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android_kernel_xiaomi_sm8350/sound/soc/soc-core.c
Frank Mandarino db2a416556 [ALSA] ASoC: core code 
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o  Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
 o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.

Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:16 +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
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 u16 soc_bfs_mult_to_div(u16 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
int i, j;
u16 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("mult --> div min bclk %d with mclk %d\n", min, mclk);
for (i = 0; i < 16; i++) {
if ((bfs >> i) & 0x1) {
j = rate * SND_SOC_FSB_REAL(1<<i);
if (j >= min) {
bfs_ |= SND_SOC_FSBD(mclk/j);
dbgc("mult --> div support mult %d\n",
SND_SOC_FSB_REAL(1<<i));
}
}
}
return bfs_;
}
/* changes a bitclk divider mask to a multiplier mask */
static u16 soc_bfs_div_to_mult(u16 bfs, int rate, unsigned int mclk,
unsigned int pcmfmt, unsigned int chn)
{
int i, j;
u16 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 mult min bclk %d with mclk %d\n", min, mclk);
for (i = 0; i < 16; i++) {
if ((bfs >> i) & 0x1) {
j = mclk / (SND_SOC_FSBD_REAL(1<<i));
if (j >= min) {
bfs_ |= SND_SOC_FSB(j/rate);
dbgc("div --> mult support div %d\n",
SND_SOC_FSBD_REAL(1<<i));
}
}
}
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, codec_bfs, cpu_bfs, rate = params_rate(params),
chn, j, k, cpu_bclk, codec_bclk, pcmrate;
u16 fmt = 0;
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.
*/
if (codec_dai_mode->flags & cpu_dai_mode->flags &
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);
} else if (codec_dai_mode->flags & SND_SOC_DAI_BFS_DIV) {
/* normalise bfs codec divider & cpu mult */
codec_bfs = soc_bfs_div_to_mult(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_mult_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);
} else if (cpu_dai_mode->flags & SND_SOC_DAI_BFS_DIV) {
/* normalise bfs codec mult & cpu divider */
codec_bfs = soc_bfs_mult_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_mult(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);
} else {
/* codec & cpu bfs rate multipliers */
rtd->cpu_dai->dai_runtime.bfs =
rtd->codec_dai->dai_runtime.bfs =
1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1);
}
/* 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 %x codec %x\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 {
codec_bclk = params_rate(params) *
SND_SOC_FSB_REAL(rtd->codec_dai->dai_runtime.bfs);
dbg("asoc: codec fs %d mclk %d bfs mult %d bclk %d\n",
rtd->codec_dai->dai_runtime.fs, mclk,
SND_SOC_FSB_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk);
}
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 {
cpu_bclk = params_rate(params) *
SND_SOC_FSB_REAL(rtd->cpu_dai->dai_runtime.bfs);
dbg("asoc: cpu fs %d mclk %d bfs mult %d bclk %d\n",
rtd->cpu_dai->dai_runtime.fs, mclk,
SND_SOC_FSB_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk);
}
/*
* 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){
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 \nasoc: min ch %d max ch %d\n
asoc: min rate %d max rate %d\n",
runtime->hw.rates, runtime->hw.channels_min,
runtime->hw.channels_max, 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)
goto out;
}
if (rtd->codec_dai->ops.prepare) {
ret = rtd->codec_dai->ops.prepare(substream);
if (ret < 0)
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;
mutex_lock(&codec->mutex);
for(i = 0; i < machine->num_links; i++) {
if (socdev->machine->dai_link[i].init)
socdev->machine->dai_link[i].init(codec);
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);
mutex_unlock(&codec->mutex);
return ret;
}
#ifdef CONFIG_SND_SOC_AC97_BUS
if (ac97)
soc_ac97_dev_register(codec);
#endif
snd_soc_dapm_sys_add(socdev->dev);
device_create_file(socdev->dev, &dev_attr_codec_reg);
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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