nsresult
VP8TrackEncoder::GetEncodedPartitions(EncodedFrameContainer& aData)
{
vpx_codec_iter_t iter = nullptr;
EncodedFrame::FrameType frameType = EncodedFrame::VP8_P_FRAME;
nsTArray<uint8_t> frameData;
nsresult rv;
const vpx_codec_cx_pkt_t *pkt = nullptr;
while ((pkt = vpx_codec_get_cx_data(mVPXContext, &iter)) != nullptr) {
switch (pkt->kind) {
case VPX_CODEC_CX_FRAME_PKT: {
// Copy the encoded data from libvpx to frameData
frameData.AppendElements((uint8_t*)pkt->data.frame.buf,
pkt->data.frame.sz);
break;
}
default: {
break;
}
}
// End of frame
if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
frameType = EncodedFrame::VP8_I_FRAME;
}
break;
}
}
if (!frameData.IsEmpty() &&
(pkt->data.frame.pts == mEncodedTimestamp)) {
// Copy the encoded data to aData.
EncodedFrame* videoData = new EncodedFrame();
videoData->SetFrameType(frameType);
// Convert the timestamp and duration to Usecs.
CheckedInt64 timestamp = FramesToUsecs(mEncodedTimestamp, mTrackRate);
if (timestamp.isValid()) {
videoData->SetTimeStamp(
(uint64_t)FramesToUsecs(mEncodedTimestamp, mTrackRate).value());
}
CheckedInt64 duration = FramesToUsecs(pkt->data.frame.duration, mTrackRate);
if (duration.isValid()) {
videoData->SetDuration(
(uint64_t)FramesToUsecs(pkt->data.frame.duration, mTrackRate).value());
}
rv = videoData->SwapInFrameData(frameData);
NS_ENSURE_SUCCESS(rv, rv);
VP8LOG("GetEncodedPartitions TimeStamp %lld Duration %lld\n",
videoData->GetTimeStamp(), videoData->GetDuration());
VP8LOG("frameType %d\n", videoData->GetFrameType());
aData.AppendEncodedFrame(videoData);
}
return NS_OK;
}
bool MediaPluginReader::DecodeAudioData()
{
NS_ASSERTION(mDecoder->OnDecodeThread(), "Should be on decode thread.");
// This is the approximate byte position in the stream.
int64_t pos = mDecoder->GetResource()->Tell();
// Read next frame
MPAPI::AudioFrame frame;
if (!mPlugin->ReadAudio(mPlugin, &frame, mAudioSeekTimeUs)) {
return false;
}
mAudioSeekTimeUs = -1;
// Ignore empty buffers which stagefright media read will sporadically return
if (frame.mSize == 0)
return true;
nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[frame.mSize/2] );
memcpy(buffer.get(), frame.mData, frame.mSize);
uint32_t frames = frame.mSize / (2 * frame.mAudioChannels);
CheckedInt64 duration = FramesToUsecs(frames, frame.mAudioSampleRate);
if (!duration.isValid()) {
return false;
}
mAudioQueue.Push(new AudioData(pos,
frame.mTimeUs,
duration.value(),
frames,
buffer.forget(),
frame.mAudioChannels));
return true;
}
int64_t
AudioSink::GetEndTime() const
{
CheckedInt64 playedUsecs = FramesToUsecs(mWritten, mInfo.mRate) + mStartTime;
if (!playedUsecs.isValid()) {
NS_WARNING("Int overflow calculating audio end time");
return -1;
}
return playedUsecs.value();
}
uint32_t
AudioSink::PushProcessedAudio(AudioData* aData)
{
if (!aData || !aData->mFrames) {
return 0;
}
mProcessedQueue.Push(aData);
mProcessedQueueLength += FramesToUsecs(aData->mFrames, mOutputRate).value();
return aData->mFrames;
}
nsresult
GonkAudioDecoderManager::CreateAudioData(int64_t aStreamOffset, AudioData **v) {
if (!(mAudioBuffer != nullptr && mAudioBuffer->data() != nullptr)) {
GADM_LOG("Audio Buffer is not valid!");
return NS_ERROR_UNEXPECTED;
}
int64_t timeUs;
if (!mAudioBuffer->meta_data()->findInt64(kKeyTime, &timeUs)) {
return NS_ERROR_UNEXPECTED;
}
if (mAudioBuffer->range_length() == 0) {
// Some decoders may return spurious empty buffers that we just want to ignore
// quoted from Android's AwesomePlayer.cpp
ReleaseAudioBuffer();
return NS_ERROR_NOT_AVAILABLE;
}
if (mLastDecodedTime > timeUs) {
ReleaseAudioBuffer();
GADM_LOG("Output decoded sample time is revert. time=%lld", timeUs);
MOZ_ASSERT(false);
return NS_ERROR_NOT_AVAILABLE;
}
mLastDecodedTime = timeUs;
const uint8_t *data = static_cast<const uint8_t*>(mAudioBuffer->data());
size_t dataOffset = mAudioBuffer->range_offset();
size_t size = mAudioBuffer->range_length();
nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[size/2]);
memcpy(buffer.get(), data+dataOffset, size);
uint32_t frames = size / (2 * mAudioChannels);
