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vdr-plugin-softhdcuvid/audio.c
jojo61 a2b52bb804 Fix jump in recordings 
Fix in CUVID Bufferhandling
2019-10-11 11:47:11 +02:00

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///
/// @file audio.c @brief Audio module
///
/// Copyright (c) 2009 - 2014 by Johns. 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: 77fa65030b179e78c13d0bf69a7cc417dae89e1a $
//////////////////////////////////////////////////////////////////////////////
///
/// @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!
///
/// OSS PCM/Mixer api is supported.
/// @see http://manuals.opensound.com/developer/
///
///
/// @todo FIXME: there can be problems with little/big endian.
///
#ifdef DEBUG
#undef DEBUG
#endif
//#define USE_ALSA ///< enable alsa support
//#define USE_OSS ///< enable OSS support
#define USE_AUDIO_THREAD ///< use thread for audio playback
#define USE_AUDIO_MIXER ///< use audio module mixer
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <inttypes.h>
#include <string.h>
#include <math.h>
#include <sys/prctl.h>
#include <libintl.h>
#define _(str) gettext(str) ///< gettext shortcut
#define _N(str) str ///< gettext_noop shortcut
#ifdef USE_ALSA
#include <alsa/asoundlib.h>
#endif
#ifdef USE_OSS
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
// SNDCTL_DSP_HALT_OUTPUT compatibility
#ifndef SNDCTL_DSP_HALT_OUTPUT
# if defined(SNDCTL_DSP_RESET_OUTPUT)
# define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET_OUTPUT
# elif defined(SNDCTL_DSP_RESET)
# define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET
# else
# error "No valid SNDCTL_DSP_HALT_OUTPUT found."
# endif
#endif
#include <poll.h>
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#endif
#ifdef USE_AUDIO_THREAD
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#ifndef HAVE_PTHREAD_NAME
/// only available with newer glibc
#define pthread_setname_np(thread, name)
#endif
#endif
#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 char AudioAppendAES; ///< flag automatic append AES
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
int AudioDelay; /// delay audio to sync to video
static const int AudioBytesProSample = 2; ///< number of bytes per sample
static int AudioBufferTime = 336; ///< audio buffer time in ms
#ifdef USE_AUDIO_THREAD
static pthread_t AudioThread; ///< audio play thread
static pthread_mutex_t AudioMutex; ///< audio condition mutex
static pthread_cond_t AudioStartCond; ///< condition variable
static char AudioThreadStop; ///< stop audio thread
#else
static const int AudioThread; ///< dummy audio thread
#endif
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
/// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
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 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;
}
Debug(4, "audio/noramlize: avg %8d, fac=%6.3f, norm=%6.3f\n",
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
}
Debug(4, "audio/compress: max %5d, fac=%6.3f, com=%6.3f\n", 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;
}
}
#ifdef USE_AUDIO_MIXER
/**
** 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\n", in_chan,
out_chan);
// play silence
memset(out, 0, frames * out_chan * AudioBytesProSample);
break;
}
}
#endif
//----------------------------------------------------------------------------
// 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\n"), sample_rate);
return -1; // unsupported sample-rate
found:
if (!AudioChannelMatrix[u][channels]) {
Error(_("audio: %d channels unsupported\n"), 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\n"));
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);
Debug(3, "audio: %d ring buffer prepared\n",
atomic_read(&AudioRingFilled) + 1);
atomic_inc(&AudioRingFilled);
#ifdef USE_AUDIO_THREAD
if (AudioThread) {
// tell thread, that there is something todo
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
Debug(3,"Start on AudioRingAdd\n");
}
#endif
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;
}
#ifdef USE_ALSA
//============================================================================
// 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;
}
Warning(_("audio/alsa: avail underrun error? '%s'\n"),
snd_strerror(n));
err = snd_pcm_recover(AlsaPCMHandle, n, 0);
if (err >= 0) {
continue;
}
Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"),
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'\n"),
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\n"),
snd_strerror(err));
}
}
usleep(5 * 1000);
}
}
Debug(4, "audio/alsa: break state '%s'\n",
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
Debug(4, "audio/alsa: empty buffers %d\n", 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\n"));
}
#endif
for (;;) {
if (AlsaUseMmap) {
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
//Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames);
if (err != frames) {
if (err < 0) {
if (err == -EAGAIN) {
continue;
}
/*
if (err == -EBADFD) {
goto again;
}
*/
Warning(_("audio/alsa: writei underrun error? '%s'\n"),
snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
continue;
}
Error(_("audio/alsa: snd_pcm_writei failed: %s\n"),
snd_strerror(err));
return -1;
}
// this could happen, if underrun happened
Warning(_("audio/alsa: not all frames written\n"));
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
}
break;
}
RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, avail);
first = 0;
}
return 0;
}
/**
** Flush alsa buffers.
*/
static void AlsaFlushBuffers(void)
{
if (AlsaPCMHandle) {
int err;
snd_pcm_state_t state;
state = snd_pcm_state(AlsaPCMHandle);
Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state));
if (state != SND_PCM_STATE_OPEN) {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
}
// ****ing alsa crash, when in open state here
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
}
}
}
}
#ifdef USE_AUDIO_THREAD
//----------------------------------------------------------------------------
// 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) {
Warning(_("audio/alsa: wait underrun error? '%s'\n"),
snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
continue;
}
Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
usleep(24 * 1000);
return -1;
}
break;
}
if (!err || 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) {
Debug(3, "audio/alsa: stopping play '%s'\n",
snd_pcm_state_name(state));
return 0;
}
usleep(24 * 1000); // let fill/empty the buffers
}
return 1;
}
#endif
//----------------------------------------------------------------------------
/**
** 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'\n"), passthrough ? "pass-through " : "", device);
}
//
// for AC3 pass-through try to set the non-audio bit, use AES0=6
//
if (passthrough && AudioAppendAES) {
#if 0
// 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");
}
Debug(3, "audio/alsa: try '%s'\n", buf);
#endif
}
// 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\n"), device, snd_strerror(err));
return NULL;
}
if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
Error(_("audio/alsa: can't set block mode: %s\n"), 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\n"),
snd_strerror(err));
}
AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
AlsaPCMHandle = handle;
}
//----------------------------------------------------------------------------
// 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";
}
}
Debug(3, "audio/alsa: mixer %s - %s open\n", 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);
Debug(3, "audio/alsa: PCM mixer found %ld - %ld ratio %d\n",
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'\n"), device);
