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https://gitlab.winehq.org/wine/wine-gecko.git
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2bb6ffd9ed
--HG-- rename : content/media/webaudio/DelayProcessor.cpp => content/media/webaudio/DelayBuffer.cpp extra : rebase_source : ebdc7404c8d27e3a24098f21a7752df529bb44c9
312 lines
9.0 KiB
C++
312 lines
9.0 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 "WaveShaperNode.h"
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#include "mozilla/dom/WaveShaperNodeBinding.h"
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#include "AudioNode.h"
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#include "AudioNodeEngine.h"
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#include "AudioNodeStream.h"
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#include "mozilla/PodOperations.h"
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#include "speex/speex_resampler.h"
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namespace mozilla {
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namespace dom {
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NS_IMPL_CYCLE_COLLECTION_CLASS(WaveShaperNode)
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NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(WaveShaperNode, AudioNode)
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NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
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tmp->ClearCurve();
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NS_IMPL_CYCLE_COLLECTION_UNLINK_END
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NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(WaveShaperNode, AudioNode)
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NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS
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NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
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NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(WaveShaperNode)
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NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER
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NS_IMPL_CYCLE_COLLECTION_TRACE_JS_MEMBER_CALLBACK(mCurve)
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NS_IMPL_CYCLE_COLLECTION_TRACE_END
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NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(WaveShaperNode)
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NS_INTERFACE_MAP_END_INHERITING(AudioNode)
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NS_IMPL_ADDREF_INHERITED(WaveShaperNode, AudioNode)
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NS_IMPL_RELEASE_INHERITED(WaveShaperNode, AudioNode)
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static uint32_t ValueOf(OverSampleType aType)
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{
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switch (aType) {
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case OverSampleType::None: return 1;
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case OverSampleType::_2x: return 2;
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case OverSampleType::_4x: return 4;
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default:
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NS_NOTREACHED("We should never reach here");
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return 1;
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}
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}
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class Resampler
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{
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public:
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Resampler()
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: mType(OverSampleType::None)
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, mUpSampler(nullptr)
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, mDownSampler(nullptr)
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, mChannels(0)
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, mSampleRate(0)
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{
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}
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~Resampler()
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{
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Destroy();
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}
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void Reset(uint32_t aChannels, TrackRate aSampleRate, OverSampleType aType)
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{
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if (aChannels == mChannels &&
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aSampleRate == mSampleRate &&
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aType == mType) {
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return;
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}
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mChannels = aChannels;
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mSampleRate = aSampleRate;
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mType = aType;
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Destroy();
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if (aType == OverSampleType::None) {
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mBuffer.Clear();
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return;
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}
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mUpSampler = speex_resampler_init(aChannels,
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aSampleRate,
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aSampleRate * ValueOf(aType),
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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nullptr);
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mDownSampler = speex_resampler_init(aChannels,
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aSampleRate * ValueOf(aType),
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aSampleRate,
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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nullptr);
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mBuffer.SetLength(WEBAUDIO_BLOCK_SIZE*ValueOf(aType));
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}
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float* UpSample(uint32_t aChannel, const float* aInputData, uint32_t aBlocks)
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{
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uint32_t inSamples = WEBAUDIO_BLOCK_SIZE;
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uint32_t outSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
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float* outputData = mBuffer.Elements();
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MOZ_ASSERT(mBuffer.Length() == outSamples);
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WebAudioUtils::SpeexResamplerProcess(mUpSampler, aChannel,
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aInputData, &inSamples,
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outputData, &outSamples);
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MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE && outSamples == WEBAUDIO_BLOCK_SIZE*aBlocks);
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return outputData;
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}
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void DownSample(uint32_t aChannel, float* aOutputData, uint32_t aBlocks)
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{
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uint32_t inSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
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uint32_t outSamples = WEBAUDIO_BLOCK_SIZE;
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const float* inputData = mBuffer.Elements();
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MOZ_ASSERT(mBuffer.Length() == inSamples);
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WebAudioUtils::SpeexResamplerProcess(mDownSampler, aChannel,
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inputData, &inSamples,
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aOutputData, &outSamples);
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MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE*aBlocks && outSamples == WEBAUDIO_BLOCK_SIZE);
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}
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private:
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void Destroy()
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{
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if (mUpSampler) {
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speex_resampler_destroy(mUpSampler);
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mUpSampler = nullptr;
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}
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if (mDownSampler) {
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speex_resampler_destroy(mDownSampler);
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mDownSampler = nullptr;
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}
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}
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private:
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OverSampleType mType;
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SpeexResamplerState* mUpSampler;
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SpeexResamplerState* mDownSampler;
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uint32_t mChannels;
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TrackRate mSampleRate;
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nsTArray<float> mBuffer;
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};
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class WaveShaperNodeEngine : public AudioNodeEngine
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{
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public:
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explicit WaveShaperNodeEngine(AudioNode* aNode)
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: AudioNodeEngine(aNode)
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, mType(OverSampleType::None)
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{
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}
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enum Parameteres {
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TYPE
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};
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virtual void SetRawArrayData(nsTArray<float>& aCurve) MOZ_OVERRIDE
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{
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mCurve.SwapElements(aCurve);
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}
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virtual void SetInt32Parameter(uint32_t aIndex, int32_t aValue) MOZ_OVERRIDE
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{
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switch (aIndex) {
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case TYPE:
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mType = static_cast<OverSampleType>(aValue);
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break;
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default:
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NS_ERROR("Bad WaveShaperNode Int32Parameter");
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}
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}
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template <uint32_t blocks>
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void ProcessCurve(const float* aInputBuffer, float* aOutputBuffer)
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{
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for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE*blocks; ++j) {
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// Index into the curve array based on the amplitude of the
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// incoming signal by clamping the amplitude to [-1, 1] and
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// performing a linear interpolation of the neighbor values.
