if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
return err;
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
return err;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
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");
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