}
}
//----------------------------------------------------------------------------
// Alsa API
//----------------------------------------------------------------------------
/**
** Get alsa audio delay in time-stamps.
**
** @returns audio delay in time-stamps.
**
** @todo FIXME: handle the case no audio running
*/
static int64_t AlsaGetDelay(void)
{
int err;
snd_pcm_sframes_t delay;
int64_t pts;
// setup error
if (!AlsaPCMHandle || !AudioRing[AudioRingRead].HwSampleRate) {
return 0L;
}
// delay in frames in alsa + kernel buffers
if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) {
//Debug(3, "audio/alsa: no hw delay\n");
delay = 0L;
#ifdef DEBUG
} else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) {
//Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay);
#endif
}
//Debug(3, "audio/alsa: %ld frames hw delay\n", delay);
// delay can be negative, when underrun occur
if (delay < 0) {
delay = 0L;
}
pts = ((int64_t) delay * 90 * 1000) / AudioRing[AudioRingRead].HwSampleRate;
return pts;
}
/**
** Setup alsa audio for requested format.
**
** @param freq sample frequency
** @param channels number of channels
** @param passthrough use pass-through (AC-3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
** @retval -1 something gone wrong
**
** @todo FIXME: remove pointer for freq + channels
*/
static int AlsaSetup(int *freq, int *channels, int passthrough)
{
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int err;
int delay;
if (!AlsaPCMHandle) { // alsa not running yet
// FIXME: if open fails for fe. pass-through, we never recover
return -1;
}
if (!AudioAlsaNoCloseOpen) { // close+open to fix HDMI no sound bug
snd_pcm_t *handle;
handle = AlsaPCMHandle;
// no lock needed, thread exit in main loop only
//Debug(3, "audio: %s [\n", __FUNCTION__);
AlsaPCMHandle = NULL; // other threads should check handle
snd_pcm_close(handle);
if (AudioAlsaCloseOpenDelay) {
usleep(50 * 1000); // 50ms delay for alsa recovery
}
// FIXME: can use multiple retries
if (!(handle = AlsaOpenPCM(passthrough))) {
return -1;
}
AlsaPCMHandle = handle;
//Debug(3, "audio: %s ]\n", __FUNCTION__);
}
for (;;) {
if ((err =
snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED :
SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
96 * 1000))) {
// try reduced buffer size (needed for sunxi)
// FIXME: alternativ make this configurable
if ((err =
snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED :
SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
72 * 1000))) {
/*
if ( err == -EBADFD ) {
snd_pcm_close(AlsaPCMHandle);
AlsaPCMHandle = NULL;
continue;
}
*/
if (!AudioDoingInit) {
Error(_("audio/alsa: set params error: %s\n"),
snd_strerror(err));
}
// FIXME: must stop sound, AudioChannels ... invalid
return -1;
}
}
break;
}
// this is disabled, no advantages!
if (0) { // no underruns allowed, play silence
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t boundary;
snd_pcm_sw_params_alloca(&sw_params);
err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params);
if (err < 0) {
Error(_("audio: snd_pcm_sw_params_current failed: %s\n"),
snd_strerror(err));
}
if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) {
Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"),
snd_strerror(err));
}
Debug(4, "audio/alsa: boundary %lu frames\n", boundary);
if ((err =
snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params,
boundary)) < 0) {
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
snd_strerror(err));
}
if ((err =
snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params,
boundary)) < 0) {
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
snd_strerror(err));
}
if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) {
Error(_("audio: snd_pcm_sw_params failed: %s\n"),
snd_strerror(err));
}
}
// update buffer
snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n",
buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
buffer_size) * 1000 / (*freq * *channels * AudioBytesProSample),
period_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
period_size) * 1000 / (*freq * *channels * AudioBytesProSample));
Debug(3, "audio/alsa: state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
AudioStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
// buffer time/delay in ms
delay = AudioBufferTime;
if (VideoAudioDelay > 0) {
delay += VideoAudioDelay / 90;
}
if (AudioStartThreshold <
(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
AudioStartThreshold = (*freq * *channels * AudioBytesProSample * delay) / 1000U;
}
// no bigger, than 1/3 the buffer
if (AudioStartThreshold > AudioRingBufferSize / 3) {
AudioStartThreshold = AudioRingBufferSize / 3;
}
if (!AudioDoingInit) {
Info(_("audio/alsa: start delay %ums\n"), (AudioStartThreshold * 1000)
/ (*freq * *channels * AudioBytesProSample));
}
return 0;
}
/**
** Play audio.
*/
static void AlsaPlay(void)
{
int err;
if (AlsaCanPause) {
if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) {
Error(_("audio/alsa: snd_pcm_pause(): %s\n"), snd_strerror(err));
}
} else {
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
Error(_("audio/alsa: snd_pcm_prepare(): %s\n"), snd_strerror(err));
}
}
#ifdef DEBUG
if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PAUSED) {
Error(_("audio/alsa: still paused\n"));
}
#endif
}
/**
** Pause audio.
*/
static void AlsaPause(void)
{
int err;
if (AlsaCanPause) {
if ((err = snd_pcm_pause(AlsaPCMHandle, 1))) {
Error(_("snd_pcm_pause(): %s\n"), snd_strerror(err));
}
} else {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
Error(_("snd_pcm_drop(): %s\n"), snd_strerror(err));
}
}
}
/**
** Empty log callback
*/
static void AlsaNoopCallback( __attribute__ ((unused))
const char *file, __attribute__ ((unused))
int line, __attribute__ ((unused))
const char *function, __attribute__ ((unused))
int err, __attribute__ ((unused))
const char *fmt, ...)
{
}
/**
** Initialize alsa audio output module.
*/
static void AlsaInit(void)
{
#ifdef DEBUG
(void)AlsaNoopCallback;
#else
// disable display of alsa error messages
snd_lib_error_set_handler(AlsaNoopCallback);
#endif
AlsaInitPCM();
AlsaInitMixer();
}
/**
** Cleanup alsa audio output module.
*/
static void AlsaExit(void)
{
if (AlsaPCMHandle) {
snd_pcm_close(AlsaPCMHandle);
AlsaPCMHandle = NULL;
}
if (AlsaMixer) {
snd_mixer_close(AlsaMixer);
AlsaMixer = NULL;
AlsaMixerElem = NULL;
}
}
/**
** Alsa module.
*/
static const AudioModule AlsaModule = {
.Name = "alsa",
#ifdef USE_AUDIO_THREAD
.Thread = AlsaThread,
#endif
.FlushBuffers = AlsaFlushBuffers,
.GetDelay = AlsaGetDelay,
.SetVolume = AlsaSetVolume,
.Setup = AlsaSetup,
.Play = AlsaPlay,
.Pause = AlsaPause,
.Init = AlsaInit,
.Exit = AlsaExit,
};
#endif // USE_ALSA
#ifdef USE_OSS
//============================================================================
// O S S
//============================================================================
//----------------------------------------------------------------------------
// OSS variables
//----------------------------------------------------------------------------
static int OssPcmFildes = -1; ///< pcm file descriptor
static int OssMixerFildes = -1; ///< mixer file descriptor
static int OssMixerChannel; ///< mixer channel index
static int OssFragmentTime; ///< fragment time in ms
//----------------------------------------------------------------------------
// OSS pcm
//----------------------------------------------------------------------------
/**
** Play samples from ringbuffer.
**
** @retval 0 ok
** @retval 1 ring buffer empty
** @retval -1 underrun error
*/
static int OssPlayRingbuffer(void)
{
int first;
first = 1;
for (;;) {
audio_buf_info bi;
const void *p;
int n;
if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
strerror(errno));
return -1;
}
Debug(4, "audio/oss: %d bytes free\n", bi.bytes);
n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
if (!n) { // ring buffer empty
if (first) { // only error on first loop
return 1;
}
return 0;
}
if (n < bi.bytes) { // not enough bytes in ring buffer
bi.bytes = n;
}
if (bi.bytes <= 0) { // full or buffer empty
break; // bi.bytes could become negative!
