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tdesktop/Telegram/SourceFiles/audio.cpp

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/*
This file is part of Telegram Desktop,
the official desktop version of Telegram messaging app, see https://telegram.org
Telegram Desktop is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
It 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 General Public License for more details.
Full license: https://github.com/telegramdesktop/tdesktop/blob/master/LICENSE
Copyright (c) 2014 John Preston, https://desktop.telegram.org
*/
#include "stdafx.h"
#include "audio.h"
#include <AL/al.h>
#include <AL/alc.h>
#include <opusfile.h>
#include <ogg/ogg.h>
#include <mpg123.h>
#define _test_HDR_SYNC 0xffe00000
#define _test_HDR_LAYER 0x00060000
#define _test_HDR_LAYER_VAL(h) (((h)&_test_HDR_LAYER) >> 17)
#define _test_HDR_BITRATE 0x0000f000
#define _test_HDR_BITRATE_VAL(h) (((h)&_test_HDR_BITRATE) >> 12)
#define _test_HDR_SAMPLERATE 0x00000c00
#define _test_HDR_SAMPLERATE_VAL(h) (((h)&_test_HDR_SAMPLERATE) >> 10)
#include <neaacdec.h>
#define HAVE_STDINT_H
#include <mp4ff.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libswresample/swresample.h>
}
namespace {
ALCdevice *audioDevice = 0;
ALCcontext *audioContext = 0;
ALuint notifySource = 0;
ALuint notifyBuffer = 0;
QMutex voicemsgsMutex;
VoiceMessages *voicemsgs = 0;
bool _mpg123 = false;
}
bool _checkALCError() {
ALenum errCode;
if ((errCode = alcGetError(audioDevice)) != ALC_NO_ERROR) {
LOG(("Audio Error: (alc) %1").arg((const char *)alcGetString(audioDevice, errCode)));
return false;
}
return true;
}
bool _checkALError() {
ALenum errCode;
if ((errCode = alGetError()) != AL_NO_ERROR) {
LOG(("Audio Error: (al) %1").arg((const char *)alGetString(errCode)));
return false;
}
return true;
}
void audioInit() {
uint64 ms = getms();
if (audioDevice) return;
audioDevice = alcOpenDevice(NULL);
if (!audioDevice) {
LOG(("Audio Error: default sound device not present."));
return;
}
ALCint attributes[] = { ALC_STEREO_SOURCES, 8, 0 };
audioContext = alcCreateContext(audioDevice, attributes);
alcMakeContextCurrent(audioContext);
if (!_checkALCError()) return audioFinish();
ALfloat v[] = { 0.f, 0.f, -1.f, 0.f, 1.f, 0.f };
alListener3f(AL_POSITION, 0.f, 0.f, 0.f);
alListener3f(AL_VELOCITY, 0.f, 0.f, 0.f);
alListenerfv(AL_ORIENTATION, v);
alDistanceModel(AL_NONE);
alGenSources(1, &notifySource);
alSourcef(notifySource, AL_PITCH, 1.f);
alSourcef(notifySource, AL_GAIN, 1.f);
alSource3f(notifySource, AL_POSITION, 0, 0, 0);
alSource3f(notifySource, AL_VELOCITY, 0, 0, 0);
alSourcei(notifySource, AL_LOOPING, 0);
alGenBuffers(1, &notifyBuffer);
if (!_checkALError()) return audioFinish();
QFile notify(st::newMsgSound);
if (!notify.open(QIODevice::ReadOnly)) return audioFinish();
QByteArray blob = notify.readAll();
const char *data = blob.constData();
if (blob.size() < 44) return audioFinish();
if (*((const uint32*)(data + 0)) != 0x46464952) return audioFinish(); // ChunkID - "RIFF"
if (*((const uint32*)(data + 4)) != uint32(blob.size() - 8)) return audioFinish(); // ChunkSize
if (*((const uint32*)(data + 8)) != 0x45564157) return audioFinish(); // Format - "WAVE"
if (*((const uint32*)(data + 12)) != 0x20746d66) return audioFinish(); // Subchunk1ID - "fmt "
uint32 subchunk1Size = *((const uint32*)(data + 16)), extra = subchunk1Size - 16;
if (subchunk1Size < 16 || (extra && extra < 2)) return audioFinish();
if (*((const uint16*)(data + 20)) != 1) return audioFinish(); // AudioFormat - PCM (1)
uint16 numChannels = *((const uint16*)(data + 22));
if (numChannels != 1 && numChannels != 2) return audioFinish();
uint32 sampleRate = *((const uint32*)(data + 24));
uint32 byteRate = *((const uint32*)(data + 28));
uint16 blockAlign = *((const uint16*)(data + 32));
uint16 bitsPerSample = *((const uint16*)(data + 34));
if (bitsPerSample % 8) return audioFinish();
uint16 bytesPerSample = bitsPerSample / 8;
if (bytesPerSample != 1 && bytesPerSample != 2) return audioFinish();
if (blockAlign != numChannels * bytesPerSample) return audioFinish();
if (byteRate != sampleRate * blockAlign) return audioFinish();
if (extra) {
uint16 extraSize = *((const uint16*)(data + 36));
if (uint32(extraSize + 2) != extra) return audioFinish();
if (uint32(blob.size()) < 44 + extra) return audioFinish();
}
if (*((const uint32*)(data + extra + 36)) != 0x61746164) return audioFinish(); // Subchunk2ID - "data"
uint32 subchunk2Size = *((const uint32*)(data + extra + 40));
if (subchunk2Size % (numChannels * bytesPerSample)) return audioFinish();
uint32 numSamples = subchunk2Size / (numChannels * bytesPerSample);
if (uint32(blob.size()) < 44 + extra + subchunk2Size) return audioFinish();
data += 44 + extra;
ALenum format = 0;
switch (bytesPerSample) {
case 1:
switch (numChannels) {
case 1: format = AL_FORMAT_MONO8; break;
case 2: format = AL_FORMAT_STEREO8; break;
}
break;
case 2:
switch (numChannels) {
case 1: format = AL_FORMAT_MONO16; break;
case 2: format = AL_FORMAT_STEREO16; break;
}
break;
}
if (!format) return audioFinish();
alBufferData(notifyBuffer, format, data, subchunk2Size, sampleRate);
alSourcei(notifySource, AL_BUFFER, notifyBuffer);
if (!_checkALError()) return audioFinish();
voicemsgs = new VoiceMessages();
alcSuspendContext(audioContext);
int mpg123res = mpg123_init();
if (mpg123res == MPG123_OK) {
_mpg123 = true;
} else {
LOG(("Could not init MPG123, result: %1").arg(mpg123res));
}
av_register_all();
avcodec_register_all();
LOG(("Audio init time: %1").arg(getms() - ms));
}
bool audioWorks() {
return !!voicemsgs;
}
void audioPlayNotify() {