CheckedInt64 duration = FramesToUsecs(frames, mAudioRate);
if (!duration.isValid()) {
return NS_ERROR_UNEXPECTED;
}
nsRefPtr<AudioData> audioData = new AudioData(aStreamOffset,
timeUs,
duration.value(),
frames,
buffer.forget(),
mAudioChannels,
mAudioRate);
ReleaseAudioBuffer();
audioData.forget(v);
return NS_OK;
}
int64_t
DecodedStream::GetEndTime(TrackType aType) const
{
AssertOwnerThread();
if (aType == TrackInfo::kAudioTrack && mInfo.HasAudio() && mData) {
CheckedInt64 t = mStartTime.ref() +
FramesToUsecs(mData->mAudioFramesWritten, mInfo.mAudio.mRate);
if (t.isValid()) {
return t.value();
}
} else if (aType == TrackInfo::kVideoTrack && mData) {
return mData->mNextVideoTime;
}
return -1;
}
nsresult
GonkAudioDecoderManager::CreateAudioData(MediaBuffer* aBuffer, int64_t aStreamOffset)
{
if (!(aBuffer != nullptr && aBuffer->data() != nullptr)) {
GADM_LOG("Audio Buffer is not valid!");
return NS_ERROR_UNEXPECTED;
}
int64_t timeUs;
if (!aBuffer->meta_data()->findInt64(kKeyTime, &timeUs)) {
return NS_ERROR_UNEXPECTED;
}
if (aBuffer->range_length() == 0) {
// Some decoders may return spurious empty buffers that we just want to ignore
// quoted from Android's AwesomePlayer.cpp
return NS_ERROR_NOT_AVAILABLE;
}
if (mLastTime > timeUs) {
GADM_LOG("Output decoded sample time is revert. time=%lld", timeUs);
MOZ_ASSERT(false);
return NS_ERROR_NOT_AVAILABLE;
}
mLastTime = timeUs;
const uint8_t *data = static_cast<const uint8_t*>(aBuffer->data());
size_t dataOffset = aBuffer->range_offset();
size_t size = aBuffer->range_length();
uint32_t frames = size / (2 * mAudioChannels);
CheckedInt64 duration = FramesToUsecs(frames, mAudioRate);
if (!duration.isValid()) {
return NS_ERROR_UNEXPECTED;
}
typedef AudioCompactor::NativeCopy OmxCopy;
mAudioCompactor.Push(aStreamOffset,
timeUs,
mAudioRate,
frames,
mAudioChannels,
OmxCopy(data+dataOffset,
size,
mAudioChannels));
return NS_OK;
}
/**
* Compares the elapsed time from the beginning of GetEncodedTrack and
* the processed frame duration in mSourceSegment
* in order to set the nextEncodeOperation for next target frame.
*/
VP8TrackEncoder::EncodeOperation
VP8TrackEncoder::GetNextEncodeOperation(TimeDuration aTimeElapsed,
TrackTicks aProcessedDuration)
{
int64_t durationInUsec =
FramesToUsecs(aProcessedDuration + mEncodedFrameDuration,
mTrackRate).value();
if (aTimeElapsed.ToMicroseconds() > (durationInUsec * SKIP_FRAME_RATIO)) {
// The encoder is too slow.
// We should skip next frame to consume the mSourceSegment.
return SKIP_FRAME;
} else if (aTimeElapsed.ToMicroseconds() > (durationInUsec * I_FRAME_RATIO)) {
// The encoder is a little slow.
// We force the encoder to encode an I-frame to accelerate.
return ENCODE_I_FRAME;
} else {
return ENCODE_NORMAL_FRAME;
}
}
nsresult
SeekTask::DropAudioUpToSeekTarget(MediaData* aSample)
{
AssertOwnerThread();
RefPtr<AudioData> audio(aSample->As<AudioData>());
MOZ_ASSERT(audio && mSeekJob.Exists() && mSeekJob.mTarget.IsAccurate());
CheckedInt64 sampleDuration = FramesToUsecs(audio->mFrames, mAudioRate);
if (!sampleDuration.isValid()) {
return NS_ERROR_FAILURE;
}
if (audio->mTime + sampleDuration.value() <= mSeekJob.mTarget.GetTime().ToMicroseconds()) {
// Our seek target lies after the frames in this AudioData. Don't
// push it onto the audio queue, and keep decoding forwards.
return NS_OK;
}
if (audio->mTime > mSeekJob.mTarget.GetTime().ToMicroseconds()) {
// The seek target doesn't lie in the audio block just after the last
// audio frames we've seen which were before the seek target. This
// could have been the first audio data we've seen after seek, i.e. the
// seek terminated after the seek target in the audio stream. Just
// abort the audio decode-to-target, the state machine will play
// silence to cover the gap. Typically this happens in poorly muxed
// files.
DECODER_WARN("Audio not synced after seek, maybe a poorly muxed file?");
mSeekedAudioData = audio;
return NS_OK;
}
// The seek target lies somewhere in this AudioData's frames, strip off
// any frames which lie before the seek target, so we'll begin playback
// exactly at the seek target.
