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ao: queue audio data to avoid deep copy for all kinds of backends

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internal queue in AudioRenderer can ensure the data is not destoryed
until it is rendered. so no data copy happens even for backends like
opensl
This commit is contained in:
wang-bin
2016-09-03 22:22:22 +08:00
parent 1537a5d1c2
commit e826c39df0
2 changed files with 29 additions and 50 deletions
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src/output/audio/AudioOutput.cpp
+27 -31
View File
@@ -152,9 +152,10 @@ public:
}
struct FrameInfo {
FrameInfo(qreal t = 0, int s = 0) : timestamp(t), data_size(s) {}
FrameInfo(const QByteArray& d = QByteArray(), qreal t = 0, int us = 0) : timestamp(t), duration(us), data(d) {}
qreal timestamp;
int data_size; // TODO: size_t
int duration; // in us
QByteArray data;
};
void resetStatus() {
@@ -178,7 +179,6 @@ public:
qreal speed;
AudioFormat format;
AudioFormat requested;
QByteArray data;
//AudioFrame audio_frame;
quint32 nb_buffers;
qint32 buffer_samples;
@@ -214,14 +214,13 @@ AudioOutputPrivate::~AudioOutputPrivate()
void AudioOutputPrivate::playInitialData()
{
if (!backend)
return;
const char c = (format.sampleFormat() == AudioFormat::SampleFormat_Unsigned8
|| format.sampleFormat() == AudioFormat::SampleFormat_Unsigned8Planar)
? 0x80 : 0;
for (quint32 i = 0; i < nb_buffers; ++i) {
backend->write(QByteArray(buffer_samples*format.bytesPerSample(), c)); // fill silence byte, not always 0. AudioFormat.silenceByte
frame_infos.push_back(FrameInfo(0, 1*format.bytesPerSample())); // initial data can be small (1 instead of buffer_samples)
const QByteArray data(backend->buffer_size, c);
backend->write(data); // fill silence byte, not always 0. AudioFormat.silenceByte
frame_infos.push_back(FrameInfo(data, 0, 0)); // initial data can be small (1 instead of buffer_samples)
}
backend->play();
}
@@ -421,33 +420,31 @@ bool AudioOutput::receiveData(const QByteArray &data, qreal pts)
DPTR_D(AudioOutput);
if (isPaused())
return false;
d.data = data;
QByteArray queue_data(data);
if (isMute() && d.sw_mute) {
char s = 0;
if (d.format.isUnsigned() && !d.format.isFloat())
s = 1<<((d.format.bytesPerSample() << 3)-1);
d.data.fill(s);
queue_data.fill(s);
} else {
if (!qFuzzyCompare(volume(), (qreal)1.0)
&& d.sw_volume
&& d.scale_samples
) {
// TODO: af_volume needs samples_align to get nb_samples
const int nb_samples = d.data.size()/d.format.bytesPerSample();
quint8 *dst = (quint8*)d.data.constData();
const int nb_samples = queue_data.size()/d.format.bytesPerSample();
quint8 *dst = (quint8*)queue_data.constData();
d.scale_samples(dst, dst, nb_samples, d.volume_i, volume());
}
}
// wait after all data processing finished to reduce time error
if (!waitForNextBuffer()) {
if (!waitForNextBuffer()) { // TODO: wait or not parameter, set by user (async)
qWarning("ao backend maybe not open");
d.resetStatus();
return false;
}
d.frame_infos.push_back(AudioOutputPrivate::FrameInfo(pts, data.size()));
if (!d.backend || !isOpen())
return false;
return d.backend->write(d.data);
d.frame_infos.push_back(AudioOutputPrivate::FrameInfo(queue_data, pts, d.format.durationForBytes(queue_data.size())));
return d.backend->write(queue_data); // backend is not null here
}
AudioFormat AudioOutput::setAudioFormat(const AudioFormat& format)
@@ -612,18 +609,17 @@ AudioOutput::DeviceFeatures AudioOutput::supportedDeviceFeatures() const