if (!audioWorks()) return;
audioVoice()->processContext();
alSourcePlay(notifySource);
emit audioVoice()->faderOnTimer();
}
void audioFinish() {
if (voicemsgs) {
delete voicemsgs;
}
alSourceStop(notifySource);
if (alIsBuffer(notifyBuffer)) {
alDeleteBuffers(1, &notifyBuffer);
notifyBuffer = 0;
}
if (alIsSource(notifySource)) {
alDeleteSources(1, &notifySource);
notifySource = 0;
}
if (audioContext) {
alcMakeContextCurrent(NULL);
alcDestroyContext(audioContext);
audioContext = 0;
}
if (audioDevice) {
alcCloseDevice(audioDevice);
audioDevice = 0;
}
if (_mpg123) mpg123_exit();
}
VoiceMessages::VoiceMessages() : _current(0),
_fader(new VoiceMessagesFader(&_faderThread)), _loader(new VoiceMessagesLoaders(&_loaderThread)) {
connect(this, SIGNAL(faderOnTimer()), _fader, SLOT(onTimer()));
connect(this, SIGNAL(loaderOnStart(AudioData*)), _loader, SLOT(onStart(AudioData*)));
connect(this, SIGNAL(loaderOnCancel(AudioData*)), _loader, SLOT(onCancel(AudioData*)));
connect(&_faderThread, SIGNAL(started()), _fader, SLOT(onInit()));
connect(&_loaderThread, SIGNAL(started()), _loader, SLOT(onInit()));
connect(&_faderThread, SIGNAL(finished()), _fader, SLOT(deleteLater()));
connect(&_loaderThread, SIGNAL(finished()), _loader, SLOT(deleteLater()));
connect(_loader, SIGNAL(needToCheck()), _fader, SLOT(onTimer()));
connect(_loader, SIGNAL(error(AudioData*)), this, SLOT(onError(AudioData*)));
connect(_fader, SIGNAL(needToPreload(AudioData*)), _loader, SLOT(onLoad(AudioData*)));
connect(_fader, SIGNAL(playPositionUpdated(AudioData*)), this, SIGNAL(updated(AudioData*)));
connect(_fader, SIGNAL(audioStopped(AudioData*)), this, SIGNAL(stopped(AudioData*)));
connect(_fader, SIGNAL(error(AudioData*)), this, SLOT(onError(AudioData*)));
_loaderThread.start();
_faderThread.start();
}
VoiceMessages::~VoiceMessages() {
{
QMutexLocker lock(&voicemsgsMutex);
voicemsgs = 0;
}
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
alSourceStop(_data[i].source);
if (alIsBuffer(_data[i].buffers[0])) {
alDeleteBuffers(3, _data[i].buffers);
for (int32 j = 0; j < 3; ++j) {
_data[i].buffers[j] = _data[i].samplesCount[j] = 0;
}
}
if (alIsSource(_data[i].source)) {
alDeleteSources(1, &_data[i].source);
_data[i].source = 0;
}
}
_faderThread.quit();
_loaderThread.quit();
_faderThread.wait();
_loaderThread.wait();
}
void VoiceMessages::onError(AudioData *audio) {
emit stopped(audio);
}
bool VoiceMessages::updateCurrentStarted(int32 pos) {
if (pos < 0) {
if (alIsSource(_data[_current].source)) {
alGetSourcei(_data[_current].source, AL_SAMPLE_OFFSET, &pos);
} else {
pos = 0;
}
}
if (!_checkALError()) {
_data[_current].state = VoiceMessageStopped;
onError(_data[_current].audio);
return false;
}
_data[_current].started = _data[_current].position = pos + _data[_current].skipStart;
return true;
}
void VoiceMessages::play(AudioData *audio) {
QMutexLocker lock(&voicemsgsMutex);
bool startNow = true;
if (_data[_current].audio != audio) {
switch (_data[_current].state) {
case VoiceMessageStarting:
case VoiceMessageResuming:
case VoiceMessagePlaying:
_data[_current].state = VoiceMessageFinishing;
updateCurrentStarted();
startNow = false;
break;
case VoiceMessagePausing: _data[_current].state = VoiceMessageFinishing; startNow = false; break;
case VoiceMessagePaused: _data[_current].state = VoiceMessageStopped; break;
}
if (_data[_current].audio) {
emit loaderOnCancel(_data[_current].audio);
emit faderOnTimer();
}
}
int32 index = 0;
for (; index < AudioVoiceMsgSimultaneously; ++index) {
if (_data[index].audio == audio) {
_current = index;
break;
}
}
if (index == AudioVoiceMsgSimultaneously && ++_current >= AudioVoiceMsgSimultaneously) {
_current -= AudioVoiceMsgSimultaneously;
}
_data[_current].audio = audio;
_data[_current].fname = audio->already(true);
_data[_current].data = audio->data;
if (_data[_current].fname.isEmpty() && _data[_current].data.isEmpty()) {
_data[_current].state = VoiceMessageStopped;
onError(audio);
} else if (updateCurrentStarted(0)) {
_data[_current].state = startNow ? VoiceMessagePlaying : VoiceMessageStarting;
_data[_current].loading = true;
emit loaderOnStart(audio);
}
}
void VoiceMessages::pauseresume() {
QMutexLocker lock(&voicemsgsMutex);
switch (_data[_current].state) {
case VoiceMessagePausing:
case VoiceMessagePaused:
if (_data[_current].state == VoiceMessagePaused) {
updateCurrentStarted();
}
_data[_current].state = VoiceMessageResuming;
processContext();
alSourcePlay(_data[_current].source);
break;
case VoiceMessageStarting:
case VoiceMessageResuming:
case VoiceMessagePlaying:
_data[_current].state = VoiceMessagePausing;
updateCurrentStarted();
break;
case VoiceMessageFinishing: _data[_current].state = VoiceMessagePausing; break;
}
emit faderOnTimer();
}
void VoiceMessages::currentState(AudioData **audio, VoiceMessageState *state, int64 *position, int64 *duration, int32 *frequency) {
QMutexLocker lock(&voicemsgsMutex);
if (audio) *audio = _data[_current].audio;
if (state) *state = _data[_current].state;
if (position) *position = _data[_current].position;
if (duration) *duration = _data[_current].duration;
if (frequency) *frequency = _data[_current].frequency;
}
void VoiceMessages::clearStoppedAtStart(AudioData *audio) {
QMutexLocker lock(&voicemsgsMutex);
if (_data[_current].audio == audio && _data[_current].state == VoiceMessageStoppedAtStart) {
_data[_current].state = VoiceMessageStopped;
}
}
void VoiceMessages::processContext() {
_fader->processContext();
}
VoiceMessages *audioVoice() {
return voicemsgs;
}
VoiceMessagesFader::VoiceMessagesFader(QThread *thread) : _timer(this), _suspendFlag(false) {
moveToThread(thread);
