The Tor Browser: comparison content/media/webaudio/blink/Reverb.cpp

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+:ee06592e7470
+/*
+* 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 "Reverb.h"
+#include "ReverbConvolverStage.h"
+#include <math.h>
+#include "ReverbConvolver.h"
+#include "mozilla/FloatingPoint.h"
+using namespace mozilla;
+namespace WebCore {
+// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
+const float GainCalibration = -58;
+const float GainCalibrationSampleRate = 44100;
+// A minimum power value to when normalizing a silent (or very quiet) impulse response
+const float MinPower = 0.000125f;
+static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
+{
+// Normalize by RMS power
+size_t numberOfChannels = response->GetChannels();
+float power = 0;
+for (size_t i = 0; i < numberOfChannels; ++i) {
+float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
+power += channelPower;
+}
+power = sqrt(power / (numberOfChannels * aLength));
+// Protect against accidental overload
+if (!IsFinite(power) || IsNaN(power) || power < MinPower)
+power = MinPower;
+float scale = 1 / power;
+scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
+// Scale depends on sample-rate.
+if (sampleRate)
+scale *= GainCalibrationSampleRate / sampleRate;
+// True-stereo compensation
+if (response->GetChannels() == 4)
+scale *= 0.5f;
+return scale;
+}
+Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
+{
+float scale = 1;
+nsAutoTArray<const float*,4> irChannels;
+for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
+irChannels.AppendElement(impulseResponse->GetData(i));
+}
+nsAutoTArray<float,1024> tempBuf;
+if (normalize) {
+scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
+if (scale) {
+tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
+for (uint32_t i = 0; i < irChannels.Length(); ++i) {
+float* buf = &tempBuf[i*impulseResponseBufferLength];
+AudioBufferCopyWithScale(irChannels[i], scale, buf,
+impulseResponseBufferLength);
+irChannels[i] = buf;
+}
+}
+}
+initialize(irChannels, impulseResponseBufferLength, renderSliceSize,
+maxFFTSize, numberOfChannels, useBackgroundThreads);
+}
+size_t Reverb::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+size_t amount = aMallocSizeOf(this);
+amount += m_convolvers.SizeOfExcludingThis(aMallocSizeOf);
+for (size_t i = 0; i < m_convolvers.Length(); i++) {
+if (m_convolvers[i]) {
+amount += m_convolvers[i]->sizeOfIncludingThis(aMallocSizeOf);
+}
+}
+amount += m_tempBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
+return amount;
+}
+void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+size_t impulseResponseBufferLength, size_t renderSliceSize,
+size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+{
+m_impulseResponseLength = impulseResponseBufferLength;
+// The reverb can handle a mono impulse response and still do stereo processing
+size_t numResponseChannels = impulseResponseBuffer.Length();
+m_convolvers.SetCapacity(numberOfChannels);
+int convolverRenderPhase = 0;
+for (size_t i = 0; i < numResponseChannels; ++i) {
+const float* channel = impulseResponseBuffer[i];
+size_t length = impulseResponseBufferLength;
+nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
+m_convolvers.AppendElement(convolver.forget());
+convolverRenderPhase += renderSliceSize;
+}
+// For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
+// It can be bad to allocate memory in a real-time thread.
+if (numResponseChannels == 4) {
+AllocateAudioBlock(2, &m_tempBuffer);
+WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
+}
+}
+void Reverb::process(const AudioChunk* sourceBus, AudioChunk* destinationBus, size_t framesToProcess)
+{
+// Do a fairly comprehensive sanity check.
+// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
+bool isSafeToProcess = sourceBus && destinationBus && sourceBus->mChannelData.Length() > 0 && destinationBus->mChannelData.Length() > 0
+&& framesToProcess <= MaxFrameSize && framesToProcess <= size_t(sourceBus->mDuration) && framesToProcess <= size_t(destinationBus->mDuration);
+MOZ_ASSERT(isSafeToProcess);
+if (!isSafeToProcess)
+return;
+// For now only handle mono or stereo output
+MOZ_ASSERT(destinationBus->mChannelData.Length() <= 2);
+float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
+const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
+// Handle input -> output matrixing...
+size_t numInputChannels = sourceBus->mChannelData.Length();
+size_t numOutputChannels = destinationBus->mChannelData.Length();
+size_t numReverbChannels = m_convolvers.Length();
+if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
+// 2 -> 2 -> 2
+const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
+m_convolvers[1]->process(sourceBusR, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
+} else  if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
+// 1 -> 2 -> 2
+for (int i = 0; i < 2; ++i) {
+float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
+m_convolvers[i]->process(sourceBusL, sourceBus->mDuration, destinationChannel, destinationBus->mDuration, framesToProcess);
+}
+} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
+// 1 -> 1 -> 2
+m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
+// simply copy L -> R
+float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->mDuration) >= framesToProcess && size_t(destinationBus->mDuration) >= framesToProcess;
+MOZ_ASSERT(isCopySafe);
+if (!isCopySafe)
+return;
+PodCopy(destinationChannelR, destinationChannelL, framesToProcess);
+} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
+// 1 -> 1 -> 1
+m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
+} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
+// 2 -> 4 -> 2 ("True" stereo)
+const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+// Process left virtual source
+m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
+m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
+// Process right virtual source
+m_convolvers[2]->process(sourceBusR, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
+m_convolvers[3]->process(sourceBusR, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
+AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
+AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
+} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
+// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
+// This is an inefficient use of a four-channel impulse response, but we should handle the case.
+float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+// Process left virtual source
+m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
+m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
+// Process right virtual source
+m_convolvers[2]->process(sourceBusL, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
+m_convolvers[3]->process(sourceBusL, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
+AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
+AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
+} else {
+// Handle gracefully any unexpected / unsupported matrixing
+// FIXME: add code for 5.1 support...
+destinationBus->SetNull(destinationBus->mDuration);
+}
+}
+void Reverb::reset()
+{
+for (size_t i = 0; i < m_convolvers.Length(); ++i)
+m_convolvers[i]->reset();
+}
+size_t Reverb::latencyFrames() const
+{
+return !m_convolvers.IsEmpty() ? m_convolvers[0]->latencyFrames() : 0;
+}
+} // namespace WebCore

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comparison: content/media/webaudio/blink/Reverb.cpp

content/media/webaudio/blink/Reverb.cpp