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audio.c
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audio.c
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/// Copyright (C) 2009 - 2014 by Johns. All Rights Reserved.
/// Copyright (C) 2018 by pesintta, rofafor.
///
/// SPDX-License-Identifier: AGPL-3.0-only
///
/// This module contains all audio output functions.
///
/// ALSA PCM/Mixer api is supported.
/// @see http://www.alsa-project.org/alsa-doc/alsa-lib
///
/// @note alsa async playback is broken, don't use it!
///
///
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <inttypes.h>
#include <string.h>
#include <math.h>
#include <sched.h>
#include <alsa/asoundlib.h>
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#include "iatomic.h" // portable atomic_t
#include "ringbuffer.h"
#include "misc.h"
#include "audio.h"
//----------------------------------------------------------------------------
// Declarations
//----------------------------------------------------------------------------
/**
** Audio output module structure and typedef.
*/
typedef struct _audio_module_
{
const char *Name; ///< audio output module name
int (*const Thread) (void); ///< module thread handler
void (*const FlushBuffers) (void); ///< flush sample buffers
int64_t(*const GetDelay) (void); ///< get current audio delay
void (*const SetVolume) (int); ///< set output volume
int (*const Setup) (int *, int *, int); ///< setup channels, samplerate
void (*const Play) (void); ///< play audio
void (*const Pause) (void); ///< pause audio
void (*const Init) (void); ///< initialize audio output module
void (*const Exit) (void); ///< cleanup audio output module
} AudioModule;
static const AudioModule NoopModule; ///< forward definition of noop module
//----------------------------------------------------------------------------
// Variables
//----------------------------------------------------------------------------
char AudioAlsaDriverBroken; ///< disable broken driver message
char AudioAlsaNoCloseOpen; ///< disable alsa close/open fix
char AudioAlsaCloseOpenDelay; ///< enable alsa close/open delay fix
static const char *AudioModuleName; ///< which audio module to use
/// Selected audio module.
static const AudioModule *AudioUsedModule = &NoopModule;
static const char *AudioPCMDevice; ///< PCM device name
static const char *AudioPassthroughDevice; ///< Passthrough device name
static const char *AudioMixerDevice; ///< mixer device name
static const char *AudioMixerChannel; ///< mixer channel name
static char AudioDoingInit; ///> flag in init, reduce error
static volatile char AudioRunning; ///< thread running / stopped
static volatile char AudioPaused; ///< audio paused
static volatile char AudioVideoIsReady; ///< video ready start early
static int AudioSkip; ///< skip audio to sync to video
static const int AudioBytesProSample = 2; ///< number of bytes per sample
static int AudioBufferTime = 336; ///< audio buffer time in ms
static pthread_t AudioThread; ///< audio play thread
static pthread_mutex_t AudioMutex; ///< audio condition mutex
pthread_mutex_t PTS_mutex; ///< PTS mutex
pthread_mutex_t ReadAdvance_mutex; ///< PTS mutex
static pthread_cond_t AudioStartCond; ///< condition variable
static char AudioThreadStop; ///< stop audio thread
static char AudioSoftVolume; ///< flag use soft volume
static char AudioNormalize; ///< flag use volume normalize
static char AudioCompression; ///< flag use compress volume
static char AudioMute; ///< flag muted
static int AudioAmplifier; ///< software volume factor
static int AudioNormalizeFactor; ///< current normalize factor
static const int AudioMinNormalize = 100; ///< min. normalize factor