_timer.moveToThread(thread);
_suspendTimer.moveToThread(thread);
_timer.setSingleShot(true);
connect(&_timer, SIGNAL(timeout()), this, SLOT(onTimer()));
_suspendTimer.setSingleShot(true);
connect(&_suspendTimer, SIGNAL(timeout()), this, SLOT(onSuspendTimer()));
connect(this, SIGNAL(stopSuspend()), this, SLOT(onSuspendTimerStop()), Qt::QueuedConnection);
}
void VoiceMessagesFader::onInit() {
}
void VoiceMessagesFader::onTimer() {
bool hasFading = false, hasPlaying = false;
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
VoiceMessages::Msg &m(voice->_data[i]);
if (m.state == VoiceMessageStopped || m.state == VoiceMessageStoppedAtStart || m.state == VoiceMessagePaused || !m.source) continue;
bool playing = false, fading = false;
ALint pos = 0;
ALint state = AL_INITIAL;
alGetSourcei(m.source, AL_SAMPLE_OFFSET, &pos);
alGetSourcei(m.source, AL_SOURCE_STATE, &state);
if (!_checkALError()) {
m.state = VoiceMessageStopped;
emit error(m.audio);
} else {
switch (m.state) {
case VoiceMessageFinishing:
case VoiceMessagePausing:
case VoiceMessageStarting:
case VoiceMessageResuming:
fading = true;
break;
case VoiceMessagePlaying:
playing = true;
break;
}
if (fading && (state == AL_PLAYING || !m.loading)) {
if (state != AL_PLAYING) {
fading = false;
if (m.source) {
alSourcef(m.source, AL_GAIN, 1);
alSourceStop(m.source);
}
m.state = VoiceMessageStopped;
emit audioStopped(m.audio);
} else if (1000 * (pos + m.skipStart - m.started) >= AudioFadeDuration * m.frequency) {
fading = false;
alSourcef(m.source, AL_GAIN, 1);
switch (m.state) {
case VoiceMessageFinishing: alSourceStop(m.source); m.state = VoiceMessageStopped; break;
case VoiceMessagePausing: alSourcePause(m.source); m.state = VoiceMessagePaused; break;
case VoiceMessageStarting:
case VoiceMessageResuming:
m.state = VoiceMessagePlaying;
playing = true;
break;
}
} else {
float64 newGain = 1000. * (pos + m.skipStart - m.started) / (AudioFadeDuration * m.frequency);
if (m.state == VoiceMessagePausing || m.state == VoiceMessageFinishing) {
newGain = 1. - newGain;
}
alSourcef(m.source, AL_GAIN, newGain);
}
} else if (playing && (state == AL_PLAYING || !m.loading)) {
if (state != AL_PLAYING) {
playing = false;
if (m.source) {
alSourceStop(m.source);
alSourcef(m.source, AL_GAIN, 1);
}
m.state = VoiceMessageStopped;
emit audioStopped(m.audio);
}
}
if (pos + m.skipStart - m.position >= AudioCheckPositionDelta) {
m.position = pos + m.skipStart;
emit playPositionUpdated(m.audio);
}
if (!m.loading && m.skipEnd > 0 && m.position + AudioPreloadSamples + m.skipEnd > m.duration) {
m.loading = true;
emit needToPreload(m.audio);
}
if (playing) hasPlaying = true;
if (fading) hasFading = true;
}
}
if (!hasPlaying) {
ALint state = AL_INITIAL;
alGetSourcei(notifySource, AL_SOURCE_STATE, &state);
if (_checkALError() && state == AL_PLAYING) {
hasPlaying = true;
}
}
if (hasFading) {
_timer.start(AudioFadeTimeout);
processContext();
} else if (hasPlaying) {
_timer.start(AudioCheckPositionTimeout);
processContext();
} else {
QMutexLocker lock(&_suspendMutex);
_suspendFlag = true;
_suspendTimer.start(AudioSuspendTimeout);
}
}
void VoiceMessagesFader::onSuspendTimer() {
QMutexLocker lock(&_suspendMutex);
if (_suspendFlag) {
alcSuspendContext(audioContext);
}
}
void VoiceMessagesFader::onSuspendTimerStop() {
if (_suspendTimer.isActive()) _suspendTimer.stop();
}
void VoiceMessagesFader::processContext() {
QMutexLocker lock(&_suspendMutex);
_suspendFlag = false;
emit stopSuspend();
alcProcessContext(audioContext);
}
class VoiceMessagesLoader {
public:
VoiceMessagesLoader(const QString &fname, const QByteArray &data) : fname(fname), data(data), dataPos(0) {
}
virtual ~VoiceMessagesLoader() {
}
bool check(const QString &fname, const QByteArray &data) {
return this->fname == fname && this->data.size() == data.size();
}
virtual bool open() = 0;
virtual int64 duration() = 0;
virtual int32 frequency() = 0;
virtual int32 format() = 0;
virtual void started() = 0;
virtual bool readMore(QByteArray &result, int64 &samplesAdded) = 0;
protected:
QString fname;
QByteArray data;
QFile f;
int32 dataPos;
bool openFile() {
if (data.isEmpty()) {
if (f.isOpen()) f.close();
f.setFileName(fname);
if (!f.open(QIODevice::ReadOnly)) {
LOG(("Audio Error: could not open file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(f.error()).arg(f.errorString()));
return false;
}
}
dataPos = 0;
return true;
}
};
class OggOpusLoader : public VoiceMessagesLoader {
public:
OggOpusLoader(const QString &fname, const QByteArray &data) : VoiceMessagesLoader(fname, data), file(0), pcm_offset(0), pcm_print_offset(0), prev_li(-1) {
}
bool open() {
if (!VoiceMessagesLoader::openFile()) {
return false;
}
OpusFileCallbacks cb = { &OggOpusLoader::_read_data, &OggOpusLoader::_seek_data, &OggOpusLoader::_tell_data, 0 };
if (data.isEmpty()) {
cb.read = &OggOpusLoader::_read_file;
cb.seek = &OggOpusLoader::_seek_file;
cb.tell = &OggOpusLoader::_tell_file;
}
int ret = 0;
file = op_open_callbacks(reinterpret_cast<void*>(this), &cb, 0, 0, &ret);
if (!file) {
LOG(("Audio Error: op_open_file failed for '%1', data size '%2', error code %3").arg(fname).arg(data.size()).arg(ret));
return false;
}
return true;
}
int64 duration() {
ogg_int64_t duration = op_pcm_total(file, -1);
if (duration < 0) {
LOG(("Audio Error: op_pcm_total failed to get full duration for '%1', data size '%2', error code %3").arg(fname).arg(data.size()).arg(duration));
}
return duration;
}
int32 frequency() {
return AudioVoiceMsgFrequency;
}
int32 format() {
return AL_FORMAT_STEREO16;
}
void started() {
pcm_offset = op_pcm_tell(file);
