/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ /* vim:set ts=2 sw=2 sts=2 et cindent: */ /* This Source Code Form is subject to the terms of the Mozilla Public * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ #include "mozilla/dom/AnalyserNode.h" #include "mozilla/dom/AnalyserNodeBinding.h" #include "AudioNodeEngine.h" #include "AudioNodeTrack.h" #include "mozilla/Mutex.h" #include "mozilla/PodOperations.h" #include "nsMathUtils.h" #include "Tracing.h" namespace mozilla { static const uint32_t MAX_FFT_SIZE = 32768; static const size_t CHUNK_COUNT = MAX_FFT_SIZE >> WEBAUDIO_BLOCK_SIZE_BITS; static_assert(MAX_FFT_SIZE == CHUNK_COUNT * WEBAUDIO_BLOCK_SIZE, "MAX_FFT_SIZE must be a multiple of WEBAUDIO_BLOCK_SIZE"); static_assert((CHUNK_COUNT & (CHUNK_COUNT - 1)) == 0, "CHUNK_COUNT must be power of 2 for remainder behavior"); namespace dom { class AnalyserNodeEngine final : public AudioNodeEngine { class TransferBuffer final : public Runnable { public: TransferBuffer(AudioNodeTrack* aTrack, const AudioChunk& aChunk) : Runnable("dom::AnalyserNodeEngine::TransferBuffer"), mTrack(aTrack), mChunk(aChunk) {} NS_IMETHOD Run() override { RefPtr node = static_cast(mTrack->Engine()->NodeMainThread()); if (node) { node->AppendChunk(mChunk); } return NS_OK; } private: RefPtr mTrack; AudioChunk mChunk; }; public: explicit AnalyserNodeEngine(AnalyserNode* aNode) : AudioNodeEngine(aNode) { MOZ_ASSERT(NS_IsMainThread()); } virtual void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom, const AudioBlock& aInput, AudioBlock* aOutput, bool* aFinished) override { TRACE("AnalyserNodeEngine::ProcessBlock"); *aOutput = aInput; if (aInput.IsNull()) { // If AnalyserNode::mChunks has only null chunks, then there is no need // to send further null chunks. if (mChunksToProcess == 0) { return; } --mChunksToProcess; if (mChunksToProcess == 0) { aTrack->ScheduleCheckForInactive(); } } else { // This many null chunks will be required to empty AnalyserNode::mChunks. mChunksToProcess = CHUNK_COUNT; } RefPtr transfer = new TransferBuffer(aTrack, aInput.AsAudioChunk()); mAbstractMainThread->Dispatch(transfer.forget()); } virtual bool IsActive() const override { return mChunksToProcess != 0; } virtual size_t SizeOfIncludingThis( MallocSizeOf aMallocSizeOf) const override { return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf); } uint32_t mChunksToProcess = 0; }; /* static */ already_AddRefed AnalyserNode::Create( AudioContext& aAudioContext, const AnalyserOptions& aOptions, ErrorResult& aRv) { RefPtr analyserNode = new AnalyserNode(&aAudioContext); analyserNode->Initialize(aOptions, aRv); if (NS_WARN_IF(aRv.Failed())) { return nullptr; } analyserNode->SetFftSize(aOptions.mFftSize, aRv); if (NS_WARN_IF(aRv.Failed())) { return nullptr; } analyserNode->SetMinAndMaxDecibels(aOptions.mMinDecibels, aOptions.mMaxDecibels, aRv); if (NS_WARN_IF(aRv.Failed())) { return nullptr; } analyserNode->SetSmoothingTimeConstant(aOptions.mSmoothingTimeConstant, aRv); if (NS_WARN_IF(aRv.Failed())) { return nullptr; } return analyserNode.forget(); } AnalyserNode::AnalyserNode(AudioContext* aContext) : AudioNode(aContext, 2, ChannelCountMode::Max, ChannelInterpretation::Speakers), mAnalysisBlock(2048), mMinDecibels(-100.), mMaxDecibels(-30.), mSmoothingTimeConstant(.8) { mTrack = AudioNodeTrack::Create(aContext, new AnalyserNodeEngine(this), AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph()); // Enough chunks must be recorded to handle the case of fftSize being // increased to maximum immediately before getFloatTimeDomainData() is // called, for example. Unused << mChunks.SetLength(CHUNK_COUNT, fallible); AllocateBuffer(); } size_t AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const { size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf); amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf); amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf); amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf); return amount; } size_t AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const { return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf); } JSObject* AnalyserNode::WrapObject(JSContext* aCx, JS::Handle aGivenProto) { return AnalyserNode_Binding::Wrap(aCx, this, aGivenProto); } void AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv) { // Disallow values that are not a power of 2 and outside the [32,32768] range if (aValue < 32 || aValue > MAX_FFT_SIZE || (aValue & (aValue - 1)) != 0) { aRv.ThrowIndexSizeError(nsPrintfCString( "FFT size %u is not a power of two in the range 32 to 32768", aValue)); return; } if (FftSize() != aValue) { mAnalysisBlock.SetFFTSize(aValue); AllocateBuffer(); } } void AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv) { if (aValue >= mMaxDecibels) { aRv.ThrowIndexSizeError(nsPrintfCString( "%g is not strictly smaller than current maxDecibels (%g)", aValue, mMaxDecibels)); return; } mMinDecibels = aValue; } void AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv) { if (aValue <= mMinDecibels) { aRv.ThrowIndexSizeError(nsPrintfCString( "%g is not strictly larger than current minDecibels (%g)", aValue, mMinDecibels)); return; } mMaxDecibels = aValue; } void