static int AudioMaxNormalize; ///< max. normalize factor
static int AudioCompressionFactor; ///< current compression factor
static int AudioMaxCompression; ///< max. compression factor
static int AudioStereoDescent; ///< volume descent for stereo
static int AudioVolume; ///< current volume (0 .. 1000)
extern int VideoAudioDelay; ///< import audio/video delay
extern volatile char SoftIsPlayingVideo; ///< stream contains video data
/// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
#define AUDIO_MIN_BUFFER_FREE (3072 * 8 * 8)
static int AudioChannelsInHw[9]; ///< table which channels are supported
enum _audio_rates
{ ///< sample rates enumeration
// HW: 32000 44100 48000 88200 96000 176400 192000
//Audio32000, ///< 32.0Khz
Audio44100, ///< 44.1Khz
Audio48000, ///< 48.0Khz
//Audio88200, ///< 88.2Khz
//Audio96000, ///< 96.0Khz
//Audio176400, ///< 176.4Khz
Audio192000, ///< 192.0Khz
AudioRatesMax ///< max index
};
/// table which rates are supported
static int AudioRatesInHw[AudioRatesMax];
/// input to hardware channel matrix
static int AudioChannelMatrix[AudioRatesMax][9];
/// rates tables (must be sorted by frequency)
static const unsigned AudioRatesTable[AudioRatesMax] = {
44100, 48000, 192000
};
//----------------------------------------------------------------------------
// filter
//----------------------------------------------------------------------------
static const int AudioNormSamples = 4096; ///< number of samples
#define AudioNormMaxIndex 128 ///< number of average values
/// average of n last sample blocks
static uint32_t AudioNormAverage[AudioNormMaxIndex];
static int AudioNormIndex; ///< index into average table
static int AudioNormReady; ///< index counter
static int AudioNormCounter; ///< sample counter
/**
** Audio normalizer.
**
** @param samples sample buffer
** @param count number of bytes in sample buffer
*/
static void AudioNormalizer(int16_t * samples, int count)
{
int i;
int l;
int factor;
int16_t *data;
// average samples
l = count / AudioBytesProSample;
data = samples;
do {
uint32_t avg;
int n = l;
if (AudioNormCounter + n > AudioNormSamples) {
n = AudioNormSamples - AudioNormCounter;
}
avg = AudioNormAverage[AudioNormIndex];
for (i = 0; i < n; ++i) {
int t;
t = data[i];
avg += (t * t) / AudioNormSamples;
}
AudioNormAverage[AudioNormIndex] = avg;
AudioNormCounter += n;
if (AudioNormCounter >= AudioNormSamples) {
if (AudioNormReady < AudioNormMaxIndex) {
AudioNormReady++;
} else {
avg = 0;
for (i = 0; i < AudioNormMaxIndex; ++i) {
avg += AudioNormAverage[i] / AudioNormMaxIndex;
}
// calculate normalize factor
if (avg > 0) {
factor = ((INT16_MAX / 8) * 1000U) / (uint32_t) sqrt(avg);
// smooth normalize
AudioNormalizeFactor = (AudioNormalizeFactor * 500 + factor * 500) / 1000;
if (AudioNormalizeFactor < AudioMinNormalize) {
AudioNormalizeFactor = AudioMinNormalize;
}
if (AudioNormalizeFactor > AudioMaxNormalize) {
AudioNormalizeFactor = AudioMaxNormalize;
}
} else {
factor = 1000;
}
Debug6("audio/normalize: avg %8d, fac=%6.3f, norm=%6.3f", avg, factor / 1000.0,
AudioNormalizeFactor / 1000.0);
}
AudioNormIndex = (AudioNormIndex + 1) % AudioNormMaxIndex;
AudioNormCounter = 0;
AudioNormAverage[AudioNormIndex] = 0U;
}
data += n;
l -= n;
} while (l > 0);
// apply normalize factor
for (i = 0; i < count / AudioBytesProSample; ++i) {
int t;
t = (samples[i] * AudioNormalizeFactor) / 1000;
if (t < INT16_MIN) {
t = INT16_MIN;
} else if (t > INT16_MAX) {
t = INT16_MAX;
}
samples[i] = t;
}
}
/**
** Reset normalizer.
*/
static void AudioResetNormalizer(void)
{
int i;
AudioNormCounter = 0;
AudioNormReady = 0;
for (i = 0; i < AudioNormMaxIndex; ++i) {
AudioNormAverage[i] = 0U;
}
AudioNormalizeFactor = 1000;
