void
MediaDecodeTask::AllocateBuffer()
{
MOZ_ASSERT(NS_IsMainThread());
if (!mDecodeJob.AllocateBuffer()) {
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
mPhase = PhaseEnum::Done;
CallbackTheResult();
}
void
MediaDecodeTask::Decode()
{
MOZ_ASSERT(!mThreadPool == NS_IsMainThread(),
"We should be on the main thread only if we don't have a thread pool");
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();
mDecoderReader->OnDecodeThreadStart();
VideoInfo videoInfo;
nsAutoPtr<MetadataTags> tags;
nsresult rv = mDecoderReader->ReadMetadata(&videoInfo, getter_Transfers(tags));
if (NS_FAILED(rv)) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!mDecoderReader->HasAudio()) {
ReportFailureOnMainThread(WebAudioDecodeJob::NoAudio);
return;
}
while (mDecoderReader->DecodeAudioData()) {
// consume all of the buffer
continue;
}
mDecoderReader->OnDecodeThreadFinish();
MediaQueue<AudioData>& audioQueue = mDecoderReader->AudioQueue();
uint32_t frameCount = audioQueue.FrameCount();
uint32_t channelCount = videoInfo.mAudioChannels;
uint32_t sampleRate = videoInfo.mAudioRate;
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;
}
nsAutoPtr<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 expectedOutSamples = static_cast<uint32_t>(
static_cast<uint64_t>(destSampleRate) *
static_cast<uint64_t>(audioData->mFrames) /
static_cast<uint64_t>(sampleRate)
);
#ifdef MOZ_SAMPLE_TYPE_S16
AudioDataValue* resampledBuffer = new(fallible) AudioDataValue[channelCount * expectedOutSamples];
#endif
for (uint32_t i = 0; i < audioData->mChannels; ++i) {
uint32_t inSamples = audioData->mFrames;
uint32_t outSamples = expectedOutSamples;
#ifdef MOZ_SAMPLE_TYPE_S16
speex_resampler_process_int(resampler, i, &bufferData[i * audioData->mFrames], &inSamples,
&resampledBuffer[i * expectedOutSamples],
&outSamples);
ConvertAudioSamples(&resampledBuffer[i * expectedOutSamples],
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
outSamples);
#else
speex_resampler_process_float(resampler, i, &bufferData[i * audioData->mFrames], &inSamples,
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
&outSamples);
#endif
if (i == audioData->mChannels - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
}
}
#ifdef MOZ_SAMPLE_TYPE_S16
delete[] resampledBuffer;
#endif
} 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) {
int inputLatency = speex_resampler_get_input_latency(resampler);
int outputLatency = speex_resampler_get_output_latency(resampler);
AudioDataValue* zero = (AudioDataValue*)calloc(inputLatency, sizeof(AudioDataValue));
#ifdef MOZ_SAMPLE_TYPE_S16
AudioDataValue* resampledBuffer = new(fallible) AudioDataValue[channelCount * outputLatency];
if (!resampledBuffer || !zero) {
#else
if (!zero) {
#endif
// Out of memory!
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
for (uint32_t i = 0; i < channelCount; ++i) {
uint32_t inSamples = inputLatency;
uint32_t outSamples = outputLatency;
#ifdef MOZ_SAMPLE_TYPE_S16
speex_resampler_process_int(resampler, i, zero, &inSamples,
&resampledBuffer[i * outputLatency],
&outSamples);
ConvertAudioSamples(&resampledBuffer[i * outputLatency],
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
outSamples);
#else
speex_resampler_process_float(resampler, i, zero, &inSamples,
mDecodeJob.mChannelBuffers[i] + mDecodeJob.mWriteIndex,
&outSamples);
#endif
if (i == channelCount - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
}
}
free(zero);
#ifdef MOZ_SAMPLE_TYPE_S16
delete[] resampledBuffer;
#endif
}
mPhase = PhaseEnum::AllocateBuffer;
RunNextPhase();
}
void
MediaDecodeTask::AllocateBuffer()
{
MOZ_ASSERT(NS_IsMainThread());
if (!mDecodeJob.AllocateBuffer()) {
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
mPhase = PhaseEnum::Done;
CallbackTheResult();
}
