static void
CopyChunkToBlock(AudioChunk& aInput, AudioBlock *aBlock,
uint32_t aOffsetInBlock)
{
uint32_t blockChannels = aBlock->ChannelCount();
AutoTArray<const T*,2> channels;
if (aInput.IsNull()) {
channels.SetLength(blockChannels);
PodZero(channels.Elements(), blockChannels);
} else {
const nsTArray<const T*>& inputChannels = aInput.ChannelData<T>();
channels.SetLength(inputChannels.Length());
PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
if (channels.Length() != blockChannels) {
// We only need to upmix here because aBlock's channel count has been
// chosen to be a superset of the channel count of every chunk.
AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
}
}
for (uint32_t c = 0; c < blockChannels; ++c) {
float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
if (channels[c]) {
ConvertAudioSamplesWithScale(channels[c], outputData, aInput.GetDuration(), aInput.mVolume);
} else {
PodZero(outputData, aInput.GetDuration());
}
}
}
// Read audio data in aChunk, resample them if needed,
// and then send the result to OMX input buffer (or buffers if one buffer is not enough).
// aSamplesRead will be the number of samples that have been read from aChunk.
BufferState ReadChunk(AudioChunk& aChunk, size_t* aSamplesRead)
{
size_t chunkSamples = aChunk.GetDuration();
size_t bytesToCopy = chunkSamples * mOMXAEncoder.mResamplingRatio
* mOMXAEncoder.mChannels * sizeof(AudioDataValue);
size_t bytesCopied = 0;
if (bytesToCopy <= AvailableSize()) {
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(chunkSamples);
} else {
bytesCopied = SendChunkToBuffer(aChunk, chunkSamples);
}
UpdateAfterSendChunk(chunkSamples, bytesCopied, aSamplesRead);
} else {
// Interleave data to a temporary buffer.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(bytesToCopy);
AudioDataValue* interleavedSource = pcm.Elements();
AudioTrackEncoder::InterleaveTrackData(aChunk, chunkSamples,
mOMXAEncoder.mChannels,
interleavedSource);
// When the data size of chunk is larger than the buffer capacity,
// we split it into sub-chunks to fill up buffers.
size_t subChunkSamples = 0;
while(GetNextSubChunk(bytesToCopy, subChunkSamples)) {
// To avoid enqueueing an empty buffer, we follow the order that
// clear up buffer first, then create one, send data to it in the end.
if (!IsEmpty()) {
// Submit the filled-up buffer and request a new buffer.
status_t result = Enqueue(mOMXAEncoder.mTimestamp,
mInputFlags & ~OMXCodecWrapper::BUFFER_EOS);
if (result != OK) {
return BUFFER_FAIL;
}
result = Dequeue();
if (result == -EAGAIN) {
return WAIT_FOR_NEW_BUFFER;
}
if (result != OK) {
return BUFFER_FAIL;
}
}
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(subChunkSamples);
} else {
bytesCopied = SendInterleavedSubChunkToBuffer(interleavedSource, subChunkSamples);
}
UpdateAfterSendChunk(subChunkSamples, bytesCopied, aSamplesRead);
// Move to the position where samples are not yet send to the buffer.
interleavedSource += subChunkSamples * mOMXAEncoder.mChannels;
}
}
return BUFFER_OK;
}
/**
* Copies the data in aInput to aOffsetInBlock within aBlock. All samples must
* be float. Both chunks must have the same number of channels (or else
* aInput is null). aBlock must have been allocated with AllocateInputBlock.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock, uint32_t aOffsetInBlock)
{
uint32_t d = aInput.GetDuration();
for (uint32_t i = 0; i < aBlock->mChannelData.Length(); ++i) {
float* out = static_cast<float*>(const_cast<void*>(aBlock->mChannelData[i])) +
aOffsetInBlock;
if (aInput.IsNull()) {
PodZero(out, d);
} else {
const float* in = static_cast<const float*>(aInput.mChannelData[i]);
ConvertAudioSamplesWithScale(in, out, d, aInput.mVolume);
}
}
}
/**
* Copies the data in aInput to aOffsetInBlock within aBlock.