}
/**
** Audio compression.
**
** @param samples sample buffer
** @param count number of bytes in sample buffer
*/
static void AudioCompressor(int16_t * samples, int count)
{
// find loudest sample
int max_sample = 0;
for (int i = 0; i < count / AudioBytesProSample; ++i) {
int t = abs(samples[i]);
if (t > max_sample) {
max_sample = t;
}
}
// calculate compression factor
if (max_sample > 0) {
int factor = (INT16_MAX * 1000) / max_sample;
// smooth compression (FIXME: make configurable?)
AudioCompressionFactor = (AudioCompressionFactor * 950 + factor * 50) / 1000;
if (AudioCompressionFactor > factor) {
AudioCompressionFactor = factor; // no clipping
}
if (AudioCompressionFactor > AudioMaxCompression) {
AudioCompressionFactor = AudioMaxCompression;
}
Debug6("audio/compress: max %5d, fac=%6.3f, com=%6.3f", max_sample, factor / 1000.0,
AudioCompressionFactor / 1000.0);
// apply compression factor
for (int i = 0; i < count / AudioBytesProSample; ++i) {
int t;
t = (samples[i] * AudioCompressionFactor) / 1000;
if (t < INT16_MIN) {
t = INT16_MIN;
} else if (t > INT16_MAX) {
t = INT16_MAX;
}
samples[i] = t;
}
}
}
/**
** Reset compressor.
*/
static void AudioResetCompressor(void)
{
AudioCompressionFactor = 2000;
if (AudioCompressionFactor > AudioMaxCompression) {
AudioCompressionFactor = AudioMaxCompression;
}
}
/**
** Audio software amplifier.
**
** @param samples sample buffer
** @param count number of bytes in sample buffer
**
** @todo FIXME: this does hard clipping
*/
static void AudioSoftAmplifier(int16_t * samples, int count)
{
int i;
// silence
if (AudioMute || !AudioAmplifier) {
memset(samples, 0, count);
return;
}
for (i = 0; i < count / AudioBytesProSample; ++i) {
int t;
t = (samples[i] * AudioAmplifier) / 1000;
if (t < INT16_MIN) {
t = INT16_MIN;
} else if (t > INT16_MAX) {
t = INT16_MAX;
}
samples[i] = t;
}
}
/**
** Upmix mono to stereo.
**
** @param in input sample buffer
** @param frames number of frames in sample buffer
** @param out output sample buffer
*/
static void AudioMono2Stereo(const int16_t * in, int frames, int16_t * out)
{
int i;
for (i = 0; i < frames; ++i) {
int t;
t = in[i];
out[i * 2 + 0] = t;
out[i * 2 + 1] = t;
}
}
/**
** Downmix stereo to mono.
**
** @param in input sample buffer
** @param frames number of frames in sample buffer
** @param out output sample buffer
*/
static void AudioStereo2Mono(const int16_t * in, int frames, int16_t * out)
{
int i;
for (i = 0; i < frames; i += 2) {
out[i / 2] = (in[i + 0] + in[i + 1]) / 2;
