bool
DirectShowReader::DecodeAudioData()
{
MOZ_ASSERT(mDecoder->OnDecodeThread(), "Should be on decode thread.");
HRESULT hr;
SampleSink* sink = mAudioSinkFilter->GetSampleSink();
if (sink->AtEOS()) {
// End of stream.
return Finish(S_OK);
}
// Get the next chunk of audio samples. This blocks until the sample
// arrives, or an error occurs (like the stream is shutdown).
RefPtr<IMediaSample> sample;
hr = sink->Extract(sample);
if (FAILED(hr) || hr == S_FALSE) {
return Finish(hr);
}
int64_t start = 0, end = 0;
sample->GetMediaTime(&start, &end);
LOG("DirectShowReader::DecodeAudioData [%4.2lf-%4.2lf]",
RefTimeToSeconds(start),
RefTimeToSeconds(end));
LONG length = sample->GetActualDataLength();
LONG numSamples = length / mBytesPerSample;
LONG numFrames = length / mBytesPerSample / mNumChannels;
BYTE* data = nullptr;
hr = sample->GetPointer(&data);
NS_ENSURE_TRUE(SUCCEEDED(hr), Finish(hr));
nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[numSamples]);
AudioDataValue* dst = buffer.get();
if (mBytesPerSample == 1) {
uint8_t* src = reinterpret_cast<uint8_t*>(data);
for (int32_t i = 0; i < numSamples; ++i) {
dst[i] = UnsignedByteToAudioSample(src[i]);
}
} else if (mBytesPerSample == 2) {
int16_t* src = reinterpret_cast<int16_t*>(data);
for (int32_t i = 0; i < numSamples; ++i) {
dst[i] = AudioSampleToFloat(src[i]);
}
}
mAudioQueue.Push(new AudioData(mDecoder->GetResource()->Tell(),
RefTimeToUsecs(start),
RefTimeToUsecs(end - start),
numFrames,
buffer.forget(),
mNumChannels));
return true;
}
uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples)
{
uint32_t maxSamples = std::min(aSamples, mSamples - mNextSample);
uint32_t frames = maxSamples / mChannels;
size_t byteOffset = mNextSample * mBytesPerSample;
if (mBytesPerSample == 1) {
for (uint32_t i = 0; i < maxSamples; ++i) {
uint8_t *sample = mSource + byteOffset;
aBuffer[i] = UnsignedByteToAudioSample(*sample);
byteOffset += mBytesPerSample;
}
} else if (mBytesPerSample == 2) {
for (uint32_t i = 0; i < maxSamples; ++i) {
int16_t *sample = reinterpret_cast<int16_t *>(mSource + byteOffset);
aBuffer[i] = AudioSampleToFloat(*sample);
byteOffset += mBytesPerSample;
}
}
mNextSample += maxSamples;
return frames;
}
