The Tor Browser: content/media/webaudio/blink/DynamicsCompressor.cpp@97036ab72558 (annotated)

content/media/webaudio/blink/DynamicsCompressor.cpp@97036ab72558 (annotated)

content/media/webaudio/blink/DynamicsCompressor.cpp

Tue, 06 Jan 2015 21:39:09 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Tue, 06 Jan 2015 21:39:09 +0100
branch: TOR_BUG_9701
changeset 8: 97036ab72558
permissions: -rw-r--r--

Conditionally force memory storage according to privacy.thirdparty.isolate;
This solves Tor bug #9701, complying with disk avoidance documented in
https://www.torproject.org/projects/torbrowser/design/#disk-avoidance.

 /*
  * Copyright (C) 2011 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 "DynamicsCompressor.h"
 #include "AudioSegment.h"
 #include <cmath>
 #include "AudioNodeEngine.h"
 #include "nsDebug.h"
 using mozilla::WEBAUDIO_BLOCK_SIZE;
 using mozilla::AudioBlockCopyChannelWithScale;
 namespace WebCore {
 DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels)
     : m_numberOfChannels(numberOfChannels)
     , m_sampleRate(sampleRate)
     , m_compressor(sampleRate, numberOfChannels)
 {
     // Uninitialized state - for parameter recalculation.
     m_lastFilterStageRatio = -1;
     m_lastAnchor = -1;
     m_lastFilterStageGain = -1;
     setNumberOfChannels(numberOfChannels);
     initializeParameters();
 }
 size_t DynamicsCompressor::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
 {
     size_t amount = aMallocSizeOf(this);
     amount += m_preFilterPacks.SizeOfExcludingThis(aMallocSizeOf);
     for (size_t i = 0; i < m_preFilterPacks.Length(); i++) {
         if (m_preFilterPacks[i]) {
             amount += m_preFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
         }
     }
     amount += m_postFilterPacks.SizeOfExcludingThis(aMallocSizeOf);
     for (size_t i = 0; i < m_postFilterPacks.Length(); i++) {
         if (m_postFilterPacks[i]) {
             amount += m_postFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
         }
     }
     amount += m_sourceChannels.SizeOfExcludingThis(aMallocSizeOf);
     amount += m_destinationChannels.SizeOfExcludingThis(aMallocSizeOf);
     amount += m_compressor.sizeOfExcludingThis(aMallocSizeOf);
     return amount;
 }
 void DynamicsCompressor::setParameterValue(unsigned parameterID, float value)
 {
     MOZ_ASSERT(parameterID < ParamLast);
     if (parameterID < ParamLast)
         m_parameters[parameterID] = value;
 }
 void DynamicsCompressor::initializeParameters()
 {
     // Initializes compressor to default values.
     m_parameters[ParamThreshold] = -24; // dB
     m_parameters[ParamKnee] = 30; // dB
     m_parameters[ParamRatio] = 12; // unit-less
     m_parameters[ParamAttack] = 0.003f; // seconds
     m_parameters[ParamRelease] = 0.250f; // seconds
     m_parameters[ParamPreDelay] = 0.006f; // seconds
     // Release zone values 0 -> 1.
     m_parameters[ParamReleaseZone1] = 0.09f;
     m_parameters[ParamReleaseZone2] = 0.16f;
     m_parameters[ParamReleaseZone3] = 0.42f;
     m_parameters[ParamReleaseZone4] = 0.98f;
     m_parameters[ParamFilterStageGain] = 4.4f; // dB
     m_parameters[ParamFilterStageRatio] = 2;
     m_parameters[ParamFilterAnchor] = 15000 / nyquist();
     m_parameters[ParamPostGain] = 0; // dB
     m_parameters[ParamReduction] = 0; // dB
     // Linear crossfade (0 -> 1).
     m_parameters[ParamEffectBlend] = 1;
 }
 float DynamicsCompressor::parameterValue(unsigned parameterID)
 {
     MOZ_ASSERT(parameterID < ParamLast);
     return m_parameters[parameterID];
 }
 void DynamicsCompressor::setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */)
 {
     float gk = 1 - gain / 20;
     float f1 = normalizedFrequency * gk;
     float f2 = normalizedFrequency / gk;
     float r1 = expf(-f1 * M_PI);
     float r2 = expf(-f2 * M_PI);
     MOZ_ASSERT(m_numberOfChannels == m_preFilterPacks.Length());
     for (unsigned i = 0; i < m_numberOfChannels; ++i) {
         // Set pre-filter zero and pole to create an emphasis filter.
