зеркало из https://github.com/mozilla/gecko-dev.git
384 строки
13 KiB
C++
384 строки
13 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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#ifndef MOZILLA_AUDIONODEENGINE_H_
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#define MOZILLA_AUDIONODEENGINE_H_
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#include "AudioSegment.h"
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#include "mozilla/dom/AudioNode.h"
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#include "mozilla/MemoryReporting.h"
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#include "mozilla/Mutex.h"
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namespace mozilla {
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namespace dom {
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struct ThreeDPoint;
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class AudioParamTimeline;
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class DelayNodeEngine;
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struct AudioTimelineEvent;
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} // namespace dom
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class AbstractThread;
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class AudioBlock;
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class AudioNodeStream;
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/**
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* This class holds onto a set of immutable channel buffers. The storage
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* for the buffers must be malloced, but the buffer pointers and the malloc
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* pointers can be different (e.g. if the buffers are contained inside
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* some malloced object).
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*/
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class ThreadSharedFloatArrayBufferList final : public ThreadSharedObject {
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public:
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/**
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* Construct with null channel data pointers.
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*/
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explicit ThreadSharedFloatArrayBufferList(uint32_t aCount) {
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mContents.SetLength(aCount);
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}
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/**
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* Create with buffers suitable for transfer to
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* JS::NewArrayBufferWithContents(). The buffer contents are uninitialized
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* and so should be set using GetDataForWrite().
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*/
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static already_AddRefed<ThreadSharedFloatArrayBufferList> Create(
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uint32_t aChannelCount, size_t aLength, const mozilla::fallible_t&);
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ThreadSharedFloatArrayBufferList* AsThreadSharedFloatArrayBufferList()
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override {
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return this;
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};
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struct Storage final {
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Storage() : mDataToFree(nullptr), mFree(nullptr), mSampleData(nullptr) {}
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~Storage() {
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if (mFree) {
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mFree(mDataToFree);
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} else {
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MOZ_ASSERT(!mDataToFree);
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}
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}
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size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
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// NB: mSampleData might not be owned, if it is it just points to
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// mDataToFree.
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return aMallocSizeOf(mDataToFree);
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}
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void* mDataToFree;
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void (*mFree)(void*);
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float* mSampleData;
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};
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/**
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* This can be called on any thread.
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*/
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uint32_t GetChannels() const { return mContents.Length(); }
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/**
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* This can be called on any thread.
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*/
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const float* GetData(uint32_t aIndex) const {
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return mContents[aIndex].mSampleData;
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}
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/**
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* This can be called on any thread, but only when the calling thread is the
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* only owner.
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*/
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float* GetDataForWrite(uint32_t aIndex) {
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MOZ_ASSERT(!IsShared());
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return mContents[aIndex].mSampleData;
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}
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/**
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* Call this only during initialization, before the object is handed to
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* any other thread.
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*/
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void SetData(uint32_t aIndex, void* aDataToFree, void (*aFreeFunc)(void*),
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float* aData) {
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Storage* s = &mContents[aIndex];
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if (s->mFree) {
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s->mFree(s->mDataToFree);
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} else {
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MOZ_ASSERT(!s->mDataToFree);
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}
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s->mDataToFree = aDataToFree;
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s->mFree = aFreeFunc;
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s->mSampleData = aData;
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}
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/**
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* Put this object into an error state where there are no channels.
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*/
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void Clear() { mContents.Clear(); }
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size_t SizeOfExcludingThis(
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mozilla::MallocSizeOf aMallocSizeOf) const override {
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size_t amount = ThreadSharedObject::SizeOfExcludingThis(aMallocSizeOf);
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amount += mContents.ShallowSizeOfExcludingThis(aMallocSizeOf);
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for (size_t i = 0; i < mContents.Length(); i++) {
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amount += mContents[i].SizeOfExcludingThis(aMallocSizeOf);
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}
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return amount;
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}
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size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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private:
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AutoTArray<Storage, 2> mContents;
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};
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/**
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* aChunk must have been allocated by AllocateAudioBlock.
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*/
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void WriteZeroesToAudioBlock(AudioBlock* aChunk, uint32_t aStart,
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uint32_t aLength);
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/**
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* Copy with scale. aScale == 1.0f should be optimized.
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*/
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void AudioBufferCopyWithScale(const float* aInput, float aScale, float* aOutput,
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uint32_t aSize);
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/**
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* Pointwise multiply-add operation. aScale == 1.0f should be optimized.
