mirror of
https://github.com/mozilla/gecko-dev.git
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344 lines
9.4 KiB
C++
344 lines
9.4 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "mozilla/dom/AnalyserNode.h"
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#include "mozilla/dom/AnalyserNodeBinding.h"
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#include "AudioNodeEngine.h"
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#include "AudioNodeStream.h"
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#include "mozilla/Mutex.h"
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#include "mozilla/PodOperations.h"
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namespace mozilla {
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namespace dom {
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NS_IMPL_ISUPPORTS_INHERITED0(AnalyserNode, AudioNode)
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class AnalyserNodeEngine : public AudioNodeEngine
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{
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class TransferBuffer : public nsRunnable
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{
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public:
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TransferBuffer(AudioNodeStream* aStream,
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const AudioChunk& aChunk)
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: mStream(aStream)
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, mChunk(aChunk)
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{
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}
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NS_IMETHOD Run()
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{
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nsRefPtr<AnalyserNode> node;
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{
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// No need to keep holding the lock for the whole duration of this
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// function, since we're holding a strong reference to it, so if
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// we can obtain the reference, we will hold the node alive in
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// this function.
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MutexAutoLock lock(mStream->Engine()->NodeMutex());
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node = static_cast<AnalyserNode*>(mStream->Engine()->Node());
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}
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if (node) {
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node->AppendChunk(mChunk);
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}
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return NS_OK;
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}
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private:
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nsRefPtr<AudioNodeStream> mStream;
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AudioChunk mChunk;
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};
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public:
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explicit AnalyserNodeEngine(AnalyserNode* aNode)
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: AudioNodeEngine(aNode)
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{
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MOZ_ASSERT(NS_IsMainThread());
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}
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virtual void ProcessBlock(AudioNodeStream* aStream,
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const AudioChunk& aInput,
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AudioChunk* aOutput,
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bool* aFinished) MOZ_OVERRIDE
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{
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*aOutput = aInput;
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MutexAutoLock lock(NodeMutex());
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if (Node() &&
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aInput.mChannelData.Length() > 0) {
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nsRefPtr<TransferBuffer> transfer = new TransferBuffer(aStream, aInput);
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NS_DispatchToMainThread(transfer);
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}
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}
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virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
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{
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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};
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AnalyserNode::AnalyserNode(AudioContext* aContext)
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: AudioNode(aContext,
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1,
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ChannelCountMode::Max,
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ChannelInterpretation::Speakers)
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, mAnalysisBlock(2048)
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, mMinDecibels(-100.)
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, mMaxDecibels(-30.)
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, mSmoothingTimeConstant(.8)
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, mWriteIndex(0)
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{
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mStream = aContext->Graph()->CreateAudioNodeStream(new AnalyserNodeEngine(this),
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MediaStreamGraph::INTERNAL_STREAM);
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AllocateBuffer();
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}
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size_t
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AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
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{
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size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
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amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf);
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amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
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amount += mOutputBuffer.SizeOfExcludingThis(aMallocSizeOf);
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return amount;
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}
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size_t
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AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
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{
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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JSObject*
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AnalyserNode::WrapObject(JSContext* aCx)
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{
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return AnalyserNodeBinding::Wrap(aCx, this);
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}
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void
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AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv)
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{
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// Disallow values that are not a power of 2 and outside the [32,32768] range
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if (aValue < 32 ||
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aValue > 32768 ||
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(aValue & (aValue - 1)) != 0) {
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aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
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return;
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}
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if (FftSize() != aValue) {
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mAnalysisBlock.SetFFTSize(aValue);
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AllocateBuffer();
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}
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}
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void
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AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv)
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{
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if (aValue >= mMaxDecibels) {
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aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
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return;
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}
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mMinDecibels = aValue;
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}
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void
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AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv)
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{
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if (aValue <= mMinDecibels) {
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aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
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return;
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}
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mMaxDecibels = aValue;
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}
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void
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AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv)
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{
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if (aValue < 0 || aValue > 1) {
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aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
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return;
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}
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mSmoothingTimeConstant = aValue;
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}
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void
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AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray)
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{
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if (!FFTAnalysis()) {
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// Might fail to allocate memory
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return;
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}
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aArray.ComputeLengthAndData();
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float* buffer = aArray.Data();
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size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
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for (size_t i = 0; i < length; ++i) {