         ZeroPole& preFilter = m_preFilterPacks[i]->filters[stageIndex];
         preFilter.setZero(r1);
         preFilter.setPole(r2);
         // Set post-filter with zero and pole reversed to create the de-emphasis filter.
         // If there were no compressor kernel in between, they would cancel each other out (allpass filter).
         ZeroPole& postFilter = m_postFilterPacks[i]->filters[stageIndex];
         postFilter.setZero(r2);
         postFilter.setPole(r1);
     }
 }
 void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio)
 {
     setEmphasisStageParameters(0, gain, anchorFreq);
     setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio);
     setEmphasisStageParameters(2, gain, anchorFreq / (filterStageRatio * filterStageRatio));
     setEmphasisStageParameters(3, gain, anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio));
 }
 void DynamicsCompressor::process(const AudioChunk* sourceChunk, AudioChunk* destinationChunk, unsigned framesToProcess)
 {
     // Though numberOfChannels is retrived from destinationBus, we still name it numberOfChannels instead of numberOfDestinationChannels.
     // It's because we internally match sourceChannels's size to destinationBus by channel up/down mix. Thus we need numberOfChannels
     // to do the loop work for both m_sourceChannels and m_destinationChannels.
     unsigned numberOfChannels = destinationChunk->mChannelData.Length();
     unsigned numberOfSourceChannels = sourceChunk->mChannelData.Length();
     MOZ_ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);
     if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
         destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
         return;
     }
     switch (numberOfChannels) {
     case 2: // stereo
         m_sourceChannels[0] = static_cast<const float*>(sourceChunk->mChannelData[0]);
         if (numberOfSourceChannels > 1)
             m_sourceChannels[1] = static_cast<const float*>(sourceChunk->mChannelData[1]);
         else
             // Simply duplicate mono channel input data to right channel for stereo processing.
             m_sourceChannels[1] = m_sourceChannels[0];
         break;
     default:
         // FIXME : support other number of channels.
         NS_WARNING("Support other number of channels");
         destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
         return;
     }
     for (unsigned i = 0; i < numberOfChannels; ++i)
         m_destinationChannels[i] = const_cast<float*>(static_cast<const float*>(
             destinationChunk->mChannelData[i]));
     float filterStageGain = parameterValue(ParamFilterStageGain);
     float filterStageRatio = parameterValue(ParamFilterStageRatio);
     float anchor = parameterValue(ParamFilterAnchor);
     if (filterStageGain != m_lastFilterStageGain || filterStageRatio != m_lastFilterStageRatio || anchor != m_lastAnchor) {
         m_lastFilterStageGain = filterStageGain;
         m_lastFilterStageRatio = filterStageRatio;
         m_lastAnchor = anchor;
         setEmphasisParameters(filterStageGain, anchor, filterStageRatio);
     }
     float sourceWithVolume[WEBAUDIO_BLOCK_SIZE];
     // Apply pre-emphasis filter.
     // Note that the final three stages are computed in-place in the destination buffer.
     for (unsigned i = 0; i < numberOfChannels; ++i) {
         const float* sourceData;
         if (sourceChunk->mVolume == 1.0f) {
           // Fast path, the volume scale doesn't need to get taken into account
           sourceData = m_sourceChannels[i];
         } else {
           AudioBlockCopyChannelWithScale(m_sourceChannels[i],
                                          sourceChunk->mVolume,
                                          sourceWithVolume);
           sourceData = sourceWithVolume;
         }
         float* destinationData = m_destinationChannels[i];
         ZeroPole* preFilters = m_preFilterPacks[i]->filters;
