void
AudioTrackEncoder::InterleaveTrackData(AudioChunk& aChunk,
int32_t aDuration,
uint32_t aOutputChannels,
AudioDataValue* aOutput)
{
if (aChunk.mChannelData.Length() < aOutputChannels) {
// Up-mix. This might make the mChannelData have more than aChannels.
AudioChannelsUpMix(&aChunk.mChannelData, aOutputChannels, gZeroChannel);
}
if (aChunk.mChannelData.Length() > aOutputChannels) {
DownmixAndInterleave(aChunk.mChannelData, aChunk.mBufferFormat, aDuration,
aChunk.mVolume, mChannels, aOutput);
} else {
InterleaveAndConvertBuffer(aChunk.mChannelData.Elements(),
aChunk.mBufferFormat, aDuration, aChunk.mVolume,
mChannels, aOutput);
}
}
void
AudioSegment::WriteTo(uint64_t aID, AudioStream* aOutput, AudioMixer* aMixer)
{
uint32_t outputChannels = aOutput->GetChannels();
nsAutoTArray<AudioDataValue,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
// Offset in the buffer that will end up sent to the AudioStream, in samples.
uint32_t offset = 0;
if (!GetDuration()) {
return;
}
uint32_t outBufferLength = GetDuration() * outputChannels;
buf.SetLength(outBufferLength);
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
uint32_t frames = c.mDuration;
// If we have written data in the past, or we have real (non-silent) data
// to write, we can proceed. Otherwise, it means we just started the
// AudioStream, and we don't have real data to write to it (just silence).
// To avoid overbuffering in the AudioStream, we simply drop the silence,
// here. The stream will underrun and output silence anyways.
if (c.mBuffer || aOutput->GetWritten()) {
if (c.mBuffer && c.mBufferFormat != AUDIO_FORMAT_SILENCE) {
channelData.SetLength(c.mChannelData.Length());
for (uint32_t i = 0; i < channelData.Length(); ++i) {
channelData[i] = c.mChannelData[i];
}
if (channelData.Length() < outputChannels) {
// Up-mix. Note that this might actually make channelData have more
// than outputChannels temporarily.
AudioChannelsUpMix(&channelData, outputChannels, gZeroChannel);
}
if (channelData.Length() > outputChannels) {
// Down-mix.
DownmixAndInterleave(channelData, c.mBufferFormat, frames,
c.mVolume, outputChannels, buf.Elements() + offset);
} else {
InterleaveAndConvertBuffer(channelData.Elements(), c.mBufferFormat,
frames, c.mVolume,
outputChannels,
buf.Elements() + offset);
}
} else {
// Assumes that a bit pattern of zeroes == 0.0f
memset(buf.Elements() + offset, 0, outputChannels * frames * sizeof(AudioDataValue));
}
offset += frames * outputChannels;
}
if (!c.mTimeStamp.IsNull()) {
TimeStamp now = TimeStamp::Now();
// would be more efficient to c.mTimeStamp to ms on create time then pass here
LogTime(AsyncLatencyLogger::AudioMediaStreamTrack, aID,
(now - c.mTimeStamp).ToMilliseconds(), c.mTimeStamp);
}
}
aOutput->Write(buf.Elements(), offset / outputChannels, &(mChunks[mChunks.Length() - 1].mTimeStamp));
if (aMixer) {
aMixer->Mix(buf.Elements(), outputChannels, GetDuration(), aOutput->GetRate());
}
aOutput->Start();
}