pcm_print_offset = pcm_offset - AudioVoiceMsgFrequency;
}
bool readMore(QByteArray &result, int64 &samplesAdded) {
DEBUG_LOG(("Audio Info: reading buffer for file '%1', data size '%2', current pcm_offset %3").arg(fname).arg(data.size()).arg(pcm_offset));
opus_int16 pcm[AudioVoiceMsgFrequency * AudioVoiceMsgChannels];
int ret = op_read_stereo(file, pcm, sizeof(pcm) / sizeof(*pcm));
if (ret < 0) {
LOG(("Audio Error: op_read_stereo failed, error code %1 (corrupted voice message?)").arg(ret));
return false;
}
int li = op_current_link(file);
if (li != prev_li) {
const OpusHead *head = op_head(file, li);
const OpusTags *tags = op_tags(file, li);
for (int32 ci = 0; ci < tags->comments; ++ci) {
const char *comment = tags->user_comments[ci];
if (opus_tagncompare("METADATA_BLOCK_PICTURE", 22, comment) == 0) {
OpusPictureTag pic;
int err = opus_picture_tag_parse(&pic, comment);
if (err >= 0) {
opus_picture_tag_clear(&pic);
}
}
}
if (!op_seekable(file)) {
pcm_offset = op_pcm_tell(file) - ret;
}
}
if (li != prev_li || pcm_offset >= pcm_print_offset + AudioVoiceMsgFrequency) {
pcm_print_offset = pcm_offset;
}
pcm_offset = op_pcm_tell(file);
if (!ret) {
DEBUG_LOG(("Audio Info: read completed"));
return false;
}
result.append((const char*)pcm, sizeof(*pcm) * ret * AudioVoiceMsgChannels);
prev_li = li;
samplesAdded += ret;
return true;
}
~OggOpusLoader() {
}
private:
OggOpusFile *file;
ogg_int64_t pcm_offset;
ogg_int64_t pcm_print_offset;
int prev_li;
static int _read_data(void *_stream, unsigned char *_ptr, int _nbytes) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
int32 nbytes = qMin(l->data.size() - l->dataPos, _nbytes);
if (nbytes <= 0) {
return 0;
}
memcpy(_ptr, l->data.constData() + l->dataPos, nbytes);
l->dataPos += nbytes;
return nbytes;
}
static int _seek_data(void *_stream, opus_int64 _offset, int _whence) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
int32 newPos = -1;
switch (_whence) {
case SEEK_SET: newPos = _offset; break;
case SEEK_CUR: newPos = l->dataPos + _offset; break;
case SEEK_END: newPos = l->data.size() + _offset; break;
}
if (newPos < 0 || newPos > l->data.size()) {
return -1;
}
l->dataPos = newPos;
return 0;
}
static opus_int64 _tell_data(void *_stream) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
return l->dataPos;
}
static int _read_file(void *_stream, unsigned char *_ptr, int _nbytes) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
return int(l->f.read((char*)(_ptr), _nbytes));
}
static int _seek_file(void *_stream, opus_int64 _offset, int _whence) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
switch (_whence) {
case SEEK_SET: return l->f.seek(_offset) ? 0 : -1;
case SEEK_CUR: return l->f.seek(l->f.pos() + _offset) ? 0 : -1;
case SEEK_END: return l->f.seek(l->f.size() + _offset) ? 0 : -1;
}
return -1;
}
static opus_int64 _tell_file(void *_stream) {
OggOpusLoader *l = reinterpret_cast<OggOpusLoader*>(_stream);
return l->f.pos();
}
};
class Mpg123Loader : public VoiceMessagesLoader {
public:
Mpg123Loader(const QString &fname, const QByteArray &data) : VoiceMessagesLoader(fname, data),
handle(0), opened(false), freq(AudioVoiceMsgFrequency), fmt(AL_FORMAT_STEREO16), channels(0) {
int ret = 0;
handle = mpg123_new(NULL, &ret);
if (!handle) {
LOG(("Audio Error: Unable to create mpg123 handle: %1\n").arg(mpg123_plain_strerror(ret)));
return;
}
mpg123_param(handle, MPG123_REMOVE_FLAGS, MPG123_FORCE_FLOAT, 0.); // not float
}
bool open() {
if (!VoiceMessagesLoader::openFile()) {
return false;
}
int res;
if (data.isEmpty()) {
res = mpg123_replace_reader_handle(handle, &Mpg123Loader::_read_file, &Mpg123Loader::_seek_file, 0);
} else {
res = mpg123_replace_reader_handle(handle, &Mpg123Loader::_read_data, &Mpg123Loader::_seek_data, 0);
}
if (res != MPG123_OK) {
LOG(("Audio Error: Unable to mpg123_replace_reader_handle() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(mpg123_strerror(handle)));
return false;
}
res = mpg123_open_handle(handle, reinterpret_cast<void*>(this));
if (res != MPG123_OK) {
LOG(("Audio Error: Unable to mpg123_open() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(mpg123_strerror(handle)));
return false;
}
opened = true;
int encoding = 0;
long rate = 0;
res = mpg123_getformat(handle, &rate, &channels, &encoding);
if (res != MPG123_OK) {
LOG(("Audio Error: Unable to mpg123_getformat() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(mpg123_strerror(handle)));
return false;
}
if (channels == 2) {
if (encoding == MPG123_ENC_SIGNED_16) {
fmt = AL_FORMAT_STEREO16;
} else if (encoding == MPG123_ENC_UNSIGNED_8) {
fmt = AL_FORMAT_STEREO8;
} else {
LOG(("Audio Error: Bad encoding for 2 channels in mpg123_getformat() file '%1', data size '%2', encoding %3").arg(fname).arg(data.size()).arg(encoding));
return false;
}
} else if (channels == 1) {
if (encoding == MPG123_ENC_SIGNED_16) {
fmt = AL_FORMAT_MONO16;
} else if (encoding == MPG123_ENC_UNSIGNED_8) {
fmt = AL_FORMAT_MONO8;
} else {
LOG(("Audio Error: Bad encoding for 1 channel in mpg123_getformat() file '%1', data size '%2', encoding %3").arg(fname).arg(data.size()).arg(encoding));
return false;
}
} else {
LOG(("Audio Error: Bad channels in mpg123_getformat() file '%1', data size '%2', channels %3").arg(fname).arg(data.size()).arg(channels));
return false;
}
freq = rate;
mpg123_format_none(handle);
res = mpg123_format(handle, freq, channels, encoding);
if (res != MPG123_OK) {
LOG(("Audio Error: Unable to mpg123_format() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(mpg123_strerror(handle)));
return false;
}
return true;
}
int64 duration() {
return mpg123_length(handle);
}
int32 frequency() {
