gecko-dev/content/media/webaudio/DelayNode.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 "DelayNode.h"
#include "mozilla/dom/DelayNodeBinding.h"
#include "AudioNodeEngine.h"
#include "AudioNodeStream.h"
#include "AudioDestinationNode.h"
#include "WebAudioUtils.h"
#include "DelayProcessor.h"
#include "PlayingRefChangeHandler.h"
namespace mozilla {
namespace dom {
NS_IMPL_CYCLE_COLLECTION_INHERITED_1(DelayNode, AudioNode,
mDelay)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(DelayNode)
NS_INTERFACE_MAP_END_INHERITING(AudioNode)
NS_IMPL_ADDREF_INHERITED(DelayNode, AudioNode)
NS_IMPL_RELEASE_INHERITED(DelayNode, AudioNode)
class DelayNodeEngine : public AudioNodeEngine
{
typedef PlayingRefChangeHandler<DelayNode> PlayingRefChanged;
public:
DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
int aMaxDelayFrames)
: AudioNodeEngine(aNode)
, mSource(nullptr)
, mDestination(static_cast<AudioNodeStream*> (aDestination->Stream()))
// Keep the default value in sync with the default value in DelayNode::DelayNode.
, mDelay(0.f)
// Use a smoothing range of 20ms
, mProcessor(aMaxDelayFrames,
WebAudioUtils::ComputeSmoothingRate(0.02,
mDestination->SampleRate()))
, mLeftOverData(INT32_MIN)
{
}
void SetSourceStream(AudioNodeStream* aSource)
{
mSource = aSource;
}
enum Parameters {
DELAY,
};
void SetTimelineParameter(uint32_t aIndex,
const AudioParamTimeline& aValue,
TrackRate aSampleRate) MOZ_OVERRIDE
{
switch (aIndex) {
case DELAY:
MOZ_ASSERT(mSource && mDestination);
mDelay = aValue;
WebAudioUtils::ConvertAudioParamToTicks(mDelay, mSource, mDestination);
break;
default:
NS_ERROR("Bad DelayNodeEngine TimelineParameter");
}
}
virtual void ProduceAudioBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished)
{
MOZ_ASSERT(mSource == aStream, "Invalid source stream");
MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate());
const uint32_t numChannels = aInput.IsNull() ?
mProcessor.BufferChannelCount() :
aInput.mChannelData.Length();
bool playedBackAllLeftOvers = false;
if (mProcessor.BufferChannelCount() &&
mLeftOverData == INT32_MIN &&
aStream->AllInputsFinished()) {
mLeftOverData = mProcessor.CurrentDelayFrames() - WEBAUDIO_BLOCK_SIZE;
if (mLeftOverData > 0) {
nsRefPtr<PlayingRefChanged> refchanged =
new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF);
NS_DispatchToMainThread(refchanged);
}
} else if (mLeftOverData != INT32_MIN) {
mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
if (mLeftOverData <= 0) {
// Continue spamming the main thread with messages until we are destroyed.
// This isn't great, but it ensures a message will get through even if
// some are ignored by DelayNode::AcceptPlayingRefRelease
mLeftOverData = 0;
playedBackAllLeftOvers = true;
nsRefPtr<PlayingRefChanged> refchanged =
new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE);
NS_DispatchToMainThread(refchanged);
}
}
AllocateAudioBlock(numChannels, aOutput);
AudioChunk input = aInput;
if (!aInput.IsNull() && aInput.mVolume != 1.0f) {
// Pre-multiply the input's volume
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);
}
}
const float* const* inputChannels = input.IsNull() ? nullptr :
reinterpret_cast<const float* const*>(input.mChannelData.Elements());
float* const* outputChannels = reinterpret_cast<float* const*>
(const_cast<void* const*>(aOutput->mChannelData.Elements()));
double sampleRate = aStream->SampleRate();
if (mDelay.HasSimpleValue()) {
double delayFrames = mDelay.GetValue() * sampleRate;
mProcessor.Process(delayFrames, inputChannels, outputChannels,
numChannels, WEBAUDIO_BLOCK_SIZE);
} else {
// Compute the delay values for the duration of the input AudioChunk
double computedDelay[WEBAUDIO_BLOCK_SIZE];
TrackTicks tick = aStream->GetCurrentPosition();
for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
computedDelay[counter] =
mDelay.GetValueAtTime(tick, counter) * sampleRate;
}
mProcessor.Process(computedDelay, inputChannels, outputChannels,
numChannels, WEBAUDIO_BLOCK_SIZE);
}
if (playedBackAllLeftOvers) {
// Delete our buffered data once we no longer need it
mProcessor.Reset();
}
}
AudioNodeStream* mSource;
AudioNodeStream* mDestination;
AudioParamTimeline mDelay;
DelayProcessor mProcessor;
// How much data we have in our buffer which needs to be flushed out when our inputs
// finish.
int32_t mLeftOverData;
};
DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
: AudioNode(aContext,
2,
ChannelCountMode::Max,
ChannelInterpretation::Speakers)
, mMediaStreamGraphUpdateIndexAtLastInputConnection(0)
, mDelay(new AudioParam(MOZ_THIS_IN_INITIALIZER_LIST(),
SendDelayToStream, 0.0f))
{
DelayNodeEngine* engine =
new DelayNodeEngine(this, aContext->Destination(),
ceil(aContext->SampleRate() * aMaxDelay));
mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM);
engine->SetSourceStream(static_cast<AudioNodeStream*> (mStream.get()));
}
JSObject*
DelayNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aScope)
{
return DelayNodeBinding::Wrap(aCx, aScope, this);
}
void
DelayNode::SendDelayToStream(AudioNode* aNode)
{
DelayNode* This = static_cast<DelayNode*>(aNode);
SendTimelineParameterToStream(This, DelayNodeEngine::DELAY, *This->mDelay);
}
}
}