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audio.c
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///
/// @file audio.c @brief Audio module
///
/// Copyright (c) 2009 - 2014 by Johns. All Rights Reserved.
/// Copyright (c) 2018 by zille. All Rights Reserved.
///
/// Contributor(s):
///
/// License: AGPLv3
///
/// This program is free software: you can redistribute it and/or modify
/// it under the terms of the GNU Affero General Public License as
/// published by the Free Software Foundation, either version 3 of the
/// License.
///
/// This program is distributed in the hope that it will be useful,
/// but WITHOUT ANY WARRANTY; without even the implied warranty of
/// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
/// GNU Affero General Public License for more details.
///
/// $Id$
//////////////////////////////////////////////////////////////////////////////
///
/// @defgroup Audio The audio module.
///
/// 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!
///
/// @todo FIXME: there can be problems with little/big endian.
///
#include <stdint.h>
#include <math.h>
#include <libintl.h>
#define _(str) gettext(str) ///< gettext shortcut
#define _N(str) str ///< gettext_noop shortcut
#include <alsa/asoundlib.h>
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavfilter/avfilter.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include "iatomic.h" // portable atomic_t
#include "ringbuffer.h"
#include "misc.h"
#include "audio.h"
#include "video.h"
#include "codec.h"
#include "softhddev.h"
//----------------------------------------------------------------------------
// Defines
//----------------------------------------------------------------------------
#define MIN_AUDIO_BUFFER 450 ///< minimal output buffer in ms
//----------------------------------------------------------------------------
// Variables
//----------------------------------------------------------------------------
static const char *AudioPCMDevice; ///< PCM device name
static const char *AudioPassthroughDevice; ///< Passthrough device name
static char AudioAppendAES; ///< flag automatic append AES
static const char *AudioMixerDevice; ///< mixer device name
static const char *AudioMixerChannel; ///< mixer channel name
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; ///< audio buffer time in ms
static pthread_t AudioThread; ///< audio play thread
static pthread_mutex_t AudioRbMutex; ///< audio condition mutex
static pthread_mutex_t AudioStartMutex; ///< audio condition mutex
static pthread_cond_t AudioStartCond; ///< condition variable
static char AudioThreadStop; ///< stop audio thread
static char AlsaPlayerStop; ///< 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)
static int AudioDownMix;
extern int VideoAudioDelay; ///< import audio/video delay
/// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
// 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
// Filter variables
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
AVFilterGraph *filter_graph;
AVFilterContext *abuffersrc_ctx, *abuffersink_ctx;
int FilterInit;
float AudioEqBand[18];
int AudioEq;
int Filterchanged;
// ring buffer variables
char Passthrough; ///< flag: use pass-through (AC-3, ...)
unsigned int HwSampleRate; ///< hardware sample rate in Hz
unsigned int HwChannels; ///< hardware number of channels
AVRational *timebase; ///< pointer to AVCodecContext pkts_timebase
int64_t PTS; ///< pts clock
RingBuffer *AudioRingBuffer; ///< sample ring buffer
static unsigned AudioStartThreshold; ///< start play, if filled
static int AlsaSetup(int channels, int sample_rate, int passthrough);
//----------------------------------------------------------------------------
// Filter
//----------------------------------------------------------------------------
/**
** Reorder audio frame.
**
** 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 buf[IN,OUT] sample buffer
** @param size size of sample buffer in bytes
** @param channels number of channels interleaved in sample buffer
*/
static void AudioReorderAudioFrame(int16_t * buf, int size, int channels)
{
int i;
int c;
int ls;
int rs;
int lfe;
switch (channels) {
case 5:
size /= 2;
for (i = 0; i < size; i += 5) {
c = buf[i + 2];
ls = buf[i + 3];
rs = buf[i + 4];
buf[i + 2] = ls;
buf[i + 3] = rs;
buf[i + 4] = c;
}
break;
case 6:
size /= 2;
for (i = 0; i < size; i += 6) {
c = buf[i + 2];
lfe = buf[i + 3];
// ls = buf[i + 4]; tested from jsffm
// rs = buf[i + 5];
// buf[i + 2] = ls;
// buf[i + 3] = rs;
buf[i + 2] = lfe;
buf[i + 3] = c;
// buf[i + 4] = c;
// buf[i + 5] = lfe;
}
break;
case 8:
size /= 2;
for (i = 0; i < size; i += 8) {
c = buf[i + 2];
lfe = buf[i + 3];
ls = buf[i + 4];
rs = buf[i + 5];
buf[i + 2] = ls;
buf[i + 3] = rs;
buf[i + 4] = c;
buf[i + 5] = lfe;
}
break;
}
}
/**
** 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 n;
uint32_t avg;
int factor;
int16_t *data;
// average samples
l = count / AudioBytesProSample;
data = samples;
do {
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;
}
Debug2(L_SOUND, "audio/noramlize: 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)
{
int max_sample;
int i;
int factor;
// find loudest sample
max_sample = 0;
for (i = 0; i < count / AudioBytesProSample; ++i) {
int t;
t = abs(samples[i]);
if (t > max_sample) {
max_sample = t;
}
}
// calculate compression factor
if (max_sample > 0) {
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;
}
} else {
return; // silent nothing todo
}
Debug2(L_SOUND, "audio/compress: max %5d, fac=%6.3f, com=%6.3f", max_sample,
factor / 1000.0, AudioCompressionFactor / 1000.0);
// apply compression factor
for (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;
