static void
SendStreamAudio(DecodedStreamData* aStream, int64_t aStartTime,
MediaData* aData, AudioSegment* aOutput, uint32_t aRate,
const PrincipalHandle& aPrincipalHandle)
{
// The amount of audio frames that is used to fuzz rounding errors.
static const int64_t AUDIO_FUZZ_FRAMES = 1;
MOZ_ASSERT(aData);
AudioData* audio = aData->As<AudioData>();
// This logic has to mimic AudioSink closely to make sure we write
// the exact same silences
CheckedInt64 audioWrittenOffset = aStream->mAudioFramesWritten +
UsecsToFrames(aStartTime, aRate);
CheckedInt64 frameOffset = UsecsToFrames(audio->mTime, aRate);
if (!audioWrittenOffset.isValid() ||
!frameOffset.isValid() ||
// ignore packet that we've already processed
audio->GetEndTime() <= aStream->mNextAudioTime) {
return;
}
if (audioWrittenOffset.value() + AUDIO_FUZZ_FRAMES < frameOffset.value()) {
int64_t silentFrames = frameOffset.value() - audioWrittenOffset.value();
// Write silence to catch up
AudioSegment silence;
silence.InsertNullDataAtStart(silentFrames);
aStream->mAudioFramesWritten += silentFrames;
audioWrittenOffset += silentFrames;
aOutput->AppendFrom(&silence);
}
// Always write the whole sample without truncation to be consistent with
// DecodedAudioDataSink::PlayFromAudioQueue()
audio->EnsureAudioBuffer();
RefPtr<SharedBuffer> buffer = audio->mAudioBuffer;
AudioDataValue* bufferData = static_cast<AudioDataValue*>(buffer->Data());
AutoTArray<const AudioDataValue*, 2> channels;
for (uint32_t i = 0; i < audio->mChannels; ++i) {
channels.AppendElement(bufferData + i * audio->mFrames);
}
aOutput->AppendFrames(buffer.forget(), channels, audio->mFrames, aPrincipalHandle);
aStream->mAudioFramesWritten += audio->mFrames;
aStream->mNextAudioTime = audio->GetEndTime();
}
static void
SendStreamAudio(DecodedStreamData* aStream, int64_t aStartTime,
MediaData* aData, AudioSegment* aOutput,
uint32_t aRate, double aVolume)
{
MOZ_ASSERT(aData);
AudioData* audio = aData->As<AudioData>();
// This logic has to mimic AudioSink closely to make sure we write
// the exact same silences
CheckedInt64 audioWrittenOffset = aStream->mAudioFramesWritten +
UsecsToFrames(aStartTime, aRate);
CheckedInt64 frameOffset = UsecsToFrames(audio->mTime, aRate);
if (!audioWrittenOffset.isValid() ||
!frameOffset.isValid() ||
// ignore packet that we've already processed
frameOffset.value() + audio->mFrames <= audioWrittenOffset.value()) {
return;
}
if (audioWrittenOffset.value() < frameOffset.value()) {
int64_t silentFrames = frameOffset.value() - audioWrittenOffset.value();
// Write silence to catch up
AudioSegment silence;
silence.InsertNullDataAtStart(silentFrames);
aStream->mAudioFramesWritten += silentFrames;
audioWrittenOffset += silentFrames;
aOutput->AppendFrom(&silence);
}
MOZ_ASSERT(audioWrittenOffset.value() >= frameOffset.value());
int64_t offset = audioWrittenOffset.value() - frameOffset.value();
size_t framesToWrite = audio->mFrames - offset;
audio->EnsureAudioBuffer();
nsRefPtr<SharedBuffer> buffer = audio->mAudioBuffer;
AudioDataValue* bufferData = static_cast<AudioDataValue*>(buffer->Data());
nsAutoTArray<const AudioDataValue*, 2> channels;
for (uint32_t i = 0; i < audio->mChannels; ++i) {
channels.AppendElement(bufferData + i * audio->mFrames + offset);
}
aOutput->AppendFrames(buffer.forget(), channels, framesToWrite);
aStream->mAudioFramesWritten += framesToWrite;
aOutput->ApplyVolume(aVolume);
aStream->mNextAudioTime = audio->GetEndTime();
}