return freq;
}
int32 format() {
return fmt;
}
void started() {
}
bool readMore(QByteArray &result, int64 &samplesAdded) {
int64 more_samples;
uchar buffer[sizeof(short) * AudioVoiceMsgFrequency * AudioVoiceMsgChannels];
size_t buffer_size = sizeof(buffer), done = 0;
int res = mpg123_read(handle, buffer, buffer_size, &done);
if (done) {
samplesAdded += done / (sizeof(short) * channels);
result.append((const char*)buffer, done);
}
if (res == MPG123_DONE) return false;
if (res != MPG123_OK) {
LOG(("Audio Error: Unable to mpg123_read() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(mpg123_strerror(handle)));
return false;
}
return true;
}
~Mpg123Loader() {
if (handle) {
if (opened) mpg123_close(handle);
mpg123_delete(handle);
}
}
private:
mpg123_handle *handle;
bool opened;
int32 freq, fmt;
int32 channels;
static ssize_t _read_data(void *_stream, void *_ptr, size_t _nbytes) {
Mpg123Loader *l = reinterpret_cast<Mpg123Loader*>(_stream);
int32 nbytes = qMin(l->data.size() - l->dataPos, int32(_nbytes));
if (nbytes <= 0) {
return 0;
}
memcpy(_ptr, l->data.constData() + l->dataPos, nbytes);
l->dataPos += nbytes;
return nbytes;
}
static off_t _seek_data(void *_stream, off_t _offset, int _whence) {
Mpg123Loader *l = reinterpret_cast<Mpg123Loader*>(_stream);
int32 newPos = -1;
switch (_whence) {
case SEEK_SET: newPos = _offset; break;
case SEEK_CUR: newPos = l->dataPos + _offset; break;
case SEEK_END: newPos = l->data.size() + _offset; break;
}
if (newPos < 0) {
return -1;
}
l->dataPos = newPos;
return l->dataPos;
}
static ssize_t _read_file(void *_stream, void *_ptr, size_t _nbytes) {
Mpg123Loader *l = reinterpret_cast<Mpg123Loader*>(_stream);
return ssize_t(l->f.read((char*)(_ptr), _nbytes));
}
static off_t _seek_file(void *_stream, off_t _offset, int _whence) {
Mpg123Loader *l = reinterpret_cast<Mpg123Loader*>(_stream);
switch (_whence) {
case SEEK_SET: return l->f.seek(_offset) ? l->f.pos() : -1;
case SEEK_CUR: return l->f.seek(l->f.pos() + _offset) ? l->f.pos() : -1;
case SEEK_END: return l->f.seek(l->f.size() + _offset) ? l->f.pos() : -1;
}
return -1;
}
};
class FAADMp4Loader : public VoiceMessagesLoader {
public:
FAADMp4Loader(const QString &fname, const QByteArray &data) : VoiceMessagesLoader(fname, data),
freq(AudioVoiceMsgFrequency), fmt(AL_FORMAT_STEREO16), len(0),
mp4f(0),
initial(true), useAacLength(false),
framesize(1024), timescale(AudioVoiceMsgFrequency),
trackId(-1), sampleId(0), samplesCount(0) {
}
bool open() {
if (!VoiceMessagesLoader::openFile()) {
return false;
}
if (data.isEmpty()) {
mp4cb.read = &FAADMp4Loader::_read_file;
mp4cb.seek = &FAADMp4Loader::_seek_file;
} else {
mp4cb.read = &FAADMp4Loader::_read_data;
mp4cb.seek = &FAADMp4Loader::_seek_data;
}
mp4cb.write = 0;
mp4cb.truncate = 0;
mp4cb.user_data = static_cast<void*>(this);
hDecoder = NeAACDecOpen();
config = NeAACDecGetCurrentConfiguration(hDecoder);
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 1; // Down matrix 5.1 to 2 channels
NeAACDecSetConfiguration(hDecoder, config);
mp4f = mp4ff_open_read(&mp4cb);
if (!mp4f) {
LOG(("Audio Error: Unable to mp4ff_open_read() file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
trackId = getAACTrack();
if (trackId < 0) {
LOG(("Audio Error: Unable to find correct AAC sound track in the MP4 file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
uchar *buffer = 0;
uint buffer_size = 0;
mp4ff_get_decoder_config(mp4f, trackId, &buffer, &buffer_size);
unsigned long samplerate = 0;
uchar channels = 2;
if (NeAACDecInit2(hDecoder, buffer, buffer_size, &samplerate, &channels) < 0) {
free(buffer);
LOG(("Audio Error: Error initializaing decoder library for file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
freq = samplerate;
switch (channels) {
case 1: fmt = AL_FORMAT_MONO16; break;
case 2: fmt = AL_FORMAT_STEREO16; break;
}
timescale = mp4ff_time_scale(mp4f, trackId);
if (buffer) {
if (NeAACDecAudioSpecificConfig(buffer, buffer_size, &mp4ASC) >= 0) {
if (mp4ASC.frameLengthFlag == 1) framesize = 960;
if (mp4ASC.sbr_present_flag == 1) framesize *= 2;
}
free(buffer);
}
samplesCount = mp4ff_num_samples(mp4f, trackId);
int32 f = 1024;
if (mp4ASC.sbr_present_flag == 1) {
f *= 2;
}
len = int64(samplesCount) * (f - 1);
return true;
}
int64 duration() {
return len;
}
int32 frequency() {
return freq;
}
int32 format() {
return fmt;
}
void started() {
}
bool readMore(QByteArray &result, int64 &samplesAdded) {
if (sampleId >= samplesCount) return false;
int32 dur = mp4ff_get_sample_duration(mp4f, trackId, sampleId);
uchar *buffer = 0;
uint32 buffer_size = 0;
if (!mp4ff_read_sample(mp4f, trackId, sampleId, &buffer, &buffer_size)) {
LOG(("Audio Error: Unable to mp4ff_read_sample() file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
void *sample_buffer = NeAACDecDecode(hDecoder, &frameInfo, buffer, buffer_size);
if (buffer) free(buffer);
if (sampleId == 0) dur = 0;
uint32 sample_count = frameInfo.samples;
if (!useAacLength && timescale == freq) {
sample_count = (uint32)(dur * frameInfo.channels);
if (sample_count > frameInfo.samples) {
sample_count = frameInfo.samples;
}
if (!initial && (sampleId < samplesCount / 2) && (sample_count != frameInfo.samples)) {
DEBUG_LOG(("Audio Warning: MP4 seems to have incorrect frame duration, using values from AAC data in file '%1', data size '%2'").arg(fname).arg(data.size()));
useAacLength = true;
sample_count = frameInfo.samples;
}
}
uint32 delay = 0;
if (initial && (sample_count < framesize * uint32(frameInfo.channels)) && (frameInfo.samples > sample_count)) {
delay = frameInfo.samples - sample_count;
}
switch (frameInfo.channels) {