}
if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
// FIXME: quick&dirty cast
AudioSoftAmplifier((int16_t *) p, bi.bytes);
// FIXME: if not all are written, we double amplify them
}
for (;;) {
n = write(OssPcmFildes, p, bi.bytes);
if (n != bi.bytes) {
if (n < 0) {
if (n == EAGAIN) {
continue;
}
Error(_("audio/oss: write error: %s\n"), strerror(errno));
return 1;
}
Warning(_("audio/oss: error not all bytes written\n"));
}
break;
}
// advance how many could written
RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, n);
first = 0;
}
return 0;
}
/**
** Flush OSS buffers.
*/
static void OssFlushBuffers(void)
{
if (OssPcmFildes != -1) {
// flush kernel buffers
if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"),
strerror(errno));
}
}
}
#ifdef USE_AUDIO_THREAD
//----------------------------------------------------------------------------
// thread playback
//----------------------------------------------------------------------------
/**
** OSS thread
**
** @retval -1 error
** @retval 0 underrun
** @retval 1 running
*/
static int OssThread(void)
{
int err;
if (!OssPcmFildes) {
usleep(OssFragmentTime * 1000);
return -1;
}
for (;;) {
struct pollfd fds[1];
if (AudioPaused) {
return 1;
}
// wait for space in kernel buffers
fds[0].fd = OssPcmFildes;
fds[0].events = POLLOUT | POLLERR;
// wait for space in kernel buffers
err = poll(fds, 1, OssFragmentTime);
if (err < 0) {
if (err == EAGAIN) {
continue;
}
Error(_("audio/oss: error poll %s\n"), strerror(errno));
usleep(OssFragmentTime * 1000);
return -1;
}
break;
}
if (!err || AudioPaused) { // timeout or some commands
return 1;
}
if ((err = OssPlayRingbuffer())) { // empty / error
if (err < 0) { // underrun error
return -1;
}
pthread_yield();
usleep(OssFragmentTime * 1000); // let fill/empty the buffers
return 0;
}
return 1;
}
#endif
//----------------------------------------------------------------------------
/**
** Open OSS pcm device.
**
** @param passthrough use pass-through (AC-3, ...) device
*/
static int OssOpenPCM(int passthrough)
{
const char *device;
int fildes;
// &&|| hell
if (!(passthrough && ((device = AudioPassthroughDevice)
|| (device = getenv("OSS_PASSTHROUGHDEV"))))
&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
device = "/dev/dsp";
}
if (!AudioDoingInit) {
Info(_("audio/oss: using %sdevice '%s'\n"),
passthrough ? "pass-through " : "", device);
}
if ((fildes = open(device, O_WRONLY)) < 0) {
Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
strerror(errno));
return -1;
}
return fildes;
}
/**
** Initialize OSS pcm device.
**
** @see AudioPCMDevice
*/
static void OssInitPCM(void)
{
int fildes;
fildes = OssOpenPCM(0);
OssPcmFildes = fildes;
}
//----------------------------------------------------------------------------
// OSS Mixer
//----------------------------------------------------------------------------
/**
** Set OSS mixer volume (0-1000)
**
** @param volume volume (0 .. 1000)
*/
static void OssSetVolume(int volume)
{
int v;
if (OssMixerFildes != -1) {
v = (volume * 255) / 1000;
v &= 0xff;
v = (v << 8) | v;
if (ioctl(OssMixerFildes, MIXER_WRITE(OssMixerChannel), &v) < 0) {
Error(_("audio/oss: ioctl(MIXER_WRITE): %s\n"), strerror(errno));
}
}
}
/**
** Mixer channel name table.
*/
static const char *OssMixerChannelNames[SOUND_MIXER_NRDEVICES] =
SOUND_DEVICE_NAMES;
/**
** Initialize OSS mixer.
*/
static void OssInitMixer(void)
{
const char *device;
const char *channel;
int fildes;
int devmask;
int i;
if (!(device = AudioMixerDevice)) {
if (!(device = getenv("OSS_MIXERDEV"))) {
device = "/dev/mixer";
}
}
if (!(channel = AudioMixerChannel)) {
if (!(channel = getenv("OSS_MIXER_CHANNEL"))) {
channel = "pcm";
}
}
Debug(3, "audio/oss: mixer %s - %s open\n", device, channel);
if ((fildes = open(device, O_RDWR)) < 0) {
Error(_("audio/oss: can't open mixer device '%s': %s\n"), device,
strerror(errno));
return;
}
// search channel name
if (ioctl(fildes, SOUND_MIXER_READ_DEVMASK, &devmask) < 0) {
Error(_("audio/oss: ioctl(SOUND_MIXER_READ_DEVMASK): %s\n"),
strerror(errno));
close(fildes);
return;
}
for (i = 0; i < SOUND_MIXER_NRDEVICES; ++i) {
if (!strcasecmp(OssMixerChannelNames[i], channel)) {
if (devmask & (1 << i)) {
OssMixerFildes = fildes;
OssMixerChannel = i;
return;
}
Error(_("audio/oss: channel '%s' not supported\n"), channel);
break;
}
}
Error(_("audio/oss: channel '%s' not found\n"), channel);
close(fildes);
}
//----------------------------------------------------------------------------
// OSS API
//----------------------------------------------------------------------------
/**
** Get OSS audio delay in time stamps.
**
** @returns audio delay in time stamps.
*/
static int64_t OssGetDelay(void)
{
int delay;
int64_t pts;
// setup failure
if (OssPcmFildes == -1 || !AudioRing[AudioRingRead].HwSampleRate) {
return 0L;
}
if (!AudioRunning) { // audio not running
Error(_("audio/oss: should not happen\n"));
return 0L;
}
// delay in bytes in kernel buffers
delay = -1;
if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &delay) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"),
strerror(errno));
return 0L;
}
if (delay < 0) {
delay = 0;
}
pts = ((int64_t) delay * 90 * 1000)
/ (AudioRing[AudioRingRead].HwSampleRate *
AudioRing[AudioRingRead].HwChannels * AudioBytesProSample);
return pts;