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buffer[i] = WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels);
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}
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}
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void
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AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray)
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{
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if (!FFTAnalysis()) {
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// Might fail to allocate memory
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return;
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}
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const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels);
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aArray.ComputeLengthAndData();
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unsigned char* buffer = aArray.Data();
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size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
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for (size_t i = 0; i < length; ++i) {
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const double decibels = WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels);
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// scale down the value to the range of [0, UCHAR_MAX]
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const double scaled = std::max(0.0, std::min(double(UCHAR_MAX),
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UCHAR_MAX * (decibels - mMinDecibels) * rangeScaleFactor));
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buffer[i] = static_cast<unsigned char>(scaled);
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}
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}
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void
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AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray)
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{
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aArray.ComputeLengthAndData();
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float* buffer = aArray.Data();
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size_t length = std::min(size_t(aArray.Length()), mBuffer.Length());
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for (size_t i = 0; i < length; ++i) {
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buffer[i] = mBuffer[(i + mWriteIndex) % mBuffer.Length()];;
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}
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}
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void
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AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray)
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{
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aArray.ComputeLengthAndData();
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unsigned char* buffer = aArray.Data();
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size_t length = std::min(size_t(aArray.Length()), mBuffer.Length());
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for (size_t i = 0; i < length; ++i) {
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const float value = mBuffer[(i + mWriteIndex) % mBuffer.Length()];
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// scale the value to the range of [0, UCHAR_MAX]
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const float scaled = std::max(0.0f, std::min(float(UCHAR_MAX),
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128.0f * (value + 1.0f)));
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buffer[i] = static_cast<unsigned char>(scaled);
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}
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}
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bool
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AnalyserNode::FFTAnalysis()
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{
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float* inputBuffer;
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bool allocated = false;
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if (mWriteIndex == 0) {
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inputBuffer = mBuffer.Elements();
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} else {
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inputBuffer = static_cast<float*>(moz_malloc(FftSize() * sizeof(float)));
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if (!inputBuffer) {
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return false;
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}
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memcpy(inputBuffer, mBuffer.Elements() + mWriteIndex, sizeof(float) * (FftSize() - mWriteIndex));
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memcpy(inputBuffer + FftSize() - mWriteIndex, mBuffer.Elements(), sizeof(float) * mWriteIndex);
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allocated = true;
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}
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ApplyBlackmanWindow(inputBuffer, FftSize());
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mAnalysisBlock.PerformFFT(inputBuffer);
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// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
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const double magnitudeScale = 1.0 / FftSize();
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for (uint32_t i = 0; i < mOutputBuffer.Length(); ++i) {
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double scalarMagnitude = NS_hypot(mAnalysisBlock.RealData(i),
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mAnalysisBlock.ImagData(i)) *
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magnitudeScale;
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mOutputBuffer[i] = mSmoothingTimeConstant * mOutputBuffer[i] +
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(1.0 - mSmoothingTimeConstant) * scalarMagnitude;
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}
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if (allocated) {
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moz_free(inputBuffer);
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}
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return true;
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}
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void
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AnalyserNode::ApplyBlackmanWindow(float* aBuffer, uint32_t aSize)
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{
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double alpha = 0.16;
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double a0 = 0.5 * (1.0 - alpha);
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double a1 = 0.5;
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double a2 = 0.5 * alpha;
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for (uint32_t i = 0; i < aSize; ++i) {
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double x = double(i) / aSize;
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double window = a0 - a1 * cos(2 * M_PI * x) + a2 * cos(4 * M_PI * x);
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aBuffer[i] *= window;
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}
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}
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bool
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AnalyserNode::AllocateBuffer()
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{
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bool result = true;
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if (mBuffer.Length() != FftSize()) {
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result = mBuffer.SetLength(FftSize());
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if (result) {
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memset(mBuffer.Elements(), 0, sizeof(float) * FftSize());
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mWriteIndex = 0;
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result = mOutputBuffer.SetLength(FrequencyBinCount());
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if (result) {
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memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount());
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}
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}
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}
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return result;
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}
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void
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AnalyserNode::AppendChunk(const AudioChunk& aChunk)
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{
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const uint32_t bufferSize = mBuffer.Length();
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const uint32_t channelCount = aChunk.mChannelData.Length();
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uint32_t chunkDuration = aChunk.mDuration;
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MOZ_ASSERT((bufferSize & (bufferSize - 1)) == 0); // Must be a power of two!
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MOZ_ASSERT(channelCount > 0);
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MOZ_ASSERT(chunkDuration == WEBAUDIO_BLOCK_SIZE);
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if (chunkDuration > bufferSize) {
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// Copy a maximum bufferSize samples.
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chunkDuration = bufferSize;
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}
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PodCopy(mBuffer.Elements() + mWriteIndex, static_cast<const float*>(aChunk.mChannelData[0]), chunkDuration);
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for (uint32_t i = 1; i < channelCount; ++i) {
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AudioBlockAddChannelWithScale(static_cast<const float*>(aChunk.mChannelData[i]), 1.0f,
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mBuffer.Elements() + mWriteIndex);
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}
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if (channelCount > 1) {
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AudioBlockInPlaceScale(mBuffer.Elements() + mWriteIndex,
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1.0f / aChunk.mChannelData.Length());
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}
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mWriteIndex += chunkDuration;
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MOZ_ASSERT(mWriteIndex <= bufferSize);
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if (mWriteIndex >= bufferSize) {
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mWriteIndex = 0;
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}
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}
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}
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}
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