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Bug 790559 - Part 1: Apply DownmixToStereo function when decode multichannel Opus and Vorbis. r=rillian
This commit is contained in:
Alexandros Chronopoulos
parent
1414e59ca9
commit
162156997a
@ -298,7 +298,7 @@ nsresult OggReader::ReadMetadata(VideoInfo* aInfo,
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if (mVorbisState && ReadHeaders(mVorbisState)) {
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mInfo.mHasAudio = true;
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mInfo.mAudioRate = mVorbisState->mInfo.rate;
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mInfo.mAudioChannels = mVorbisState->mInfo.channels;
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mInfo.mAudioChannels = mVorbisState->mInfo.channels > 2 ? 2 : mVorbisState->mInfo.channels;
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// Copy Vorbis info data for time computations on other threads.
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memcpy(&mVorbisInfo, &mVorbisState->mInfo, sizeof(mVorbisInfo));
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mVorbisInfo.codec_setup = NULL;
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@ -404,6 +404,15 @@ nsresult OggReader::DecodeVorbis(ogg_packet* aPacket) {
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}
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}
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// More than 2 decoded channels must be downmixed to stereo.
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if (channels > 2) {
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if (channels > 8) {
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// No channel mapping for more than 8 channels.
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return NS_ERROR_FAILURE;
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}
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DownmixToStereo(buffer, channels, frames);
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}
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int64_t duration = mVorbisState->Time((int64_t)frames);
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int64_t startTime = mVorbisState->Time(endFrame - frames);
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mAudioQueue.Push(new AudioData(mPageOffset,
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@ -520,57 +529,7 @@ nsresult OggReader::DecodeOpus(ogg_packet* aPacket) {
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// so we can't downmix more than that.
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if (channels > 8)
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return NS_ERROR_FAILURE;
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uint32_t out_channels;
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out_channels = 2;
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// dBuffer stores the downmixed sample data.
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nsAutoArrayPtr<AudioDataValue> dBuffer(new AudioDataValue[frames * out_channels]);
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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static const float dmatrix[6][8][2]= {
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/*3*/{ {0.5858f,0}, {0.4142f,0.4142f}, {0,0.5858f}},
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/*4*/{ {0.4226f,0}, {0,0.4226f}, {0.366f,0.2114f}, {0.2114f,0.366f}},
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/*5*/{ {0.651f,0}, {0.46f,0.46f}, {0,0.651f}, {0.5636f,0.3254f}, {0.3254f,0.5636f}},
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/*6*/{ {0.529f,0}, {0.3741f,0.3741f}, {0,0.529f}, {0.4582f,0.2645f}, {0.2645f,0.4582f}, {0.3741f,0.3741f}},
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/*7*/{ {0.4553f,0}, {0.322f,0.322f}, {0,0.4553f}, {0.3943f,0.2277f}, {0.2277f,0.3943f}, {0.2788f,0.2788f}, {0.322f,0.322f}},
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/*8*/{ {0.3886f,0}, {0.2748f,0.2748f}, {0,0.3886f}, {0.3366f,0.1943f}, {0.1943f,0.3366f}, {0.3366f,0.1943f}, {0.1943f,0.3366f}, {0.2748f,0.2748f}},
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};
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for (int32_t i = 0; i < frames; i++) {
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float sampL = 0.0;
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float sampR = 0.0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL+=buffer[i*channels+j]*dmatrix[channels-3][j][0];
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sampR+=buffer[i*channels+j]*dmatrix[channels-3][j][1];
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}
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dBuffer[i*out_channels]=sampL;
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dBuffer[i*out_channels+1]=sampR;
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}
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#else
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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// Coefficients in Q14.
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static const int16_t dmatrix[6][8][2]= {
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/*3*/{{9598, 0},{6786,6786},{0, 9598}},
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/*4*/{{6925, 0},{0, 6925},{5997,3462},{3462,5997}},
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/*5*/{{10663,0},{7540,7540},{0, 10663},{9234,5331},{5331,9234}},
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/*6*/{{8668, 0},{6129,6129},{0, 8668},{7507,4335},{4335,7507},{6129,6129}},
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/*7*/{{7459, 0},{5275,5275},{0, 7459},{6460,3731},{3731,6460},{4568,4568},{5275,5275}},
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/*8*/{{6368, 0},{4502,4502},{0, 6368},{5514,3184},{3184,5514},{5514,3184},{3184,5514},{4502,4502}}
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};
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for (int32_t i = 0; i < frames; i++) {
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int32_t sampL = 0;
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int32_t sampR = 0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL+=buffer[i*channels+j]*dmatrix[channels-3][j][0];
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sampR+=buffer[i*channels+j]*dmatrix[channels-3][j][1];
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}
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sampL = (sampL + 8192)>>14;
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dBuffer[i*out_channels] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(sampL));
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sampR = (sampR + 8192)>>14;
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dBuffer[i*out_channels+1] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(sampR));
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}
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#endif
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channels = out_channels;
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buffer = dBuffer;
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DownmixToStereo(buffer, channels, frames);
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}
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LOG(PR_LOG_DEBUG, ("Opus decoder pushing %d frames", frames));
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@ -589,6 +548,61 @@ nsresult OggReader::DecodeOpus(ogg_packet* aPacket) {
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}
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#endif /* MOZ_OPUS */
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void OggReader::DownmixToStereo(nsAutoArrayPtr<AudioDataValue>& buffer,
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uint32_t& channels, int32_t frames)
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{
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uint32_t out_channels;
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out_channels = 2;
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// dBuffer stores the downmixed samples.