}
}
/**
** Set filter bands.
**
** @param band setting frequenz bands
** @param onoff set using equalizer
*/
void AudioSetEq(int band[17], int onoff)
{
int i;
/* Debug2(L_SOUND, "AudioSetEq %i %i %i %i %i %i %i %i %i %i %i %i %i %i %i %i %i %i onoff %d",
band[0], band[1], band[2], band[3], band[4], band[5], band[6], band[7],
band[8], band[9], band[10], band[11], band[12], band[13], band[14],
band[15], band[16], band[17], onoff);*/
for (i = 0; i < 18; i++) {
switch (band[i]) {
case 1:
AudioEqBand[i] = 1.5;
break;
case 0:
AudioEqBand[i] = 1;
break;
case -1:
AudioEqBand[i] = 0.95;
break;
case -2:
AudioEqBand[i] = 0.9;
break;
case -3:
AudioEqBand[i] = 0.85;
break;
case -4:
AudioEqBand[i] = 0.8;
break;
case -5:
AudioEqBand[i] = 0.75;
break;
case -6:
AudioEqBand[i] = 0.7;
break;
case -7:
AudioEqBand[i] = 0.65;
break;
case -8:
AudioEqBand[i] = 0.6;
break;
case -9:
AudioEqBand[i] = 0.55;
break;
case -10:
AudioEqBand[i] = 0.5;
break;
case -11:
AudioEqBand[i] = 0.45;
break;
case -12:
AudioEqBand[i] = 0.4;
break;
case -13:
AudioEqBand[i] = 0.35;
break;
case -14:
AudioEqBand[i] = 0.3;
break;
case -15:
AudioEqBand[i] = 0.25;
break;
}
}
Filterchanged = 1;
AudioEq = onoff;
}
/**
** Filter init.
**
** @retval 0 everything ok
** @retval 1 didn't support channels, CodecDownmix set > scrap this frame, test next
** @retval -1 something gone wrong
*/
static int AudioFilterInit(AVCodecContext *AudioCtx)
{
const AVFilter *abuffer;
AVFilterContext *filter_ctx[3];
const AVFilter *eq;
const AVFilter *aformat;
const AVFilter *abuffersink;
char ch_layout[64];
char options_str[1024];
int err, i, n_filter = 0;
// Before filter init set HW parameter.
if (AudioCtx->sample_rate != (int)HwSampleRate ||
(AudioCtx->ch_layout.nb_channels != (int)HwChannels &&
!(AudioDownMix && HwChannels == 2))) {
err = AlsaSetup(AudioCtx->ch_layout.nb_channels, AudioCtx->sample_rate, 0);
if (err)
return err;
}
timebase = &AudioCtx->pkt_timebase;
#if LIBAVFILTER_VERSION_INT < AV_VERSION_INT(7,16,100)
avfilter_register_all();
#endif
if (!(filter_graph = avfilter_graph_alloc()))
Error("Unable to create filter graph.");
// input buffer
if (!(abuffer = avfilter_get_by_name("abuffer")))
Warning("AudioFilterInit: Could not find the abuffer filter.");
if (!(abuffersrc_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src")))
Warning("AudioFilterInit: Could not allocate the abuffersrc_ctx instance.");
av_channel_layout_describe(&AudioCtx->ch_layout, ch_layout, sizeof(ch_layout));
Debug2(L_SOUND, "AudioFilterInit: IN ch_layout %s sample_fmt %s sample_rate %d channels %d",
ch_layout, av_get_sample_fmt_name(AudioCtx->sample_fmt), AudioCtx->sample_rate, AudioCtx->ch_layout.nb_channels);
av_opt_set (abuffersrc_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffersrc_ctx, "sample_fmt", av_get_sample_fmt_name(AudioCtx->sample_fmt), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffersrc_ctx, "time_base", (AVRational){ 1, AudioCtx->sample_rate }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffersrc_ctx, "sample_rate", AudioCtx->sample_rate, AV_OPT_SEARCH_CHILDREN);
// av_opt_set_int(abuffersrc_ctx, "channel_counts", AudioCtx->channels, AV_OPT_SEARCH_CHILDREN);
// initialize the filter with NULL options, set all options above.