return d.backend->supportedFeatures();
}
bool AudioOutput::waitForNextBuffer()
bool AudioOutput::waitForNextBuffer() // parameter bool wait: if no wait and no next buffer, return false
{
DPTR_D(AudioOutput);
if (d.frame_infos.empty())
return true;
if (!d.backend)
return false;
//don't return even if we can add buffer because we don't know when a buffer is processed and we have /to update dequeue index
// openal need enqueue to a dequeued buffer! why sl crash
bool no_wait = false;//d.canAddBuffer();
const AudioOutputBackend::BufferControl f = d.backend->bufferControl();
int remove = 0;
const AudioOutputPrivate::FrameInfo &fi(d.frame_infos.front());
if (f & AudioOutputBackend::Blocking) {
remove = 1;
} else if (f & AudioOutputBackend::CountCallback) {
@@ -636,10 +632,10 @@ bool AudioOutput::waitForNextBuffer()
d.processed_remain = d.backend->getWritableBytes();
if (d.processed_remain < 0)
return false;
const int next = d.frame_infos.front().data_size;
const int next = fi.data.size();
//qDebug("remain: %d-%d, size: %d, next: %d", processed, d.processed_remain, d.data.size(), next);
qint64 last_wait = 0LL;
while (d.processed_remain - processed < next || d.processed_remain < d.data.size()) { //implies next > 0
while (d.processed_remain - processed < next || d.processed_remain < fi.data.size()) { //implies next > 0
const qint64 us = d.format.durationForBytes(next - (d.processed_remain - processed));
d.uwait(us);
d.processed_remain = d.backend->getWritableBytes();
@@ -661,13 +657,13 @@ bool AudioOutput::waitForNextBuffer()
last_wait = us;
}
processed = d.processed_remain - processed;
d.processed_remain -= d.data.size(); //ensure d.processed_remain later is greater
d.processed_remain -= fi.data.size(); //ensure d.processed_remain later is greater
remove = -processed; // processed_this_period
} else if (f & AudioOutputBackend::PlayedBytes) {
d.processed_remain = d.backend->getPlayedBytes();
const int next = d.frame_infos.front().data_size;
const int next = fi.data.size();
// TODO: avoid always 0
// TODO: compare processed_remain with d.data.size because input chuncks can be in different sizes
// TODO: compare processed_remain with fi.data.size because input chuncks can be in different sizes
while (!no_wait && d.processed_remain < next) {
const qint64 us = d.format.durationForBytes(next - d.processed_remain);
if (us < 1000LL)
@@ -689,7 +685,7 @@ bool AudioOutput::waitForNextBuffer()
qint64 us = 0;
while (!no_wait && c < 1) {
if (us <= 0)
us = d.format.durationForBytes(d.frame_infos.front().data_size);
us = fi.duration;
#if AO_USE_TIMER
elapsed = d.timer.restart();
if (elapsed > 0 && us > elapsed*1000LL)
@@ -709,9 +705,9 @@ bool AudioOutput::waitForNextBuffer()
if (processed < 0)
processed += bufferSizeTotal();
d.play_pos = s;
const int next = d.frame_infos.front().data_size;
const int next = fi.data.size();
int writable_size = d.processed_remain + processed;
while (!no_wait && (/*processed < next ||*/ writable_size < d.data.size()) && next > 0) {
while (!no_wait && (/*processed < next ||*/ writable_size < fi.data.size()) && next > 0) {
const qint64 us = d.format.durationForBytes(next - writable_size);
d.uwait(us);
s = d.backend->getOffsetByBytes();
@@ -722,7 +718,7 @@ bool AudioOutput::waitForNextBuffer()
d.play_pos = s;
}
d.processed_remain += processed;
d.processed_remain -= d.data.size(); //ensure d.processed_remain later is greater