NS_ASSERTION(mSeekJob.mTarget.GetTime().ToMicroseconds() >= audio->mTime,
"Target must at or be after data start.");
NS_ASSERTION(mSeekJob.mTarget.GetTime().ToMicroseconds() < audio->mTime + sampleDuration.value(),
"Data must end after target.");
CheckedInt64 framesToPrune =
UsecsToFrames(mSeekJob.mTarget.GetTime().ToMicroseconds() - audio->mTime, mAudioRate);
if (!framesToPrune.isValid()) {
return NS_ERROR_FAILURE;
}
if (framesToPrune.value() > audio->mFrames) {
// We've messed up somehow. Don't try to trim frames, the |frames|
// variable below will overflow.
DECODER_WARN("Can't prune more frames that we have!");
return NS_ERROR_FAILURE;
}
uint32_t frames = audio->mFrames - static_cast<uint32_t>(framesToPrune.value());
uint32_t channels = audio->mChannels;
AlignedAudioBuffer audioData(frames * channels);
if (!audioData) {
return NS_ERROR_OUT_OF_MEMORY;
}
memcpy(audioData.get(),
audio->mAudioData.get() + (framesToPrune.value() * channels),
frames * channels * sizeof(AudioDataValue));
CheckedInt64 duration = FramesToUsecs(frames, mAudioRate);
if (!duration.isValid()) {
return NS_ERROR_FAILURE;
}
RefPtr<AudioData> data(new AudioData(audio->mOffset,
mSeekJob.mTarget.GetTime().ToMicroseconds(),
duration.value(),
frames,
Move(audioData),
channels,
audio->mRate));
MOZ_ASSERT(!mSeekedAudioData, "Should be the 1st sample after seeking");
mSeekedAudioData = data;
return NS_OK;
}
bool
WMFReader::DecodeAudioData()
{
MOZ_ASSERT(OnTaskQueue());
HRESULT hr;
hr = mSourceReader->ReadSample(MF_SOURCE_READER_FIRST_AUDIO_STREAM,
0, // control flags
0, // read stream index
nullptr,
nullptr,
nullptr);
if (FAILED(hr)) {
DECODER_LOG("WMFReader::DecodeAudioData() ReadSample failed with hr=0x%x", hr);
// End the stream.
return false;
}
DWORD flags = 0;
LONGLONG timestampHns = 0;
RefPtr<IMFSample> sample;
hr = mSourceReaderCallback->Wait(&flags, ×tampHns, byRef(sample));
if (FAILED(hr) ||
(flags & MF_SOURCE_READERF_ERROR) ||
(flags & MF_SOURCE_READERF_ENDOFSTREAM) ||
(flags & MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED)) {
DECODER_LOG("WMFReader::DecodeAudioData() ReadSample failed with hr=0x%x flags=0x%x",
hr, flags);
// End the stream.
return false;
}
if (!sample) {
// Not enough data? Try again...
return true;
}
RefPtr<IMFMediaBuffer> buffer;
hr = sample->ConvertToContiguousBuffer(byRef(buffer));
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
BYTE* data = nullptr; // Note: *data will be owned by the IMFMediaBuffer, we don't need to free it.
DWORD maxLength = 0, currentLength = 0;
hr = buffer->Lock(&data, &maxLength, ¤tLength);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
uint32_t numFrames = currentLength / mAudioBytesPerSample / mAudioChannels;
NS_ASSERTION(sizeof(AudioDataValue) == mAudioBytesPerSample, "Size calculation is wrong");
nsAutoArrayPtr<AudioDataValue> pcmSamples(new AudioDataValue[numFrames * mAudioChannels]);
memcpy(pcmSamples.get(), data, currentLength);
buffer->Unlock();
// We calculate the timestamp and the duration based on the number of audio
// frames we've already played. We don't trust the timestamp stored on the
// IMFSample, as sometimes it's wrong, possibly due to buggy encoders?
// If this sample block comes after a discontinuity (i.e. a gap or seek)
// reset the frame counters, and capture the timestamp. Future timestamps
// will be offset from this block's timestamp.