case 1: fmt = AL_FORMAT_MONO16; break;
case 2: fmt = AL_FORMAT_STEREO16; break;
}
if (sample_count > 0) initial = false;
if (frameInfo.error) {
DEBUG_LOG(("Audio Warning: Read frame error in file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(frameInfo.error).arg(NeAACDecGetErrorMessage(frameInfo.error)));
} else if (sample_count > 0) {
samplesAdded += sample_count / frameInfo.channels;
result.append((const char*)sample_buffer + delay * sizeof(short), sample_count * sizeof(short)); // delay
}
++sampleId;
return true;
}
~FAADMp4Loader() {
NeAACDecClose(hDecoder);
mp4ff_close(mp4f);
}
private:
int32 freq, fmt;
int64 len;
NeAACDecHandle hDecoder;
NeAACDecConfigurationPtr config;
NeAACDecFrameInfo frameInfo;
mp4AudioSpecificConfig mp4ASC;
mp4ff_t *mp4f;
mp4ff_callback_t mp4cb;
bool initial, useAacLength;
uint32 framesize;
int32 timescale;
int32 trackId, sampleId, samplesCount;
int32 getAACTrack() {
int32 rc;
for (int32 i = 0, numTracks = mp4ff_total_tracks(mp4f); i < numTracks; i++)
{
uchar *buff = 0;
uint32 buff_size = 0;
mp4ff_get_decoder_config(mp4f, i, &buff, &buff_size);
if (buff) {
mp4AudioSpecificConfig mp4ASC;
rc = NeAACDecAudioSpecificConfig(buff, buff_size, &mp4ASC);
free(buff);
if (rc < 0) continue;
return i;
}
}
return -1;
}
static uint32_t _read_data(void *_stream, void *_ptr, uint32_t _nbytes) {
FAADMp4Loader *l = reinterpret_cast<FAADMp4Loader*>(_stream);
int32 nbytes = qMin(l->data.size() - l->dataPos, int32(_nbytes));
if (nbytes <= 0) {
return 0;
}
memcpy(_ptr, l->data.constData() + l->dataPos, nbytes);
l->dataPos += nbytes;
return nbytes;
}
static uint32_t _seek_data(void *_stream, uint64_t _offset) {
FAADMp4Loader *l = reinterpret_cast<FAADMp4Loader*>(_stream);
int32 newPos = _offset;
if (newPos < 0) {
return (uint32_t)-1;
}
l->dataPos = newPos;
return 0;
}
static uint32_t _read_file(void *_stream, void *_ptr, uint32_t _nbytes) {
FAADMp4Loader *l = reinterpret_cast<FAADMp4Loader*>(_stream);
return ssize_t(l->f.read((char*)(_ptr), _nbytes));
}
static uint32_t _seek_file(void *_stream, uint64_t _offset) {
FAADMp4Loader *l = reinterpret_cast<FAADMp4Loader*>(_stream);
return l->f.seek(_offset) ? 0 : (uint32_t)-1;
}
};
static const uint32 AVBlockSize = 4096; // 4Kb
static const AVSampleFormat _toFormat = AV_SAMPLE_FMT_S16;
static const int64_t _toChannelLayout = AV_CH_LAYOUT_STEREO;
static const int32 _toChannels = 2;
class FFMpegLoader : public VoiceMessagesLoader {
public:
FFMpegLoader(const QString &fname, const QByteArray &data) : VoiceMessagesLoader(fname, data),
freq(AudioVoiceMsgFrequency), fmt(AL_FORMAT_STEREO16),
sampleSize(2 * sizeof(short)), srcRate(AudioVoiceMsgFrequency), dstRate(AudioVoiceMsgFrequency),
maxResampleSamples(1024), resampleData(0), len(0),
ioBuffer(0), ioContext(0), fmtContext(0), codec(0), codecContext(0), streamId(0), frame(0), swrContext(0),
_opened(false) {
frame = av_frame_alloc();
}
bool open() {
if (!VoiceMessagesLoader::openFile()) {
return false;
}
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
ioBuffer = (uchar*)av_malloc(AVBlockSize);
if (data.isEmpty()) {
ioContext = avio_alloc_context(ioBuffer, AVBlockSize, 0, static_cast<void*>(this), &FFMpegLoader::_read_file, 0, &FFMpegLoader::_seek_file);
} else {
ioContext = avio_alloc_context(ioBuffer, AVBlockSize, 0, static_cast<void*>(this), &FFMpegLoader::_read_data, 0, &FFMpegLoader::_seek_data);
}
fmtContext = avformat_alloc_context();
fmtContext->pb = ioContext;
if (!fmtContext) {
LOG(("Audio Error: Unable to avformat_alloc_context for file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
int res = 0;
if ((res = avformat_open_input(&fmtContext, 0, 0, 0)) < 0) {
LOG(("Audio Error: Unable to avformat_open_input for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
_opened = true;
if ((res = avformat_find_stream_info(fmtContext, 0)) < 0) {
LOG(("Audio Error: Unable to avformat_find_stream_info for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
streamId = av_find_best_stream(fmtContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (streamId < 0) {
LOG(("Audio Error: Unable to av_find_best_stream for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(streamId).arg(av_make_error_string(err, sizeof(err), streamId)));
return false;
}
// Get a pointer to the codec context for the audio stream
codecContext = fmtContext->streams[streamId]->codec;
av_opt_set_int(codecContext, "refcounted_frames", 1, 0);
if ((res = avcodec_open2(codecContext, codec, 0)) < 0) {
LOG(("Audio Error: Unable to avcodec_open2 for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
freq = fmtContext->streams[streamId]->codec->sample_rate;
len = (fmtContext->streams[streamId]->duration * freq) / fmtContext->streams[streamId]->time_base.den;
uint64_t layout = fmtContext->streams[streamId]->codec->channel_layout;
inputFormat = fmtContext->streams[streamId]->codec->sample_fmt;
switch (layout) {
case AV_CH_LAYOUT_MONO:
switch (inputFormat) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_U8P: fmt = AL_FORMAT_MONO8; sampleSize = 1; break;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P: fmt = AL_FORMAT_MONO16; sampleSize = 2; break;
default:
sampleSize = -1; // convert needed
break;
}
break;
case AV_CH_LAYOUT_STEREO:
switch (inputFormat) {
case AV_SAMPLE_FMT_U8: fmt = AL_FORMAT_STEREO8; sampleSize = sizeof(short); break;
case AV_SAMPLE_FMT_S16: fmt = AL_FORMAT_STEREO16; sampleSize = 2 * sizeof(short); break;
default:
sampleSize = -1; // convert needed
break;
}
break;
default:
sampleSize = -1; // convert needed
break;
}
if (freq != 44100 && freq != 48000) {
sampleSize = -1; // convert needed
}
if (sampleSize < 0) {
