void
MediaEngineWebRTCMicrophoneSource::InsertInGraph(const T* aBuffer,
size_t aFrames,
uint32_t aChannels)
{
if (mState != kStarted) {
return;
}
if (MOZ_LOG_TEST(AudioLogModule(), LogLevel::Debug)) {
mTotalFrames += aFrames;
if (mTotalFrames > mLastLogFrames + mSampleFrequency) { // ~ 1 second
MOZ_LOG(AudioLogModule(), LogLevel::Debug,
("%p: Inserting %" PRIuSIZE " samples into graph, total frames = %" PRIu64,
(void*)this, aFrames, mTotalFrames));
mLastLogFrames = mTotalFrames;
}
}
size_t len = mSources.Length();
for (size_t i = 0; i < len; i++) {
if (!mSources[i]) {
continue;
}
RefPtr<SharedBuffer> buffer =
SharedBuffer::Create(aFrames * aChannels * sizeof(T));
PodCopy(static_cast<T*>(buffer->Data()),
aBuffer, aFrames * aChannels);
TimeStamp insertTime;
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion,
LATENCY_STREAM_ID(mSources[i].get(), mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
nsAutoPtr<AudioSegment> segment(new AudioSegment());
AutoTArray<const T*, 1> channels;
// XXX Bug 971528 - Support stereo capture in gUM
MOZ_ASSERT(aChannels == 1,
"GraphDriver only supports us stereo audio for now");
channels.AppendElement(static_cast<T*>(buffer->Data()));
segment->AppendFrames(buffer.forget(), channels, aFrames,
mPrincipalHandles[i]);
segment->GetStartTime(insertTime);
mSources[i]->AppendToTrack(mTrackID, segment);
}
}
void
MediaEngineWebRTCMicrophoneSource::InsertInGraph(const T* aBuffer,
size_t aFrames,
uint32_t aChannels)
{
if (mState != kStarted) {
return;
}
size_t len = mSources.Length();
for (size_t i = 0; i < len; i++) {
if (!mSources[i]) {
continue;
}
RefPtr<SharedBuffer> buffer =
SharedBuffer::Create(aFrames * aChannels * sizeof(T));
PodCopy(static_cast<T*>(buffer->Data()),
aBuffer, aFrames * aChannels);
TimeStamp insertTime;
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion,
LATENCY_STREAM_ID(mSources[i].get(), mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
nsAutoPtr<AudioSegment> segment(new AudioSegment());
AutoTArray<const T*, 1> channels;
// XXX Bug 971528 - Support stereo capture in gUM
MOZ_ASSERT(aChannels == 1,
"GraphDriver only supports us stereo audio for now");
channels.AppendElement(static_cast<T*>(buffer->Data()));
segment->AppendFrames(buffer.forget(), channels, aFrames,
mPrincipalHandles[i]);
segment->GetStartTime(insertTime);
RUN_ON_THREAD(mThread,
WrapRunnable(mSources[i], &SourceMediaStream::AppendToTrack,
mTrackID, segment,
static_cast<AudioSegment*>(nullptr)),
NS_DISPATCH_NORMAL);
}
}
void
MediaEngineWebRTCAudioSource::Process(int channel,
webrtc::ProcessingTypes type, sample* audio10ms,
int length, int samplingFreq, bool isStereo)
{
MonitorAutoLock lock(mMonitor);
if (mState != kStarted)
return;
uint32_t len = mSources.Length();
for (uint32_t i = 0; i < len; i++) {
nsRefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));
sample* dest = static_cast<sample*>(buffer->Data());
memcpy(dest, audio10ms, length * sizeof(sample));
AudioSegment segment;
nsAutoTArray<const sample*,1> channels;
channels.AppendElement(dest);
segment.AppendFrames(buffer.forget(), channels, length);
TimeStamp insertTime;
segment.GetStartTime(insertTime);
SourceMediaStream *source = mSources[i];
if (source) {
// This is safe from any thread, and is safe if the track is Finished
// or Destroyed.