UINT32 discontinuity = false;
sample->GetUINT32(MFSampleExtension_Discontinuity, &discontinuity);
if (mMustRecaptureAudioPosition || discontinuity) {
mAudioFrameSum = 0;
hr = HNsToFrames(timestampHns, mAudioRate, &mAudioFrameOffset);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
mMustRecaptureAudioPosition = false;
}
int64_t timestamp;
hr = FramesToUsecs(mAudioFrameOffset + mAudioFrameSum, mAudioRate, ×tamp);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
mAudioFrameSum += numFrames;
int64_t duration;
hr = FramesToUsecs(numFrames, mAudioRate, &duration);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
mAudioQueue.Push(new AudioData(mDecoder->GetResource()->Tell(),
timestamp,
duration,
numFrames,
pcmSamples.forget(),
mAudioChannels,
mAudioRate));
#ifdef LOG_SAMPLE_DECODE
DECODER_LOG("Decoded audio sample! timestamp=%lld duration=%lld currentLength=%u",
timestamp, duration, currentLength);
#endif
return true;
}
MediaResult
VorbisDataDecoder::DoDecode(MediaRawData* aSample)
{
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
const unsigned char* aData = aSample->Data();
size_t aLength = aSample->Size();
int64_t aOffset = aSample->mOffset;
uint64_t aTstampUsecs = aSample->mTime;
int64_t aTotalFrames = 0;
MOZ_ASSERT(mPacketCount >= 3);
if (!mLastFrameTime || mLastFrameTime.ref() != aSample->mTime) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime);
}
ogg_packet pkt = InitVorbisPacket(aData, aLength, false, aSample->mEOS,
aSample->mTimecode, mPacketCount++);
int err = vorbis_synthesis(&mVorbisBlock, &pkt);
if (err) {
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis:%d", err));
}
err = vorbis_synthesis_blockin(&mVorbisDsp, &mVorbisBlock);
if (err) {
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_blockin:%d", err));
}
VorbisPCMValue** pcm = 0;
int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
if (frames == 0) {
return NS_OK;
}
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
uint32_t rate = mVorbisDsp.vi->rate;
AlignedAudioBuffer buffer(frames*channels);
if (!buffer) {
return MediaResult(NS_ERROR_OUT_OF_MEMORY, __func__);
}
for (uint32_t j = 0; j < channels; ++j) {
VorbisPCMValue* channel = pcm[j];
for (uint32_t i = 0; i < uint32_t(frames); ++i) {
buffer[i*channels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
}
}
CheckedInt64 duration = FramesToUsecs(frames, rate);
if (!duration.isValid()) {
return MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio duration"));
}
CheckedInt64 total_duration = FramesToUsecs(mFrames, rate);
if (!total_duration.isValid()) {
return MediaResult(
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio total_duration"));
}
CheckedInt64 time = total_duration + aTstampUsecs;
if (!time.isValid()) {
return MediaResult(
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow adding total_duration and aTstampUsecs"));
};
if (!mAudioConverter) {
AudioConfig in(AudioConfig::ChannelLayout(channels, VorbisLayout(channels)),
rate);
AudioConfig out(channels, rate);
if (!in.IsValid() || !out.IsValid()) {
return MediaResult(
NS_ERROR_DOM_MEDIA_FATAL_ERR,
RESULT_DETAIL("Invalid channel layout:%u", channels));
}
mAudioConverter = MakeUnique<AudioConverter>(in, out);
}
MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
AudioSampleBuffer data(Move(buffer));
data = mAudioConverter->Process(Move(data));
aTotalFrames += frames;
mCallback->Output(new AudioData(aOffset,
time.value(),
duration.value(),
frames,
data.Forget(),
channels,
rate));
mFrames += frames;
err = vorbis_synthesis_read(&mVorbisDsp, frames);
if (err) {
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_read:%d", err));
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return NS_OK;
}
UniquePtr<AudioStream::Chunk>
AudioSink::PopFrames(uint32_t aFrames)
{
class Chunk : public AudioStream::Chunk {
public:
Chunk(AudioData* aBuffer, uint32_t aFrames, AudioDataValue* aData)
: mBuffer(aBuffer), mFrames(aFrames), mData(aData) {}
Chunk() : mFrames(0), mData(nullptr) {}
const AudioDataValue* Data() const override { return mData; }
uint32_t Frames() const override { return mFrames; }
uint32_t Channels() const override { return mBuffer ? mBuffer->mChannels: 0; }
uint32_t Rate() const override { return mBuffer ? mBuffer->mRate : 0; }
AudioDataValue* GetWritable() const override { return mData; }
private:
const RefPtr<AudioData> mBuffer;
const uint32_t mFrames;
AudioDataValue* const mData;
};
bool needPopping = false;
if (!mCurrentData) {
// No data in the queue. Return an empty chunk.
if (!mProcessedQueue.GetSize()) {
return MakeUnique<Chunk>();
}
// We need to update our values prior popping the processed queue in
// order to prevent the pop event to fire too early (prior
// mProcessedQueueLength being updated) or prevent HasUnplayedFrames
// to incorrectly return true during the time interval betweeen the
// when mProcessedQueue is read and mWritten is updated.
needPopping = true;
mCurrentData = mProcessedQueue.PeekFront();
{
MonitorAutoLock mon(mMonitor);
mCursor = MakeUnique<AudioBufferCursor>(mCurrentData->mAudioData.get(),
mCurrentData->mChannels,
mCurrentData->mFrames);
}
MOZ_ASSERT(mCurrentData->mFrames > 0);
mProcessedQueueLength -=
FramesToUsecs(mCurrentData->mFrames, mOutputRate).value();
}
auto framesToPop = std::min(aFrames, mCursor->Available());
SINK_LOG_V("playing audio at time=%" PRId64 " offset=%u length=%u",
mCurrentData->mTime.ToMicroseconds(),
mCurrentData->mFrames - mCursor->Available(), framesToPop);
UniquePtr<AudioStream::Chunk> chunk =
MakeUnique<Chunk>(mCurrentData, framesToPop, mCursor->Ptr());
{
MonitorAutoLock mon(mMonitor);
mWritten += framesToPop;
mCursor->Advance(framesToPop);
}
// All frames are popped. Reset mCurrentData so we can pop new elements from
// the audio queue in next calls to PopFrames().
if (!mCursor->Available()) {
mCurrentData = nullptr;
}
if (needPopping) {
// We can now safely pop the audio packet from the processed queue.