if (avfilter_init_str(abuffersrc_ctx, NULL) < 0)
Warning("AudioFilterInit: Could not initialize the abuffer filter.");
if (AudioEq) {
// superequalizer
if (!(eq = avfilter_get_by_name("superequalizer")))
Warning("AudioFilterInit: Could not find the superequalizer filter.");
if (!(filter_ctx[n_filter] = avfilter_graph_alloc_filter(filter_graph, eq, "superequalizer")))
Warning("AudioFilterInit: Could not allocate the superequalizer instance.");
snprintf(options_str, sizeof(options_str),"1b=%.2f:2b=%.2f:3b=%.2f:4b=%.2f:5b=%.2f"
":6b=%.2f:7b=%.2f:8b=%.2f:9b=%.2f:10b=%.2f:11b=%.2f:12b=%.2f:13b=%.2f:14b=%.2f:"
"15b=%.2f:16b=%.2f:17b=%.2f:18b=%.2f ", AudioEqBand[0], AudioEqBand[1],
AudioEqBand[2], AudioEqBand[3], AudioEqBand[4], AudioEqBand[5],
AudioEqBand[6], AudioEqBand[7], AudioEqBand[8], AudioEqBand[9],
AudioEqBand[10], AudioEqBand[11], AudioEqBand[12], AudioEqBand[13],
AudioEqBand[14], AudioEqBand[15], AudioEqBand[16], AudioEqBand[17]);
if (avfilter_init_str(filter_ctx[n_filter], options_str) < 0)
Warning("AudioFilterInit: Could not initialize the superequalizer filter.");
n_filter++;
}
// aformat
AVChannelLayout channel_layout;
av_channel_layout_default(&channel_layout, HwChannels);
av_channel_layout_describe(&channel_layout, ch_layout, sizeof(ch_layout));
av_channel_layout_uninit(&channel_layout);
// should use IN layout if more then 2 ch!?
Debug2(L_SOUND, "AudioFilterInit: OUT AudioDownMix %d HwChannels %d HwSampleRate %d ch_layout %s bytes_per_sample %d",
AudioDownMix, HwChannels, HwSampleRate,
ch_layout, av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
if (!(aformat = avfilter_get_by_name("aformat")))
Warning("AudioFilterInit: Could not find the aformat filter.");
if (!(filter_ctx[n_filter] = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat")))
Warning("AudioFilterInit: Could not allocate the aformat instance.");
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=%s",
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), HwSampleRate, ch_layout);
if (avfilter_init_str(filter_ctx[n_filter], options_str) < 0)
Warning("AudioFilterInit: Could not initialize the aformat filter.");
n_filter++;
// abuffersink
if (!(abuffersink = avfilter_get_by_name("abuffersink")))
Warning("AudioFilterInit: Could not find the abuffersink filter.");
if (!(filter_ctx[n_filter] = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink")))
Warning("AudioFilterInit: Could not allocate the abuffersink instance.");
if (avfilter_init_str(filter_ctx[n_filter], NULL) < 0)
Warning("AudioFilterInit: Could not initialize the abuffersink instance.");
n_filter++;
// Connect the filters
for (i = 0; i < n_filter; i++) {
if (i == 0) {
err = avfilter_link(abuffersrc_ctx, 0, filter_ctx[i], 0);
} else {
err = avfilter_link(filter_ctx[i - 1], 0, filter_ctx[i], 0);
}
}
if (err < 0)
Warning("AudioFilterInit: Error connecting audio filters");
// Configure the graph.