swrContext = swr_alloc();
if (!swrContext) {
LOG(("Audio Error: Unable to swr_alloc for file '%1', data size '%2'").arg(fname).arg(data.size()));
return false;
}
int64_t src_ch_layout = layout, dst_ch_layout = _toChannelLayout;
srcRate = freq;
AVSampleFormat src_sample_fmt = inputFormat, dst_sample_fmt = _toFormat;
dstRate = (freq != 44100 && freq != 48000) ? AudioVoiceMsgFrequency : freq;
av_opt_set_int(swrContext, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swrContext, "in_sample_rate", srcRate, 0);
av_opt_set_sample_fmt(swrContext, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swrContext, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swrContext, "out_sample_rate", dstRate, 0);
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", dst_sample_fmt, 0);
if ((res = swr_init(swrContext)) < 0) {
LOG(("Audio Error: Unable to swr_init for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
sampleSize = _toChannels * sizeof(short);
freq = dstRate;
len = av_rescale_rnd(len, dstRate, srcRate, AV_ROUND_UP);
fmt = AL_FORMAT_STEREO16;
maxResampleSamples = av_rescale_rnd(AVBlockSize / sampleSize, dstRate, srcRate, AV_ROUND_UP);
if ((res = av_samples_alloc_array_and_samples(&resampleData, 0, _toChannels, maxResampleSamples, _toFormat, 0)) < 0) {
LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
}
return true;
}
int64 duration() {
return len;
}
int32 frequency() {
return freq;
}
int32 format() {
return fmt;
}
void started() {
}
bool readMore(QByteArray &result, int64 &samplesAdded) {
int res;
if ((res = av_read_frame(fmtContext, &avpkt)) < 0) {
if (res != AVERROR_EOF) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to av_read_frame() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
}
return false;
}
if (avpkt.stream_index == streamId) {
avcodec_get_frame_defaults(frame);
int got_frame = 0;
if ((res = avcodec_decode_audio4(codecContext, frame, &got_frame, &avpkt)) < 0) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to avcodec_decode_audio4() file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
if (got_frame) {
if (resampleData) { // convert needed
int64_t dstSamples = av_rescale_rnd(swr_get_delay(swrContext, srcRate) + frame->nb_samples, dstRate, srcRate, AV_ROUND_UP);
if (dstSamples > maxResampleSamples) {
maxResampleSamples = dstSamples;
av_free(resampleData[0]);
if ((res = av_samples_alloc(resampleData, 0, _toChannels, maxResampleSamples, _toFormat, 1)) < 0) {
resampleData[0] = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
}
if ((res = swr_convert(swrContext, resampleData, dstSamples, (const uint8_t**)frame->extended_data, frame->nb_samples)) < 0) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to swr_convert for file '%1', data size '%2', error %3, %4").arg(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
int32 resultLen = av_samples_get_buffer_size(0, _toChannels, res, _toFormat, 1);
result.append((const char*)resampleData[0], resultLen);
samplesAdded += resultLen / sampleSize;
} else {
result.append((const char*)frame->extended_data[0], frame->nb_samples * sampleSize);
samplesAdded += frame->nb_samples;
}
}
}
av_free_packet(&avpkt);
return true;
}
~FFMpegLoader() {
if (ioContext) av_free(ioContext);
if (codecContext) avcodec_close(codecContext);
if (swrContext) swr_free(&swrContext);
if (resampleData) {
if (resampleData[0]) {
av_freep(&resampleData[0]);
}
av_freep(&resampleData);
}
if (_opened) {
avformat_close_input(&fmtContext);
} else {
av_free(ioBuffer);
}
if (fmtContext) avformat_free_context(fmtContext);
av_frame_free(&frame);
}
private:
int32 freq, fmt, channels;
int32 sampleSize, srcRate, dstRate, maxResampleSamples;
uint8_t **resampleData;
int64 len;
uchar *ioBuffer;
AVIOContext *ioContext;
AVFormatContext *fmtContext;
AVCodec *codec;
AVCodecContext *codecContext;
AVPacket avpkt;
int32 streamId;
AVSampleFormat inputFormat;
AVFrame *frame;
SwrContext *swrContext;
bool _opened;
static int _read_data(void *opaque, uint8_t *buf, int buf_size) {
FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(opaque);
int32 nbytes = qMin(l->data.size() - l->dataPos, int32(buf_size));
if (nbytes <= 0) {
return 0;
}
memcpy(buf, l->data.constData() + l->dataPos, nbytes);
l->dataPos += nbytes;
return nbytes;
}
static int64_t _seek_data(void *opaque, int64_t offset, int whence) {
FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(opaque);
int32 newPos = -1;
switch (whence) {
case SEEK_SET: newPos = offset; break;
case SEEK_CUR: newPos = l->dataPos + offset; break;
case SEEK_END: newPos = l->data.size() + offset; break;
}
if (newPos < 0 || newPos > l->data.size()) {
return -1;
}
l->dataPos = newPos;
return l->dataPos;
}
static int _read_file(void *opaque, uint8_t *buf, int buf_size) {
FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(opaque);
return ssize_t(l->f.read((char*)(buf), buf_size));
}
static int64_t _seek_file(void *opaque, int64_t offset, int whence) {
FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(opaque);
switch (whence) {
case SEEK_SET: return l->f.seek(offset) ? l->f.pos() : -1;
case SEEK_CUR: return l->f.seek(l->f.pos() + offset) ? l->f.pos() : -1;
case SEEK_END: return l->f.seek(l->f.size() + offset) ? l->f.pos() : -1;
}
return -1;
}
};
VoiceMessagesLoaders::VoiceMessagesLoaders(QThread *thread) {
moveToThread(thread);
}
VoiceMessagesLoaders::~VoiceMessagesLoaders() {
for (Loaders::iterator i = _loaders.begin(), e = _loaders.end(); i != e; ++i) {
delete i.value();
}
_loaders.clear();
}
void VoiceMessagesLoaders::onInit() {
}
void VoiceMessagesLoaders::onStart(AudioData *audio) {
Loaders::iterator i = _loaders.find(audio);