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*/
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void AudioBufferAddWithScale(const float* aInput, float aScale, float* aOutput,
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uint32_t aSize);
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/**
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* Pointwise multiply-add operation. aScale == 1.0f should be optimized.
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*/
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void AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
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float aScale,
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float aOutput[WEBAUDIO_BLOCK_SIZE]);
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/**
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* Pointwise copy-scaled operation. aScale == 1.0f should be optimized.
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*
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* Buffer size is implicitly assumed to be WEBAUDIO_BLOCK_SIZE.
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*/
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void AudioBlockCopyChannelWithScale(const float* aInput, float aScale,
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float* aOutput);
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/**
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* Vector copy-scaled operation.
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*/
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void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
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const float aScale[WEBAUDIO_BLOCK_SIZE],
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float aOutput[WEBAUDIO_BLOCK_SIZE]);
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/**
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* Vector complex multiplication on arbitrary sized buffers.
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*/
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void BufferComplexMultiply(const float* aInput, const float* aScale,
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float* aOutput, uint32_t aSize);
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/**
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* Vector maximum element magnitude ( max(abs(aInput)) ).
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*/
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float AudioBufferPeakValue(const float* aInput, uint32_t aSize);
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/**
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* In place gain. aScale == 1.0f should be optimized.
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*/
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void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE], float aScale);
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/**
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* In place gain. aScale == 1.0f should be optimized.
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*/
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void AudioBufferInPlaceScale(float* aBlock, float aScale, uint32_t aSize);
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/**
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* a-rate in place gain.
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*/
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void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
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float aScale[WEBAUDIO_BLOCK_SIZE]);
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/**
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* a-rate in place gain.
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*/
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void AudioBufferInPlaceScale(float* aBlock, float* aScale, uint32_t aSize);
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/**
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* Upmix a mono input to a stereo output, scaling the two output channels by two
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* different gain value.
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* This algorithm is specified in the WebAudio spec.
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*/
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void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
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float aGainL, float aGainR,
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float aOutputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputR[WEBAUDIO_BLOCK_SIZE]);
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void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
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float aGainL[WEBAUDIO_BLOCK_SIZE],
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float aGainR[WEBAUDIO_BLOCK_SIZE],
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float aOutputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputR[WEBAUDIO_BLOCK_SIZE]);
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/**
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* Pan a stereo source according to right and left gain, and the position
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* (whether the listener is on the left of the source or not).
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* This algorithm is specified in the WebAudio spec.
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*/
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void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
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const float aInputR[WEBAUDIO_BLOCK_SIZE],
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float aGainL, float aGainR, bool aIsOnTheLeft,
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float aOutputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputR[WEBAUDIO_BLOCK_SIZE]);
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void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
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const float aInputR[WEBAUDIO_BLOCK_SIZE],
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float aGainL[WEBAUDIO_BLOCK_SIZE],
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float aGainR[WEBAUDIO_BLOCK_SIZE],
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bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
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float aOutputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputR[WEBAUDIO_BLOCK_SIZE]);
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/**
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* Return the sum of squares of all of the samples in the input.
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*/
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float AudioBufferSumOfSquares(const float* aInput, uint32_t aLength);
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/**
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* All methods of this class and its subclasses are called on the
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* MediaStreamGraph thread.
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*/
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class AudioNodeEngine {
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public:
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// This should be compatible with AudioNodeStream::OutputChunks.
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typedef AutoTArray<AudioBlock, 1> OutputChunks;
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explicit AudioNodeEngine(dom::AudioNode* aNode);
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virtual ~AudioNodeEngine() {
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MOZ_ASSERT(!mNode, "The node reference must be already cleared");
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MOZ_COUNT_DTOR(AudioNodeEngine);
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}
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virtual dom::DelayNodeEngine* AsDelayNodeEngine() { return nullptr; }
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virtual void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) {
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NS_ERROR("Invalid SetStreamTimeParameter index");
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}
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virtual void SetDoubleParameter(uint32_t aIndex, double aParam) {
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NS_ERROR("Invalid SetDoubleParameter index");
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}
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virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam) {
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NS_ERROR("Invalid SetInt32Parameter index");
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}
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virtual void RecvTimelineEvent(uint32_t aIndex,
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dom::AudioTimelineEvent& aValue) {
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NS_ERROR("Invalid RecvTimelineEvent index");
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}
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virtual void SetThreeDPointParameter(uint32_t aIndex,
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const dom::ThreeDPoint& aValue) {
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NS_ERROR("Invalid SetThreeDPointParameter index");
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}
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virtual void SetBuffer(AudioChunk&& aBuffer) {
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NS_ERROR("SetBuffer called on engine that doesn't support it");
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}
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// This consumes the contents of aData. aData will be emptied after this
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// returns.