}
}
/**
** Downmix surround to stereo.
**
** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
**
** @param in input sample buffer
** @param in_chan nr. of input channels
** @param frames number of frames in sample buffer
** @param out output sample buffer
*/
static void AudioSurround2Stereo(const int16_t * in, int in_chan, int frames, int16_t * out)
{
while (frames--) {
int l;
int r;
switch (in_chan) {
case 3: // stereo or surround? =>stereo
l = in[0] * 600; // L
r = in[1] * 600; // R
l += in[2] * 400; // C
r += in[2] * 400;
break;
case 4: // quad or surround? =>quad
l = in[0] * 600; // L
r = in[1] * 600; // R
l += in[2] * 400; // Ls
r += in[3] * 400; // Rs
break;
case 5: // 5.0
l = in[0] * 500; // L
r = in[1] * 500; // R
l += in[2] * 200; // Ls
r += in[3] * 200; // Rs
l += in[4] * 300; // C
r += in[4] * 300;
break;
case 6: // 5.1
l = in[0] * 400; // L
r = in[1] * 400; // R
l += in[2] * 200; // Ls
r += in[3] * 200; // Rs
l += in[4] * 300; // C
r += in[4] * 300;
l += in[5] * 100; // LFE
r += in[5] * 100;
break;
case 7: // 7.0
l = in[0] * 400; // L
r = in[1] * 400; // R
l += in[2] * 200; // Ls
r += in[3] * 200; // Rs
l += in[4] * 300; // C
r += in[4] * 300;
l += in[5] * 100; // RL
r += in[6] * 100; // RR
break;
case 8: // 7.1
l = in[0] * 400; // L
r = in[1] * 400; // R
l += in[2] * 150; // Ls
r += in[3] * 150; // Rs
l += in[4] * 250; // C
r += in[4] * 250;
l += in[5] * 100; // LFE
r += in[5] * 100;
l += in[6] * 100; // RL
r += in[7] * 100; // RR
break;
default:
abort();
}
in += in_chan;
out[0] = l / 1000;
out[1] = r / 1000;
out += 2;
}
}
/**
** Upmix @a in_chan channels to @a out_chan.
**
** @param in input sample buffer
** @param in_chan nr. of input channels
** @param frames number of frames in sample buffer
** @param out output sample buffer
** @param out_chan nr. of output channels
*/
static void AudioUpmix(const int16_t * in, int in_chan, int frames, int16_t * out, int out_chan)
{
while (frames--) {
int i;
for (i = 0; i < in_chan; ++i) { // copy existing channels
*out++ = *in++;
}
for (; i < out_chan; ++i) { // silents missing channels
*out++ = 0;
}
}
}
/**
** Resample ffmpeg sample format to hardware format.
**
** FIXME: use libswresample for this and move it to codec.
** FIXME: ffmpeg to alsa conversion is already done in codec.c.
**
** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
**
** @param in input sample buffer
** @param in_chan nr. of input channels
** @param frames number of frames in sample buffer
** @param out output sample buffer
** @param out_chan nr. of output channels
*/
static void AudioResample(const int16_t * in, int in_chan, int frames, int16_t * out, int out_chan)
{
switch (in_chan * 8 + out_chan) {
case 1 * 8 + 1:
case 2 * 8 + 2:
case 3 * 8 + 3:
case 4 * 8 + 4:
case 5 * 8 + 5:
case 6 * 8 + 6:
case 7 * 8 + 7:
case 8 * 8 + 8: // input = output channels
memcpy(out, in, frames * in_chan * AudioBytesProSample);
break;
case 2 * 8 + 1:
AudioStereo2Mono(in, frames, out);
break;
case 1 * 8 + 2:
AudioMono2Stereo(in, frames, out);
break;
case 3 * 8 + 2:
case 4 * 8 + 2:
case 5 * 8 + 2:
case 6 * 8 + 2:
case 7 * 8 + 2:
case 8 * 8 + 2:
AudioSurround2Stereo(in, in_chan, frames, out);
break;
case 5 * 8 + 6:
case 3 * 8 + 8:
case 5 * 8 + 8:
case 6 * 8 + 8:
AudioUpmix(in, in_chan, frames, out, out_chan);
break;
default:
Error("audio: unsupported %d -> %d channels resample", in_chan, out_chan);
// play silence
memset(out, 0, frames * out_chan * AudioBytesProSample);
break;
}
}
//----------------------------------------------------------------------------
// ring buffer
//----------------------------------------------------------------------------
#define AUDIO_RING_MAX 8 ///< number of audio ring buffers
/**
** Audio ring buffer.
*/
typedef struct _audio_ring_ring_
{
char FlushBuffers; ///< flag: flush buffers
char Passthrough; ///< flag: use pass-through (AC-3, ...)