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion, LATENCY_STREAM_ID(source, mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
source->AppendToTrack(mTrackID, &segment);
}
}
return;
}
void
MediaEngineWebRTCMicrophoneSource::Process(int channel,
webrtc::ProcessingTypes type,
sample *audio10ms, int length,
int samplingFreq, bool isStereo)
{
// On initial capture, throw away all far-end data except the most recent sample
// since it's already irrelevant and we want to keep avoid confusing the AEC far-end
// input code with "old" audio.
if (!mStarted) {
mStarted = true;
while (gFarendObserver->Size() > 1) {
free(gFarendObserver->Pop()); // only call if size() > 0
}
}
while (gFarendObserver->Size() > 0) {
FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
if (buffer) {
int length = buffer->mSamples;
int res = mVoERender->ExternalPlayoutData(buffer->mData,
gFarendObserver->PlayoutFrequency(),
gFarendObserver->PlayoutChannels(),
mPlayoutDelay,
length);
free(buffer);
if (res == -1) {
return;
}
}
}
MonitorAutoLock lock(mMonitor);
if (mState != kStarted)
return;
uint32_t len = mSources.Length();
for (uint32_t i = 0; i < len; i++) {
RefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));
sample* dest = static_cast<sample*>(buffer->Data());
memcpy(dest, audio10ms, length * sizeof(sample));
nsAutoPtr<AudioSegment> segment(new AudioSegment());
nsAutoTArray<const sample*,1> channels;
channels.AppendElement(dest);
segment->AppendFrames(buffer.forget(), channels, length);
TimeStamp insertTime;
segment->GetStartTime(insertTime);
if (mSources[i]) {
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion, LATENCY_STREAM_ID(mSources[i].get(), mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
// This is safe from any thread, and is safe if the track is Finished
// or Destroyed.
// Note: due to evil magic, the nsAutoPtr<AudioSegment>'s ownership transfers to
// the Runnable (AutoPtr<> = AutoPtr<>)
RUN_ON_THREAD(mThread, WrapRunnable(mSources[i], &SourceMediaStream::AppendToTrack,
mTrackID, segment, (AudioSegment *) nullptr),
NS_DISPATCH_NORMAL);
}
}
return;
}
void
MediaEngineWebRTCAudioSource::Process(int channel,
webrtc::ProcessingTypes type, sample* audio10ms,
int length, int samplingFreq, bool isStereo)
{
// On initial capture, throw away all far-end data except the most recent sample
// since it's already irrelevant and we want to keep avoid confusing the AEC far-end
// input code with "old" audio.
if (!mStarted) {
mStarted = true;
while (gFarendObserver->Size() > 1) {
FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
free(buffer);
}
}
while (gFarendObserver->Size() > 0) {
FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
if (buffer) {
int length = buffer->mSamples;
if (mVoERender->ExternalPlayoutData(buffer->mData,
gFarendObserver->PlayoutFrequency(),
gFarendObserver->PlayoutChannels(),
mPlayoutDelay,
length) == -1) {
return;
}
}
free(buffer);
}
#ifdef PR_LOGGING
mSamples += length;
if (mSamples > samplingFreq) {
mSamples %= samplingFreq; // just in case mSamples >> samplingFreq
if (PR_LOG_TEST(GetMediaManagerLog(), PR_LOG_DEBUG)) {
webrtc::EchoStatistics echo;
mVoECallReport->GetEchoMetricSummary(echo);
#define DUMP_STATVAL(x) (x).min, (x).max, (x).average
LOG(("Echo: ERL: %d/%d/%d, ERLE: %d/%d/%d, RERL: %d/%d/%d, NLP: %d/%d/%d",
DUMP_STATVAL(echo.erl),
DUMP_STATVAL(echo.erle),
DUMP_STATVAL(echo.rerl),
DUMP_STATVAL(echo.a_nlp)));
}
}
#endif
MonitorAutoLock lock(mMonitor);
if (mState != kStarted)
return;
uint32_t len = mSources.Length();
for (uint32_t i = 0; i < len; i++) {
nsRefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));
sample* dest = static_cast<sample*>(buffer->Data());
memcpy(dest, audio10ms, length * sizeof(sample));
AudioSegment segment;
nsAutoTArray<const sample*,1> channels;
channels.AppendElement(dest);
segment.AppendFrames(buffer.forget(), channels, length);
TimeStamp insertTime;
segment.GetStartTime(insertTime);
SourceMediaStream *source = mSources[i];
if (source) {
// This is safe from any thread, and is safe if the track is Finished
// or Destroyed.
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion, LATENCY_STREAM_ID(source, mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
source->AppendToTrack(mTrackID, &segment);
}
}
return;
}