// This will fire the popped event, triggering a call to NotifyAudioNeeded.
RefPtr<AudioData> releaseMe = mProcessedQueue.PopFront();
CheckIsAudible(releaseMe);
}
return chunk;
}
void
AudioCallbackAdapter::Decoded(const nsTArray<int16_t>& aPCM, uint64_t aTimeStamp, uint32_t aChannels, uint32_t aRate)
{
MOZ_ASSERT(IsOnGMPThread());
if (aRate == 0 || aChannels == 0) {
NS_WARNING("Invalid rate or num channels returned on GMP audio samples");
mCallback->Error();
return;
}
size_t numFrames = aPCM.Length() / aChannels;
MOZ_ASSERT((aPCM.Length() % aChannels) == 0);
nsAutoArrayPtr<AudioDataValue> audioData(new AudioDataValue[aPCM.Length()]);
for (size_t i = 0; i < aPCM.Length(); ++i) {
audioData[i] = AudioSampleToFloat(aPCM[i]);
}
if (mMustRecaptureAudioPosition) {
mAudioFrameSum = 0;
auto timestamp = UsecsToFrames(aTimeStamp, aRate);
if (!timestamp.isValid()) {
NS_WARNING("Invalid timestamp");
mCallback->Error();
return;
}
mAudioFrameOffset = timestamp.value();
MOZ_ASSERT(mAudioFrameOffset >= 0);
mMustRecaptureAudioPosition = false;
}
auto timestamp = FramesToUsecs(mAudioFrameOffset + mAudioFrameSum, aRate);
if (!timestamp.isValid()) {
NS_WARNING("Invalid timestamp on audio samples");
mCallback->Error();
return;
}
mAudioFrameSum += numFrames;
auto duration = FramesToUsecs(numFrames, aRate);
if (!duration.isValid()) {
NS_WARNING("Invalid duration on audio samples");
mCallback->Error();
return;
}
nsRefPtr<AudioData> audio(new AudioData(mLastStreamOffset,
timestamp.value(),
duration.value(),
numFrames,
audioData.forget(),
aChannels,
aRate));
#ifdef LOG_SAMPLE_DECODE
LOG("Decoded audio sample! timestamp=%lld duration=%lld currentLength=%u",
timestamp, duration, currentLength);
#endif
mCallback->Output(audio);
}
nsresult
AppleATDecoder::DecodeSample(mp4_demuxer::MP4Sample* aSample)
{
// Array containing the queued decoded audio frames, about to be output.
nsTArray<AudioDataValue> outputData;
UInt32 channels = mOutputFormat.mChannelsPerFrame;
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state.
PassthroughUserData userData =
{ channels, (UInt32)aSample->size, aSample->data };
// Decompressed audio buffer
nsAutoArrayPtr<AudioDataValue> decoded(new AudioDataValue[maxDecodedSamples]);
do {
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = channels;
decBuffer.mBuffers[0].mDataByteSize =
maxDecodedSamples * sizeof(AudioDataValue);
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNoMoreDataErr) {
LOG("Error decoding audio stream: %d\n", rv);
return NS_ERROR_FAILURE;
}
if (numFrames) {
outputData.AppendElements(decoded.get(), numFrames * channels);
}
if (rv == kNoMoreDataErr) {
break;
}
} while (true);
if (outputData.IsEmpty()) {
return NS_OK;
}
size_t numFrames = outputData.Length() / channels;
int rate = mOutputFormat.mSampleRate;
CheckedInt<Microseconds> duration = FramesToUsecs(numFrames, rate);
if (!duration.isValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
return NS_ERROR_FAILURE;
}
#ifdef LOG_SAMPLE_DECODE
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)aSample->composition_timestamp / USECS_PER_S,
(double)duration.value() / USECS_PER_S);
#endif
nsAutoArrayPtr<AudioDataValue> data(new AudioDataValue[outputData.Length()]);
PodCopy(data.get(), &outputData[0], outputData.Length());
nsRefPtr<AudioData> audio = new AudioData(aSample->byte_offset,
aSample->composition_timestamp,
duration.value(),
numFrames,
data.forget(),
channels,
rate);
mCallback->Output(audio);
return NS_OK;
}
void
AppleATDecoder::SampleCallback(uint32_t aNumBytes,
uint32_t aNumPackets,
const void* aData,
AudioStreamPacketDescription* aPackets)
{
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t decodedSize = MAX_AUDIO_FRAMES * mConfig.channel_count *
sizeof(AudioDataValue);
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state and the result of the
// parser on to the callback above.