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nsAutoArrayPtr<AudioDataValue> dBuffer(new AudioDataValue[frames * out_channels]);
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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static const float dmatrix[6][8][2]= {
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/*3*/{{0.5858f,0},{0.4142f,0.4142f},{0, 0.5858f}},
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/*4*/{{0.4226f,0},{0, 0.4226f},{0.366f,0.2114f},{0.2114f,0.366f}},
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/*5*/{{0.6510f,0},{0.4600f,0.4600f},{0, 0.6510f},{0.5636f,0.3254f},{0.3254f,0.5636f}},
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/*6*/{{0.5290f,0},{0.3741f,0.3741f},{0, 0.5290f},{0.4582f,0.2645f},{0.2645f,0.4582f},{0.3741f,0.3741f}},
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/*7*/{{0.4553f,0},{0.3220f,0.3220f},{0, 0.4553f},{0.3943f,0.2277f},{0.2277f,0.3943f},{0.2788f,0.2788f},{0.3220f,0.3220f}},
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/*8*/{{0.3886f,0},{0.2748f,0.2748f},{0, 0.3886f},{0.3366f,0.1943f},{0.1943f,0.3366f},{0.3366f,0.1943f},{0.1943f,0.3366f},{0.2748f,0.2748f}},
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};
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for (int32_t i = 0; i < frames; i++) {
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float sampL = 0.0;
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float sampR = 0.0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL+=buffer[i*channels+j]*dmatrix[channels-3][j][0];
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sampR+=buffer[i*channels+j]*dmatrix[channels-3][j][1];
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}
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dBuffer[i*out_channels]=sampL;
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dBuffer[i*out_channels+1]=sampR;
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}
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#else
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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// Coefficients in Q14.
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static const int16_t dmatrix[6][8][2]= {
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/*3*/{{9598, 0},{6786,6786},{0, 9598}},
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/*4*/{{6925, 0},{0, 6925},{5997,3462},{3462,5997}},
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/*5*/{{10663,0},{7540,7540},{0, 10663},{9234,5331},{5331,9234}},
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/*6*/{{8668, 0},{6129,6129},{0, 8668},{7507,4335},{4335,7507},{6129,6129}},
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/*7*/{{7459, 0},{5275,5275},{0, 7459},{6460,3731},{3731,6460},{4568,4568},{5275,5275}},
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/*8*/{{6368, 0},{4502,4502},{0, 6368},{5514,3184},{3184,5514},{5514,3184},{3184,5514},{4502,4502}}
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};
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for (int32_t i = 0; i < frames; i++) {
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int32_t sampL = 0;
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int32_t sampR = 0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL+=buffer[i*channels+j]*dmatrix[channels-3][j][0];
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sampR+=buffer[i*channels+j]*dmatrix[channels-3][j][1];
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}
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sampL = (sampL + 8192)>>14;
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dBuffer[i*out_channels] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(sampL));
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sampR = (sampR + 8192)>>14;
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dBuffer[i*out_channels+1] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(sampR));
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}
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#endif
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channels = out_channels;
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buffer = dBuffer;
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}
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bool OggReader::DecodeAudioData()
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{
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NS_ASSERTION(mDecoder->OnDecodeThread(), "Should be on decode thread.");
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@ -220,6 +220,13 @@ private:
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// audio queue.
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nsresult DecodeOpus(ogg_packet* aPacket);
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// Downmix multichannel Audio samples to Stereo.
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// It is used from Vorbis and Opus decoders.
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// Input are the buffer contains multichannel data,
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// the number of channels and the number of frames.
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void DownmixToStereo(nsAutoArrayPtr<AudioDataValue>& buffer,
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uint32_t& channel, int32_t frames);
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// Decodes a packet of Theora data, and inserts its frame into the
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// video queue. May return NS_ERROR_OUT_OF_MEMORY. Caller must have obtained
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// the reader's monitor. aTimeThreshold is the current playback position
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