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virtual void SetRawArrayData(nsTArray<float>& aData) {
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NS_ERROR("SetRawArrayData called on an engine that doesn't support it");
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}
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/**
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* Produce the next block of audio samples, given input samples aInput
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* (the mixed data for input 0).
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* aInput is guaranteed to have float sample format (if it has samples at all)
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* and to have been resampled to the sampling rate for the stream, and to have
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* exactly WEBAUDIO_BLOCK_SIZE samples.
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* *aFinished is set to false by the caller. The callee must not set this to
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* true unless silent output is produced. If set to true, we'll finish the
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* stream, consider this input inactive on any downstream nodes, and not
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* call this again.
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*/
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virtual void ProcessBlock(AudioNodeStream* aStream, GraphTime aFrom,
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const AudioBlock& aInput, AudioBlock* aOutput,
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bool* aFinished);
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/**
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* Produce the next block of audio samples, before input is provided.
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* ProcessBlock() will be called later, and it then should not change
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* aOutput. This is used only for DelayNodeEngine in a feedback loop.
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*/
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virtual void ProduceBlockBeforeInput(AudioNodeStream* aStream,
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GraphTime aFrom, AudioBlock* aOutput) {
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MOZ_ASSERT_UNREACHABLE("ProduceBlockBeforeInput called on wrong engine");
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}
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/**
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* Produce the next block of audio samples, given input samples in the aInput
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* array. There is one input sample per active port in aInput, in order.
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* This is the multi-input/output version of ProcessBlock. Only one kind
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* of ProcessBlock is called on each node, depending on whether the
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* number of inputs and outputs are both 1 or not.
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*
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* aInput is always guaranteed to not contain more input AudioChunks than the
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* maximum number of inputs for the node. It is the responsibility of the
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* overrides of this function to make sure they will only add a maximum number
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* of AudioChunks to aOutput as advertized by the AudioNode implementation.
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* An engine may choose to produce fewer inputs than advertizes by the
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* corresponding AudioNode, in which case it will be interpreted as a channel
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* of silence.
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*/
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virtual void ProcessBlocksOnPorts(AudioNodeStream* aStream,
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const OutputChunks& aInput,
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OutputChunks& aOutput, bool* aFinished);
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// IsActive() returns true if the engine needs to continue processing an
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// unfinished stream even when it has silent or no input connections. This
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// includes tail-times and when sources have been scheduled to start. If
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// returning false, then the stream can be suspended.
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virtual bool IsActive() const { return false; }
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bool HasNode() const {
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MOZ_ASSERT(NS_IsMainThread());
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return !!mNode;
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}
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dom::AudioNode* NodeMainThread() const {
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MOZ_ASSERT(NS_IsMainThread());
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return mNode;
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}
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void ClearNode() {
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MOZ_ASSERT(NS_IsMainThread());
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MOZ_ASSERT(mNode != nullptr);
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mNode = nullptr;
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}
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uint16_t InputCount() const { return mInputCount; }
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uint16_t OutputCount() const { return mOutputCount; }
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virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
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// NB: |mNode| is tracked separately so it is excluded here.
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return 0;
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}
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virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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void SizeOfIncludingThis(MallocSizeOf aMallocSizeOf,
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AudioNodeSizes& aUsage) const {
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aUsage.mEngine = SizeOfIncludingThis(aMallocSizeOf);
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aUsage.mNodeType = mNodeType;
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}
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private:
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// This is cleared from AudioNode::DestroyMediaStream()
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dom::AudioNode* MOZ_NON_OWNING_REF mNode; // main thread only
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const char* const mNodeType;
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const uint16_t mInputCount;
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const uint16_t mOutputCount;
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protected:
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const RefPtr<AbstractThread> mAbstractMainThread;
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};
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} // namespace mozilla
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#endif /* MOZILLA_AUDIONODEENGINE_H_ */
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