PassthroughUserData userData =
{ this, aNumPackets, aNumBytes, aData, aPackets, false };
do {
// Decompressed audio buffer
nsAutoArrayPtr<uint8_t> decoded(new uint8_t[decodedSize]);
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = mOutputFormat.mChannelsPerFrame;
decBuffer.mBuffers[0].mDataByteSize = decodedSize;
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNeedMoreData) {
LOG("Error decoding audio stream: %#x\n", rv);
mCallback->Error();
break;
}
LOG("%d frames decoded", numFrames);
// If we decoded zero frames then AudioConverterFillComplexBuffer is out
// of data to provide. We drained its internal buffer completely on the
// last pass.
if (numFrames == 0 && rv == kNeedMoreData) {
LOG("FillComplexBuffer out of data exactly\n");
mCallback->InputExhausted();
break;
}
const int rate = mOutputFormat.mSampleRate;
const int channels = mOutputFormat.mChannelsPerFrame;
int64_t time = mCurrentAudioTimestamp;
int64_t duration = FramesToUsecs(numFrames, rate).value();
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)time / USECS_PER_S, (double)duration / USECS_PER_S);
AudioData* audio = new AudioData(mSamplePosition,
time, duration, numFrames,
reinterpret_cast<AudioDataValue*>(decoded.forget()),
channels, rate);
mCallback->Output(audio);
mHaveOutput = true;
if (rv == kNeedMoreData) {
// No error; we just need more data.
LOG("FillComplexBuffer out of data\n");
mCallback->InputExhausted();
break;
}
} while (true);
}
/*
* This callback is called when |AudioFileStreamParseBytes| has enough data to
* extract one or more MP3 packets.
*/
void
AppleMP3Reader::AudioSampleCallback(UInt32 aNumBytes,
UInt32 aNumPackets,
const void *aData,
AudioStreamPacketDescription *aPackets)
{
LOGD("got %u bytes, %u packets\n", aNumBytes, aNumPackets);
// 1 frame per packet * num channels * 32-bit float
uint32_t decodedSize = MAX_AUDIO_FRAMES * mAudioChannels *
sizeof(AudioDataValue);
// descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having MP3 packets spoon-fed to it from a callback.
// This structure exists only to pass our state and the result of the parser
// on to the callback above.
PassthroughUserData userData = { this, aNumPackets, aNumBytes, aData, aPackets, false };
do {
// Decompressed audio buffer
nsAutoArrayPtr<uint8_t> decoded(new uint8_t[decodedSize]);
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = mAudioChannels;
decBuffer.mBuffers[0].mDataByteSize = decodedSize;
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mAudioConverter,
PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNeedMoreData) {
LOGE("Error decoding audio stream: %x\n", rv);
break;
}
// If we decoded zero frames then AudiOConverterFillComplexBuffer is out
// of data to provide. We drained its internal buffer completely on the
// last pass.
if (numFrames == 0 && rv == kNeedMoreData) {
LOGD("FillComplexBuffer out of data exactly\n");
break;
}
int64_t time = FramesToUsecs(mCurrentAudioFrame, mAudioSampleRate).value();
int64_t duration = FramesToUsecs(numFrames, mAudioSampleRate).value();
LOGD("pushed audio at time %lfs; duration %lfs\n",
(double)time / USECS_PER_S, (double)duration / USECS_PER_S);
AudioData *audio = new AudioData(mDecoder->GetResource()->Tell(),
time, duration, numFrames,
reinterpret_cast<AudioDataValue *>(decoded.forget()),
mAudioChannels, mAudioSampleRate);
mAudioQueue.Push(audio);
mCurrentAudioFrame += numFrames;
if (rv == kNeedMoreData) {
// No error; we just need more data.
LOGD("FillComplexBuffer out of data\n");
break;
}
} while (true);
}
HRESULT
WMFAudioMFTManager::Output(int64_t aStreamOffset,
nsAutoPtr<MediaData>& aOutData)
{
aOutData = nullptr;
RefPtr<IMFSample> sample;
HRESULT hr;
while (true) {
hr = mDecoder->Output(&sample);
if (hr == MF_E_TRANSFORM_NEED_MORE_INPUT) {
return hr;
}
if (hr == MF_E_TRANSFORM_STREAM_CHANGE) {
hr = UpdateOutputType();
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
continue;
}
break;
}
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
RefPtr<IMFMediaBuffer> buffer;
hr = sample->ConvertToContiguousBuffer(byRef(buffer));
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
BYTE* data = nullptr; // Note: *data will be owned by the IMFMediaBuffer, we don't need to free it.
DWORD maxLength = 0, currentLength = 0;
hr = buffer->Lock(&data, &maxLength, ¤tLength);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
// Sometimes when starting decoding, the AAC decoder gives us samples
// with a negative timestamp. AAC does usually have preroll (or encoder
// delay) encoded into its bitstream, but the amount encoded to the stream
// is variable, and it not signalled in-bitstream. There is sometimes
// signalling in the MP4 container what the preroll amount, but it's
// inconsistent. It looks like WMF's AAC encoder may take this into
// account, so strip off samples with a negative timestamp to get us
// to a 0-timestamp start. This seems to maintain A/V sync, so we can run
// with this until someone complains...
// We calculate the timestamp and the duration based on the number of audio
// frames we've already played. We don't trust the timestamp stored on the
// IMFSample, as sometimes it's wrong, possibly due to buggy encoders?
// If this sample block comes after a discontinuity (i.e. a gap or seek)
// reset the frame counters, and capture the timestamp. Future timestamps
// will be offset from this block's timestamp.