d.processed_remain -= fi.data.size(); //ensure d.processed_remain later is greater
remove = -processed;
} else if (f & AudioOutputBackend::OffsetIndex) {
int n = d.backend->getOffset();
@@ -733,7 +729,7 @@ bool AudioOutput::waitForNextBuffer()
// TODO: timer
// TODO: avoid always 0
while (!no_wait && processed < 1) {
d.uwait(d.format.durationForBytes(d.frame_infos.front().data_size));
d.uwait(fi.duration);
n = d.backend->getOffset();
processed = n - d.play_pos;
if (processed < 0)
@@ -746,7 +742,7 @@ bool AudioOutput::waitForNextBuffer()
return false;
}
if (remove < 0) {
int next = d.frame_infos.front().data_size;
int next = fi.data.size();
int free_bytes = -remove;//d.processed_remain;
while (free_bytes >= next && next > 0) {
free_bytes -= next;
@@ -755,7 +751,7 @@ bool AudioOutput::waitForNextBuffer()
break;
}
d.frame_infos.pop_front();
next = d.frame_infos.front().data_size;
next = d.frame_infos.front().data.size();
}
//qDebug("remove: %d, unremoved bytes < %d, writable_bytes: %d", remove, free_bytes, d.processed_remain);
return true;
src/output/audio/AudioOutputOpenSL.cpp
+2 -19
View File
@@ -94,10 +94,6 @@ private:
SLint32 m_streamType;
quint32 buffers_queued;
QSemaphore sem;
// Enqueue does not copy data. We MUST keep the data until it is played out
int queue_data_write;
QByteArray queue_data;
};
typedef AudioOutputOpenSL AudioOutputBackendOpenSL;
@@ -193,7 +189,6 @@ AudioOutputOpenSL::AudioOutputOpenSL(QObject *parent)
, m_sl_step(0)
, m_streamType(-1)
, buffers_queued(0)
, queue_data_write(0)
{
#ifdef Q_OS_ANDROID
char v[PROP_VALUE_MAX+1];
@@ -255,7 +250,6 @@ void AudioOutputOpenSL::onCallback()
bool AudioOutputOpenSL::open()
{
queue_data.resize(buffer_size*buffer_count);
SLDataLocator_BufferQueue bufferQueueLocator = { SL_DATALOCATOR_BUFFERQUEUE, (SLuint32)buffer_count };
SLDataFormat_PCM_EX pcmFormat = audioFormatToSL(format);
SLDataSource audioSrc = { &bufferQueueLocator, &pcmFormat };
@@ -348,8 +342,6 @@ bool AudioOutputOpenSL::close()
m_playItf = NULL;
m_volumeItf = NULL;
m_bufferQueueItf = NULL;
queue_data.clear();
queue_data_write = 0;
return true;
}
@@ -357,24 +349,15 @@ bool AudioOutputOpenSL::write(const QByteArray& data)
{
if (bufferControl() & CountCallback)
sem.acquire();
const int s = qMin(queue_data.size() - queue_data_write, data.size());
// assume data.size() <= buffer_size. It's true in QtAV
if (s < data.size())
queue_data_write = 0;
memcpy((char*)queue_data.constData() + queue_data_write, data.constData(), data.size());
//qDebug("enqueue %p, queue_data_write: %d/%d available:%d", data.constData(), queue_data_write, queue_data.size(), sem.available());
#ifdef Q_OS_ANDROID
if (m_android)
SL_ENSURE((*m_bufferQueueItf_android)->Enqueue(m_bufferQueueItf_android, queue_data.constData() + queue_data_write, data.size()), false);
SL_ENSURE((*m_bufferQueueItf_android)->Enqueue(m_bufferQueueItf_android, data.constData(), data.size()), false);
else
SL_ENSURE((*m_bufferQueueItf)->Enqueue(m_bufferQueueItf, queue_data.constData() + queue_data_write, data.size()), false);
#else
SL_ENSURE((*m_bufferQueueItf)->Enqueue(m_bufferQueueItf, queue_data.constData() + queue_data_write, data.size()), false);
SL_ENSURE((*m_bufferQueueItf)->Enqueue(m_bufferQueueItf, data.constData(), data.size()), false);
#endif
buffers_queued++;
queue_data_write += data.size();
if (queue_data_write == queue_data.size())
queue_data_write = 0;
return true;
}