gecko-dev/dom/media/webaudio/DelayBuffer.h
Karl Tomlinson e74d20e2da bug 1197028 use AudioBlock for web audio processing to reuse buffers shared downstream r=padenot
--HG--
extra : rebase_source : d2e403ae64a314177cba4d596ea235eb351ad3bc
2015-09-03 19:01:50 +12:00

116 lines
4.1 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 DelayBuffer_h_
#define DelayBuffer_h_
#include "nsTArray.h"
#include "AudioSegment.h"
#include "mozilla/dom/AudioNodeBinding.h" // for ChannelInterpretation
namespace mozilla {
class DelayBuffer final
{
typedef dom::ChannelInterpretation ChannelInterpretation;
public:
// See WebAudioUtils::ComputeSmoothingRate() for frame to frame exponential
// |smoothingRate| multiplier.
DelayBuffer(double aMaxDelayTicks, double aSmoothingRate)
: mSmoothingRate(aSmoothingRate)
, mCurrentDelay(-1.0)
// Round the maximum delay up to the next tick.
, mMaxDelayTicks(ceil(aMaxDelayTicks))
, mCurrentChunk(0)
// mLastReadChunk is initialized in EnsureBuffer
#ifdef DEBUG
, mHaveWrittenBlock(false)
#endif
{
// The 180 second limit in AudioContext::CreateDelay() and the
// 1 << MEDIA_TIME_FRAC_BITS limit on sample rate provide a limit on the
// maximum delay.
MOZ_ASSERT(aMaxDelayTicks <=
std::numeric_limits<decltype(mMaxDelayTicks)>::max());
}
// Write a WEBAUDIO_BLOCK_SIZE block for aChannelCount channels.
void Write(const AudioBlock& aInputChunk);
// Read a block with an array of delays, in ticks, for each sample frame.
// Each delay should be >= 0 and <= MaxDelayTicks().
void Read(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk,
ChannelInterpretation aChannelInterpretation);
// Read a block with a constant delay, which will be smoothed with the
// previous delay. The delay should be >= 0 and <= MaxDelayTicks().
void Read(double aDelayTicks, AudioBlock* aOutputChunk,
ChannelInterpretation aChannelInterpretation);
// Read into one of the channels of aOutputChunk, given an array of
// delays in ticks. This is useful when delays are different on different
// channels. aOutputChunk must have already been allocated with at least as
// many channels as were in any of the blocks passed to Write().
void ReadChannel(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk, uint32_t aChannel,
ChannelInterpretation aChannelInterpretation);
// Advance the buffer pointer
void NextBlock()
{
mCurrentChunk = (mCurrentChunk + 1) % mChunks.Length();
#ifdef DEBUG
MOZ_ASSERT(mHaveWrittenBlock);
mHaveWrittenBlock = false;
#endif
}
void Reset() {
mChunks.Clear();
mCurrentDelay = -1.0;
};
int MaxDelayTicks() const { return mMaxDelayTicks; }
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const;
private:
void ReadChannels(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk,
uint32_t aFirstChannel, uint32_t aNumChannelsToRead,
ChannelInterpretation aChannelInterpretation);
bool EnsureBuffer();
int PositionForDelay(int aDelay);
int ChunkForPosition(int aPosition);
int OffsetForPosition(int aPosition);
int ChunkForDelay(int aDelay);
void UpdateUpmixChannels(int aNewReadChunk, uint32_t channelCount,
ChannelInterpretation aChannelInterpretation);
// Circular buffer for capturing delayed samples.
FallibleTArray<AudioChunk> mChunks;
// Cache upmixed channel arrays.
nsAutoTArray<const float*,GUESS_AUDIO_CHANNELS> mUpmixChannels;
double mSmoothingRate;
// Current delay, in fractional ticks
double mCurrentDelay;
// Maximum delay, in ticks
int mMaxDelayTicks;
// The current position in the circular buffer. The next write will be to
// this chunk, and the next read may begin before this chunk.
int mCurrentChunk;
// The chunk owning the pointers in mUpmixChannels
int mLastReadChunk;
#ifdef DEBUG
bool mHaveWrittenBlock;
#endif
};
} // namespace mozilla
#endif // DelayBuffer_h_