зеркало из https://github.com/mozilla/gecko-dev.git
b=815643 Refactor DelayNodeEngine delay processing into a shareable class r=ehsan
--HG-- extra : rebase_source : e25ff3e490c2cbce5ed7cdf9419ccc2850ea16e5
This commit is contained in:
Karl Tomlinson
Родитель
705047473b
Коммит
bfa9fc4aa2
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@ -10,6 +10,7 @@
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#include "AudioNodeStream.h"
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#include "AudioDestinationNode.h"
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#include "WebAudioUtils.h"
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#include "DelayProcessor.h"
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namespace mozilla {
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namespace dom {
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@ -27,16 +28,18 @@ class DelayNodeEngine : public AudioNodeEngine
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{
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typedef PlayingRefChangeHandler<DelayNode> PlayingRefChanged;
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public:
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DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
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DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
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int aMaxDelayFrames)
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: AudioNodeEngine(aNode)
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, mSource(nullptr)
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, mDestination(static_cast<AudioNodeStream*> (aDestination->Stream()))
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// Keep the default value in sync with the default value in DelayNode::DelayNode.
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, mDelay(0.f)
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, mMaxDelay(0.)
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, mWriteIndex(0)
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// Use a smoothing range of 20ms
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, mProcessor(aMaxDelayFrames,
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WebAudioUtils::ComputeSmoothingRate(0.02,
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mDestination->SampleRate()))
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, mLeftOverData(INT32_MIN)
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, mCurrentDelayTime(0.)
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{
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}
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@ -47,7 +50,6 @@ public:
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enum Parameters {
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DELAY,
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MAX_DELAY
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};
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void SetTimelineParameter(uint32_t aIndex,
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const AudioParamTimeline& aValue,
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@ -63,37 +65,6 @@ public:
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NS_ERROR("Bad DelayNodeEngine TimelineParameter");
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}
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}
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void SetDoubleParameter(uint32_t aIndex, double aValue) MOZ_OVERRIDE
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{
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switch (aIndex) {
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case MAX_DELAY: mMaxDelay = aValue; break;
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default:
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NS_ERROR("Bad DelayNodeEngine DoubleParameter");
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}
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}
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bool EnsureBuffer(uint32_t aNumberOfChannels, TrackRate aSampleRate)
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{
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if (aNumberOfChannels == 0) {
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return false;
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}
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if (mBuffer.Length() == 0) {
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if (!mBuffer.SetLength(aNumberOfChannels)) {
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return false;
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}
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const int32_t numFrames = ceil(mMaxDelay * aSampleRate);
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for (uint32_t channel = 0; channel < aNumberOfChannels; ++channel) {
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if (!mBuffer[channel].SetLength(numFrames)) {
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return false;
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}
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memset(mBuffer[channel].Elements(), 0, numFrames * sizeof(float));
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}
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} else if (mBuffer.Length() != aNumberOfChannels) {
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// TODO: Handle changes in the channel count
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return false;
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}
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return true;
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}
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virtual void ProduceAudioBlock(AudioNodeStream* aStream,
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const AudioChunk& aInput,
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@ -101,17 +72,17 @@ public:
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bool* aFinished)
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{
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MOZ_ASSERT(mSource == aStream, "Invalid source stream");
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MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate());
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const bool firstTime = !!!mBuffer.Length();
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const uint32_t numChannels = aInput.IsNull() ?
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mBuffer.Length() :
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mProcessor.BufferChannelCount() :
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aInput.mChannelData.Length();
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bool playedBackAllLeftOvers = false;
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if (!mBuffer.IsEmpty() &&
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if (mProcessor.BufferChannelCount() &&
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mLeftOverData == INT32_MIN &&
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aStream->AllInputsFinished()) {
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mLeftOverData = static_cast<int32_t>(mCurrentDelayTime * aStream->SampleRate()) - WEBAUDIO_BLOCK_SIZE;
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mLeftOverData = mProcessor.CurrentDelayFrames() - WEBAUDIO_BLOCK_SIZE;
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if (mLeftOverData > 0) {
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nsRefPtr<PlayingRefChanged> refchanged =
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@ -133,113 +104,55 @@ public:
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}
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}
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if (!EnsureBuffer(numChannels, aStream->SampleRate())) {
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aOutput->SetNull(0);
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return;
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}
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AllocateAudioBlock(numChannels, aOutput);
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double delayTime = 0;
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double computedDelay[WEBAUDIO_BLOCK_SIZE];
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// Use a smoothing range of 20ms
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const double smoothingRate = WebAudioUtils::ComputeSmoothingRate(0.02, aStream->SampleRate());
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if (mDelay.HasSimpleValue()) {
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delayTime = std::max(0.0, std::min(mMaxDelay, double(mDelay.GetValue())));
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if (firstTime) {
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// Initialize this only the first time to make sure that mCurrentDelayTime
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// has a valid value when we try to change the delay time further below.
