michael@0: /*
michael@0:  * Copyright (C) 2011 Google Inc. All rights reserved.
michael@0:  *
michael@0:  * Redistribution and use in source and binary forms, with or without
michael@0:  * modification, are permitted provided that the following conditions
michael@0:  * are met:
michael@0:  *
michael@0:  * 1.  Redistributions of source code must retain the above copyright
michael@0:  *     notice, this list of conditions and the following disclaimer.
michael@0:  * 2.  Redistributions in binary form must reproduce the above copyright
michael@0:  *     notice, this list of conditions and the following disclaimer in the
michael@0:  *     documentation and/or other materials provided with the distribution.
michael@0:  * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
michael@0:  *     its contributors may be used to endorse or promote products derived
michael@0:  *     from this software without specific prior written permission.
michael@0:  *
michael@0:  * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
michael@0:  * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
michael@0:  * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
michael@0:  * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
michael@0:  * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
michael@0:  * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
michael@0:  * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
michael@0:  * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
michael@0:  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
michael@0:  * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
michael@0:  */
michael@0: 
michael@0: #include "DynamicsCompressor.h"
michael@0: #include "AudioSegment.h"
michael@0: 
michael@0: #include <cmath>
michael@0: #include "AudioNodeEngine.h"
michael@0: #include "nsDebug.h"
michael@0: 
michael@0: using mozilla::WEBAUDIO_BLOCK_SIZE;
michael@0: using mozilla::AudioBlockCopyChannelWithScale;
michael@0: 
michael@0: namespace WebCore {
michael@0: 
michael@0: DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels)
michael@0:     : m_numberOfChannels(numberOfChannels)
michael@0:     , m_sampleRate(sampleRate)
michael@0:     , m_compressor(sampleRate, numberOfChannels)
michael@0: {
michael@0:     // Uninitialized state - for parameter recalculation.
michael@0:     m_lastFilterStageRatio = -1;
michael@0:     m_lastAnchor = -1;
michael@0:     m_lastFilterStageGain = -1;
michael@0: 
michael@0:     setNumberOfChannels(numberOfChannels);
michael@0:     initializeParameters();
michael@0: }
michael@0: 
michael@0: size_t DynamicsCompressor::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
michael@0: {
michael@0:     size_t amount = aMallocSizeOf(this);
michael@0:     amount += m_preFilterPacks.SizeOfExcludingThis(aMallocSizeOf);
michael@0:     for (size_t i = 0; i < m_preFilterPacks.Length(); i++) {
michael@0:         if (m_preFilterPacks[i]) {
michael@0:             amount += m_preFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
michael@0:         }
michael@0:     }
michael@0: 
michael@0:     amount += m_postFilterPacks.SizeOfExcludingThis(aMallocSizeOf);
michael@0:     for (size_t i = 0; i < m_postFilterPacks.Length(); i++) {
michael@0:         if (m_postFilterPacks[i]) {
michael@0:             amount += m_postFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
michael@0:         }
michael@0:     }
michael@0: 
michael@0:     amount += m_sourceChannels.SizeOfExcludingThis(aMallocSizeOf);
michael@0:     amount += m_destinationChannels.SizeOfExcludingThis(aMallocSizeOf);
michael@0:     amount += m_compressor.sizeOfExcludingThis(aMallocSizeOf);
michael@0:     return amount;
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::setParameterValue(unsigned parameterID, float value)
michael@0: {
michael@0:     MOZ_ASSERT(parameterID < ParamLast);
michael@0:     if (parameterID < ParamLast)
michael@0:         m_parameters[parameterID] = value;
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::initializeParameters()
michael@0: {
michael@0:     // Initializes compressor to default values.
michael@0:     
michael@0:     m_parameters[ParamThreshold] = -24; // dB
michael@0:     m_parameters[ParamKnee] = 30; // dB
michael@0:     m_parameters[ParamRatio] = 12; // unit-less
michael@0:     m_parameters[ParamAttack] = 0.003f; // seconds
michael@0:     m_parameters[ParamRelease] = 0.250f; // seconds
michael@0:     m_parameters[ParamPreDelay] = 0.006f; // seconds
michael@0: 
michael@0:     // Release zone values 0 -> 1.
