/*
*TODO:
* if output is null or dummy, the use duration to wait
*/
void AudioThread::run()
{
DPTR_D(AudioThread);
//No decoder or output. No audio output is ok, just display picture
if (!d.dec || !d.dec->isAvailable() || !d.outputSet)
return;
resetState();
Q_ASSERT(d.clock != 0);
AudioDecoder *dec = static_cast<AudioDecoder*>(d.dec);
AudioOutput *ao = 0;
// first() is not null even if list empty
if (!d.outputSet->outputs().isEmpty())
ao = static_cast<AudioOutput*>(d.outputSet->outputs().first()); //TODO: not here
d.init();
//TODO: bool need_sync in private class
bool is_external_clock = d.clock->clockType() == AVClock::ExternalClock;
Packet pkt;
while (!d.stop) {
processNextTask();
//TODO: why put it at the end of loop then playNextFrame() not work?
if (tryPause()) { //DO NOT continue, or playNextFrame() will fail
if (d.stop)
break; //the queue is empty and may block. should setBlocking(false) wake up cond empty?
} else {
if (isPaused())
continue;
}
if (d.packets.isEmpty() && !d.stop) {
d.stop = d.demux_end;
}
if (d.stop) {
qDebug("audio thread stop before take packet");
break;
}
if (!pkt.isValid()) {
pkt = d.packets.take(); //wait to dequeue
}
if (!pkt.isValid()) {
qDebug("Invalid packet! flush audio codec context!!!!!!!! audio queue size=%d", d.packets.size());
dec->flush();
continue;
}
bool skip_render = pkt.pts < d.render_pts0;
// audio has no key frame, skip rendering equals to skip decoding
if (skip_render) {
d.clock->updateValue(pkt.pts);
/*
* audio may be too fast than video if skip without sleep
* a frame is about 20ms. sleep time must be << frame time
*/
qreal a_v = pkt.pts - d.clock->videoPts();
//qDebug("skip audio decode at %f/%f v=%f a-v=%f", pkt.pts, d.render_pts0, d.clock->videoPts(), a_v);
if (a_v > 0)
msleep(qMin((ulong)300, ulong(a_v*1000.0)));
else
msleep(2);
pkt = Packet(); //mark invalid to take next
continue;
}
d.render_pts0 = 0;
if (is_external_clock) {
d.delay = pkt.pts - d.clock->value();
/*
*after seeking forward, a packet may be the old, v packet may be
*the new packet, then the d.delay is very large, omit it.
*TODO: 1. how to choose the value
* 2. use last delay when seeking
*/
if (qAbs(d.delay) < 2.718) {
if (d.delay < -kSyncThreshold) { //Speed up. drop frame?
//continue;
}
while (d.delay > kSyncThreshold) { //Slow down
//d.delay_cond.wait(&d.mutex, d.delay*1000); //replay may fail. why?
//qDebug("~~~~~wating for %f msecs", d.delay*1000);
usleep(kSyncThreshold * 1000000UL);
if (d.stop)
d.delay = 0;
else
d.delay -= kSyncThreshold;
}
if (d.delay > 0)
usleep(d.delay * 1000000UL);
} else { //when to drop off?
if (d.delay > 0) {
msleep(64);
} else {
//audio packet not cleaned up?
continue;
}
}
} else {
d.clock->updateValue(pkt.pts);
}
//DO NOT decode and convert if ao is not available or mute!
bool has_ao = ao && ao->isAvailable();
//if (!has_ao) {//do not decode?