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mCurrentDelayTime = delayTime;
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AudioChunk input = aInput;
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if (!aInput.IsNull() && aInput.mVolume != 1.0f) {
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// Pre-multiply the input's volume
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AllocateAudioBlock(numChannels, &input);
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for (uint32_t i = 0; i < numChannels; ++i) {
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const float* src = static_cast<const float*>(aInput.mChannelData[i]);
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float* dest = static_cast<float*>(const_cast<void*>(input.mChannelData[i]));
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AudioBlockCopyChannelWithScale(src, aInput.mVolume, dest);
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}
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}
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const float* const* inputChannels = input.IsNull() ? nullptr :
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reinterpret_cast<const float* const*>(input.mChannelData.Elements());
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float* const* outputChannels = reinterpret_cast<float* const*>
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(const_cast<void* const*>(aOutput->mChannelData.Elements()));
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double sampleRate = aStream->SampleRate();
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if (mDelay.HasSimpleValue()) {
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double delayFrames = mDelay.GetValue() * sampleRate;
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mProcessor.Process(delayFrames, inputChannels, outputChannels,
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numChannels, WEBAUDIO_BLOCK_SIZE);
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} else {
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// Compute the delay values for the duration of the input AudioChunk
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double computedDelay[WEBAUDIO_BLOCK_SIZE];
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TrackTicks tick = aStream->GetCurrentPosition();
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for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
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computedDelay[counter] = std::max(0.0, std::min(mMaxDelay,
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double(mDelay.GetValueAtTime(tick, counter))));
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computedDelay[counter] =
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mDelay.GetValueAtTime(tick, counter) * sampleRate;
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}
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mProcessor.Process(computedDelay, inputChannels, outputChannels,
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numChannels, WEBAUDIO_BLOCK_SIZE);
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}
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for (uint32_t channel = 0; channel < numChannels; ++channel) {
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double currentDelayTime = mCurrentDelayTime;
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uint32_t writeIndex = mWriteIndex;
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float* buffer = mBuffer[channel].Elements();
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const uint32_t bufferLength = mBuffer[channel].Length();
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const float* input = static_cast<const float*>(aInput.mChannelData.SafeElementAt(channel));
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float* output = static_cast<float*>(const_cast<void*>(aOutput->mChannelData[channel]));
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for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
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if (mDelay.HasSimpleValue()) {
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// If the simple value has changed, smoothly approach it
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currentDelayTime += (delayTime - currentDelayTime) * smoothingRate;
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} else {
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currentDelayTime = computedDelay[i];
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}
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// Write the input sample to the correct location in our buffer
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if (input) {
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buffer[writeIndex] = input[i] * aInput.mVolume;
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}
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// Now, determine the correct read position. We adjust the read position to be
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// from currentDelayTime seconds in the past. We also interpolate the two input
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// frames in case the read position does not match an integer index.
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double readPosition = writeIndex + bufferLength -
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(currentDelayTime * aStream->SampleRate());
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if (readPosition >= bufferLength) {
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readPosition -= bufferLength;
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}
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MOZ_ASSERT(readPosition >= 0.0, "Why are we reading before the beginning of the buffer?");
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// Here is a the reason why readIndex1 and readIndex will never be out
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// of bounds. The maximum value for bufferLength is 180 * 48000 (see
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// AudioContext::CreateDelay). The maximum value for mCurrentDelay is
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// 180.0, so initially readPosition cannot be more than bufferLength +
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// a fraction less than 1. Then we take care of that case by
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// subtracting bufferLength from it if needed. So, if
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// |bufferLength-readPosition<1.0|, readIndex1 will end up being zero.
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// If |1.0<=bufferLength-readPosition<2.0|, readIndex1 will be
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// bufferLength-1 and readIndex2 will be 0.
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int readIndex1 = int(readPosition);
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int readIndex2 = (readIndex1 + 1) % bufferLength;
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double interpolationFactor = readPosition - readIndex1;
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output[i] = (1.0 - interpolationFactor) * buffer[readIndex1] +
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interpolationFactor * buffer[readIndex2];
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writeIndex = (writeIndex + 1) % bufferLength;
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}
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// Remember currentDelayTime and writeIndex for the next ProduceAudioBlock
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// call when processing the last channel.