int16_t PacketSize; ///< packet size
unsigned HwSampleRate; ///< hardware sample rate in Hz
unsigned HwChannels; ///< hardware number of channels
unsigned InSampleRate; ///< input sample rate in Hz
unsigned InChannels; ///< input number of channels
int64_t PTS; ///< pts clock
RingBuffer *RingBuffer; ///< sample ring buffer
} AudioRingRing;
/// ring of audio ring buffers
static AudioRingRing AudioRing[AUDIO_RING_MAX];
static int AudioRingWrite; ///< audio ring write pointer
static int AudioRingRead; ///< audio ring read pointer
static atomic_t AudioRingFilled; ///< how many of the ring is used
static unsigned AudioStartThreshold; ///< start play, if filled
/**
** Add sample-rate, number of channels change to ring.
**
** @param sample_rate sample-rate frequency
** @param channels number of channels
** @param passthrough use /pass-through (AC-3, ...) device
**
** @retval -1 error
** @retval 0 okay
**
** @note this function shouldn't fail. Checks are done during AudoInit.
*/
static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough)
{
unsigned u;
// search supported sample-rates
for (u = 0; u < AudioRatesMax; ++u) {
if (AudioRatesTable[u] == sample_rate) {
goto found;
}
if (AudioRatesTable[u] > sample_rate) {
break;
}
}
Error("audio: %dHz sample-rate unsupported", sample_rate);
return -1; // unsupported sample-rate
found:
if (!AudioChannelMatrix[u][channels]) {
Error("audio: %d channels unsupported", channels);
return -1; // unsupported nr. of channels
}
if (atomic_read(&AudioRingFilled) == AUDIO_RING_MAX) { // no free slot
// FIXME: can wait for ring buffer empty
Error("audio: out of ring buffers");
return -1;
}
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
AudioRing[AudioRingWrite].FlushBuffers = 0;
AudioRing[AudioRingWrite].Passthrough = passthrough;
AudioRing[AudioRingWrite].PacketSize = 0;
AudioRing[AudioRingWrite].InSampleRate = sample_rate;
AudioRing[AudioRingWrite].InChannels = channels;
AudioRing[AudioRingWrite].HwSampleRate = sample_rate;
AudioRing[AudioRingWrite].HwChannels = AudioChannelMatrix[u][channels];
AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000);
RingBufferReset(AudioRing[AudioRingWrite].RingBuffer);
Debug5("audio: %d ring buffer prepared", atomic_read(&AudioRingFilled) + 1);
atomic_inc(&AudioRingFilled);
if (AudioThread) {
// tell thread, that there is something todo
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
}
return 0;
}
/**
** Setup audio ring.
*/
static void AudioRingInit(void)
{
int i;
for (i = 0; i < AUDIO_RING_MAX; ++i) {
// ~2s 8ch 16bit
AudioRing[i].RingBuffer = RingBufferNew(AudioRingBufferSize);
}
atomic_set(&AudioRingFilled, 0);
}
/**
** Cleanup audio ring.
*/
static void AudioRingExit(void)
{
int i;
for (i = 0; i < AUDIO_RING_MAX; ++i) {
if (AudioRing[i].RingBuffer) {
RingBufferDel(AudioRing[i].RingBuffer);
AudioRing[i].RingBuffer = NULL;
}
AudioRing[i].HwSampleRate = 0; // checked for valid setup
AudioRing[i].InSampleRate = 0;
}
AudioRingRead = 0;
AudioRingWrite = 0;
}
//============================================================================
// A L S A
//============================================================================
//----------------------------------------------------------------------------
// Alsa variables
//----------------------------------------------------------------------------
static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle
static char AlsaCanPause; ///< hw supports pause
static int AlsaUseMmap; ///< use mmap
static snd_mixer_t *AlsaMixer; ///< alsa mixer handle
static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element
static int AlsaRatio; ///< internal -> mixer ratio * 1000
//----------------------------------------------------------------------------
// alsa pcm
//----------------------------------------------------------------------------
/**
** Play samples from ringbuffer.
**
** Fill the kernel buffer, as much as possible.
**
** @retval 0 ok
** @retval 1 ring buffer empty
** @retval -1 underrun error
*/
static int AlsaPlayRingbuffer(void)
{
int first;
first = 1;
for (;;) { // loop for ring buffer wrap
int avail;
int n;
int err;
int frames;
const void *p;
// how many bytes can be written?
