void AudioNodeStream::AccumulateInputChunk(uint32_t aInputIndex, const AudioBlock& aChunk, AudioBlock* aBlock, nsTArray<float>* aDownmixBuffer) { nsAutoTArray<const float*,GUESS_AUDIO_CHANNELS> channels; UpMixDownMixChunk(&aChunk, aBlock->ChannelCount(), channels, *aDownmixBuffer); for (uint32_t c = 0; c < channels.Length(); ++c) { const float* inputData = static_cast<const float*>(channels[c]); float* outputData = aBlock->ChannelFloatsForWrite(c); if (inputData) { if (aInputIndex == 0) { AudioBlockCopyChannelWithScale(inputData, aChunk.mVolume, outputData); } else { AudioBlockAddChannelWithScale(inputData, aChunk.mVolume, outputData); } } else { if (aInputIndex == 0) { PodZero(outputData, WEBAUDIO_BLOCK_SIZE); } } } }
void AnalyserNode::AppendChunk(const AudioChunk& aChunk) { const uint32_t bufferSize = mBuffer.Length(); const uint32_t channelCount = aChunk.mChannelData.Length(); uint32_t chunkDuration = aChunk.mDuration; MOZ_ASSERT((bufferSize & (bufferSize - 1)) == 0); // Must be a power of two! MOZ_ASSERT(channelCount > 0); MOZ_ASSERT(chunkDuration == WEBAUDIO_BLOCK_SIZE); if (chunkDuration > bufferSize) { // Copy a maximum bufferSize samples. chunkDuration = bufferSize; } PodCopy(mBuffer.Elements() + mWriteIndex, static_cast<const float*>(aChunk.mChannelData[0]), chunkDuration); for (uint32_t i = 1; i < channelCount; ++i) { AudioBlockAddChannelWithScale(static_cast<const float*>(aChunk.mChannelData[i]), 1.0f, mBuffer.Elements() + mWriteIndex); } if (channelCount > 1) { AudioBlockInPlaceScale(mBuffer.Elements() + mWriteIndex, 1.0f / aChunk.mChannelData.Length()); } mWriteIndex += chunkDuration; MOZ_ASSERT(mWriteIndex <= bufferSize); if (mWriteIndex >= bufferSize) { mWriteIndex = 0; } }
AudioChunk* AudioNodeStream::ObtainInputBlock(AudioChunk* aTmpChunk) { uint32_t inputCount = mInputs.Length(); uint32_t outputChannelCount = 0; nsAutoTArray<AudioChunk*,250> inputChunks; for (uint32_t i = 0; i < inputCount; ++i) { MediaStream* s = mInputs[i]->GetSource(); AudioNodeStream* a = static_cast<AudioNodeStream*>(s); MOZ_ASSERT(a == s->AsAudioNodeStream()); if (a->IsFinishedOnGraphThread()) { continue; } AudioChunk* chunk = &a->mLastChunk; // XXX when we implement DelayNode, this will no longer be true and we'll // need to treat a null chunk (when the DelayNode hasn't had a chance // to produce data yet) as silence here. MOZ_ASSERT(chunk); if (chunk->IsNull()) { continue; } inputChunks.AppendElement(chunk); outputChannelCount = GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length()); } uint32_t inputChunkCount = inputChunks.Length(); if (inputChunkCount == 0) { aTmpChunk->SetNull(WEBAUDIO_BLOCK_SIZE); return aTmpChunk; } if (inputChunkCount == 1) { return inputChunks[0]; } AllocateAudioBlock(outputChannelCount, aTmpChunk); for (uint32_t i = 0; i < inputChunkCount; ++i) { AudioChunk* chunk = inputChunks[i]; nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels; channels.AppendElements(chunk->mChannelData); if (channels.Length() < outputChannelCount) { AudioChannelsUpMix(&channels, outputChannelCount, nullptr); NS_ASSERTION(outputChannelCount == channels.Length(), "We called GetAudioChannelsSuperset to avoid this"); } for (uint32_t c = 0; c < channels.Length(); ++c) { const float* inputData = static_cast<const float*>(channels[c]); float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk->mChannelData[c])); if (inputData) { if (i == 0) { AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData); } else { AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData); } } else { if (i == 0) { memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float)); } } } } return aTmpChunk; }
void AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex) { uint32_t inputCount = mInputs.Length(); uint32_t outputChannelCount = 1; nsAutoTArray<AudioChunk*,250> inputChunks; for (uint32_t i = 0; i < inputCount; ++i) { if (aPortIndex != mInputs[i]->InputNumber()) { // This input is connected to a different port continue; } MediaStream* s = mInputs[i]->GetSource(); AudioNodeStream* a = static_cast<AudioNodeStream*>(s); MOZ_ASSERT(a == s->AsAudioNodeStream()); if (a->IsFinishedOnGraphThread() || a->IsAudioParamStream()) { continue; } AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()]; MOZ_ASSERT(chunk); if (chunk->IsNull()) { continue; } inputChunks.AppendElement(chunk); outputChannelCount = GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length()); } switch (mChannelCountMode) { case ChannelCountMode::Explicit: // Disregard the output channel count that we've calculated, and just use // mNumberOfInputChannels. outputChannelCount = mNumberOfInputChannels; break; case ChannelCountMode::Clamped_max: // Clamp the computed output channel count to mNumberOfInputChannels. outputChannelCount = std::min(outputChannelCount, mNumberOfInputChannels); break; case ChannelCountMode::Max: // Nothing to do here, just shut up the compiler warning. break; } uint32_t inputChunkCount = inputChunks.Length(); if (inputChunkCount == 0 || (inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) { aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE); return; } if (inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == outputChannelCount) { aTmpChunk = *inputChunks[0]; return; } AllocateAudioBlock(outputChannelCount, &aTmpChunk); float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f}; // The static storage here should be 1KB, so it's fine nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer; for (uint32_t i = 0; i < inputChunkCount; ++i) { AudioChunk* chunk = inputChunks[i]; nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels; channels.AppendElements(chunk->mChannelData); if (channels.Length() < outputChannelCount) { if (mChannelInterpretation == ChannelInterpretation::Speakers) { AudioChannelsUpMix(&channels, outputChannelCount, nullptr); NS_ASSERTION(outputChannelCount == channels.Length(), "We called GetAudioChannelsSuperset to avoid this"); } else { // Fill up the remaining channels by zeros for (uint32_t j = channels.Length(); j < outputChannelCount; ++j) { channels.AppendElement(silenceChannel); } } } else if (channels.Length() > outputChannelCount) { if (mChannelInterpretation == ChannelInterpretation::Speakers) { nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels; outputChannels.SetLength(outputChannelCount); downmixBuffer.SetLength(outputChannelCount * WEBAUDIO_BLOCK_SIZE); for (uint32_t j = 0; j < outputChannelCount; ++j) { outputChannels[j] = &downmixBuffer[j * WEBAUDIO_BLOCK_SIZE]; } AudioChannelsDownMix(channels, outputChannels.Elements(), outputChannelCount, WEBAUDIO_BLOCK_SIZE); channels.SetLength(outputChannelCount); for (uint32_t j = 0; j < channels.Length(); ++j) { channels[j] = outputChannels[j]; } } else { // Drop the remaining channels channels.RemoveElementsAt(outputChannelCount, channels.Length() - outputChannelCount); } } for (uint32_t c = 0; c < channels.Length(); ++c) { const float* inputData = static_cast<const float*>(channels[c]); float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk.mChannelData[c])); if (inputData) { if (i == 0) { AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData); } else { AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData); } } else { if (i == 0) { memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float)); } } } } }