// TODO: move resampler to AudioFrame, like VideoFrame does
if (has_ao && dec->resampler()) {
if (dec->resampler()->speed() != ao->speed()
|| dec->resampler()->outAudioFormat() != ao->audioFormat()) {
//resample later to ensure thread safe. TODO: test
if (d.resample) {
qDebug("decoder set speed: %.2f", ao->speed());
dec->resampler()->setOutAudioFormat(ao->audioFormat());
dec->resampler()->setSpeed(ao->speed());
dec->resampler()->prepare();
d.resample = false;
} else {
d.resample = true;
}
}
} else {
if (dec->resampler() && dec->resampler()->speed() != d.clock->speed()) {
if (d.resample) {
qDebug("decoder set speed: %.2f", d.clock->speed());
dec->resampler()->setSpeed(d.clock->speed());
dec->resampler()->prepare();
d.resample = false;
} else {
d.resample = true;
}
}
}
if (d.stop) {
qDebug("audio thread stop before decode()");
break;
}
QMutexLocker locker(&d.mutex);
Q_UNUSED(locker);
if (!dec->decode(pkt.data)) {
qWarning("Decode audio failed");
qreal dt = pkt.pts - d.last_pts;
if (dt > 0.618 || dt < 0) {
dt = 0;
}
//qDebug("sleep %f", dt);
//TODO: avoid acummulative error. External clock?
msleep((unsigned long)(dt*1000.0));
pkt = Packet();
d.last_pts = d.clock->value(); //not pkt.pts! the delay is updated!
continue;
}
QByteArray decoded(dec->data());
int decodedSize = decoded.size();
int decodedPos = 0;
qreal delay = 0;
//AudioFormat.durationForBytes() calculates int type internally. not accurate
AudioFormat &af = dec->resampler()->inAudioFormat();
qreal byte_rate = af.bytesPerSecond();
while (decodedSize > 0) {
if (d.stop) {
qDebug("audio thread stop after decode()");
break;
}
// TODO: set to format.bytesPerFrame()*1024?
const int chunk = qMin(decodedSize, has_ao ? ao->bufferSize() : 1024*4);//int(max_len*byte_rate));
//AudioFormat.bytesForDuration
const qreal chunk_delay = (qreal)chunk/(qreal)byte_rate;
pkt.pts += chunk_delay;
QByteArray decodedChunk(chunk, 0); //volume == 0 || mute
if (has_ao) {
//TODO: volume filter and other filters!!!
if (!ao->isMute()) {
decodedChunk = QByteArray::fromRawData(decoded.constData() + decodedPos, chunk);
qreal vol = ao->volume();
if (vol != 1.0) {
int len = decodedChunk.size()/ao->audioFormat().bytesPerSample();
switch (ao->audioFormat().sampleFormat()) {
case AudioFormat::SampleFormat_Unsigned8:
case AudioFormat::SampleFormat_Unsigned8Planar: {
quint8 *data = (quint8*)decodedChunk.data(); //TODO: other format?
for (int i = 0; i < len; data[i++] *= vol) {}
}
break;
case AudioFormat::SampleFormat_Signed16:
case AudioFormat::SampleFormat_Signed16Planar: {
qint16 *data = (qint16*)decodedChunk.data(); //TODO: other format?
for (int i = 0; i < len; data[i++] *= vol) {}
}
break;
case AudioFormat::SampleFormat_Signed32:
case AudioFormat::SampleFormat_Signed32Planar: {
qint32 *data = (qint32*)decodedChunk.data(); //TODO: other format?
for (int i = 0; i < len; data[i++] *= vol) {}
}
break;
case AudioFormat::SampleFormat_Float:
case AudioFormat::SampleFormat_FloatPlanar: {
float *data = (float*)decodedChunk.data(); //TODO: other format?
for (int i = 0; i < len; data[i++] *= vol) {}
}
break;
case AudioFormat::SampleFormat_Double:
case AudioFormat::SampleFormat_DoublePlanar: {
double *data = (double*)decodedChunk.data(); //TODO: other format?
for (int i = 0; i < len; data[i++] *= vol) {}
}
break;
default:
break;
}
}
}
ao->waitForNextBuffer();
ao->receiveData(decodedChunk, pkt.pts);
ao->play();
d.clock->updateValue(ao->timestamp());
} else {
d.clock->updateDelay(delay += chunk_delay);
/*
* why need this even if we add delay? and usleep sounds weird
* the advantage is if no audio device, the play speed is ok too
* So is portaudio blocking the thread when playing?