* aBlock must have been allocated with AllocateInputBlock and have a channel
* count that's a superset of the channels in aInput.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock,
uint32_t aOffsetInBlock)
{
uint32_t blockChannels = aBlock->ChannelCount();
nsAutoTArray<const void*,2> channels;
if (aInput.IsNull()) {
channels.SetLength(blockChannels);
PodZero(channels.Elements(), blockChannels);
} else {
channels.SetLength(aInput.ChannelCount());
PodCopy(channels.Elements(), aInput.mChannelData.Elements(), channels.Length());
if (channels.Length() != blockChannels) {
// We only need to upmix here because aBlock's channel count has been
// chosen to be a superset of the channel count of every chunk.
AudioChannelsUpMix(&channels, blockChannels, nullptr);
}
}
uint32_t duration = aInput.GetDuration();
for (uint32_t c = 0; c < blockChannels; ++c) {
float* outputData =
static_cast<float*>(const_cast<void*>(aBlock->mChannelData[c])) + aOffsetInBlock;
if (channels[c]) {
switch (aInput.mBufferFormat) {
case AUDIO_FORMAT_FLOAT32:
ConvertAudioSamplesWithScale(
static_cast<const float*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
case AUDIO_FORMAT_S16:
ConvertAudioSamplesWithScale(
static_cast<const int16_t*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
default:
NS_ERROR("Unhandled format");
}
} else {
PodZero(outputData, duration);
}
}
}
// The MediaStreamGraph guarantees that this is actually one block, for
// AudioNodeStreams.
void
AudioNodeStream::ProduceOutput(GraphTime aFrom, GraphTime aTo)
{
StreamBuffer::Track* track = EnsureTrack();
AudioChunk outputChunk;
AudioSegment* segment = track->Get<AudioSegment>();
outputChunk.SetNull(0);
if (mInCycle) {
// XXX DelayNode not supported yet so just produce silence
outputChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
} else {
AudioChunk tmpChunk;
AudioChunk* inputChunk = ObtainInputBlock(&tmpChunk);
bool finished = false;
mEngine->ProduceAudioBlock(this, *inputChunk, &outputChunk, &finished);
if (finished) {
FinishOutput();
}
}
mLastChunk = outputChunk;
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&outputChunk);
} else {
segment->AppendNullData(outputChunk.GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = outputChunk;
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_NODE_STREAM_TRACK_ID,
IdealAudioRate(), segment->GetDuration(), 0,
tmpSegment);
}
}
nsresult
VorbisTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
if (mEosSetInEncoder) {
return NS_OK;
}
PROFILER_LABEL("VorbisTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
nsAutoPtr<AudioSegment> sourceSegment;
sourceSegment = new AudioSegment();
{
// Move all the samples from mRawSegment to sourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait if mEncoder is not initialized, or when not enough raw data, but is
// not the end of stream nor is being canceled.
while (!mCanceled && mRawSegment.GetDuration() < GetPacketDuration() &&
!mEndOfStream) {
mon.Wait();
}
VORBISLOG("GetEncodedTrack passes wait, duration is %lld\n",
mRawSegment.GetDuration());
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
sourceSegment->AppendFrom(&mRawSegment);
}
if (mEndOfStream && (sourceSegment->GetDuration() == 0)
&& !mEosSetInEncoder) {
mEncodingComplete = true;
mEosSetInEncoder = true;
VORBISLOG("[Vorbis] Done encoding.");
vorbis_analysis_wrote(&mVorbisDsp, 0);
GetEncodedFrames(aData);
return NS_OK;
}
// Start encoding data.