if (i != _loaders.end()) {
delete (*i);
_loaders.erase(i);
}
onLoad(audio);
}
void VoiceMessagesLoaders::loadError(Loaders::iterator i) {
emit error(i.key());
delete (*i);
_loaders.erase(i);
}
void VoiceMessagesLoaders::onLoad(AudioData *audio) {
bool started = false;
int32 audioindex = -1;
VoiceMessagesLoader *l = 0;
Loaders::iterator j = _loaders.end();
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
VoiceMessages::Msg &m(voice->_data[i]);
if (m.audio != audio || !m.loading) continue;
audioindex = i;
j = _loaders.find(audio);
if (j != _loaders.end() && !j.value()->check(m.fname, m.data)) {
delete j.value();
_loaders.erase(j);
j = _loaders.end();
}
if (j == _loaders.end()) {
QByteArray header = m.data.mid(0, 8);
if (header.isEmpty()) {
QFile f(m.fname);
if (!f.open(QIODevice::ReadOnly)) {
LOG(("Audio Error: could not open file '%1'").arg(m.fname));
m.state = VoiceMessageStoppedAtStart;
emit error(audio);
return;
}
header = f.read(8);
}
if (header.size() < 8) {
LOG(("Audio Error: could not read header from file '%1', data size %2").arg(m.fname).arg(m.data.isEmpty() ? QFileInfo(m.fname).size() : m.data.size()));
m.state = VoiceMessageStoppedAtStart;
emit error(audio);
return;
}
uint32 mpegHead = (uint32(uchar(header.at(0))) << 24) | (uint32(uchar(header.at(1))) << 16) | (uint32(uchar(header.at(2))) << 8) | uint32(uchar(header.at(3)));
bool validMpegHead = ((mpegHead & _test_HDR_SYNC) == _test_HDR_SYNC) && !!(_test_HDR_LAYER_VAL(mpegHead)) && (_test_HDR_BITRATE_VAL(mpegHead) != 0x0F) && (_test_HDR_SAMPLERATE_VAL(mpegHead) != 0x03);
if (true) {
j = _loaders.insert(audio, new FFMpegLoader(m.fname, m.data));
} else if (header.at(0) == 'O' && header.at(1) == 'g' && header.at(2) == 'g' && header.at(3) == 'S') {
j = _loaders.insert(audio, new OggOpusLoader(m.fname, m.data));
} else if (header.at(4) == 'f' && header.at(5) == 't' && header.at(6) == 'y' && header.at(7) == 'p') {
j = _loaders.insert(audio, new FAADMp4Loader(m.fname, m.data));
} else if ((header.at(0) == 'I' && header.at(1) == 'D' && header.at(2) == '3') || validMpegHead) {
if (m.data.isEmpty()) {
QFile f(m.fname);
f.open(QIODevice::ReadOnly);
m.data = f.readAll();
m.fname = QString();
}
j = _loaders.insert(audio, new Mpg123Loader(m.fname, m.data));
} else {
LOG(("Audio Error: could not guess file format from header, header %1 file '%2', data size %3").arg(mb(header.constData(), header.size()).str()).arg(m.fname).arg(m.data.isEmpty() ? QFileInfo(m.fname).size() : m.data.size()));
m.state = VoiceMessageStoppedAtStart;
emit error(audio);
return;
}
l = j.value();
int ret;
if (!l->open()) {
m.state = VoiceMessageStoppedAtStart;
return loadError(j);
}
int64 duration = l->duration();
if (duration <= 0) {
m.state = VoiceMessageStoppedAtStart;
return loadError(j);
}
m.duration = duration;
m.frequency = l->frequency();
if (!m.frequency) m.frequency = AudioVoiceMsgFrequency;
m.skipStart = 0;
m.skipEnd = duration;
m.position = 0;
m.started = 0;
started = true;
} else {
if (!m.skipEnd) continue;
l = j.value();
}
break;
}
}
if (j == _loaders.end()) {
LOG(("Audio Error: trying to load part of audio, that is not playing at the moment"));
emit error(audio);
return;
}
if (started) {
l->started();
}
bool finished = false;
QByteArray result;
int64 samplesAdded = 0, frequency = l->frequency(), format = l->format();
while (result.size() < AudioVoiceMsgBufferSize) {
if (!l->readMore(result, samplesAdded)) {
finished = true;
break;
}
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || !l->check(m.fname, m.data)) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
}
}
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || !l->check(m.fname, m.data)) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
if (started) {
if (m.source) {
alSourceStop(m.source);
for (int32 i = 0; i < 3; ++i) {
if (m.samplesCount[i]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + i);
m.samplesCount[i] = 0;
}
}
m.nextBuffer = 0;
}
}
if (samplesAdded) {
if (!m.source) {
alGenSources(1, &m.source);
alSourcef(m.source, AL_PITCH, 1.f);
alSourcef(m.source, AL_GAIN, 1.f);
alSource3f(m.source, AL_POSITION, 0, 0, 0);
alSource3f(m.source, AL_VELOCITY, 0, 0, 0);
alSourcei(m.source, AL_LOOPING, 0);
}
if (!m.buffers[m.nextBuffer]) alGenBuffers(3, m.buffers);
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
if (m.samplesCount[m.nextBuffer]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipStart += m.samplesCount[m.nextBuffer];
}
m.samplesCount[m.nextBuffer] = samplesAdded;
alBufferData(m.buffers[m.nextBuffer], format, result.constData(), result.size(), frequency);
alSourceQueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipEnd -= samplesAdded;
m.nextBuffer = (m.nextBuffer + 1) % 3;
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
} else {
finished = true;
}
if (finished) {
m.skipEnd = 0;
m.duration = m.skipStart + m.samplesCount[0] + m.samplesCount[1] + m.samplesCount[2];
}
m.loading = false;
if (m.state == VoiceMessageResuming || m.state == VoiceMessagePlaying || m.state == VoiceMessageStarting) {
ALint state = AL_INITIAL;
alGetSourcei(m.source, AL_SOURCE_STATE, &state);
if (_checkALError()) {
if (state != AL_PLAYING) {
voice->processContext();
alSourcePlay(m.source);
emit needToCheck();
}
}
}
}
void VoiceMessagesLoaders::onCancel(AudioData *audio) {
Loaders::iterator i = _loaders.find(audio);
if (i != _loaders.end()) {
delete (*i);
_loaders.erase(i);
}
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
VoiceMessages::Msg &m(voice->_data[i]);
if (m.audio == audio) {
m.loading = false;
}
}
}
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