}
/**
** Setup OSS audio for requested format.
**
** @param sample_rate sample rate/frequency
** @param channels number of channels
** @param passthrough use pass-through (AC-3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
** @retval -1 something gone wrong
*/
static int OssSetup(int *sample_rate, int *channels, int passthrough)
{
int ret;
int tmp;
int delay;
audio_buf_info bi;
if (OssPcmFildes == -1) { // OSS not ready
// FIXME: if open fails for fe. pass-through, we never recover
return -1;
}
if (1) { // close+open for pcm / AC-3
int fildes;
fildes = OssPcmFildes;
OssPcmFildes = -1;
close(fildes);
if (!(fildes = OssOpenPCM(passthrough))) {
return -1;
}
OssPcmFildes = fildes;
}
ret = 0;
tmp = AFMT_S16_NE; // native 16 bits
if (ioctl(OssPcmFildes, SNDCTL_DSP_SETFMT, &tmp) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"), strerror(errno));
// FIXME: stop player, set setup failed flag
return -1;
}
if (tmp != AFMT_S16_NE) {
Error(_("audio/oss: device doesn't support 16 bit sample format.\n"));
// FIXME: stop player, set setup failed flag
return -1;
}
tmp = *channels;
if (ioctl(OssPcmFildes, SNDCTL_DSP_CHANNELS, &tmp) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"),
strerror(errno));
return -1;
}
if (tmp != *channels) {
Warning(_("audio/oss: device doesn't support %d channels.\n"),
*channels);
*channels = tmp;
ret = 1;
}
tmp = *sample_rate;
if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno));
return -1;
}
if (tmp != *sample_rate) {
Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"),
*sample_rate);
*sample_rate = tmp;
ret = 1;
}
#ifdef SNDCTL_DSP_POLICY
tmp = 3;
if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno));
} else {
Info("audio/oss: set policy to %d\n", tmp);
}
#endif
if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
strerror(errno));
bi.fragsize = 4096;
bi.fragstotal = 16;
} else {
Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
}
OssFragmentTime = (bi.fragsize * 1000)
/ (*sample_rate * *channels * AudioBytesProSample);
Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n",
bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000)
/ (*sample_rate * *channels * AudioBytesProSample), bi.fragsize,
OssFragmentTime);
// start when enough bytes for initial write
AudioStartThreshold = (bi.fragsize - 1) * bi.fragstotal;
// buffer time/delay in ms
delay = AudioBufferTime + 300;
if (VideoAudioDelay > 0) {
delay += VideoAudioDelay / 90;
}
if (AudioStartThreshold <
(*sample_rate * *channels * AudioBytesProSample * delay) / 1000U) {
AudioStartThreshold =
(*sample_rate * *channels * AudioBytesProSample * delay) / 1000U;
}
// no bigger, than 1/3 the buffer
if (AudioStartThreshold > AudioRingBufferSize / 3) {
AudioStartThreshold = AudioRingBufferSize / 3;
}
if (!AudioDoingInit) {
Info(_("audio/oss: delay %ums\n"), (AudioStartThreshold * 1000)
/ (*sample_rate * *channels * AudioBytesProSample));
}
return ret;
}
/**
** Play audio.
*/
static void OssPlay(void)
{
}
/**
** Pause audio.
*/
void OssPause(void)
{
}
/**
** Initialize OSS audio output module.
*/
static void OssInit(void)
{
OssInitPCM();
OssInitMixer();
}
/**
** Cleanup OSS audio output module.
*/
static void OssExit(void)
{
if (OssPcmFildes != -1) {
close(OssPcmFildes);
OssPcmFildes = -1;
}
if (OssMixerFildes != -1) {
close(OssMixerFildes);
OssMixerFildes = -1;
}
}
/**
** OSS module.
*/
static const AudioModule OssModule = {
.Name = "oss",
#ifdef USE_AUDIO_THREAD
.Thread = OssThread,
#endif
.FlushBuffers = OssFlushBuffers,
.GetDelay = OssGetDelay,
.SetVolume = OssSetVolume,
.Setup = OssSetup,
.Play = OssPlay,
.Pause = OssPause,
.Init = OssInit,
.Exit = OssExit,
};
#endif // USE_OSS
//============================================================================
// Noop
//============================================================================
/**
** Get audio delay in time stamps.
**
** @returns audio delay in time stamps.
*/
static int64_t NoopGetDelay(void)
{
return 0L;
}
/**
** Set mixer volume (0-1000)
**
** @param volume volume (0 .. 1000)
*/
static void NoopSetVolume( __attribute__ ((unused))
int volume)
{
}
/**
** Noop setup.
**
** @param freq sample frequency
** @param channels number of channels
** @param passthrough use pass-through (AC-3, ...) device
*/
static int NoopSetup( __attribute__ ((unused))
int *channels, __attribute__ ((unused))
int *freq, __attribute__ ((unused))
int passthrough)
{
return -1;
}
/**
** Noop void
*/
static void NoopVoid(void)
{
}
/**
** Noop module.
*/
static const AudioModule NoopModule = {
.Name = "noop",
.FlushBuffers = NoopVoid,
.GetDelay = NoopGetDelay,
.SetVolume = NoopSetVolume,
.Setup = NoopSetup,
.Play = NoopVoid,
.Pause = NoopVoid,
.Init = NoopVoid,
.Exit = NoopVoid,
};
//----------------------------------------------------------------------------
// thread playback
//----------------------------------------------------------------------------
#ifdef USE_AUDIO_THREAD
/**
** Prepare next ring buffer.
*/
static int AudioNextRing(void)
{
int passthrough;
int sample_rate;
int channels;
size_t used;
// update audio format
// not always needed, but check if needed is too complex
passthrough = AudioRing[AudioRingRead].Passthrough;
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
channels = AudioRing[AudioRingRead].HwChannels;
if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) {
Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels,
sample_rate);
// FIXME: handle error
AudioRing[AudioRingRead].HwSampleRate = 0;
AudioRing[AudioRingRead].InSampleRate = 0;
return -1;
}
AudioSetVolume(AudioVolume); // update channel delta
AudioResetCompressor();
AudioResetNormalizer();
Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled),
(RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
// stop, if not enough in next buffer
used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer);
if (AudioStartThreshold * 10 < used || (AudioVideoIsReady
&& AudioStartThreshold < used)) {
return 0;
}
return 1;
}
/**
** Audio play thread.
**
** @param dummy unused thread argument
*/
static void *AudioPlayHandlerThread(void *dummy)
{
Debug(3, "audio: play thread started\n");
prctl(PR_SET_NAME,"cuvid audio",0,0,0);
for (;;) {
// check if we should stop the thread
if (AudioThreadStop) {
Debug(3, "audio: play thread stopped\n");
return PTHREAD_CANCELED;
}
Debug(3, "audio: wait on start condition\n");
pthread_mutex_lock(&AudioMutex);
AudioRunning = 0;
do {
pthread_cond_wait(&AudioStartCond, &AudioMutex);
// cond_wait can return, without signal!
