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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,
+ bool* aAllocationFailure)
+ : m_impulseResponseLength(impulseResponseLength),
+ m_accumulationBuffer(),
+ m_inputBuffer(InputBufferSize),
+ m_backgroundThread("ConvolverWorker"),
+ m_backgroundThreadMonitor("ConvolverMonitor"),
+ m_useBackgroundThreads(useBackgroundThreads),
+ m_wantsToExit(false),
+ m_moreInputBuffered(false) {
+ *aAllocationFailure = !m_accumulationBuffer.allocate(impulseResponseLength +
+ WEBAUDIO_BLOCK_SIZE);
+ if (*aAllocationFailure) {
+ return;
+ }
+ // 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;
+
+ UniquePtr<ReverbConvolverStage> stage(new ReverbConvolverStage(
+ response, totalResponseLength, reverbTotalLatency, stageOffset,
+ stageSize, fftSize, renderPhase, &m_accumulationBuffer));
+
+ bool isBackgroundStage = false;
+
+ if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
+ m_backgroundStages.AppendElement(std::move(stage));
+ isBackgroundStage = true;
+ } else
+ m_stages.AppendElement(std::move(stage));
+
+ // 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(
+ "WebCore::ReverbConvolver::backgroundThreadEntry", 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
+ {
+ MonitorAutoLock locker(m_backgroundThreadMonitor);
+ m_moreInputBuffered = true;
+ m_backgroundThreadMonitor.Notify();
+ }
+
+ 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_backgroundThreadMonitor
+ return amount;
+}
+
+void ReverbConvolver::backgroundThreadEntry() {
+ while (!m_wantsToExit) {
+ // Wait for realtime thread to give us more input
+ m_moreInputBuffered = false;
+ {
+ MonitorAutoLock locker(m_backgroundThreadMonitor);
+ while (!m_moreInputBuffered && !m_wantsToExit)
+ m_backgroundThreadMonitor.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 MonitorAutoLock 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_backgroundThreadMonitor.TryLock()) {
+ m_moreInputBuffered = true;
+ m_backgroundThreadMonitor.Notify();
+ m_backgroundThreadMonitor.Unlock();
+ }
+}
+
+} // namespace WebCore