michael@0:     m_parameters[ParamReleaseZone1] = 0.09f;
michael@0:     m_parameters[ParamReleaseZone2] = 0.16f;
michael@0:     m_parameters[ParamReleaseZone3] = 0.42f;
michael@0:     m_parameters[ParamReleaseZone4] = 0.98f;
michael@0: 
michael@0:     m_parameters[ParamFilterStageGain] = 4.4f; // dB
michael@0:     m_parameters[ParamFilterStageRatio] = 2;
michael@0:     m_parameters[ParamFilterAnchor] = 15000 / nyquist();
michael@0:     
michael@0:     m_parameters[ParamPostGain] = 0; // dB
michael@0:     m_parameters[ParamReduction] = 0; // dB
michael@0: 
michael@0:     // Linear crossfade (0 -> 1).
michael@0:     m_parameters[ParamEffectBlend] = 1;
michael@0: }
michael@0: 
michael@0: float DynamicsCompressor::parameterValue(unsigned parameterID)
michael@0: {
michael@0:     MOZ_ASSERT(parameterID < ParamLast);
michael@0:     return m_parameters[parameterID];
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */)
michael@0: {
michael@0:     float gk = 1 - gain / 20;
michael@0:     float f1 = normalizedFrequency * gk;
michael@0:     float f2 = normalizedFrequency / gk;
michael@0:     float r1 = expf(-f1 * M_PI);
michael@0:     float r2 = expf(-f2 * M_PI);
michael@0: 
michael@0:     MOZ_ASSERT(m_numberOfChannels == m_preFilterPacks.Length());
michael@0: 
michael@0:     for (unsigned i = 0; i < m_numberOfChannels; ++i) {
michael@0:         // Set pre-filter zero and pole to create an emphasis filter.
michael@0:         ZeroPole& preFilter = m_preFilterPacks[i]->filters[stageIndex];
michael@0:         preFilter.setZero(r1);
michael@0:         preFilter.setPole(r2);
michael@0: 
michael@0:         // Set post-filter with zero and pole reversed to create the de-emphasis filter.
michael@0:         // If there were no compressor kernel in between, they would cancel each other out (allpass filter).
michael@0:         ZeroPole& postFilter = m_postFilterPacks[i]->filters[stageIndex];
michael@0:         postFilter.setZero(r2);
michael@0:         postFilter.setPole(r1);
michael@0:     }
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio)
michael@0: {
michael@0:     setEmphasisStageParameters(0, gain, anchorFreq);
michael@0:     setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio);
michael@0:     setEmphasisStageParameters(2, gain, anchorFreq / (filterStageRatio * filterStageRatio));
michael@0:     setEmphasisStageParameters(3, gain, anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio));
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::process(const AudioChunk* sourceChunk, AudioChunk* destinationChunk, unsigned framesToProcess)
michael@0: {
michael@0:     // Though numberOfChannels is retrived from destinationBus, we still name it numberOfChannels instead of numberOfDestinationChannels.
michael@0:     // It's because we internally match sourceChannels's size to destinationBus by channel up/down mix. Thus we need numberOfChannels
michael@0:     // to do the loop work for both m_sourceChannels and m_destinationChannels.
michael@0: 
michael@0:     unsigned numberOfChannels = destinationChunk->mChannelData.Length();
michael@0:     unsigned numberOfSourceChannels = sourceChunk->mChannelData.Length();
michael@0: 
michael@0:     MOZ_ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);
michael@0: 
michael@0:     if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
michael@0:         destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
michael@0:         return;
michael@0:     }
michael@0: 
michael@0:     switch (numberOfChannels) {
michael@0:     case 2: // stereo
michael@0:         m_sourceChannels[0] = static_cast<const float*>(sourceChunk->mChannelData[0]);
michael@0: 
michael@0:         if (numberOfSourceChannels > 1)
michael@0:             m_sourceChannels[1] = static_cast<const float*>(sourceChunk->mChannelData[1]);
michael@0:         else
michael@0:             // Simply duplicate mono channel input data to right channel for stereo processing.