if (avfilter_graph_config(filter_graph, NULL) < 0)
Warning("AudioFilterInit: Error configuring the audio filter graph");
abuffersink_ctx = filter_ctx[n_filter - 1];
Filterchanged = 0;
FilterInit = 1;
return 0;
}
//----------------------------------------------------------------------------
// ring buffer
//----------------------------------------------------------------------------
/**
** Setup audio ring.
*/
static void AudioRingInit(void)
{
// ~2s 8ch 16bit
AudioRingBuffer = RingBufferNew(AudioRingBufferSize);
}
/**
** Cleanup audio ring.
*/
static void AudioRingExit(void)
{
if (AudioRingBuffer) {
RingBufferDel(AudioRingBuffer);
AudioRingBuffer = NULL;
}
HwSampleRate = 0; // checked for valid setup
}
//============================================================================
// A L S A
//============================================================================
//----------------------------------------------------------------------------
// alsa pcm
//----------------------------------------------------------------------------
/**
** xrun recovery
**/
static void xrun_recovery(void)
{
int err;
snd_pcm_state_t state;
err = snd_pcm_prepare(AlsaPCMHandle);
if (err < 0) {
state = snd_pcm_state(AlsaPCMHandle);
Error("audio/alsa: Can't recovery from xrun: %s pcm state: %s",
snd_strerror(err), snd_pcm_state_name(state));
}
}
/**
** Flush alsa buffers.
*/
static void AlsaFlushBuffers(void)
{
int err;
snd_pcm_state_t state;
Debug2(L_SOUND, "audio: AlsaFlushBuffers: AlsaFlushBuffers");
state = snd_pcm_state(AlsaPCMHandle);
if (state != SND_PCM_STATE_OPEN) {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0)
Error("audio: AlsaFlushBuffers: snd_pcm_drop(): %s", snd_strerror(err));
// ****ing alsa crash, when in open state here
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0)
Error("audio: AlsaFlushBuffers: snd_pcm_prepare(): %s", snd_strerror(err));
state = snd_pcm_state(AlsaPCMHandle);
Debug2(L_SOUND, "audio: AlsaFlushBuffers: pcm state %s", snd_pcm_state_name(state));
}
RingBufferReset(AudioRingBuffer);
AudioSkip = 0;
PTS = AV_NOPTS_VALUE;
AudioVideoIsReady = 0;
}
//----------------------------------------------------------------------------
// thread playback
//----------------------------------------------------------------------------
/**
** Alsa thread
**
** Play some samples and return.
**
** @retval -1 error
** @retval 1 running
*/
static int AlsaPlayer(void)
{
for (;;) {
int avail;
int n;
int err;
int frames;
const void *p;
if (AudioPaused || AlsaPlayerStop) {
return 1;
}
// wait for space in kernel buffers
if ((err = snd_pcm_wait(AlsaPCMHandle, 150)) < 0) {
// Error("AlsaPlayer: snd_pcm_wait error? '%s'", snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
// Error("AlsaPlayer: snd_pcm_wait error: snd_pcm_recover %s", snd_strerror(err));
}
if (AudioPaused || AlsaPlayerStop) {
return 1;
}
// how many bytes can be written?
// n = snd_pcm_avail_update(AlsaPCMHandle);
n = snd_pcm_avail(AlsaPCMHandle);
if (n < 0) {
if (n == -EAGAIN) {
continue;
}
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
Debug2(L_SOUND, "audio/alsa: break state '%s'",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
break;
}
n = RingBufferGetReadPointer(AudioRingBuffer, &p);
if (!n) { // ring buffer empty
Warning("AlsaPlayer: ring buffer empty Videopkts: %d",
VideoGetPackets());
}
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
&& !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);
pthread_mutex_lock(&AudioRbMutex);
if (AlsaUseMmap) {
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
RingBufferReadAdvance(AudioRingBuffer, avail);
pthread_mutex_unlock(&AudioRbMutex);
if (err != frames) {
if (err < 0) {
if (err == -EAGAIN) {
continue;
}
Warning("audio/alsa: writei underrun error? '%s'",
snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
continue;
}
Error("audio/alsa: snd_pcm_writei failed: %s",
snd_strerror(err));
return -1;
}
// this could happen, if underrun happened
Warning("audio/alsa: not all frames written");
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
break;
}
}
return 0;