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float index = std::max(0.0f, std::min(float(mCurve.Length() - 1),
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mCurve.Length() * (aInputBuffer[j] + 1) / 2));
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uint32_t indexLower = uint32_t(index);
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uint32_t indexHigher = uint32_t(index + 1.0f);
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if (indexHigher == mCurve.Length()) {
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aOutputBuffer[j] = mCurve[indexLower];
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} else {
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float interpolationFactor = index - indexLower;
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aOutputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] +
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interpolationFactor * mCurve[indexHigher];
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}
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}
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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)
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{
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uint32_t channelCount = aInput.mChannelData.Length();
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if (!mCurve.Length() || !channelCount) {
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// Optimize the case where we don't have a curve buffer,
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// or the input is null.
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*aOutput = aInput;
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return;
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}
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AllocateAudioBlock(channelCount, aOutput);
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for (uint32_t i = 0; i < channelCount; ++i) {
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const float* inputBuffer = static_cast<const float*>(aInput.mChannelData[i]);
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float* outputBuffer = const_cast<float*> (static_cast<const float*>(aOutput->mChannelData[i]));
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float* sampleBuffer;
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switch (mType) {
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case OverSampleType::None:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::None);
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ProcessCurve<1>(inputBuffer, outputBuffer);
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break;
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case OverSampleType::_2x:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_2x);
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sampleBuffer = mResampler.UpSample(i, inputBuffer, 2);
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ProcessCurve<2>(sampleBuffer, sampleBuffer);
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mResampler.DownSample(i, outputBuffer, 2);
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break;
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case OverSampleType::_4x:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_4x);
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sampleBuffer = mResampler.UpSample(i, inputBuffer, 4);
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ProcessCurve<4>(sampleBuffer, sampleBuffer);
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mResampler.DownSample(i, outputBuffer, 4);
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break;
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default:
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NS_NOTREACHED("We should never reach here");
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}
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}
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}
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private:
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nsTArray<float> mCurve;
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OverSampleType mType;
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Resampler mResampler;
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};
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WaveShaperNode::WaveShaperNode(AudioContext* aContext)
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: AudioNode(aContext,
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2,
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ChannelCountMode::Max,
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ChannelInterpretation::Speakers)
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, mCurve(nullptr)
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, mType(OverSampleType::None)
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{
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mozilla::HoldJSObjects(this);
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WaveShaperNodeEngine* engine = new WaveShaperNodeEngine(this);
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mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM);
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}
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WaveShaperNode::~WaveShaperNode()
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{
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ClearCurve();
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}
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void
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WaveShaperNode::ClearCurve()
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{
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mCurve = nullptr;
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mozilla::DropJSObjects(this);
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}
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JSObject*
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WaveShaperNode::WrapObject(JSContext *aCx, JS::Handle<JSObject*> aScope)
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{
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return WaveShaperNodeBinding::Wrap(aCx, aScope, this);
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}
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void
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WaveShaperNode::SetCurve(const Nullable<Float32Array>& aCurve)
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{
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nsTArray<float> curve;
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if (!aCurve.IsNull()) {
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mCurve = aCurve.Value().Obj();
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curve.SetLength(aCurve.Value().Length());
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PodCopy(curve.Elements(), aCurve.Value().Data(), aCurve.Value().Length());
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} else {
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mCurve = nullptr;
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}
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AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
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MOZ_ASSERT(ns, "Why don't we have a stream here?");
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ns->SetRawArrayData(curve);
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}
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void
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WaveShaperNode::SetOversample(OverSampleType aType)
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{
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mType = aType;
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SendInt32ParameterToStream(WaveShaperNodeEngine::TYPE, static_cast<int32_t>(aType));
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}
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}
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}
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