         preFilters[0].process(sourceData, destinationData, framesToProcess);
         preFilters[1].process(destinationData, destinationData, framesToProcess);
         preFilters[2].process(destinationData, destinationData, framesToProcess);
         preFilters[3].process(destinationData, destinationData, framesToProcess);
     }
     float dbThreshold = parameterValue(ParamThreshold);
     float dbKnee = parameterValue(ParamKnee);
     float ratio = parameterValue(ParamRatio);
     float attackTime = parameterValue(ParamAttack);
     float releaseTime = parameterValue(ParamRelease);
     float preDelayTime = parameterValue(ParamPreDelay);
     // This is effectively a master volume on the compressed signal (pre-blending).
     float dbPostGain = parameterValue(ParamPostGain);
     // Linear blending value from dry to completely processed (0 -> 1)
     // 0 means the signal is completely unprocessed.
     // 1 mixes in only the compressed signal.
     float effectBlend = parameterValue(ParamEffectBlend);
     float releaseZone1 = parameterValue(ParamReleaseZone1);
     float releaseZone2 = parameterValue(ParamReleaseZone2);
     float releaseZone3 = parameterValue(ParamReleaseZone3);
     float releaseZone4 = parameterValue(ParamReleaseZone4);
     // Apply compression to the pre-filtered signal.
     // The processing is performed in place.
     m_compressor.process(m_destinationChannels.get(),
                          m_destinationChannels.get(),
                          numberOfChannels,
                          framesToProcess,
                          dbThreshold,
                          dbKnee,
                          ratio,
                          attackTime,
                          releaseTime,
                          preDelayTime,
                          dbPostGain,
                          effectBlend,
                          releaseZone1,
                          releaseZone2,
                          releaseZone3,
                          releaseZone4
                          );
     // Update the compression amount.
     setParameterValue(ParamReduction, m_compressor.meteringGain());
     // Apply de-emphasis filter.
     for (unsigned i = 0; i < numberOfChannels; ++i) {
         float* destinationData = m_destinationChannels[i];
         ZeroPole* postFilters = m_postFilterPacks[i]->filters;
         postFilters[0].process(destinationData, destinationData, framesToProcess);
         postFilters[1].process(destinationData, destinationData, framesToProcess);
         postFilters[2].process(destinationData, destinationData, framesToProcess);
         postFilters[3].process(destinationData, destinationData, framesToProcess);
     }
 }
 void DynamicsCompressor::reset()
 {
     m_lastFilterStageRatio = -1; // for recalc
     m_lastAnchor = -1;
     m_lastFilterStageGain = -1;
     for (unsigned channel = 0; channel < m_numberOfChannels; ++channel) {
         for (unsigned stageIndex = 0; stageIndex < 4; ++stageIndex) {
             m_preFilterPacks[channel]->filters[stageIndex].reset();
             m_postFilterPacks[channel]->filters[stageIndex].reset();
         }
     }
     m_compressor.reset();
 }
 void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels)
 {
     if (m_preFilterPacks.Length() == numberOfChannels)
         return;
     m_preFilterPacks.Clear();
     m_postFilterPacks.Clear();
     for (unsigned i = 0; i < numberOfChannels; ++i) {
         m_preFilterPacks.AppendElement(new ZeroPoleFilterPack4());
         m_postFilterPacks.AppendElement(new ZeroPoleFilterPack4());
     }
     m_sourceChannels = new const float* [numberOfChannels];
     m_destinationChannels = new float* [numberOfChannels];
     m_compressor.setNumberOfChannels(numberOfChannels);
     m_numberOfChannels = numberOfChannels;
 }
 } // namespace WebCore

The Tor Browser / annotate

content/media/webaudio/blink/DynamicsCompressor.cpp@97036ab72558 (annotated)

content/media/webaudio/blink/DynamicsCompressor.cpp