n = snd_pcm_avail_update(AlsaPCMHandle);
if (n < 0) {
if (n == -EAGAIN) {
continue;
}
Error("audio/alsa: avail underrun error? '%s'", snd_strerror(n));
err = snd_pcm_recover(AlsaPCMHandle, n, 0);
if (err >= 0) {
continue;
}
Error("audio/alsa: snd_pcm_avail_update(): %s", snd_strerror(n));
return -1;
}
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
if (avail < 256) { // too much overhead
if (first) {
// happens with broken alsa drivers
if (AudioThread) {
if (!AudioAlsaDriverBroken) {
Error("audio/alsa: broken driver %d state '%s'", avail,
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
}
// try to recover
if (snd_pcm_state(AlsaPCMHandle)
== SND_PCM_STATE_PREPARED) {
if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) {
Error("audio/alsa: snd_pcm_start(): %s", snd_strerror(err));
}
}
usleep(5 * 1000);
}
}
Debug6("audio/alsa: break state '%s'", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
break;
}
n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
if (!n) { // ring buffer empty
if (first) { // only error on first loop
Debug6("audio/alsa: empty buffers %d", avail);
// ring buffer empty
// AlsaLowWaterMark = 1;
return 1;
}
return 0;
}
if (n < avail) { // not enough bytes in ring buffer
avail = n;
}
if (!avail) { // full or buffer empty
break;
}
// muting pass-through AC-3, can produce disturbance
if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough)) {
// FIXME: quick&dirty cast
AudioSoftAmplifier((int16_t *) p, avail);
// FIXME: if not all are written, we double amplify them
}
frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
#ifdef DEBUG
if (avail != snd_pcm_frames_to_bytes(AlsaPCMHandle, frames)) {
Error("audio/alsa: bytes lost -> out of sync");
}
#endif
for (;;) {
pthread_mutex_lock(&ReadAdvance_mutex);
if (AlsaUseMmap) {
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
if (err != frames) {
if (err < 0) {
pthread_mutex_unlock(&ReadAdvance_mutex);
if (err == -EAGAIN) {
continue;
}
Error("audio/alsa: writei underrun error? '%s'", snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
return 0;
}
Error("audio/alsa: snd_pcm_writei failed: %s", snd_strerror(err));
return -1;
}
// this could happen, if underrun happened
Error("audio/alsa: not all frames written");
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
}
break;
}
RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, avail);
pthread_mutex_unlock(&ReadAdvance_mutex);
first = 0;
}
return 0;
}
/**
** Flush alsa buffers.
*/
static void AlsaFlushBuffers(void)
{
if (AlsaPCMHandle) {
snd_pcm_state_t state;
state = snd_pcm_state(AlsaPCMHandle);
Debug5("audio/alsa: flush state %s", snd_pcm_state_name(state));
if (state != SND_PCM_STATE_OPEN) {
int err;
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
Error("audio: snd_pcm_drop(): %s", snd_strerror(err));
}
// ****ing alsa crash, when in open state here
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
Error("audio: snd_pcm_prepare(): %s", snd_strerror(err));