UINT32 discontinuity = false;
int32_t numFramesToStrip = 0;
sample->GetUINT32(MFSampleExtension_Discontinuity, &discontinuity);
if (mMustRecaptureAudioPosition || discontinuity) {
// Update the output type, in case this segment has a different
// rate. This also triggers on the first sample, which can have a
// different rate than is advertised in the container, and sometimes we
// don't get a MF_E_TRANSFORM_STREAM_CHANGE when the rate changes.
hr = UpdateOutputType();
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
mAudioFrameSum = 0;
LONGLONG timestampHns = 0;
hr = sample->GetSampleTime(×tampHns);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
hr = HNsToFrames(timestampHns, mAudioRate, &mAudioFrameOffset);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
if (mAudioFrameOffset < 0) {
// First sample has a negative timestamp. Strip off the samples until
// we reach positive territory.
numFramesToStrip = -mAudioFrameOffset;
mAudioFrameOffset = 0;
}
mMustRecaptureAudioPosition = false;
}
MOZ_ASSERT(numFramesToStrip >= 0);
int32_t numSamples = currentLength / mAudioBytesPerSample;
int32_t numFrames = numSamples / mAudioChannels;
int32_t offset = std::min<int32_t>(numFramesToStrip, numFrames);
numFrames -= offset;
numSamples -= offset * mAudioChannels;
MOZ_ASSERT(numFrames >= 0);
MOZ_ASSERT(numSamples >= 0);
if (numFrames == 0) {
// All data from this chunk stripped, loop back and try to output the next
// frame, if possible.
return S_OK;
}
nsAutoArrayPtr<AudioDataValue> audioData(new AudioDataValue[numSamples]);
// Just assume PCM output for now...
MOZ_ASSERT(mAudioBytesPerSample == 2);
int16_t* pcm = ((int16_t*)data) + (offset * mAudioChannels);
MOZ_ASSERT(pcm >= (int16_t*)data);
MOZ_ASSERT(pcm <= (int16_t*)(data + currentLength));
MOZ_ASSERT(pcm+numSamples <= (int16_t*)(data + currentLength));
for (int32_t i = 0; i < numSamples; ++i) {
audioData[i] = AudioSampleToFloat(pcm[i]);
}
buffer->Unlock();
int64_t timestamp;
hr = FramesToUsecs(mAudioFrameOffset + mAudioFrameSum, mAudioRate, ×tamp);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
mAudioFrameSum += numFrames;
int64_t duration;
hr = FramesToUsecs(numFrames, mAudioRate, &duration);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
aOutData = new AudioData(aStreamOffset,
timestamp,
duration,
numFrames,
audioData.forget(),
mAudioChannels,
mAudioRate);
#ifdef LOG_SAMPLE_DECODE
LOG("Decoded audio sample! timestamp=%lld duration=%lld currentLength=%u",
timestamp, duration, currentLength);
#endif
return S_OK;
}
int
VorbisDataDecoder::DoDecode(MediaRawData* aSample)
{
const unsigned char* aData = aSample->Data();
size_t aLength = aSample->Size();
int64_t aOffset = aSample->mOffset;
uint64_t aTstampUsecs = aSample->mTime;
int64_t aTotalFrames = 0;
MOZ_ASSERT(mPacketCount >= 3);
if (!mLastFrameTime || mLastFrameTime.ref() != aSample->mTime) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime);
}
ogg_packet pkt = InitVorbisPacket(aData, aLength, false, false, -1, mPacketCount++);
bool first_packet = mPacketCount == 4;
if (vorbis_synthesis(&mVorbisBlock, &pkt) != 0) {
return -1;
}
if (vorbis_synthesis_blockin(&mVorbisDsp,
&mVorbisBlock) != 0) {
return -1;
}
VorbisPCMValue** pcm = 0;
int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
// If the first packet of audio in the media produces no data, we
// still need to produce an AudioData for it so that the correct media
// start time is calculated. Otherwise we'd end up with a media start
// time derived from the timecode of the first packet that produced
// data.
if (frames == 0 && first_packet) {
mCallback->Output(new AudioData(aOffset,
aTstampUsecs,
0,
0,
nullptr,
mVorbisDsp.vi->channels,
mVorbisDsp.vi->rate));
}
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
auto buffer = MakeUnique<AudioDataValue[]>(frames*channels);
for (uint32_t j = 0; j < channels; ++j) {
VorbisPCMValue* channel = pcm[j];
for (uint32_t i = 0; i < uint32_t(frames); ++i) {
buffer[i*channels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
}
}
CheckedInt64 duration = FramesToUsecs(frames, mVorbisDsp.vi->rate);
if (!duration.isValid()) {
NS_WARNING("Int overflow converting WebM audio duration");
return -1;
}
CheckedInt64 total_duration = FramesToUsecs(mFrames,
mVorbisDsp.vi->rate);
if (!total_duration.isValid()) {
NS_WARNING("Int overflow converting WebM audio total_duration");
return -1;
}
CheckedInt64 time = total_duration + aTstampUsecs;
if (!time.isValid()) {
NS_WARNING("Int overflow adding total_duration and aTstampUsecs");
return -1;
};
aTotalFrames += frames;
mCallback->Output(new AudioData(aOffset,
time.value(),
duration.value(),
frames,
Move(buffer),
mVorbisDsp.vi->channels,
mVorbisDsp.vi->rate));
mFrames += frames;
if (vorbis_synthesis_read(&mVorbisDsp, frames) != 0) {
return -1;
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return aTotalFrames > 0 ? 1 : 0;
}
nsresult MediaDecoderReader::DecodeToTarget(int64_t aTarget)
{
DECODER_LOG(PR_LOG_DEBUG, ("MediaDecoderReader::DecodeToTarget(%lld) Begin", aTarget));
// Decode forward to the target frame. Start with video, if we have it.
