void
MediaDecodeTask::Decode()
{
MOZ_ASSERT(!NS_IsMainThread());
mBufferDecoder->BeginDecoding(mDecoderReader->GetTaskQueue());
// Tell the decoder reader that we are not going to play the data directly,
// and that we should not reject files with more channels than the audio
// backend support.
mDecoderReader->SetIgnoreAudioOutputFormat();
nsAutoPtr<MetadataTags> tags;
nsresult rv = mDecoderReader->ReadMetadata(&mMediaInfo, getter_Transfers(tags));
if (NS_FAILED(rv)) {
mDecoderReader->Shutdown();
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!mDecoderReader->HasAudio()) {
mDecoderReader->Shutdown();
ReportFailureOnMainThread(WebAudioDecodeJob::NoAudio);
return;
}
RequestSample();
}
void
MediaDecodeTask::Decode()
{
MOZ_ASSERT(!NS_IsMainThread());
mBufferDecoder->BeginDecoding(NS_GetCurrentThread());
// Tell the decoder reader that we are not going to play the data directly,
// and that we should not reject files with more channels than the audio
// bakend support.
mDecoderReader->SetIgnoreAudioOutputFormat();
MediaInfo mediaInfo;
nsAutoPtr<MetadataTags> tags;
nsresult rv = mDecoderReader->ReadMetadata(&mediaInfo, getter_Transfers(tags));
if (NS_FAILED(rv)) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!mDecoderReader->HasAudio()) {
ReportFailureOnMainThread(WebAudioDecodeJob::NoAudio);
return;
}
MediaQueue<AudioData> audioQueue;
nsRefPtr<AudioDecodeRendezvous> barrier(new AudioDecodeRendezvous());
mDecoderReader->SetCallback(barrier);
while (1) {
mDecoderReader->RequestAudioData();
nsRefPtr<AudioData> audio;
if (NS_FAILED(barrier->Await(audio))) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!audio) {
// End of stream.
break;
}
audioQueue.Push(audio);
}
mDecoderReader->Shutdown();
mDecoderReader->BreakCycles();
uint32_t frameCount = audioQueue.FrameCount();
uint32_t channelCount = mediaInfo.mAudio.mChannels;
uint32_t sampleRate = mediaInfo.mAudio.mRate;
if (!frameCount || !channelCount || !sampleRate) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
const uint32_t destSampleRate = mDecodeJob.mContext->SampleRate();
AutoResampler resampler;
uint32_t resampledFrames = frameCount;
if (sampleRate != destSampleRate) {
resampledFrames = static_cast<uint32_t>(
static_cast<uint64_t>(destSampleRate) *
static_cast<uint64_t>(frameCount) /
static_cast<uint64_t>(sampleRate)
);
resampler = speex_resampler_init(channelCount,
sampleRate,
destSampleRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
speex_resampler_skip_zeros(resampler);
resampledFrames += speex_resampler_get_output_latency(resampler);
}
// Allocate the channel buffers. Note that if we end up resampling, we may
// write fewer bytes than mResampledFrames to the output buffer, in which
// case mWriteIndex will tell us how many valid samples we have.
static const fallible_t fallible = fallible_t();
bool memoryAllocationSuccess = true;
if (!mDecodeJob.mChannelBuffers.SetLength(channelCount)) {
memoryAllocationSuccess = false;
} else {
for (uint32_t i = 0; i < channelCount; ++i) {
mDecodeJob.mChannelBuffers[i] = new(fallible) float[resampledFrames];
if (!mDecodeJob.mChannelBuffers[i]) {
memoryAllocationSuccess = false;
break;
}
}
}
if (!memoryAllocationSuccess) {
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
nsRefPtr<AudioData> audioData;
while ((audioData = audioQueue.PopFront())) {
audioData->EnsureAudioBuffer(); // could lead to a copy :(
AudioDataValue* bufferData = static_cast<AudioDataValue*>
(audioData->mAudioBuffer->Data());
if (sampleRate != destSampleRate) {
const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
for (uint32_t i = 0; i < audioData->mChannels; ++i) {
uint32_t inSamples = audioData->mFrames;
uint32_t outSamples = maxOutSamples;
WebAudioUtils::SpeexResamplerProcess(
resampler, i, &bufferData[i * audioData->mFrames], &inSamples,
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
&outSamples);
if (i == audioData->mChannels - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
MOZ_ASSERT(inSamples == audioData->mFrames);
}
}
} else {
for (uint32_t i = 0; i < audioData->mChannels; ++i) {
ConvertAudioSamples(&bufferData[i * audioData->mFrames],
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
audioData->mFrames);
if (i == audioData->mChannels - 1) {
mDecodeJob.mWriteIndex += audioData->mFrames;
}
}
}
}
if (sampleRate != destSampleRate) {
uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
for (uint32_t i = 0; i < channelCount; ++i) {
uint32_t inSamples = inputLatency;
uint32_t outSamples = maxOutSamples;
WebAudioUtils::SpeexResamplerProcess(
resampler, i, (AudioDataValue*)nullptr, &inSamples,
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
&outSamples);
if (i == channelCount - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
MOZ_ASSERT(inSamples == inputLatency);
}
}
}
mPhase = PhaseEnum::AllocateBuffer;
NS_DispatchToMainThread(this);
}