AudioSegment::ChunkIterator iter(*sourceSegment);
AudioDataValue **vorbisBuffer =
vorbis_analysis_buffer(&mVorbisDsp, (int)sourceSegment->GetDuration());
int framesCopied = 0;
AutoTArray<AudioDataValue, 9600> interleavedPcm;
AutoTArray<AudioDataValue, 9600> nonInterleavedPcm;
interleavedPcm.SetLength(sourceSegment->GetDuration() * mChannels);
nonInterleavedPcm.SetLength(sourceSegment->GetDuration() * mChannels);
while (!iter.IsEnded()) {
AudioChunk chunk = *iter;
int frameToCopy = chunk.GetDuration();
if (!chunk.IsNull()) {
InterleaveTrackData(chunk, frameToCopy, mChannels,
interleavedPcm.Elements() + framesCopied * mChannels);
} else { // empty data
memset(interleavedPcm.Elements() + framesCopied * mChannels, 0,
frameToCopy * mChannels * sizeof(AudioDataValue));
}
framesCopied += frameToCopy;
iter.Next();
}
// De-interleave the interleavedPcm.
DeInterleaveTrackData(interleavedPcm.Elements(), framesCopied, mChannels,
nonInterleavedPcm.Elements());
// Copy the nonInterleavedPcm to vorbis buffer.
for(uint8_t i = 0; i < mChannels; ++i) {
memcpy(vorbisBuffer[i], nonInterleavedPcm.Elements() + framesCopied * i,
framesCopied * sizeof(AudioDataValue));
}
// Now the vorbisBuffer contain the all data in non-interleaved.
// Tell the library how much we actually submitted.
vorbis_analysis_wrote(&mVorbisDsp, framesCopied);
VORBISLOG("vorbis_analysis_wrote framesCopied %d\n", framesCopied);
GetEncodedFrames(aData);
return NS_OK;
}
nsresult
OpusTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
PROFILER_LABEL("OpusTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
{
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until initialized or cancelled.
while (!mCanceled && !mInitialized) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
}
// calculation below depends on the truth that mInitialized is true.
MOZ_ASSERT(mInitialized);
// re-sampled frames left last time which didn't fit into an Opus packet duration.
const int framesLeft = mResampledLeftover.Length() / mChannels;
// When framesLeft is 0, (GetPacketDuration() - framesLeft) is a multiple
// of kOpusSamplingRate. There is not precision loss in the integer division
// in computing framesToFetch. If frameLeft > 0, we need to add 1 to
// framesToFetch to ensure there will be at least n frames after re-sampling.
const int frameRoundUp = framesLeft ? 1 : 0;
MOZ_ASSERT(GetPacketDuration() >= framesLeft);
// Try to fetch m frames such that there will be n frames
// where (n + frameLeft) >= GetPacketDuration() after re-sampling.
const int framesToFetch = !mResampler ? GetPacketDuration()
: (GetPacketDuration() - framesLeft) * mSamplingRate / kOpusSamplingRate
+ frameRoundUp;
{
// Move all the samples from mRawSegment to mSourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until enough raw data, end of stream or cancelled.
while (!mCanceled && mRawSegment.GetDuration() +
mSourceSegment.GetDuration() < framesToFetch &&
!mEndOfStream) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
mSourceSegment.AppendFrom(&mRawSegment);
// Pad |mLookahead| samples to the end of source stream to prevent lost of
// original data, the pcm duration will be calculated at rate 48K later.
if (mEndOfStream && !mEosSetInEncoder) {
mEosSetInEncoder = true;
mSourceSegment.AppendNullData(mLookahead);
}
}
// Start encoding data.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(GetPacketDuration() * mChannels);
AudioSegment::ChunkIterator iter(mSourceSegment);
int frameCopied = 0;
while (!iter.IsEnded() && frameCopied < framesToFetch) {
AudioChunk chunk = *iter;
// Chunk to the required frame size.
int frameToCopy = chunk.GetDuration();
if (frameCopied + frameToCopy > framesToFetch) {
frameToCopy = framesToFetch - frameCopied;
}
if (!chunk.IsNull()) {
// Append the interleaved data to the end of pcm buffer.
