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Diffstat (limited to 'dom/media/webaudio/WaveShaperNode.cpp')
| -rw-r--r-- | dom/media/webaudio/WaveShaperNode.cpp | 390 |
1 files changed, 390 insertions, 0 deletions
diff --git a/dom/media/webaudio/WaveShaperNode.cpp b/dom/media/webaudio/WaveShaperNode.cpp new file mode 100644 index 0000000000..ced9a81b20 --- /dev/null +++ b/dom/media/webaudio/WaveShaperNode.cpp @@ -0,0 +1,390 @@ +/* -*- 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 "WaveShaperNode.h" +#include "mozilla/dom/WaveShaperNodeBinding.h" +#include "AlignmentUtils.h" +#include "AudioNode.h" +#include "AudioNodeEngine.h" +#include "AudioNodeTrack.h" +#include "mozilla/PodOperations.h" +#include "Tracing.h" + +namespace mozilla::dom { + +NS_IMPL_CYCLE_COLLECTION_CLASS(WaveShaperNode) + +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(WaveShaperNode, AudioNode) + NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_UNLINK_END + +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(WaveShaperNode, AudioNode) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END + +NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(WaveShaperNode) + NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_TRACE_END + +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(WaveShaperNode) +NS_INTERFACE_MAP_END_INHERITING(AudioNode) + +NS_IMPL_ADDREF_INHERITED(WaveShaperNode, AudioNode) +NS_IMPL_RELEASE_INHERITED(WaveShaperNode, AudioNode) + +static uint32_t ValueOf(OverSampleType aType) { + switch (aType) { + case OverSampleType::None: + return 1; + case OverSampleType::_2x: + return 2; + case OverSampleType::_4x: + return 4; + default: + MOZ_ASSERT_UNREACHABLE("We should never reach here"); + return 1; + } +} + +class Resampler final { + public: + Resampler() + : mType(OverSampleType::None), + mUpSampler(nullptr), + mDownSampler(nullptr), + mChannels(0), + mSampleRate(0) {} + + ~Resampler() { Destroy(); } + + void Reset(uint32_t aChannels, TrackRate aSampleRate, OverSampleType aType) { + if (aChannels == mChannels && aSampleRate == mSampleRate && + aType == mType) { + return; + } + + mChannels = aChannels; + mSampleRate = aSampleRate; + mType = aType; + + Destroy(); + + if (aType == OverSampleType::None) { + mBuffer.Clear(); + return; + } + + mUpSampler = speex_resampler_init(aChannels, aSampleRate, + aSampleRate * ValueOf(aType), + SPEEX_RESAMPLER_QUALITY_MIN, nullptr); + mDownSampler = + speex_resampler_init(aChannels, aSampleRate * ValueOf(aType), + aSampleRate, SPEEX_RESAMPLER_QUALITY_MIN, nullptr); + mBuffer.SetLength(WEBAUDIO_BLOCK_SIZE * ValueOf(aType)); + } + + float* UpSample(uint32_t aChannel, const float* aInputData, + uint32_t aBlocks) { + uint32_t inSamples = WEBAUDIO_BLOCK_SIZE; + uint32_t outSamples = WEBAUDIO_BLOCK_SIZE * aBlocks; + float* outputData = mBuffer.Elements(); + + MOZ_ASSERT(mBuffer.Length() == outSamples); + + WebAudioUtils::SpeexResamplerProcess(mUpSampler, aChannel, aInputData, + &inSamples, outputData, &outSamples); + + MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE && + outSamples == WEBAUDIO_BLOCK_SIZE * aBlocks); + + return outputData; + } + + void DownSample(uint32_t aChannel, float* aOutputData, uint32_t aBlocks) { + uint32_t inSamples = WEBAUDIO_BLOCK_SIZE * aBlocks; + uint32_t outSamples = WEBAUDIO_BLOCK_SIZE; + const float* inputData = mBuffer.Elements(); + + MOZ_ASSERT(mBuffer.Length() == inSamples); + + WebAudioUtils::SpeexResamplerProcess(mDownSampler, aChannel, inputData, + &inSamples, aOutputData, &outSamples); + + MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE * aBlocks && + outSamples == WEBAUDIO_BLOCK_SIZE); + } + + size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const { + size_t amount = 0; + // Future: properly measure speex memory + amount += aMallocSizeOf(mUpSampler); + amount += aMallocSizeOf(mDownSampler); + amount += mBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf); + return amount; + } + + private: + void Destroy() { + if (mUpSampler) { + speex_resampler_destroy(mUpSampler); + mUpSampler = nullptr; + } + if (mDownSampler) { + speex_resampler_destroy(mDownSampler); + mDownSampler = nullptr; + } + } + + private: + OverSampleType mType; + SpeexResamplerState* mUpSampler; + SpeexResamplerState* mDownSampler; + uint32_t mChannels; + TrackRate mSampleRate; + nsTArray<float> mBuffer; +}; + +class WaveShaperNodeEngine final : public AudioNodeEngine { + public: + explicit WaveShaperNodeEngine(AudioNode* aNode) + : AudioNodeEngine(aNode), mType(OverSampleType::None) {} + + enum Parameters { TYPE }; + + void SetRawArrayData(nsTArray<float>&& aCurve) override { + mCurve = std::move(aCurve); + } + + void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override { + switch (aIndex) { + case TYPE: + mType = static_cast<OverSampleType>(aValue); + break; + default: + NS_ERROR("Bad WaveShaperNode Int32Parameter"); + } + } + + template <uint32_t blocks> + void ProcessCurve(const float* aInputBuffer, float* aOutputBuffer) { + for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE * blocks; ++j) { + // Index into the curve array based on the amplitude of the + // incoming signal by using an amplitude range of [-1, 1] and + // performing a linear interpolation of the neighbor values. + float index = (mCurve.Length() - 1) * (aInputBuffer[j] + 1.0f) / 2.0f; + if (index < 0.0f) { + aOutputBuffer[j] = mCurve[0]; + } else { + int32_t indexLower = index; + if (static_cast<uint32_t>(indexLower) >= mCurve.Length() - 1) { + aOutputBuffer[j] = mCurve[mCurve.Length() - 1]; + } else { + uint32_t indexHigher = indexLower + 1; + float interpolationFactor = index - indexLower; + aOutputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] + + interpolationFactor * mCurve[indexHigher]; + } + } + } + } + + void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom, + const AudioBlock& aInput, AudioBlock* aOutput, + bool* aFinished) override { + TRACE("WaveShaperNodeEngine::ProcessBlock"); + + uint32_t channelCount = aInput.ChannelCount(); + if (!mCurve.Length()) { + // Optimize the case where we don't have a curve buffer + *aOutput = aInput; + return; + } + + // If the input is null, check to see if non-null output will be produced + bool nullInput = false; + if (channelCount == 0) { + float index = (mCurve.Length() - 1) * 0.5; + uint32_t indexLower = index; + uint32_t indexHigher = indexLower + 1; + float interpolationFactor = index - indexLower; + if ((1.0f - interpolationFactor) * mCurve[indexLower] + + interpolationFactor * mCurve[indexHigher] == + 0.0) { + *aOutput = aInput; + return; + } + nullInput = true; + channelCount = 1; + } + + aOutput->AllocateChannels(channelCount); + for (uint32_t i = 0; i < channelCount; ++i) { + const float* inputSamples; + float scaledInput[WEBAUDIO_BLOCK_SIZE + 4]; + float* alignedScaledInput = ALIGNED16(scaledInput); + ASSERT_ALIGNED16(alignedScaledInput); + if (!nullInput) { + if (aInput.mVolume != 1.0f) { + AudioBlockCopyChannelWithScale( + static_cast<const float*>(aInput.mChannelData[i]), aInput.mVolume, + alignedScaledInput); + inputSamples = alignedScaledInput; + } else { + inputSamples = static_cast<const float*>(aInput.mChannelData[i]); + } + } else { + PodZero(alignedScaledInput, WEBAUDIO_BLOCK_SIZE); + inputSamples = alignedScaledInput; + } + float* outputBuffer = aOutput->ChannelFloatsForWrite(i); + float* sampleBuffer; + + switch (mType) { + case OverSampleType::None: + mResampler.Reset(channelCount, aTrack->mSampleRate, + OverSampleType::None); + ProcessCurve<1>(inputSamples, outputBuffer); + break; + case OverSampleType::_2x: + mResampler.Reset(channelCount, aTrack->mSampleRate, + OverSampleType::_2x); + sampleBuffer = mResampler.UpSample(i, inputSamples, 2); + ProcessCurve<2>(sampleBuffer, sampleBuffer); + mResampler.DownSample(i, outputBuffer, 2); + break; + case OverSampleType::_4x: + mResampler.Reset(channelCount, aTrack->mSampleRate, + OverSampleType::_4x); + sampleBuffer = mResampler.UpSample(i, inputSamples, 4); + ProcessCurve<4>(sampleBuffer, sampleBuffer); + mResampler.DownSample(i, outputBuffer, 4); + break; + default: + MOZ_ASSERT_UNREACHABLE("We should never reach here"); + } + } + } + + size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override { + size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf); + amount += mCurve.ShallowSizeOfExcludingThis(aMallocSizeOf); + amount += mResampler.SizeOfExcludingThis(aMallocSizeOf); + return amount; + } + + size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override { + return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf); + } + + private: + nsTArray<float> mCurve; + OverSampleType mType; + Resampler mResampler; +}; + +WaveShaperNode::WaveShaperNode(AudioContext* aContext) + : AudioNode(aContext, 2, ChannelCountMode::Max, + ChannelInterpretation::Speakers), + mType(OverSampleType::None) { + WaveShaperNodeEngine* engine = new WaveShaperNodeEngine(this); + mTrack = AudioNodeTrack::Create( + aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph()); +} + +/* static */ +already_AddRefed<WaveShaperNode> WaveShaperNode::Create( + AudioContext& aAudioContext, const WaveShaperOptions& aOptions, + ErrorResult& aRv) { + RefPtr<WaveShaperNode> audioNode = new WaveShaperNode(&aAudioContext); + + audioNode->Initialize(aOptions, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + + if (aOptions.mCurve.WasPassed()) { + audioNode->SetCurveInternal(aOptions.mCurve.Value(), aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + } + + audioNode->SetOversample(aOptions.mOversample); + return audioNode.forget(); +} + +JSObject* WaveShaperNode::WrapObject(JSContext* aCx, + JS::Handle<JSObject*> aGivenProto) { + return WaveShaperNode_Binding::Wrap(aCx, this, aGivenProto); +} + +void WaveShaperNode::SetCurve(const Nullable<Float32Array>& aCurve, + ErrorResult& aRv) { + // Let's purge the cached value for the curve attribute. + WaveShaperNode_Binding::ClearCachedCurveValue(this); + + if (aCurve.IsNull()) { + CleanCurveInternal(); + return; + } + + const Float32Array& floats = aCurve.Value(); + floats.ComputeState(); + + nsTArray<float> curve; + uint32_t argLength = floats.Length(); + if (!curve.SetLength(argLength, fallible)) { + aRv.Throw(NS_ERROR_OUT_OF_MEMORY); + return; + } + + PodCopy(curve.Elements(), floats.Data(), argLength); + SetCurveInternal(curve, aRv); +} + +void WaveShaperNode::SetCurveInternal(const nsTArray<float>& aCurve, + ErrorResult& aRv) { + if (aCurve.Length() < 2) { + aRv.ThrowInvalidStateError("Must have at least two entries"); + return; + } + + mCurve = aCurve.Clone(); + SendCurveToTrack(); +} + +void WaveShaperNode::CleanCurveInternal() { + mCurve.Clear(); + SendCurveToTrack(); +} + +void WaveShaperNode::SendCurveToTrack() { + AudioNodeTrack* ns = mTrack; + MOZ_ASSERT(ns, "Why don't we have a track here?"); + + nsTArray<float> copyCurve(mCurve.Clone()); + ns->SetRawArrayData(std::move(copyCurve)); +} + +void WaveShaperNode::GetCurve(JSContext* aCx, + JS::MutableHandle<JSObject*> aRetval) { + // Let's return a null value if the list is empty. + if (mCurve.IsEmpty()) { + aRetval.set(nullptr); + return; + } + + MOZ_ASSERT(mCurve.Length() >= 2); + aRetval.set( + Float32Array::Create(aCx, this, mCurve.Length(), mCurve.Elements())); +} + +void WaveShaperNode::SetOversample(OverSampleType aType) { + mType = aType; + SendInt32ParameterToTrack(WaveShaperNodeEngine::TYPE, + static_cast<int32_t>(aType)); +} + +} // namespace mozilla::dom |