} while (!AudioRunning);
pthread_mutex_unlock(&AudioMutex);
Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000)
/ (!AudioRing[AudioRingWrite].HwSampleRate +
!AudioRing[AudioRingWrite].HwChannels +
AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
do {
int filled;
int read;
int flush;
int err;
int i;
// check if we should stop the thread
if (AudioThreadStop) {
Debug(3, "audio: play thread stopped\n");
return PTHREAD_CANCELED;
}
// look if there is a flush command in the queue
flush = 0;
filled = atomic_read(&AudioRingFilled);
read = AudioRingRead;
i = filled;
while (i--) {
read = (read + 1) % AUDIO_RING_MAX;
if (AudioRing[read].FlushBuffers) {
AudioRing[read].FlushBuffers = 0;
AudioRingRead = read;
// handle all flush in queue
flush = filled - i;
}
}
if (flush) {
Debug(3, "audio: flush %d ring buffer(s)\n", flush);
AudioUsedModule->FlushBuffers();
atomic_sub(flush, &AudioRingFilled);
if (AudioNextRing()) {
Debug(3, "audio: HandlerThread break after flush\n");
break;
}
Debug(3, "audio: continue after flush\n");
}
// try to play some samples
err = 0;
if (RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) {
err = AudioUsedModule->Thread();
}
// underrun, check if new ring buffer is available
if (!err) {
int passthrough;
int sample_rate;
int channels;
int old_passthrough;
int old_sample_rate;
int old_channels;
// underrun, and no new ring buffer, goto sleep.
if (!atomic_read(&AudioRingFilled)) {
Debug(3,"audio: HandlerThread Underrun with no new data\n");
break;
}
Debug(3, "audio: next ring buffer\n");
old_passthrough = AudioRing[AudioRingRead].Passthrough;
old_sample_rate = AudioRing[AudioRingRead].HwSampleRate;
old_channels = AudioRing[AudioRingRead].HwChannels;
atomic_dec(&AudioRingFilled);
AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
passthrough = AudioRing[AudioRingRead].Passthrough;
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
channels = AudioRing[AudioRingRead].HwChannels;
Debug(3, "audio: thread channels %d frequency %dHz %s\n",
channels, sample_rate, passthrough ? "pass-through" : "");
// audio config changed?
if (old_passthrough != passthrough
|| old_sample_rate != sample_rate
|| old_channels != channels) {
// FIXME: wait for buffer drain
if (AudioNextRing()) {
Debug(3,"audio: HandlerThread break on nextring");
break;
}
} else {
AudioResetCompressor();
AudioResetNormalizer();
}
}
// FIXME: check AudioPaused ...Thread()
if (AudioPaused) {
Debug(3,"audio: HandlerThread break on paused");
break;
}
} while (AudioRing[AudioRingRead].HwSampleRate);
}
return dummy;
}
/**
** Initialize audio thread.
*/
static void AudioInitThread(void)
{
AudioThreadStop = 0;
pthread_mutex_init(&AudioMutex, NULL);
pthread_cond_init(&AudioStartCond, NULL);
pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL);
pthread_setname_np(AudioThread, "softhddev audio");
}
/**
** Cleanup audio thread.
*/
static void AudioExitThread(void)
{
void *retval;
Debug(3, "audio: %s\n", __FUNCTION__);
if (AudioThread) {
AudioThreadStop = 1;
AudioRunning = 1; // wakeup thread, if needed
pthread_cond_signal(&AudioStartCond);
if (pthread_join(AudioThread, &retval) || retval != PTHREAD_CANCELED) {
Error(_("audio: can't cancel play thread\n"));
}
pthread_cond_destroy(&AudioStartCond);
pthread_mutex_destroy(&AudioMutex);
AudioThread = 0;
}
}
#endif
//----------------------------------------------------------------------------
//----------------------------------------------------------------------------
/**
** Table of all audio modules.
*/
static const AudioModule *AudioModules[] = {
#ifdef USE_ALSA
&AlsaModule,
#endif
#ifdef USE_OSS
&OssModule,
#endif
&NoopModule,
};
void AudioDelayms(int delayms) {
int count;
unsigned char *p;
#ifdef DEBUG
printf("Try Delay Audio for %d ms Samplerate %d Channels %d bps %d\n",
delayms,AudioRing[AudioRingWrite].HwSampleRate,AudioRing[AudioRingWrite].HwChannels,AudioBytesProSample);
#endif
count = delayms * AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample / 1000;
if (delayms < 5000 && delayms > 0) { // not more than 5seconds
p = calloc(1,count);
RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, p, count);
free(p);
}
}
/**
** Place samples in audio output queue.
**
** @param samples sample buffer
** @param count number of bytes in sample buffer
*/
void AudioEnqueue(const void *samples, int count)
{
size_t n;
int16_t *buffer;
#ifdef noDEBUG
static uint32_t last_tick;
uint32_t tick;
tick = GetMsTicks();
if (tick - last_tick > 101) {
Debug(3, "audio: enqueue %4d %dms\n", count, tick - last_tick);
}
last_tick = tick;
#endif
if (!AudioRing[AudioRingWrite].HwSampleRate) {
Debug(3, "audio: enqueue not ready\n");
return; // no setup yet
}
// save packet size
if (!AudioRing[AudioRingWrite].PacketSize) {
AudioRing[AudioRingWrite].PacketSize = count;
Debug(3, "audio: a/v packet size %d bytes\n", count);
}
// audio sample modification allowed and needed?
buffer = (void *)samples;
if (!AudioRing[AudioRingWrite].Passthrough && (AudioCompression
|| AudioNormalize
|| AudioRing[AudioRingWrite].InChannels !=
AudioRing[AudioRingWrite].HwChannels)) {
int frames;
// resample into ring-buffer is too complex in the case of a roundabout
// just use a temporary buffer
frames =
count / (AudioRing[AudioRingWrite].InChannels *
AudioBytesProSample);
buffer =
alloca(frames * AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample);
#ifdef USE_AUDIO_MIXER
// Convert / resample input to hardware format
AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames,
buffer, AudioRing[AudioRingWrite].HwChannels);
#else
#ifdef DEBUG
if (AudioRing[AudioRingWrite].InChannels !=
AudioRing[AudioRingWrite].HwChannels) {
Debug(3, "audio: internal failure channels mismatch\n");
return;
}
#endif
memcpy(buffer, samples, count);
#endif
count =
frames * AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample;
if (AudioCompression) { // in place operation
AudioCompressor(buffer, count);
}
if (AudioNormalize) { // in place operation
AudioNormalizer(buffer, count);
}
}
n = RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, buffer, count);
if (n != (size_t) count) {
Error(_("audio: can't place %d samples in ring buffer\n"), count);
// too many bytes are lost
// FIXME: caller checks buffer full.
// FIXME: should skip more, longer skip, but less often?