if (HasVideo()) {
bool eof = false;
int64_t startTime = -1;
nsAutoPtr<VideoData> video;
while (HasVideo() && !eof) {
while (VideoQueue().GetSize() == 0 && !eof) {
bool skip = false;
eof = !DecodeVideoFrame(skip, 0);
{
ReentrantMonitorAutoEnter decoderMon(mDecoder->GetReentrantMonitor());
if (mDecoder->IsShutdown()) {
return NS_ERROR_FAILURE;
}
}
}
if (VideoQueue().GetSize() == 0) {
// Hit end of file, we want to display the last frame of the video.
if (video) {
VideoQueue().PushFront(video.forget());
}
break;
}
video = VideoQueue().PeekFront();
// If the frame end time is less than the seek target, we won't want
// to display this frame after the seek, so discard it.
if (video && video->GetEndTime() <= aTarget) {
if (startTime == -1) {
startTime = video->mTime;
}
VideoQueue().PopFront();
} else {
video.forget();
break;
}
}
{
ReentrantMonitorAutoEnter decoderMon(mDecoder->GetReentrantMonitor());
if (mDecoder->IsShutdown()) {
return NS_ERROR_FAILURE;
}
}
DECODER_LOG(PR_LOG_DEBUG, ("First video frame after decode is %lld", startTime));
}
if (HasAudio()) {
// Decode audio forward to the seek target.
bool eof = false;
while (HasAudio() && !eof) {
while (!eof && AudioQueue().GetSize() == 0) {
eof = !DecodeAudioData();
{
ReentrantMonitorAutoEnter decoderMon(mDecoder->GetReentrantMonitor());
if (mDecoder->IsShutdown()) {
return NS_ERROR_FAILURE;
}
}
}
const AudioData* audio = AudioQueue().PeekFront();
if (!audio)
break;
CheckedInt64 startFrame = UsecsToFrames(audio->mTime, mInfo.mAudio.mRate);
CheckedInt64 targetFrame = UsecsToFrames(aTarget, mInfo.mAudio.mRate);
if (!startFrame.isValid() || !targetFrame.isValid()) {
return NS_ERROR_FAILURE;
}
if (startFrame.value() + audio->mFrames <= targetFrame.value()) {
// Our seek target lies after the frames in this AudioData. Pop it
// off the queue, and keep decoding forwards.
delete AudioQueue().PopFront();
audio = nullptr;
continue;
}
if (startFrame.value() > targetFrame.value()) {
// The seek target doesn't lie in the audio block just after the last
// audio frames we've seen which were before the seek target. This
// could have been the first audio data we've seen after seek, i.e. the
// seek terminated after the seek target in the audio stream. Just
// abort the audio decode-to-target, the state machine will play
// silence to cover the gap. Typically this happens in poorly muxed
// files.
NS_WARNING("Audio not synced after seek, maybe a poorly muxed file?");
break;
}
// The seek target lies somewhere in this AudioData's frames, strip off
// any frames which lie before the seek target, so we'll begin playback
// exactly at the seek target.
NS_ASSERTION(targetFrame.value() >= startFrame.value(),
"Target must at or be after data start.");
NS_ASSERTION(targetFrame.value() < startFrame.value() + audio->mFrames,
"Data must end after target.");
int64_t framesToPrune = targetFrame.value() - startFrame.value();
if (framesToPrune > audio->mFrames) {
// We've messed up somehow. Don't try to trim frames, the |frames|
// variable below will overflow.
NS_WARNING("Can't prune more frames that we have!");
break;
}
uint32_t frames = audio->mFrames - static_cast<uint32_t>(framesToPrune);
uint32_t channels = audio->mChannels;
nsAutoArrayPtr<AudioDataValue> audioData(new AudioDataValue[frames * channels]);
memcpy(audioData.get(),
audio->mAudioData.get() + (framesToPrune * channels),
frames * channels * sizeof(AudioDataValue));
CheckedInt64 duration = FramesToUsecs(frames, mInfo.mAudio.mRate);
if (!duration.isValid()) {
return NS_ERROR_FAILURE;
}
nsAutoPtr<AudioData> data(new AudioData(audio->mOffset,
aTarget,
duration.value(),
frames,
audioData.forget(),
channels));
delete AudioQueue().PopFront();
AudioQueue().PushFront(data.forget());
break;
}
}
DECODER_LOG(PR_LOG_DEBUG, ("MediaDecoderReader::DecodeToTarget(%lld) End", aTarget));
return NS_OK;
}