}
}
}
}
//----------------------------------------------------------------------------
// thread playback
//----------------------------------------------------------------------------
/**
** Alsa thread
**
** Play some samples and return.
**
** @retval -1 error
** @retval 0 underrun
** @retval 1 running
*/
static int AlsaThread(void)
{
int err;
if (!AlsaPCMHandle) {
usleep(24 * 1000);
return -1;
}
for (;;) {
if (AudioPaused) {
return 1;
}
// wait for space in kernel buffers
if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) {
Error("audio/alsa: wait underrun error? '%s'", snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
continue;
}
Error("audio/alsa: snd_pcm_wait(): %s", snd_strerror(err));
usleep(24 * 1000);
return -1;
}
break;
}
if (AudioPaused) { // timeout or some commands
return 1;
}
if ((err = AlsaPlayRingbuffer())) { // empty or error
snd_pcm_state_t state;
if (err < 0) { // underrun error
return -1;
}
state = snd_pcm_state(AlsaPCMHandle);
if (state != SND_PCM_STATE_RUNNING) {
Debug5("audio/alsa: stopping play '%s'", snd_pcm_state_name(state));
return 0;
}
usleep(24 * 1000); // let fill/empty the buffers
}
return 1;
}
//----------------------------------------------------------------------------
/**
** Open alsa pcm device.
**
** @param passthrough use pass-through (AC-3, ...) device
*/
static snd_pcm_t *AlsaOpenPCM(int passthrough)
{
const char *device;
snd_pcm_t *handle;
int err;
// &&|| hell
if (!(passthrough && ((device = AudioPassthroughDevice)
|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
if (!AudioDoingInit) { // reduce blabla during init
Info("audio/alsa: using %sdevice '%s'", passthrough ? "pass-through " : "", device);
}
// open none blocking; if device is already used, we don't want wait
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) {
Error("audio/alsa: playback open '%s' error: %s", device, snd_strerror(err));
return NULL;
}
if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
Error("audio/alsa: can't set block mode: %s", snd_strerror(err));
}
return handle;
}
/**
** Initialize alsa pcm device.
**
** @see AudioPCMDevice
*/
static void AlsaInitPCM(void)
{
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
if (!(handle = AlsaOpenPCM(0))) {
return;
}
// FIXME: pass-through and pcm out can support different features
snd_pcm_hw_params_alloca(&hw_params);
// choose all parameters
if ((err = snd_pcm_hw_params_any(handle, hw_params)) < 0) {
Error("audio: snd_pcm_hw_params_any: no configurations available: %s", snd_strerror(err));
}
AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
Info("audio/alsa: supports pause: %s", AlsaCanPause ? "yes" : "no");
AlsaPCMHandle = handle;
}
//----------------------------------------------------------------------------
// Alsa Mixer
//----------------------------------------------------------------------------
/**
** Set alsa mixer volume (0-1000)
**
** @param volume volume (0 .. 1000)
*/
static void AlsaSetVolume(int volume)
{
if (AlsaMixer && AlsaMixerElem) {
int v = (volume * AlsaRatio) / (1000 * 1000);
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 0, v);
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 1, v);
}
}
/**
** Initialize alsa mixer.
*/
static void AlsaInitMixer(void)
{
const char *device;
const char *channel;
snd_mixer_t *alsa_mixer;
snd_mixer_elem_t *alsa_mixer_elem;
long alsa_mixer_elem_min;
long alsa_mixer_elem_max;
if (!(device = AudioMixerDevice)) {
if (!(device = getenv("ALSA_MIXER"))) {
device = "default";
}
}
if (!(channel = AudioMixerChannel)) {
if (!(channel = getenv("ALSA_MIXER_CHANNEL"))) {
channel = "PCM";
}
}
Debug5("audio/alsa: mixer %s - %s open", device, channel);
snd_mixer_open(&alsa_mixer, 0);
if (alsa_mixer && snd_mixer_attach(alsa_mixer, device) >= 0
&& snd_mixer_selem_register(alsa_mixer, NULL, NULL) >= 0 && snd_mixer_load(alsa_mixer) >= 0) {
const char *const alsa_mixer_elem_name = channel;
alsa_mixer_elem = snd_mixer_first_elem(alsa_mixer);