AudioTrackEncoder::InterleaveTrackData(chunk, frameToCopy, mChannels,
pcm.Elements() + frameCopied * mChannels);
} else {
memset(pcm.Elements() + frameCopied * mChannels, 0,
frameToCopy * mChannels * sizeof(AudioDataValue));
}
frameCopied += frameToCopy;
iter.Next();
}
RefPtr<EncodedFrame> audiodata = new EncodedFrame();
audiodata->SetFrameType(EncodedFrame::OPUS_AUDIO_FRAME);
int framesInPCM = frameCopied;
if (mResampler) {
nsAutoTArray<AudioDataValue, 9600> resamplingDest;
// We want to consume all the input data, so we slightly oversize the
// resampled data buffer so we can fit the output data in. We cannot really
// predict the output frame count at each call.
uint32_t outframes = frameCopied * kOpusSamplingRate / mSamplingRate + 1;
uint32_t inframes = frameCopied;
resamplingDest.SetLength(outframes * mChannels);
#if MOZ_SAMPLE_TYPE_S16
short* in = reinterpret_cast<short*>(pcm.Elements());
short* out = reinterpret_cast<short*>(resamplingDest.Elements());
speex_resampler_process_interleaved_int(mResampler, in, &inframes,
out, &outframes);
#else
float* in = reinterpret_cast<float*>(pcm.Elements());
float* out = reinterpret_cast<float*>(resamplingDest.Elements());
speex_resampler_process_interleaved_float(mResampler, in, &inframes,
out, &outframes);
#endif
MOZ_ASSERT(pcm.Length() >= mResampledLeftover.Length());
PodCopy(pcm.Elements(), mResampledLeftover.Elements(),
mResampledLeftover.Length());
uint32_t outframesToCopy = std::min(outframes,
static_cast<uint32_t>(GetPacketDuration() - framesLeft));
MOZ_ASSERT(pcm.Length() - mResampledLeftover.Length() >=
outframesToCopy * mChannels);
PodCopy(pcm.Elements() + mResampledLeftover.Length(),
resamplingDest.Elements(), outframesToCopy * mChannels);
int frameLeftover = outframes - outframesToCopy;
mResampledLeftover.SetLength(frameLeftover * mChannels);
PodCopy(mResampledLeftover.Elements(),
resamplingDest.Elements() + outframesToCopy * mChannels,
mResampledLeftover.Length());
// This is always at 48000Hz.
framesInPCM = framesLeft + outframesToCopy;
audiodata->SetDuration(framesInPCM);
} else {
// The ogg time stamping and pre-skip is always timed at 48000.
audiodata->SetDuration(frameCopied * (kOpusSamplingRate / mSamplingRate));
}
// Remove the raw data which has been pulled to pcm buffer.
// The value of frameCopied should equal to (or smaller than, if eos)
// GetPacketDuration().
mSourceSegment.RemoveLeading(frameCopied);
// Has reached the end of input stream and all queued data has pulled for
// encoding.
if (mSourceSegment.GetDuration() == 0 && mEndOfStream) {
mEncodingComplete = true;
LOG("[Opus] Done encoding.");
}
MOZ_ASSERT(mEndOfStream || framesInPCM == GetPacketDuration());
// Append null data to pcm buffer if the leftover data is not enough for
// opus encoder.
if (framesInPCM < GetPacketDuration() && mEndOfStream) {
PodZero(pcm.Elements() + framesInPCM * mChannels,
(GetPacketDuration() - framesInPCM) * mChannels);
}
nsTArray<uint8_t> frameData;
// Encode the data with Opus Encoder.
frameData.SetLength(MAX_DATA_BYTES);
// result is returned as opus error code if it is negative.
int result = 0;
#ifdef MOZ_SAMPLE_TYPE_S16
const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#else
const float* pcmBuf = static_cast<float*>(pcm.Elements());
result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#endif
frameData.SetLength(result >= 0 ? result : 0);
if (result < 0) {
LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
}
if (mEncodingComplete) {
if (mResampler) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
}
mResampledLeftover.SetLength(0);
}
audiodata->SwapInFrameData(frameData);
aData.AppendEncodedFrame(audiodata);
return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
}