*/
static bool sWarn_no_ao = true; //FIXME: no warning when replay. warn only once
if (sWarn_no_ao) {
qDebug("Audio output not available! msleep(%lu)", (unsigned long)((qreal)chunk/(qreal)byte_rate * 1000));
sWarn_no_ao = false;
}
//TODO: avoid acummulative error. External clock?
msleep((unsigned long)(chunk_delay * 1000.0));
}
decodedPos += chunk;
decodedSize -= chunk;
}
int undecoded = dec->undecodedSize();
if (undecoded > 0) {
pkt.data.remove(0, pkt.data.size() - undecoded);
} else {
pkt = Packet();
}
d.last_pts = d.clock->value(); //not pkt.pts! the delay is updated!
}
qDebug("Audio thread stops running...");
}
//TODO: if output is null or dummy, the use duration to wait
void AudioThread::run()
{
DPTR_D(AudioThread);
//No decoder or output. No audio output is ok, just display picture
if (!d.dec || !d.dec->isAvailable())
return;
resetState();
Q_ASSERT(d.clock != 0);
AudioDecoder *dec = static_cast<AudioDecoder*>(d.dec);
AudioOutput *ao = static_cast<AudioOutput*>(d.writer);
int sample_rate = dec->codecContext()->sample_rate;
int channels = dec->codecContext()->channels;
int csf = channels * sample_rate * sizeof(float);
static const double max_len = 0.02;
d.last_pts = 0;
while (!d.stop) {
//TODO: why put it at the end of loop then playNextFrame() not work?
if (tryPause()) { //DO NOT continue, or playNextFrame() will fail
if (d.stop)
break; //the queue is empty and may block. should setBlocking(false) wake up cond empty?
}
QMutexLocker locker(&d.mutex);
Q_UNUSED(locker);
if (d.packets.isEmpty() && !d.stop) {
d.stop = d.demux_end;
if (d.stop) {
break;
}
}
Packet pkt = d.packets.take(); //wait to dequeue
if (!pkt.isValid()) {
qDebug("Invalid packet! flush audio codec context!!!!!!!!");
dec->flush();
continue;
}
d.clock->updateValue(pkt.pts);
//DO NOT decode and convert if ao is not available or mute!
if (dec->decode(pkt.data)) {
QByteArray decoded(dec->data());
int decodedSize = decoded.size();
int decodedPos = 0;
qreal delay =0;
while (decodedSize > 0) {
int chunk = qMin(decodedSize, int(max_len*csf));
d.clock->updateDelay(delay += (qreal)chunk/(qreal)csf);
QByteArray decodedChunk(chunk, 0); //volume == 0 || mute
if (ao && ao->isAvailable()) {
if (!ao->isMute()) {
decodedChunk = QByteArray::fromRawData(decoded.constData() + decodedPos, chunk);
qreal vol = ao->volume();
if (vol != 1.0) {
int len = decodedChunk.size()/sizeof(float); //TODO: why???
float *data = (float*)decodedChunk.data();
for (int i = 0; i < len; ++i)
data[i] *= vol;
}
}
ao->writeData(decodedChunk);
} else {
/*
* why need this even if we add delay? and usleep sounds weird
* the advantage is if no audio device, the play speed is ok too
* So is portaudio blocking the thread when playing?
*/
static bool sWarn_no_ao = true; //FIXME: no warning when replay. warn only once
if (sWarn_no_ao) {
qDebug("Audio output not available! msleep(%lu)", (unsigned long)((qreal)chunk/(qreal)csf * 1000));
sWarn_no_ao = false;
}
//TODO: avoid acummulative error. External clock?
msleep((unsigned long)((qreal)chunk/(qreal)csf * 1000.0));
}
decodedPos += chunk;
decodedSize -= chunk;
}
} else {
//qWarning("Decode audio failed");
qreal dt = pkt.pts - d.last_pts;
if (abs(dt) > 0.618 || dt < 0) {
dt = 0;
}
//qDebug("sleep %f", dt);
//TODO: avoid acummulative error. External clock?
msleep((unsigned long)(dt*1000.0));
}
d.last_pts = d.clock->value(); //not pkt.pts! the delay is updated!
}
qDebug("Audio thread stops running...");
}