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if (channel == numChannels - 1) {
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mCurrentDelayTime = currentDelayTime;
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mWriteIndex = writeIndex;
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}
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}
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if (playedBackAllLeftOvers) {
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// Delete our buffered data once we no longer need it
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mBuffer.Clear();
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mProcessor.Reset();
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}
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}
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AudioNodeStream* mSource;
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AudioNodeStream* mDestination;
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AudioParamTimeline mDelay;
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// Maximum delay time in seconds
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double mMaxDelay;
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// Circular buffer for capturing delayed samples.
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AutoFallibleTArray<FallibleTArray<float>, 2> mBuffer;
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// Write index for the buffer, to write the frames to the correct index of the buffer
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// given the current delay.
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uint32_t mWriteIndex;
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DelayProcessor mProcessor;
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// How much data we have in our buffer which needs to be flushed out when our inputs
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// finish.
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int32_t mLeftOverData;
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// Current delay time, in seconds
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double mCurrentDelayTime;
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};
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DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
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@ -251,11 +164,11 @@ DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
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, mDelay(new AudioParam(MOZ_THIS_IN_INITIALIZER_LIST(),
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SendDelayToStream, 0.0f))
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{
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DelayNodeEngine* engine = new DelayNodeEngine(this, aContext->Destination());
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DelayNodeEngine* engine =
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new DelayNodeEngine(this, aContext->Destination(),
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ceil(aContext->SampleRate() * aMaxDelay));
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mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM);
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engine->SetSourceStream(static_cast<AudioNodeStream*> (mStream.get()));
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AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
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ns->SetDoubleParameter(DelayNodeEngine::MAX_DELAY, aMaxDelay);
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}
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JSObject*
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@ -0,0 +1,126 @@
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/* -*- 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 "DelayProcessor.h"
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#include "mozilla/PodOperations.h"
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#include "AudioSegment.h"
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namespace mozilla {
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void
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DelayProcessor::Process(const double *aPerFrameDelays,
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const float* const* aInputChannels,
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float* const* aOutputChannels,
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int aChannelCount, int aFramesToProcess)
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{
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if (!EnsureBuffer(aChannelCount)) {
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for (int channel = 0; channel < aChannelCount; ++channel) {
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PodZero(aOutputChannels[channel], aFramesToProcess);
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}
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return;
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}
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for (int channel = 0; channel < aChannelCount; ++channel) {
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double currentDelayFrames = mCurrentDelay;
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int writeIndex = mWriteIndex;
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float* buffer = mBuffer[channel].Elements();
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const uint32_t bufferLength = mBuffer[channel].Length();
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const float* input = aInputChannels ? aInputChannels[channel] : nullptr;
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float* output = aOutputChannels[channel];
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for (int i = 0; i < aFramesToProcess; ++i) {
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currentDelayFrames = clamped(aPerFrameDelays[i],
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0.0, static_cast<double>(mMaxDelayFrames));
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// Write the input sample to the correct location in our buffer
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if (input) {
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buffer[writeIndex] = input[i];
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}
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// Now, determine the correct read position. We adjust the read position to be
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// from currentDelayFrames frames in the past. We also interpolate the two input
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// frames in case the read position does not match an integer index.
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double readPosition = writeIndex + bufferLength - currentDelayFrames;
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if (readPosition >= bufferLength) {
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readPosition -= bufferLength;
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}
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MOZ_ASSERT(readPosition >= 0.0, "Why are we reading before the beginning of the buffer?");
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// Here is a the reason why readIndex1 and readIndex will never be out
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// of bounds. The maximum value for bufferLength is 180 * 48000 (see
|
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// AudioContext::CreateDelay). The maximum value for mCurrentDelay is
|
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// 180.0, so initially readPosition cannot be more than bufferLength +
|
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// a fraction less than 1. Then we take care of that case by
|
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// subtracting bufferLength from it if needed. So, if
|
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// |bufferLength-readPosition<1.0|, readIndex1 will end up being zero.
|
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// If |1.0<=bufferLength-readPosition<2.0|, readIndex1 will be
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// bufferLength-1 and readIndex2 will be 0.
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int readIndex1 = int(readPosition);
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int readIndex2 = (readIndex1 + 1) % bufferLength;
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double interpolationFactor = readPosition - readIndex1;
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output[i] = (1.0 - interpolationFactor) * buffer[readIndex1] +
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interpolationFactor * buffer[readIndex2];
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writeIndex = (writeIndex + 1) % bufferLength;
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}
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// Remember currentDelayFrames and writeIndex for the next ProduceAudioBlock
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// call when processing the last channel.