}
//----------------------------------------------------------------------------
/**
** Open alsa pcm device.
*/
static void AlsaInitPCM(void)
{
const char *device;
int err;
if (!(Passthrough && ((device = AudioPassthroughDevice)
|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
Info("audio/alsa: using %sdevice '%s'",
Passthrough ? "pass-through " : "", device);
#if 0
// for AC3 pass-through try to set the non-audio bit, use AES0=6 to set spdif in raw mode
if (Passthrough && AudioAppendAES) {
// FIXME: not yet finished
char *buf;
const char *s;
int n;
n = strlen(device);
buf = alloca(n + sizeof(":AES0=6") + 1);
strcpy(buf, device);
if (!(s = strchr(buf, ':'))) {
// no alsa parameters
strcpy(buf + n, ":AES=6");
}
Debug2(L_SOUND, "audio/alsa: try '%s'", buf);
}
#endif
// open none blocking; if device is already used, we don't want wait
if ((err = snd_pcm_open(&AlsaPCMHandle, device, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK)) < 0) {
Fatal("audio: AlsaOpenPCM: playback open '%s' error: %s", device,
snd_strerror(err));
}
if ((err = snd_pcm_nonblock(AlsaPCMHandle, 0)) < 0) {
Error("audio/alsa: can't set block mode: %s", snd_strerror(err));
}
}
//----------------------------------------------------------------------------
// Alsa Mixer
//----------------------------------------------------------------------------
/**
** Set alsa mixer volume (0-1000)
**
** @param volume volume (0 .. 1000)
*/
static void AlsaSetVolume(int volume)
{
int v;
if (AlsaMixer && AlsaMixerElem) {
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";
}
}
Debug2(L_SOUND, "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);
while (alsa_mixer_elem) {
const char *name;
name = snd_mixer_selem_get_name(alsa_mixer_elem);
if (!strcasecmp(name, alsa_mixer_elem_name)) {
snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
&alsa_mixer_elem_min, &alsa_mixer_elem_max);
AlsaRatio = 1000 * (alsa_mixer_elem_max - alsa_mixer_elem_min);
Debug2(L_SOUND, "audio/alsa: PCM mixer found %ld - %ld ratio %d",
alsa_mixer_elem_min, alsa_mixer_elem_max, AlsaRatio);
break;
}
alsa_mixer_elem = snd_mixer_elem_next(alsa_mixer_elem);
}
AlsaMixer = alsa_mixer;
AlsaMixerElem = alsa_mixer_elem;
} else {
Error("audio/alsa: can't open mixer '%s'", device);
}
}
//----------------------------------------------------------------------------
// Alsa API
//----------------------------------------------------------------------------
/**
** Setup alsa audio for requested format.
**
** @param channels Channels requested
** @param sample_rate SampleRate requested
** @param passthrough use pass-through (AC-3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support hw channels, CodecDownmix set > retest
** @retval -1 something gone wrong
**
** @todo FIXME: remove pointer for freq + channels
*/
static int AlsaSetup(int channels, int sample_rate, __attribute__ ((unused)) int passthrough)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_state_t state;
int err;
int delay;
unsigned buffer_time = 100000; // 100ms
AudioDownMix = 0;
if (AudioRunning) {
Debug2(L_SOUND, "AlsaSetup: Audio is Running => AudioFlushBuffers");
AudioFlushBuffers();
}
state = snd_pcm_state(AlsaPCMHandle);
if (state == SND_PCM_STATE_XRUN) {
Error("audio/AlsaSetup: recover from xrun pcm state: %s",
snd_pcm_state_name(state));
xrun_recovery();
}
snd_pcm_hw_params_alloca(&hwparams);
if ((err = snd_pcm_hw_params_any(AlsaPCMHandle, hwparams)) < 0) {
Error("AlsaSetup: Read HW config failed! %s", snd_strerror(err));
return -1;
}
if (!snd_pcm_hw_params_test_access(AlsaPCMHandle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) {
AlsaUseMmap = 1;
}
HwSampleRate = sample_rate;
if ((err = snd_pcm_hw_params_set_rate_near(AlsaPCMHandle, hwparams, &HwSampleRate, 0) < 0)) {
Error("AlsaSetup: SampleRate %d not supported! %s",
sample_rate, snd_strerror(err));
return -1;
}
if ((int)HwSampleRate != sample_rate) {
Debug2(L_SOUND, "AlsaSetup: sample_rate %d HwSampleRate %d",
sample_rate, HwSampleRate);
}
HwChannels = channels;
if ((err = snd_pcm_hw_params_set_channels_near(AlsaPCMHandle, hwparams, &HwChannels)) < 0) {
Warning("AlsaSetup: %d channels not supported! %s",
HwChannels, snd_strerror(err));
}
if ((int)HwChannels != channels) {
AudioDownMix = 1;
}
if ((err = snd_pcm_hw_params_set_buffer_time_near(AlsaPCMHandle, hwparams, &buffer_time, NULL)) < 0) {
Warning("AlsaSetup: buffer_time %d not supported! %s",