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/*
* Copyright (C) 2010 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "platform/audio/ReverbConvolver.h"
#include "platform/CrossThreadFunctional.h"
#include "platform/WebTaskRunner.h"
#include "platform/audio/AudioBus.h"
#include "platform/audio/VectorMath.h"
#include "public/platform/Platform.h"
#include "public/platform/WebThread.h"
#include "public/platform/WebTraceLocation.h"
#include "wtf/PtrUtil.h"
#include <memory>
namespace blink {
using namespace VectorMath;
const int InputBufferSize = 8 * 16384;
// We only process the leading portion of the impulse response in the real-time
// thread. We don't exceed this length. It turns out then, that the
// background thread has about 278msec of scheduling slop. Empirically, this
// has been found to be a good compromise between giving enough time for
// scheduling slop, while still minimizing the amount of processing done in the
// primary (high-priority) thread. This was found to be a good value on Mac OS
// X, and may work well on other platforms as well, assuming the very rough
// scheduling latencies are similar on these time-scales. Of course, this code
// may need to be tuned for individual platforms if this assumption is found to
// be incorrect.
const size_t RealtimeFrameLimit = 8192 + 4096; // ~278msec @ 44.1KHz
const size_t MinFFTSize = 128;
const size_t MaxRealtimeFFTSize = 2048;
ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse,
size_t renderSliceSize,
size_t maxFFTSize,
size_t convolverRenderPhase,
bool useBackgroundThreads)
: m_impulseResponseLength(impulseResponse->length()),
m_accumulationBuffer(impulseResponse->length() + renderSliceSize),
m_inputBuffer(InputBufferSize),
m_minFFTSize(MinFFTSize), // First stage will have this size - successive
// stages will double in size each time
m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize
{
// If we are using background threads then don't exceed this FFT size for the
// stages which run in the real-time thread. This avoids having only one or
// two large stages (size 16384 or so) at the end which take a lot of time
// every several processing slices. This way we amortize the cost over more
// processing slices.
m_maxRealtimeFFTSize = MaxRealtimeFFTSize;
const float* response = impulseResponse->data();
size_t totalResponseLength = impulseResponse->length();
// The total latency is zero because the direct-convolution is used in the
// leading portion.
size_t reverbTotalLatency = 0;
size_t stageOffset = 0;
int i = 0;
size_t fftSize = m_minFFTSize;
while (stageOffset < totalResponseLength) {
size_t stageSize = fftSize / 2;
// For the last stage, it's possible that stageOffset is such that we're
// straddling the end of the impulse response buffer (if we use stageSize),
// so reduce the last stage's length...
if (stageSize + stageOffset > totalResponseLength)
stageSize = totalResponseLength - stageOffset;
// This "staggers" the time when each FFT happens so they don't all happen
// at the same time
int renderPhase = convolverRenderPhase + i * renderSliceSize;
bool useDirectConvolver = !stageOffset;
std::unique_ptr<ReverbConvolverStage> stage =
WTF::wrapUnique(new ReverbConvolverStage(
response, totalResponseLength, reverbTotalLatency, stageOffset,
stageSize, fftSize, renderPhase, renderSliceSize,
&m_accumulationBuffer, useDirectConvolver));
bool isBackgroundStage = false;
if (useBackgroundThreads && stageOffset > RealtimeFrameLimit) {
m_backgroundStages.push_back(std::move(stage));
isBackgroundStage = true;
} else {
m_stages.push_back(std::move(stage));
}
stageOffset += stageSize;
++i;
if (!useDirectConvolver) {
// Figure out next FFT size
fftSize *= 2;
}
if (useBackgroundThreads && !isBackgroundStage &&
fftSize > m_maxRealtimeFFTSize)
fftSize = m_maxRealtimeFFTSize;
if (fftSize > m_maxFFTSize)
fftSize = m_maxFFTSize;
}
// Start up background thread
// FIXME: would be better to up the thread priority here. It doesn't need to
// be real-time, but higher than the default...
if (useBackgroundThreads && m_backgroundStages.size() > 0)
m_backgroundThread = WTF::wrapUnique(Platform::current()->createThread(
"Reverb convolution background thread"));
}
ReverbConvolver::~ReverbConvolver() {
// Wait for background thread to stop
m_backgroundThread.reset();
}
void ReverbConvolver::processInBackground() {
// Process all of the stages until their read indices reach the input buffer's
// write index
int writeIndex = m_inputBuffer.writeIndex();
// Even though it doesn't seem like every stage needs to maintain its own
// version of readIndex we do this in case we want to run in more than one
// background thread.
int readIndex;
while ((readIndex = m_backgroundStages[0]->inputReadIndex()) !=
writeIndex) { // FIXME: do better to detect buffer overrun...
// The ReverbConvolverStages need to process in amounts which evenly divide
// half the FFT size
const int SliceSize = MinFFTSize / 2;
// Accumulate contributions from each stage
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
m_backgroundStages[i]->processInBackground(this, SliceSize);
}
}
void ReverbConvolver::process(const AudioChannel* sourceChannel,
AudioChannel* destinationChannel,
size_t framesToProcess) {
bool isSafe = sourceChannel && destinationChannel &&
sourceChannel->length() >= framesToProcess &&
destinationChannel->length() >= framesToProcess;
ASSERT(isSafe);
if (!isSafe)
return;
const float* source = sourceChannel->data();
float* destination = destinationChannel->mutableData();
bool isDataSafe = source && destination;
ASSERT(isDataSafe);
if (!isDataSafe)
return;
// Feed input buffer (read by all threads)
m_inputBuffer.write(source, framesToProcess);
// Accumulate contributions from each stage
for (size_t i = 0; i < m_stages.size(); ++i)
m_stages[i]->process(source, framesToProcess);
// Finally read from accumulation buffer
m_accumulationBuffer.readAndClear(destination, framesToProcess);
// Now that we've buffered more input, post another task to the background
// thread.
if (m_backgroundThread)
m_backgroundThread->getWebTaskRunner()->postTask(
BLINK_FROM_HERE, crossThreadBind(&ReverbConvolver::processInBackground,
crossThreadUnretained(this)));
}
void ReverbConvolver::reset() {
for (size_t i = 0; i < m_stages.size(); ++i)
m_stages[i]->reset();
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
m_backgroundStages[i]->reset();
m_accumulationBuffer.reset();
m_inputBuffer.reset();
}
size_t ReverbConvolver::latencyFrames() const {
return 0;
}
} // namespace blink
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