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if (channel == aChannelCount - 1) {
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mCurrentDelay = currentDelayFrames;
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mWriteIndex = writeIndex;
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}
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}
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}
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void
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DelayProcessor::Process(double aDelayFrames, const float* const* aInputChannels,
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float* const* aOutputChannels, int aChannelCount,
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int aFramesToProcess)
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{
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const bool firstTime = !mBuffer.Length();
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double currentDelay = firstTime ? aDelayFrames : mCurrentDelay;
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nsAutoTArray<double, WEBAUDIO_BLOCK_SIZE> computedDelay;
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computedDelay.SetLength(aFramesToProcess);
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for (int i = 0; i < aFramesToProcess; ++i) {
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// If the value has changed, smoothly approach it
|
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currentDelay += (aDelayFrames - currentDelay) * mSmoothingRate;
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computedDelay[i] = currentDelay;
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}
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Process(computedDelay.Elements(), aInputChannels, aOutputChannels,
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aChannelCount, aFramesToProcess);
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}
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bool
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DelayProcessor::EnsureBuffer(uint32_t aNumberOfChannels)
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{
|
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if (aNumberOfChannels == 0) {
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return false;
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}
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if (mBuffer.Length() == 0) {
|
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if (!mBuffer.SetLength(aNumberOfChannels)) {
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return false;
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}
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const int numFrames = mMaxDelayFrames;
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for (uint32_t channel = 0; channel < aNumberOfChannels; ++channel) {
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if (!mBuffer[channel].SetLength(numFrames)) {
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return false;
|
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}
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PodZero(mBuffer[channel].Elements(), numFrames);
|
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}
|
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} else if (mBuffer.Length() != aNumberOfChannels) {
|
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// TODO: Handle changes in the channel count
|
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return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
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} // mozilla
|
||||
|
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@ -0,0 +1,60 @@
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/* -*- 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: */
|
||||
/* 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/. */
|
||||
|
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#ifndef DelayProcessor_h_
|
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#define DelayProcessor_h_
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#include "nsTArray.h"
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|
||||
namespace mozilla {
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|
||||
class DelayProcessor {
|
||||
public:
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// See WebAudioUtils::ComputeSmoothingRate() for frame to frame exponential
|
||||
// |smoothingRate| multiplier.
|
||||
DelayProcessor(int aMaxDelayFrames, double aSmoothingRate)
|
||||
: mSmoothingRate(aSmoothingRate)
|
||||
, mCurrentDelay(0.)
|
||||
, mMaxDelayFrames(aMaxDelayFrames)
|
||||
, mWriteIndex(0)
|
||||
{
|
||||
}
|
||||
|
||||
// Process with an array of delays, in frames, for each frame.
|
||||
void Process(const double *aPerFrameDelays,
|
||||
const float* const* aInputChannels,
|
||||
float* const* aOutputChannels,
|
||||
int aChannelCount, int aFramesToProcess);
|
||||
|
||||
// Process with a constant delay, which will be smoothed with the previous
|
||||
// delay.
|
||||
void Process(double aDelayFrames, const float* const* aInputChannels,
|
||||
float* const* aOutputChannels, int aChannelCount,
|
||||
int aFramesToProcess);
|
||||
|
||||
void Reset() { mBuffer.Clear(); };
|
||||
|
||||
double CurrentDelayFrames() const { return mCurrentDelay; }
|
||||
int BufferChannelCount() const { return mBuffer.Length(); }
|
||||
|
||||
private:
|
||||
bool EnsureBuffer(uint32_t aNumberOfChannels);
|
||||
|
||||
// Circular buffer for capturing delayed samples.
|
||||
AutoFallibleTArray<FallibleTArray<float>, 2> mBuffer;
|
||||
double mSmoothingRate;
|
||||
// Current delay, in fractional frames
|
||||
double mCurrentDelay;
|
||||
// Maximum delay, in frames
|
||||
int mMaxDelayFrames;
|
||||
// Write index for the buffer, to write the frames to the correct index of the buffer
|
||||
// given the current delay.
|
||||
int mWriteIndex;
|
||||
};
|
||||
|
||||
} // mozilla
|
||||
|
||||
#endif // DelayProcessor_h_
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||||
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@ -64,6 +64,7 @@ CPP_SOURCES += [
|
|||
'ChannelSplitterNode.cpp',
|
||||
'ConvolverNode.cpp',
|
||||
'DelayNode.cpp',
|
||||
'DelayProcessor.cpp',
|
||||
'DynamicsCompressorNode.cpp',
|
||||
'EnableWebAudioCheck.cpp',
|
||||
'FFTBlock.cpp',
|
||||
|
|
|
|||
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