gecko/content/media/AudioNodeEngine.h
Paul Adenot b7bfb58bff Bug 849918 - Initial support for PannerNode's 3D positional audio (equalpower panning model). r=ehsan,roc
--HG--
extra : rebase_source : d729da50f97b94ea0434a93ad481ac35954bfa9a
2013-03-21 17:45:53 +01:00

201 lines
6.3 KiB
C++

/* -*- 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/. */
#ifndef MOZILLA_AUDIONODEENGINE_H_
#define MOZILLA_AUDIONODEENGINE_H_
#include "AudioSegment.h"
#include "mozilla/dom/AudioParam.h"
namespace mozilla {
namespace dom {
struct ThreeDPoint;
}
class AudioNodeStream;
// We ensure that the graph advances in steps that are multiples of the Web
// Audio block size
const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
/**
* This class holds onto a set of immutable channel buffers. The storage
* for the buffers must be malloced, but the buffer pointers and the malloc
* pointers can be different (e.g. if the buffers are contained inside
* some malloced object).
*/
class ThreadSharedFloatArrayBufferList : public ThreadSharedObject {
public:
/**
* Construct with null data.
*/
ThreadSharedFloatArrayBufferList(uint32_t aCount)
{
mContents.SetLength(aCount);
}
struct Storage {
Storage()
{
mDataToFree = nullptr;
mSampleData = nullptr;
}
~Storage() { free(mDataToFree); }
void* mDataToFree;
const float* mSampleData;
};
/**
* This can be called on any thread.
*/
uint32_t GetChannels() const { return mContents.Length(); }
/**
* This can be called on any thread.
*/
const float* GetData(uint32_t aIndex) const { return mContents[aIndex].mSampleData; }
/**
* Call this only during initialization, before the object is handed to
* any other thread.
*/
void SetData(uint32_t aIndex, void* aDataToFree, const float* aData)
{
Storage* s = &mContents[aIndex];
free(s->mDataToFree);
s->mDataToFree = aDataToFree;
s->mSampleData = aData;
}
/**
* Put this object into an error state where there are no channels.
*/
void Clear() { mContents.Clear(); }
private:
AutoFallibleTArray<Storage,2> mContents;
};
/**
* Allocates an AudioChunk with fresh buffers of WEBAUDIO_BLOCK_SIZE float samples.
* AudioChunk::mChannelData's entries can be cast to float* for writing.
*/
void AllocateAudioBlock(uint32_t aChannelCount, AudioChunk* aChunk);
/**
* aChunk must have been allocated by AllocateAudioBlock.
*/
void WriteZeroesToAudioBlock(AudioChunk* aChunk, uint32_t aStart, uint32_t aLength);
/**
* Pointwise multiply-add operation. aScale == 1.0f should be optimized.
*/
void AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
float aScale,
float aOutput[WEBAUDIO_BLOCK_SIZE]);
/**
* Pointwise copy-scaled operation. aScale == 1.0f should be optimized.
*/
void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
float aScale,
float aOutput[WEBAUDIO_BLOCK_SIZE]);
/**
* Vector copy-scaled operation.
*/
void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
const float aScale[WEBAUDIO_BLOCK_SIZE],
float aOutput[WEBAUDIO_BLOCK_SIZE]);
/**
* In place gain. aScale == 1.0f should be optimized.
*/
void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
uint32_t aChannelCount,
float aScale);
/**
* Upmix a mono input to a stereo output, scaling the two output channels by two
* different gain value.
* This algorithm is specified in the WebAudio spec.
*/
void
AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
float aGainL, float aGainR,
float aOutputL[WEBAUDIO_BLOCK_SIZE],
float aOutputR[WEBAUDIO_BLOCK_SIZE]);
/**
* Pan a stereo source according to right and left gain, and the position
* (whether the listener is on the left of the source or not).
* This algorithm is specified in the WebAudio spec.
*/
void
AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
const float aInputR[WEBAUDIO_BLOCK_SIZE],
float aGainL, float aGainR, bool aIsOnTheLeft,
float aOutputL[WEBAUDIO_BLOCK_SIZE],
float aOutputR[WEBAUDIO_BLOCK_SIZE]);
/**
* All methods of this class and its subclasses are called on the
* MediaStreamGraph thread.
*/
class AudioNodeEngine {
public:
AudioNodeEngine() {}
virtual ~AudioNodeEngine() {}
virtual void SetStreamTimeParameter(uint32_t aIndex, TrackTicks aParam)
{
NS_ERROR("Invalid SetStreamTimeParameter index");
}
virtual void SetDoubleParameter(uint32_t aIndex, double aParam)
{
NS_ERROR("Invalid SetDoubleParameter index");
}
virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam)
{
NS_ERROR("Invalid SetInt32Parameter index");
}
virtual void SetTimelineParameter(uint32_t aIndex,
const dom::AudioParamTimeline& aValue)
{
NS_ERROR("Invalid SetTimelineParameter index");
}
virtual void SetThreeDPointParameter(uint32_t aIndex,
const dom::ThreeDPoint& aValue)
{
NS_ERROR("Invalid SetThreeDPointParameter index");
}
virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
{
NS_ERROR("SetBuffer called on engine that doesn't support it");
}
/**
* Produce the next block of audio samples, given input samples aInput
* (the mixed data for input 0).
* By default, simply returns the mixed input.
* aInput is guaranteed to have float sample format (if it has samples at all)
* and to have been resampled to IdealAudioRate(), and to have exactly
* WEBAUDIO_BLOCK_SIZE samples.
* *aFinished is set to false by the caller. If the callee sets it to true,
* we'll finish the stream and not call this again.
*/
virtual void ProduceAudioBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished)
{
*aOutput = aInput;
}
};
}
#endif /* MOZILLA_AUDIONODEENGINE_H_ */