// FIXME: round to channel + sample border
}
if (!AudioRunning) { // check, if we can start the thread
int skip;
n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
skip = AudioSkip;
// FIXME: round to packet size
Debug(4, "audio: start? %4zdms skip %dms\n", (n * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
(skip * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
if (skip) {
if (n < (unsigned)skip) {
skip = n;
}
AudioSkip -= skip;
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
}
// forced start or enough video + audio buffered
// for some exotic channels * 4 too small
if (AudioStartThreshold * 10 < n || (AudioVideoIsReady
// if ((AudioVideoIsReady
&& AudioStartThreshold < n)) {
// restart play-back
// no lock needed, can wakeup next time
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
Debug(3,"Start on AudioEnque\n");
}
}
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
if (AudioRing[AudioRingWrite].PTS != (int64_t) INT64_C(0x8000000000000000)) {
AudioRing[AudioRingWrite].PTS += ((int64_t) count * 90 * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
}
}
/**
** Video is ready.
**
** @param pts video presentation timestamp
*/
void AudioVideoReady(int64_t pts)
{
int64_t audio_pts;
size_t used;
if (pts == (int64_t) INT64_C(0x8000000000000000)) {
Debug(3, "audio: a/v start, no valid video\n");
return;
}
// no valid audio known
if (!AudioRing[AudioRingWrite].HwSampleRate
|| !AudioRing[AudioRingWrite].HwChannels
|| AudioRing[AudioRingWrite].PTS ==
(int64_t) INT64_C(0x8000000000000000)) {
Debug(3, "audio: a/v start, no valid audio\n");
AudioVideoIsReady = 1;
return;
}
// Audio.PTS = next written sample time stamp
used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
audio_pts =
AudioRing[AudioRingWrite].PTS -
(used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
Debug(3, "audio: a/v sync buf(%d,%4zdms) %s | %s = %dms %s\n",
atomic_read(&AudioRingFilled),
(used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
Timestamp2String(pts), Timestamp2String(audio_pts),
(int)(pts - audio_pts) / 90, AudioRunning ? "running" : "ready");
if (!AudioRunning) {
int skip;
// buffer ~15 video frames
// FIXME: HDTV can use smaller video buffer
skip =
pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay;
#ifdef DEBUG
fprintf(stderr, "%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
VideoAudioDelay / 90, skip / 90);
#endif
// guard against old PTS
if (skip > 0 && skip < 4000 * 90) {
skip = (((int64_t) skip * AudioRing[AudioRingWrite].HwSampleRate) / (1000 * 90))
* AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample;
// FIXME: round to packet size
if ((unsigned)skip > used) {
AudioSkip = skip - used;
skip = used;
}
Debug(3, "audio: sync advance %dms %d/%zd\n",
(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample), skip, used);
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
}
else {
Debug(3,"No audio skip -> should skip %d\n",skip/90);
}
// FIXME: skip<0 we need bigger audio buffer
// enough video + audio buffered
if (AudioStartThreshold < used) {
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
Debug(3,"Start on AudioVideoReady\n");
}
}
AudioVideoIsReady = 1;
#if 0
if (AudioRing[AudioRingWrite].HwSampleRate
&& AudioRing[AudioRingWrite].HwChannels) {
if (pts != (int64_t) INT64_C(0x8000000000000000)
&& AudioRing[AudioRingWrite].PTS !=
(int64_t) INT64_C(0x8000000000000000)) {
Debug(3, "audio: a/v %d %s\n",
(int)(pts - AudioRing[AudioRingWrite].PTS) / 90,
AudioRunning ? "running" : "stopped");
}
Debug(3, "audio: start %4zdms %s|%s video ready\n",
(RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
Timestamp2String(pts),
Timestamp2String(AudioRing[AudioRingWrite].PTS));
if (!AudioRunning) {
size_t used;
used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
// enough video + audio buffered
if (AudioStartThreshold < used) {
// too much audio buffered, skip it
if (AudioStartThreshold < used) {
Debug(3, "audio: start %4zdms skip video ready\n",
((used - AudioStartThreshold) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample));
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer,
used - AudioStartThreshold);
}
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
}
}
}
AudioVideoIsReady = 1;
#endif
}
/**
** Flush audio buffers.
*/
void AudioFlushBuffers(void)
{
int old;
int i;
if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) {
// wait for space in ring buffer, should never happen
for (i = 0; i < 24 * 2; ++i) {
if (atomic_read(&AudioRingFilled) < AUDIO_RING_MAX) {
break;
}
Debug(3, "audio: flush out of ring buffers\n");
usleep(1 * 1000); // avoid hot polling
}
if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) {
// FIXME: We can set the flush flag in the last wrote ring buffer
Error(_("audio: flush out of ring buffers\n"));
return;
}
}
old = AudioRingWrite;
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
AudioRing[AudioRingWrite].FlushBuffers = 1;
AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough;
AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate;
AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
AudioRing[AudioRingWrite].InChannels = AudioRing[old].InChannels;
AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000);
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer,
RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer));
Debug(3, "audio: reset video ready\n");
AudioVideoIsReady = 0;
AudioSkip = 0;
atomic_inc(&AudioRingFilled);
// FIXME: wait for flush complete needed?
for (i = 0; i < 24 * 2; ++i) {
if (!AudioRunning) { // wakeup thread to flush buffers
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
Debug(3,"Start on Flush\n");
}
// FIXME: waiting on zero isn't correct, but currently works
if (!atomic_read(&AudioRingFilled)) {
break;
}
usleep(1 * 1000); // avoid hot polling
}
Debug(3, "audio: audio flush %dms\n", i);
}
/**
** Call back to play audio polled.
*/
void AudioPoller(void)
{
// FIXME: write poller
}
/**
** Get free bytes in audio output.
*/
int AudioFreeBytes(void)
{
return AudioRing[AudioRingWrite].RingBuffer ?
RingBufferFreeBytes(AudioRing[AudioRingWrite].RingBuffer)
: INT32_MAX;
}
/**
** Get used bytes in audio output.
*/
int AudioUsedBytes(void)
{
// FIXME: not correct, if multiple buffer are in use
return AudioRing[AudioRingWrite].RingBuffer ?
RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) : 0;
}
/**
** Get audio delay in time stamps.
**
** @returns audio delay in time stamps.
*/
int64_t AudioGetDelay(void)
{
int64_t pts;
if (!AudioRunning) {
return 0L; // audio not running
}
if (!AudioRing[AudioRingRead].HwSampleRate) {
return 0L; // audio not setup
}
if (atomic_read(&AudioRingFilled)) {
return 0L; // multiple buffers, invalid delay
}
pts = AudioUsedModule->GetDelay();
pts += ((int64_t) RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)
* 90 * 1000) / (AudioRing[AudioRingRead].HwSampleRate *
AudioRing[AudioRingRead].HwChannels * AudioBytesProSample);
Debug(4, "audio: hw+sw delay %zd %" PRId64 "ms\n",
RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer), pts / 90);
return pts;
}
/**
** Set audio clock base.
**
** @param pts audio presentation timestamp
*/
void AudioSetClock(int64_t pts)
{
if (AudioRing[AudioRingWrite].PTS != pts) {
Debug(4, "audio: set clock %s -> %s pts\n",
Timestamp2String(AudioRing[AudioRingWrite].PTS),
Timestamp2String(pts));
}
AudioRing[AudioRingWrite].PTS = pts;
}
/**
** Get current audio clock.
**
** @returns the audio clock in time stamps.
*/
int64_t AudioGetClock(void)
{
// (cast) needed for the evil gcc
if (AudioRing[AudioRingRead].PTS != (int64_t) INT64_C(0x8000000000000000)) {
int64_t delay;
// delay zero, if no valid time stamp
if ((delay = AudioGetDelay())) {
if (AudioRing[AudioRingRead].Passthrough) {
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
}
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
}
}
return INT64_C(0x8000000000000000);
}
/**
** Set mixer volume (0-1000)
**
** @param volume volume (0 .. 1000)
*/
void AudioSetVolume(int volume)
{
AudioVolume = volume;
AudioMute = !volume;
// reduce loudness for stereo output
if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2
&& !AudioRing[AudioRingRead].Passthrough) {
volume -= AudioStereoDescent;
if (volume < 0) {
volume = 0;
} else if (volume > 1000) {
volume = 1000;
}
}
AudioAmplifier = volume;
if (!AudioSoftVolume) {
AudioUsedModule->SetVolume(volume);
}
}
/**
** Setup audio for requested format.
**
** @param freq sample frequency
** @param channels number of channels
** @param passthrough use pass-through (AC-3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
** @retval -1 something gone wrong
**
** @todo add support to report best fitting format.
*/
int AudioSetup(int *freq, int *channels, int passthrough)
{
Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq,
passthrough ? "pass-through" : "");
// invalid parameter
if (!freq || !channels || !*freq || !*channels) {
Debug(3, "audio: bad channels or frequency parameters\n");
// FIXME: set flag invalid setup
return -1;
}
return AudioRingAdd(*freq, *channels, passthrough);
}
/**
** Play audio.
*/
void AudioPlay(void)
{
if (!AudioPaused) {
Debug(3, "audio: not paused, check the code\n");
return;
}
Debug(3, "audio: resumed\n");
AudioPaused = 0;
AudioEnqueue(NULL, 0); // wakeup thread
}
/**
** Pause audio.
*/
void AudioPause(void)
{
if (AudioPaused) {
Debug(3, "audio: already paused, check the code\n");
return;
}
Debug(3, "audio: paused\n");
AudioPaused = 1;
}
/**
** Set audio buffer time.
**
** PES audio packets have a max distance of 300 ms.
** TS audio packet have a max distance of 100 ms.
** The period size of the audio buffer is 24 ms.
** With streamdev sometimes extra +100ms are needed.
*/
void AudioSetBufferTime(int delay)
{
if (!delay) {
delay = 336;
}
AudioBufferTime = delay;
}
/**
** Enable/disable software volume.
**
** @param onoff -1 toggle, true turn on, false turn off
*/
void AudioSetSoftvol(int onoff)
{
if (onoff < 0) {
AudioSoftVolume ^= 1;
} else {
AudioSoftVolume = onoff;
}
}
/**
** Set normalize volume parameters.
**
** @param onoff -1 toggle, true turn on, false turn off
** @param maxfac max. factor of normalize /1000
*/
void AudioSetNormalize(int onoff, int maxfac)
{
if (onoff < 0) {
AudioNormalize ^= 1;
} else {
AudioNormalize = onoff;
}
AudioMaxNormalize = maxfac;
}
/**
** Set volume compression parameters.
**
** @param onoff -1 toggle, true turn on, false turn off
** @param maxfac max. factor of compression /1000
*/
void AudioSetCompression(int onoff, int maxfac)
{
if (onoff < 0) {
AudioCompression ^= 1;
} else {
AudioCompression = onoff;
}
AudioMaxCompression = maxfac;
if (!AudioCompressionFactor) {
AudioCompressionFactor = 1000;
}
if (AudioCompressionFactor > AudioMaxCompression) {
AudioCompressionFactor = AudioMaxCompression;
}
}
/**
** Set stereo loudness descent.
**
** @param delta value (/1000) to reduce stereo volume
*/
void AudioSetStereoDescent(int delta)
{
AudioStereoDescent = delta;
AudioSetVolume(AudioVolume); // update channel delta
}
/**
** Set pcm audio device.
**
** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp")
**
** @note this is currently used to select alsa/OSS output module.
*/
void AudioSetDevice(const char *device)
{
if (!AudioModuleName) {
AudioModuleName = "alsa"; // detect alsa/OSS
if (!device[0]) {
AudioModuleName = "noop";
} else if (device[0] == '/') {
AudioModuleName = "oss";
}
}
AudioPCMDevice = device;
}
/**
** Set pass-through audio device.
**
** @param device name of pass-through device (fe. "hw:0,1")
**
** @note this is currently usable with alsa only.
*/
void AudioSetPassthroughDevice(const char *device)
{
if (!AudioModuleName) {
AudioModuleName = "alsa"; // detect alsa/OSS
if (!device[0]) {
AudioModuleName = "noop";
} else if (device[0] == '/') {
AudioModuleName = "oss";
}
}
AudioPassthroughDevice = device;
}
/**
** Set pcm audio mixer channel.
**
** @param channel name of the mixer channel (fe. PCM or Master)
**
** @note this is currently used to select alsa/OSS output module.
*/
void AudioSetChannel(const char *channel)
{
AudioMixerChannel = channel;
}
/**
** Set automatic AES flag handling.
**
** @param onoff turn setting AES flag on or off
*/
void AudioSetAutoAES(int onoff)
{
if (onoff < 0) {
AudioAppendAES ^= 1;
} else {
AudioAppendAES = onoff;
}
}
/**
** Initialize audio output module.
**
** @todo FIXME: make audio output module selectable.
*/
void AudioInit(void)
{
unsigned u;
const char *name;
int freq;
int chan;
name = "noop";
#ifdef USE_OSS
name = "oss";
#endif
#ifdef USE_ALSA
name = "alsa";
#endif
if (AudioModuleName) {
name = AudioModuleName;
}
//
// search selected audio module.
