gecko/content/media/webaudio/ConvolverNode.cpp

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/* -*- 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 "ConvolverNode.h"
#include "mozilla/dom/ConvolverNodeBinding.h"
#include "AudioNodeEngine.h"
#include "AudioNodeStream.h"
#include "blink/Reverb.h"
#include "PlayingRefChangeHandler.h"
namespace mozilla {
namespace dom {
NS_IMPL_CYCLE_COLLECTION_INHERITED_1(ConvolverNode, AudioNode, mBuffer)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(ConvolverNode)
NS_INTERFACE_MAP_END_INHERITING(AudioNode)
NS_IMPL_ADDREF_INHERITED(ConvolverNode, AudioNode)
NS_IMPL_RELEASE_INHERITED(ConvolverNode, AudioNode)
class ConvolverNodeEngine : public AudioNodeEngine
{
typedef PlayingRefChangeHandler PlayingRefChanged;
public:
ConvolverNodeEngine(AudioNode* aNode, bool aNormalize)
: AudioNodeEngine(aNode)
, mBufferLength(0)
, mLeftOverData(INT32_MIN)
, mSampleRate(0.0f)
, mUseBackgroundThreads(!aNode->Context()->IsOffline())
, mNormalize(aNormalize)
{
}
enum Parameters {
BUFFER_LENGTH,
SAMPLE_RATE,
NORMALIZE
};
virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam) MOZ_OVERRIDE
{
switch (aIndex) {
case BUFFER_LENGTH:
// BUFFER_LENGTH is the first parameter that we set when setting a new buffer,
// so we should be careful to invalidate the rest of our state here.
mBuffer = nullptr;
mSampleRate = 0.0f;
mBufferLength = aParam;
mLeftOverData = INT32_MIN;
break;
case SAMPLE_RATE:
mSampleRate = aParam;
break;
case NORMALIZE:
mNormalize = !!aParam;
break;
default:
NS_ERROR("Bad ConvolverNodeEngine Int32Parameter");
}
}
virtual void SetDoubleParameter(uint32_t aIndex, double aParam) MOZ_OVERRIDE
{
switch (aIndex) {
case SAMPLE_RATE:
mSampleRate = aParam;
AdjustReverb();
break;
default:
NS_ERROR("Bad ConvolverNodeEngine DoubleParameter");
}
}
virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
{
mBuffer = aBuffer;
AdjustReverb();
}
void AdjustReverb()
{
// Note about empirical tuning (this is copied from Blink)
// The maximum FFT size affects reverb performance and accuracy.
// If the reverb is single-threaded and processes entirely in the real-time audio thread,
// it's important not to make this too high. In this case 8192 is a good value.
// But, the Reverb object is multi-threaded, so we want this as high as possible without losing too much accuracy.
// Very large FFTs will have worse phase errors. Given these constraints 32768 is a good compromise.
const size_t MaxFFTSize = 32768;
if (!mBuffer || !mBufferLength || !mSampleRate) {
mReverb = nullptr;
mLeftOverData = INT32_MIN;
return;
}
mReverb = new WebCore::Reverb(mBuffer, mBufferLength,
WEBAUDIO_BLOCK_SIZE,
MaxFFTSize, 2, mUseBackgroundThreads,
mNormalize, mSampleRate);
}
virtual void ProduceAudioBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished)
{
if (!mReverb) {
*aOutput = aInput;
return;
}
AudioChunk input = aInput;
if (aInput.IsNull()) {
if (mLeftOverData > 0) {
mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
AllocateAudioBlock(1, &input);
WriteZeroesToAudioBlock(&input, 0, WEBAUDIO_BLOCK_SIZE);
} else {
if (mLeftOverData != INT32_MIN) {
mLeftOverData = INT32_MIN;
nsRefPtr<PlayingRefChanged> refchanged =
new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE);
aStream->Graph()->
DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
}
aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
} else {
if (aInput.mVolume != 1.0f) {
// Pre-multiply the input's volume
uint32_t numChannels = aInput.mChannelData.Length();
AllocateAudioBlock(numChannels, &input);
for (uint32_t i = 0; i < numChannels; ++i) {
const float* src = static_cast<const float*>(aInput.mChannelData[i]);
float* dest = static_cast<float*>(const_cast<void*>(input.mChannelData[i]));
AudioBlockCopyChannelWithScale(src, aInput.mVolume, dest);
}
}
if (mLeftOverData <= 0) {
nsRefPtr<PlayingRefChanged> refchanged =
new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF);
aStream->Graph()->
DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
}
mLeftOverData = mBufferLength;
MOZ_ASSERT(mLeftOverData > 0);
}
AllocateAudioBlock(2, aOutput);
mReverb->process(&input, aOutput, WEBAUDIO_BLOCK_SIZE);
}
private:
nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
nsAutoPtr<WebCore::Reverb> mReverb;
int32_t mBufferLength;
int32_t mLeftOverData;
float mSampleRate;
bool mUseBackgroundThreads;
bool mNormalize;
};
ConvolverNode::ConvolverNode(AudioContext* aContext)
: AudioNode(aContext,
2,
ChannelCountMode::Clamped_max,
ChannelInterpretation::Speakers)
, mNormalize(true)
{
ConvolverNodeEngine* engine = new ConvolverNodeEngine(this, mNormalize);
mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM);
}
JSObject*
ConvolverNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aScope)
{
return ConvolverNodeBinding::Wrap(aCx, aScope, this);
}
void
ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv)
{
if (aBuffer) {
switch (aBuffer->NumberOfChannels()) {
case 1:
case 2:
case 4:
// Supported number of channels
break;
default:
aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
return;
}
}
mBuffer = aBuffer;
// Send the buffer to the stream
AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
MOZ_ASSERT(ns, "Why don't we have a stream here?");
if (mBuffer) {
uint32_t length = mBuffer->Length();
nsRefPtr<ThreadSharedFloatArrayBufferList> data =
mBuffer->GetThreadSharedChannelsForRate(aCx);
if (data && length < WEBAUDIO_BLOCK_SIZE) {
// For very small impulse response buffers, we need to pad the
// buffer with 0 to make sure that the Reverb implementation
// has enough data to compute FFTs from.
length = WEBAUDIO_BLOCK_SIZE;
nsRefPtr<ThreadSharedFloatArrayBufferList> paddedBuffer =
new ThreadSharedFloatArrayBufferList(data->GetChannels());
float* channelData = (float*) malloc(sizeof(float) * length * data->GetChannels());
for (uint32_t i = 0; i < data->GetChannels(); ++i) {
PodCopy(channelData + length * i, data->GetData(i), mBuffer->Length());
PodZero(channelData + length * i + mBuffer->Length(), WEBAUDIO_BLOCK_SIZE - mBuffer->Length());
paddedBuffer->SetData(i, (i == 0) ? channelData : nullptr, channelData);
}
data = paddedBuffer;
}
SendInt32ParameterToStream(ConvolverNodeEngine::BUFFER_LENGTH, length);
SendDoubleParameterToStream(ConvolverNodeEngine::SAMPLE_RATE,
mBuffer->SampleRate());
ns->SetBuffer(data.forget());
} else {
ns->SetBuffer(nullptr);
}
}
void
ConvolverNode::SetNormalize(bool aNormalize)
{
mNormalize = aNormalize;
SendInt32ParameterToStream(ConvolverNodeEngine::NORMALIZE, aNormalize);
}
}
}