AnalyserNode::SetMinAndMaxDecibels(double aMinValue, double aMaxValue, ErrorResult& aRv) { if (aMinValue >= aMaxValue) { aRv.ThrowIndexSizeError(nsPrintfCString( "minDecibels value (%g) must be smaller than maxDecibels value (%g)", aMinValue, aMaxValue)); return; } mMinDecibels = aMinValue; mMaxDecibels = aMaxValue; } void AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv) { if (aValue < 0 || aValue > 1) { aRv.ThrowIndexSizeError( nsPrintfCString("%g is not in the range [0, 1]", aValue)); return; } mSmoothingTimeConstant = aValue; } void AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray) { if (!FFTAnalysis()) { // Might fail to allocate memory return; } aArray.ComputeState(); float* buffer = aArray.Data(); size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length()); for (size_t i = 0; i < length; ++i) { buffer[i] = WebAudioUtils::ConvertLinearToDecibels( mOutputBuffer[i], -std::numeric_limits::infinity()); } } void AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray) { if (!FFTAnalysis()) { // Might fail to allocate memory return; } const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels); aArray.ComputeState(); unsigned char* buffer = aArray.Data(); size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length()); for (size_t i = 0; i < length; ++i) { const double decibels = WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels); // scale down the value to the range of [0, UCHAR_MAX] const double scaled = std::max( 0.0, std::min(double(UCHAR_MAX), UCHAR_MAX * (decibels - mMinDecibels) * rangeScaleFactor)); buffer[i] = static_cast(scaled); } } void AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray) { aArray.ComputeState(); float* buffer = aArray.Data(); size_t length = std::min(aArray.Length(), FftSize()); GetTimeDomainData(buffer, length); } void AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray) { aArray.ComputeState(); size_t length = std::min(aArray.Length(), FftSize()); AlignedTArray tmpBuffer; if (!tmpBuffer.SetLength(length, fallible)) { return; } GetTimeDomainData(tmpBuffer.Elements(), length); unsigned char* buffer = aArray.Data(); for (size_t i = 0; i < length; ++i) { const float value = tmpBuffer[i]; // scale the value to the range of [0, UCHAR_MAX] const float scaled = std::max(0.0f, std::min(float(UCHAR_MAX), 128.0f * (value + 1.0f))); buffer[i] = static_cast(scaled); } } bool AnalyserNode::FFTAnalysis() { AlignedTArray tmpBuffer; size_t fftSize = FftSize(); if (!tmpBuffer.SetLength(fftSize, fallible)) { return false; } float* inputBuffer = tmpBuffer.Elements(); GetTimeDomainData(inputBuffer, fftSize); ApplyBlackmanWindow(inputBuffer, fftSize); mAnalysisBlock.PerformFFT(inputBuffer); // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT // scaling factor). const double magnitudeScale = 1.0 / fftSize; for (uint32_t i = 0; i < mOutputBuffer.Length(); ++i) { double scalarMagnitude = NS_hypot(mAnalysisBlock.RealData(i), mAnalysisBlock.ImagData(i)) * magnitudeScale; mOutputBuffer[i] = mSmoothingTimeConstant * mOutputBuffer[i] + (1.0 - mSmoothingTimeConstant) * scalarMagnitude; } return true; } void AnalyserNode::ApplyBlackmanWindow(float* aBuffer, uint32_t aSize) { double alpha = 0.16; double a0 = 0.5 * (1.0 - alpha); double a1 = 0.5; double a2 = 0.5 * alpha; for (uint32_t i = 0; i < aSize; ++i) { double x = double(i) / aSize; double window = a0 - a1 * cos(2 * M_PI * x) + a2 * cos(4 * M_PI * x); aBuffer[i] *= window; } } bool AnalyserNode::AllocateBuffer() { bool result = true; if (mOutputBuffer.Length() != FrequencyBinCount()) { if (!mOutputBuffer.SetLength(FrequencyBinCount(), fallible)) { return false; } memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount()); } return result; } void AnalyserNode::AppendChunk(const AudioChunk& aChunk) { if (mChunks.Length() == 0) { return; } ++mCurrentChunk; mChunks[mCurrentChunk & (CHUNK_COUNT - 1)] = aChunk; } // Reads into aData the oldest aLength samples of the fftSize most recent // samples. void AnalyserNode::GetTimeDomainData(float* aData, size_t aLength) { size_t fftSize = FftSize(); MOZ_ASSERT(aLength <= fftSize); if (mChunks.Length() == 0) { PodZero(aData, aLength); return; } size_t readChunk = mCurrentChunk - ((fftSize - 1) >> WEBAUDIO_BLOCK_SIZE_BITS); size_t readIndex = (0 - fftSize) & (WEBAUDIO_BLOCK_SIZE - 1); MOZ_ASSERT(readIndex == 0 || readIndex + fftSize == WEBAUDIO_BLOCK_SIZE); for (size_t writeIndex = 0; writeIndex < aLength;) { const AudioChunk& chunk = mChunks[readChunk & (CHUNK_COUNT - 1)]; const size_t channelCount = chunk.ChannelCount(); size_t copyLength = std::min(aLength - writeIndex, WEBAUDIO_BLOCK_SIZE); float* dataOut = &aData[writeIndex]; if (channelCount == 0) { PodZero(dataOut, copyLength); } else { float scale = chunk.mVolume / channelCount; { // channel 0 auto channelData = static_cast(chunk.mChannelData[0]) + readIndex; AudioBufferCopyWithScale(channelData, scale, dataOut, copyLength); } for (uint32_t i = 1; i < channelCount; ++i) { auto channelData = static_cast(chunk.mChannelData[i]) + readIndex; AudioBufferAddWithScale(channelData, scale, dataOut, copyLength); } } readChunk++; writeIndex += copyLength; } } } // namespace dom } // namespace mozilla