//
for (u = 0; u < sizeof(AudioModules) / sizeof(*AudioModules); ++u) {
if (!strcasecmp(name, AudioModules[u]->Name)) {
AudioUsedModule = AudioModules[u];
Info(_("audio: '%s' output module used\n"), AudioUsedModule->Name);
goto found;
}
}
Error(_("audio: '%s' output module isn't supported\n"), name);
AudioUsedModule = &NoopModule;
return;
found:
AudioDoingInit = 1;
AudioRingInit();
AudioUsedModule->Init();
//
// Check which channels/rates/formats are supported
// FIXME: we force 44.1Khz and 48Khz must be supported equal
// FIXME: should use bitmap of channels supported in RatesInHw
// FIXME: use loop over sample-rates
freq = 44100;
AudioRatesInHw[Audio44100] = 0;
for (chan = 1; chan < 9; ++chan) {
int tchan;
int tfreq;
tchan = chan;
tfreq = freq;
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
AudioChannelsInHw[chan] = 0;
} else {
AudioChannelsInHw[chan] = chan;
AudioRatesInHw[Audio44100] |= (1 << chan);
}
}
freq = 48000;
AudioRatesInHw[Audio48000] = 0;
for (chan = 1; chan < 9; ++chan) {
int tchan;
int tfreq;
if (!AudioChannelsInHw[chan]) {
continue;
}
tchan = chan;
tfreq = freq;
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
//AudioChannelsInHw[chan] = 0;
} else {
AudioChannelsInHw[chan] = chan;
AudioRatesInHw[Audio48000] |= (1 << chan);
}
}
freq = 192000;
AudioRatesInHw[Audio192000] = 0;
for (chan = 1; chan < 9; ++chan) {
int tchan;
int tfreq;
if (!AudioChannelsInHw[chan]) {
continue;
}
tchan = chan;
tfreq = freq;
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
//AudioChannelsInHw[chan] = 0;
} else {
AudioChannelsInHw[chan] = chan;
AudioRatesInHw[Audio192000] |= (1 << chan);
}
}
// build channel support and conversion table
for (u = 0; u < AudioRatesMax; ++u) {
for (chan = 1; chan < 9; ++chan) {
AudioChannelMatrix[u][chan] = 0;
if (!AudioRatesInHw[u]) { // rate unsupported
continue;
}
if (AudioChannelsInHw[chan]) {
AudioChannelMatrix[u][chan] = chan;
} else {
switch (chan) {
case 1:
if (AudioChannelsInHw[2]) {
AudioChannelMatrix[u][chan] = 2;
}
break;
case 2:
case 3:
if (AudioChannelsInHw[4]) {
AudioChannelMatrix[u][chan] = 4;
break;
}
case 4:
if (AudioChannelsInHw[5]) {
AudioChannelMatrix[u][chan] = 5;
break;
}
case 5:
if (AudioChannelsInHw[6]) {
AudioChannelMatrix[u][chan] = 6;
break;
}
case 6:
if (AudioChannelsInHw[7]) {
AudioChannelMatrix[u][chan] = 7;
break;
}
case 7:
if (AudioChannelsInHw[8]) {
AudioChannelMatrix[u][chan] = 8;
break;
}
case 8:
if (AudioChannelsInHw[6]) {
AudioChannelMatrix[u][chan] = 6;
break;
}
if (AudioChannelsInHw[2]) {
AudioChannelMatrix[u][chan] = 2;
break;
}
if (AudioChannelsInHw[1]) {
AudioChannelMatrix[u][chan] = 1;
break;
}
break;
}
}
}
}
for (u = 0; u < AudioRatesMax; ++u) {
Info(_("audio: %6dHz supports %d %d %d %d %d %d %d %d channels\n"),
AudioRatesTable[u], AudioChannelMatrix[u][1],
AudioChannelMatrix[u][2], AudioChannelMatrix[u][3],
AudioChannelMatrix[u][4], AudioChannelMatrix[u][5],
AudioChannelMatrix[u][6], AudioChannelMatrix[u][7],
AudioChannelMatrix[u][8]);
}
#ifdef USE_AUDIO_THREAD
if (AudioUsedModule->Thread) { // supports threads
AudioInitThread();
}
#endif
AudioDoingInit = 0;
}
/**
** Cleanup audio output module.
*/
void AudioExit(void)
{
const AudioModule *module;
Debug(3, "audio: %s\n", __FUNCTION__);
#ifdef USE_AUDIO_THREAD
if (AudioUsedModule->Thread) { // supports threads
AudioExitThread();
}
#endif
module = AudioUsedModule;
AudioUsedModule = &NoopModule;
module->Exit();
AudioRingExit();
AudioRunning = 0;
AudioPaused = 0;
}
#ifdef AUDIO_TEST
//----------------------------------------------------------------------------
// Test
//----------------------------------------------------------------------------
void AudioTest(void)
{
for (;;) {
unsigned u;
uint8_t buffer[16 * 1024]; // some random data
int i;
for (u = 0; u < sizeof(buffer); u++) {
buffer[u] = random() & 0xffff;
}
Debug(3, "audio/test: loop\n");
for (i = 0; i < 100; ++i) {
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
AlsaEnqueue(buffer, sizeof(buffer));
}
usleep(20 * 1000);
}
break;
}
}
#include <getopt.h>
int SysLogLevel; ///< show additional debug informations
/**
** Print version.
*/
static void PrintVersion(void)
{
printf("audio_test: audio tester Version " VERSION
#ifdef GIT_REV
"(GIT-" GIT_REV ")"
#endif
",\n\t(c) 2009 - 2013 by Johns\n"
"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
}
/**
** Print usage.
*/
static void PrintUsage(void)
{
printf("Usage: audio_test [-?dhv]\n"
"\t-d\tenable debug, more -d increase the verbosity\n"
"\t-? -h\tdisplay this message\n" "\t-v\tdisplay version information\n"
"Only idiots print usage on stderr!\n");
}
/**
** Main entry point.
**
** @param argc number of arguments
** @param argv arguments vector
**
** @returns -1 on failures, 0 clean exit.
*/
int main(int argc, char *const argv[])
{
SysLogLevel = 0;
//
// Parse command line arguments
//
for (;;) {
switch (getopt(argc, argv, "hv?-c:d")) {
case 'd': // enabled debug
++SysLogLevel;
continue;
case EOF:
break;
case 'v': // print version
PrintVersion();
return 0;
case '?':
case 'h': // help usage
PrintVersion();
PrintUsage();
return 0;
case '-':
PrintVersion();
PrintUsage();
fprintf(stderr, "\nWe need no long options\n");
return -1;
case ':':
PrintVersion();
fprintf(stderr, "Missing argument for option '%c'\n", optopt);
return -1;
default:
PrintVersion();
fprintf(stderr, "Unknown option '%c'\n", optopt);
return -1;
}
break;
}
if (optind < argc) {
PrintVersion();
while (optind < argc) {
fprintf(stderr, "Unhandled argument '%s'\n", argv[optind++]);
}
return -1;
}
//
// main loop
//
AudioInit();
for (;;) {
unsigned u;
uint8_t buffer[16 * 1024]; // some random data
for (u = 0; u < sizeof(buffer); u++) {
buffer[u] = random() & 0xffff;
}
Debug(3, "audio/test: loop\n");
for (;;) {
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
AlsaEnqueue(buffer, sizeof(buffer));
}
}
}
AudioExit();
return 0;
}
#endif
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