gecko-dev/content/media/webaudio/AnalyserNode.cpp

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Исходник Обычный вид История

/* -*- 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 "AudioNodeStream.h"
#include "mozilla/Mutex.h"
#include "mozilla/PodOperations.h"
namespace mozilla {
namespace dom {
NS_IMPL_ISUPPORTS_INHERITED0(AnalyserNode, AudioNode)
class AnalyserNodeEngine : public AudioNodeEngine
{
class TransferBuffer : public nsRunnable
{
public:
TransferBuffer(AudioNodeStream* aStream,
const AudioChunk& aChunk)
: mStream(aStream)
, mChunk(aChunk)
{
}
NS_IMETHOD Run()
{
nsRefPtr<AnalyserNode> node;
{
// No need to keep holding the lock for the whole duration of this
// function, since we're holding a strong reference to it, so if
// we can obtain the reference, we will hold the node alive in
// this function.
MutexAutoLock lock(mStream->Engine()->NodeMutex());
node = static_cast<AnalyserNode*>(mStream->Engine()->Node());
}
if (node) {
node->AppendChunk(mChunk);
}
return NS_OK;
}
private:
nsRefPtr<AudioNodeStream> mStream;
AudioChunk mChunk;
};
public:
explicit AnalyserNodeEngine(AnalyserNode* aNode)
: AudioNodeEngine(aNode)
{
MOZ_ASSERT(NS_IsMainThread());
}
virtual void ProcessBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished) MOZ_OVERRIDE
{
*aOutput = aInput;
MutexAutoLock lock(NodeMutex());
if (Node() &&
aInput.mChannelData.Length() > 0) {
nsRefPtr<TransferBuffer> transfer = new TransferBuffer(aStream, aInput);
NS_DispatchToMainThread(transfer);
}
}
virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
{
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
};
AnalyserNode::AnalyserNode(AudioContext* aContext)
: AudioNode(aContext,
1,
ChannelCountMode::Explicit,
ChannelInterpretation::Speakers)
, mAnalysisBlock(2048)
, mMinDecibels(-100.)
, mMaxDecibels(-30.)
, mSmoothingTimeConstant(.8)
, mWriteIndex(0)
{
mStream = aContext->Graph()->CreateAudioNodeStream(new AnalyserNodeEngine(this),
MediaStreamGraph::INTERNAL_STREAM);
AllocateBuffer();
}
size_t
AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf);
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
amount += mOutputBuffer.SizeOfExcludingThis(aMallocSizeOf);
return amount;
}
size_t
AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
{
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
JSObject*
AnalyserNode::WrapObject(JSContext* aCx)
{
return AnalyserNodeBinding::Wrap(aCx, this);
}
void
AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv)
{
// Disallow values that are not a power of 2 and outside the [32,2048] range
if (aValue < 32 ||
aValue > 2048 ||
(aValue & (aValue - 1)) != 0) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
if (FftSize() != aValue) {
mAnalysisBlock.SetFFTSize(aValue);
AllocateBuffer();
}
}
void
AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv)
{
if (aValue >= mMaxDecibels) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
mMinDecibels = aValue;
}
void
AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv)
{
if (aValue <= mMinDecibels) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
mMaxDecibels = aValue;
}
void
AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv)
{
if (aValue < 0 || aValue > 1) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
mSmoothingTimeConstant = aValue;
}
void
AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray)
{
if (!FFTAnalysis()) {
// Might fail to allocate memory
return;
}
aArray.ComputeLengthAndData();
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], mMinDecibels);
}
}
void
AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray)
{
if (!FFTAnalysis()) {
// Might fail to allocate memory
return;
}
const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels);
aArray.ComputeLengthAndData();
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<unsigned char>(scaled);
}
}
void
AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray)
{
aArray.ComputeLengthAndData();
float* buffer = aArray.Data();
size_t length = std::min(size_t(aArray.Length()), mBuffer.Length());
for (size_t i = 0; i < length; ++i) {
buffer[i] = mBuffer[(i + mWriteIndex) % mBuffer.Length()];;
}
}
void
AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray)
{
aArray.ComputeLengthAndData();
unsigned char* buffer = aArray.Data();
size_t length = std::min(size_t(aArray.Length()), mBuffer.Length());
for (size_t i = 0; i < length; ++i) {
const float value = mBuffer[(i + mWriteIndex) % mBuffer.Length()];
// 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<unsigned char>(scaled);
}
}
bool
AnalyserNode::FFTAnalysis()
{
float* inputBuffer;
bool allocated = false;
if (mWriteIndex == 0) {
inputBuffer = mBuffer.Elements();
} else {
inputBuffer = static_cast<float*>(moz_malloc(FftSize() * sizeof(float)));
if (!inputBuffer) {
return false;
}
memcpy(inputBuffer, mBuffer.Elements() + mWriteIndex, sizeof(float) * (FftSize() - mWriteIndex));
memcpy(inputBuffer + FftSize() - mWriteIndex, mBuffer.Elements(), sizeof(float) * mWriteIndex);
allocated = true;
}
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;
}
if (allocated) {
moz_free(inputBuffer);
}
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 (mBuffer.Length() != FftSize()) {
result = mBuffer.SetLength(FftSize());
if (result) {
memset(mBuffer.Elements(), 0, sizeof(float) * FftSize());
mWriteIndex = 0;
result = mOutputBuffer.SetLength(FrequencyBinCount());
if (result) {
memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount());
}
}
}
return result;
}
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;
}
}
}
}