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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 "ReverbConvolver.h"
#include "ReverbConvolverStage.h"
using namespace mozilla;
namespace WebCore {
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
- WEBAUDIO_BLOCK_SIZE;
// First stage will have size MinFFTSize - successive stages will double in
// size each time until we hit the maximum size.
const size_t MinFFTSize = 256;
// 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.
const size_t MaxRealtimeFFTSize = 4096;
ReverbConvolver::ReverbConvolver(const float* impulseResponseData,
size_t impulseResponseLength,
size_t maxFFTSize,
size_t convolverRenderPhase,
bool useBackgroundThreads)
: m_impulseResponseLength(impulseResponseLength)
, m_accumulationBuffer(impulseResponseLength + WEBAUDIO_BLOCK_SIZE)
, m_inputBuffer(InputBufferSize)
, m_backgroundThread("ConvolverWorker")
, m_backgroundThreadCondition(&m_backgroundThreadLock)
, m_useBackgroundThreads(useBackgroundThreads)
, m_wantsToExit(false)
, m_moreInputBuffered(false)
{
// For the moment, a good way to know if we have real-time constraint is to check if we're using background threads.
// Otherwise, assume we're being run from a command-line tool.
bool hasRealtimeConstraint = useBackgroundThreads;
const float* response = impulseResponseData;
size_t totalResponseLength = impulseResponseLength;
// The total latency is zero because the first FFT stage is small enough
// to return output in the first block.
size_t reverbTotalLatency = 0;
size_t stageOffset = 0;
size_t stagePhase = 0;
size_t fftSize = 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;
// Use smallest FFT that is large enough to cover the last stage.
fftSize = MinFFTSize;
while (stageSize * 2 > fftSize) {
fftSize *= 2;
}
}
// This "staggers" the time when each FFT happens so they don't all happen at the same time
int renderPhase = convolverRenderPhase + stagePhase;
nsAutoPtr<ReverbConvolverStage> stage
(new ReverbConvolverStage(response, totalResponseLength,
reverbTotalLatency, stageOffset, stageSize,
fftSize, renderPhase,
&m_accumulationBuffer));
bool isBackgroundStage = false;
if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
m_backgroundStages.AppendElement(stage.forget());
isBackgroundStage = true;
} else
m_stages.AppendElement(stage.forget());
// Figure out next FFT size
fftSize *= 2;
stageOffset += stageSize;
if (hasRealtimeConstraint && !isBackgroundStage
&& fftSize > MaxRealtimeFFTSize) {
fftSize = MaxRealtimeFFTSize;
// Custom phase positions for all but the first of the realtime
// stages of largest size. These spread out the work of the
// larger realtime stages. None of the FFTs of size 1024, 2048 or
// 4096 are performed when processing the same block. The first
// MaxRealtimeFFTSize = 4096 stage, at the end of the doubling,
// performs its FFT at block 7. The FFTs of size 2048 are
// performed in blocks 3 + 8 * n and size 1024 at 1 + 4 * n.
const uint32_t phaseLookup[] = { 14, 0, 10, 4 };
stagePhase = WEBAUDIO_BLOCK_SIZE *
phaseLookup[m_stages.Length() % ArrayLength(phaseLookup)];
} else if (fftSize > maxFFTSize) {
fftSize = maxFFTSize;
// A prime offset spreads out FFTs in a way that all
// available phase positions will be used if there are sufficient
// stages.
stagePhase += 5 * WEBAUDIO_BLOCK_SIZE;
} else if (stageSize > WEBAUDIO_BLOCK_SIZE) {
// As the stages are doubling in size, the next FFT will occur
// mid-way between FFTs for this stage.
stagePhase = stageSize - WEBAUDIO_BLOCK_SIZE;
}
}
// 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 (this->useBackgroundThreads() && m_backgroundStages.Length() > 0) {
if (!m_backgroundThread.Start()) {
NS_WARNING("Cannot start convolver thread.");
return;
}
m_backgroundThread.message_loop()->PostTask(NewNonOwningRunnableMethod(this,
&ReverbConvolver::backgroundThreadEntry));
}
}
ReverbConvolver::~ReverbConvolver()
{
// Wait for background thread to stop
if (useBackgroundThreads() && m_backgroundThread.IsRunning()) {
m_wantsToExit = true;
// Wake up thread so it can return
{
AutoLock locker(m_backgroundThreadLock);
m_moreInputBuffered = true;
m_backgroundThreadCondition.Signal();
}
m_backgroundThread.Stop();
}
}
size_t ReverbConvolver::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
{
size_t amount = aMallocSizeOf(this);
amount += m_stages.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < m_stages.Length(); i++) {
if (m_stages[i]) {
amount += m_stages[i]->sizeOfIncludingThis(aMallocSizeOf);
}
}
amount += m_backgroundStages.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < m_backgroundStages.Length(); i++) {
if (m_backgroundStages[i]) {
amount += m_backgroundStages[i]->sizeOfIncludingThis(aMallocSizeOf);
}
}
// NB: The buffer sizes are static, so even though they might be accessed
// in another thread it's safe to measure them.
amount += m_accumulationBuffer.sizeOfExcludingThis(aMallocSizeOf);
amount += m_inputBuffer.sizeOfExcludingThis(aMallocSizeOf);
// Possible future measurements:
// - m_backgroundThread
// - m_backgroundThreadLock
// - m_backgroundThreadCondition
return amount;
}
void ReverbConvolver::backgroundThreadEntry()
{
while (!m_wantsToExit) {
// Wait for realtime thread to give us more input
m_moreInputBuffered = false;
{
AutoLock locker(m_backgroundThreadLock);
while (!m_moreInputBuffered && !m_wantsToExit)
m_backgroundThreadCondition.Wait();
}
// 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...
// Accumulate contributions from each stage
for (size_t i = 0; i < m_backgroundStages.Length(); ++i)
m_backgroundStages[i]->processInBackground(this);
}
}
}
void ReverbConvolver::process(const float* sourceChannelData,
float* destinationChannelData)
{
const float* source = sourceChannelData;
float* destination = destinationChannelData;
bool isDataSafe = source && destination;
MOZ_ASSERT(isDataSafe);
if (!isDataSafe)
return;
// Feed input buffer (read by all threads)
m_inputBuffer.write(source, WEBAUDIO_BLOCK_SIZE);
// Accumulate contributions from each stage
for (size_t i = 0; i < m_stages.Length(); ++i)
m_stages[i]->process(source);
// Finally read from accumulation buffer
m_accumulationBuffer.readAndClear(destination, WEBAUDIO_BLOCK_SIZE);
// Now that we've buffered more input, wake up our background thread.
// Not using a MutexLocker looks strange, but we use a tryLock() instead because this is run on the real-time
// thread where it is a disaster for the lock to be contended (causes audio glitching). It's OK if we fail to
// signal from time to time, since we'll get to it the next time we're called. We're called repeatedly
// and frequently (around every 3ms). The background thread is processing well into the future and has a considerable amount of
// leeway here...
if (m_backgroundThreadLock.Try()) {
m_moreInputBuffered = true;
m_backgroundThreadCondition.Signal();
m_backgroundThreadLock.Release();
}
}
} // namespace WebCore
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