long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
int64_t insertTime;
// NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
// Bug 996162
// callback tells us cubeb succeeded initializing
if (mState == STARTED) {
// For low-latency streams, we want to minimize any built-up data when
// we start getting callbacks.
// Simple version - contract on first callback only.
if (mLatencyRequest == LowLatency) {
#ifdef PR_LOGGING
uint32_t old_len = mBuffer.Length();
#endif
available = mBuffer.ContractTo(FramesToBytes(aFrames));
#ifdef PR_LOGGING
TimeStamp now = TimeStamp::Now();
if (!mStartTime.IsNull()) {
int64_t timeMs = (now - mStartTime).ToMilliseconds();
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
} else {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("Stream started before Write() @ %u", mOutRate));
}
if (old_len != available) {
// Note that we may have dropped samples in Write() as well!
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p dropped %u + %u initial frames @ %u", this,
mReadPoint, BytesToFrames(old_len - available), mOutRate));
mReadPoint += BytesToFrames(old_len - available);
}
#endif
}
mState = RUNNING;
}
if (available) {
// When we are playing a low latency stream, and it is the first time we are
// getting data from the buffer, we prefer to add the silence for an
// underrun at the beginning of the buffer, so the first buffer is not cut
// in half by the silence inserted to compensate for the underrun.
if (mInRate == mOutRate) {
if (mLatencyRequest == LowLatency && !mWritten) {
servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
} else {
servicedFrames = GetUnprocessed(output, aFrames, insertTime);
}
} else {
servicedFrames = GetTimeStretched(output, aFrames, insertTime);
}
float scaled_volume = float(GetVolumeScale() * mVolume);
ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
} else {
GetBufferInsertTime(insertTime);
}
underrunFrames = aFrames - servicedFrames;
// Always send audible frames first, and silent frames later.
// Otherwise it will break the assumption of FrameHistory.
if (mState != DRAINING) {
mAudioClock.UpdateFrameHistory(servicedFrames, underrunFrames);
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
memset(rpos, 0, FramesToBytes(underrunFrames));
if (underrunFrames) {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p lost %d frames", this, underrunFrames));
}
servicedFrames += underrunFrames;
} else {
mAudioClock.UpdateFrameHistory(servicedFrames, 0);
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
// Don't log if we're not interested or if the stream is inactive
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
mState != SHUTDOWN &&
insertTime != INT64_MAX && servicedFrames > underrunFrames) {
uint32_t latency = UINT32_MAX;
if (cubeb_stream_get_latency(mCubebStream, &latency)) {
NS_WARNING("Could not get latency from cubeb.");
}
TimeStamp now = TimeStamp::Now();
mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
insertTime, now);
mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
(latency * 1000) / mOutRate, now);
}
return servicedFrames;
}
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
int64_t insertTime;
if (available) {
// When we are playing a low latency stream, and it is the first time we are
// getting data from the buffer, we prefer to add the silence for an
// underrun at the beginning of the buffer, so the first buffer is not cut
// in half by the silence inserted to compensate for the underrun.
if (mInRate == mOutRate) {
if (mLatencyRequest == LowLatency && !mWritten) {
servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
} else {
servicedFrames = GetUnprocessed(output, aFrames, insertTime);
}
} else {
servicedFrames = GetTimeStretched(output, aFrames, insertTime);
}
float scaled_volume = float(GetVolumeScale() * mVolume);
ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
} else {
GetBufferInsertTime(insertTime);
}
underrunFrames = aFrames - servicedFrames;
if (mState != DRAINING) {
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
memset(rpos, 0, FramesToBytes(underrunFrames));
if (underrunFrames) {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p lost %d frames", this, underrunFrames));
}
mLostFrames += underrunFrames;
servicedFrames += underrunFrames;
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
// Don't log if we're not interested or if the stream is inactive
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
insertTime != INT64_MAX && servicedFrames > underrunFrames) {
uint32_t latency = UINT32_MAX;
if (cubeb_stream_get_latency(mCubebStream, &latency)) {
NS_WARNING("Could not get latency from cubeb.");
}
TimeStamp now = TimeStamp::Now();
mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
insertTime, now);
mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
(latency * 1000) / mOutRate, now);
}
mAudioClock.UpdateWritePosition(servicedFrames);
return servicedFrames;
}