michael@0:             m_sourceChannels[1] = m_sourceChannels[0];
michael@0: 
michael@0:         break;
michael@0:     default:
michael@0:         // FIXME : support other number of channels.
michael@0:         NS_WARNING("Support other number of channels");
michael@0:         destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
michael@0:         return;
michael@0:     }
michael@0: 
michael@0:     for (unsigned i = 0; i < numberOfChannels; ++i)
michael@0:         m_destinationChannels[i] = const_cast<float*>(static_cast<const float*>(
michael@0:             destinationChunk->mChannelData[i]));
michael@0: 
michael@0:     float filterStageGain = parameterValue(ParamFilterStageGain);
michael@0:     float filterStageRatio = parameterValue(ParamFilterStageRatio);
michael@0:     float anchor = parameterValue(ParamFilterAnchor);
michael@0: 
michael@0:     if (filterStageGain != m_lastFilterStageGain || filterStageRatio != m_lastFilterStageRatio || anchor != m_lastAnchor) {
michael@0:         m_lastFilterStageGain = filterStageGain;
michael@0:         m_lastFilterStageRatio = filterStageRatio;
michael@0:         m_lastAnchor = anchor;
michael@0: 
michael@0:         setEmphasisParameters(filterStageGain, anchor, filterStageRatio);
michael@0:     }
michael@0: 
michael@0:     float sourceWithVolume[WEBAUDIO_BLOCK_SIZE];
michael@0: 
michael@0:     // Apply pre-emphasis filter.
michael@0:     // Note that the final three stages are computed in-place in the destination buffer.
michael@0:     for (unsigned i = 0; i < numberOfChannels; ++i) {
michael@0:         const float* sourceData;
michael@0:         if (sourceChunk->mVolume == 1.0f) {
michael@0:           // Fast path, the volume scale doesn't need to get taken into account
michael@0:           sourceData = m_sourceChannels[i];
michael@0:         } else {
michael@0:           AudioBlockCopyChannelWithScale(m_sourceChannels[i],
michael@0:                                          sourceChunk->mVolume,
michael@0:                                          sourceWithVolume);
michael@0:           sourceData = sourceWithVolume;
michael@0:         }
michael@0: 
michael@0:         float* destinationData = m_destinationChannels[i];
michael@0:         ZeroPole* preFilters = m_preFilterPacks[i]->filters;
michael@0: 
michael@0:         preFilters[0].process(sourceData, destinationData, framesToProcess);
michael@0:         preFilters[1].process(destinationData, destinationData, framesToProcess);
michael@0:         preFilters[2].process(destinationData, destinationData, framesToProcess);
michael@0:         preFilters[3].process(destinationData, destinationData, framesToProcess);
michael@0:     }
michael@0: 
michael@0:     float dbThreshold = parameterValue(ParamThreshold);
michael@0:     float dbKnee = parameterValue(ParamKnee);
michael@0:     float ratio = parameterValue(ParamRatio);
michael@0:     float attackTime = parameterValue(ParamAttack);
michael@0:     float releaseTime = parameterValue(ParamRelease);
michael@0:     float preDelayTime = parameterValue(ParamPreDelay);
michael@0: 
michael@0:     // This is effectively a master volume on the compressed signal (pre-blending).
michael@0:     float dbPostGain = parameterValue(ParamPostGain);
michael@0: 
michael@0:     // Linear blending value from dry to completely processed (0 -> 1)
michael@0:     // 0 means the signal is completely unprocessed.
michael@0:     // 1 mixes in only the compressed signal.
michael@0:     float effectBlend = parameterValue(ParamEffectBlend);
michael@0: 
michael@0:     float releaseZone1 = parameterValue(ParamReleaseZone1);
michael@0:     float releaseZone2 = parameterValue(ParamReleaseZone2);
michael@0:     float releaseZone3 = parameterValue(ParamReleaseZone3);
michael@0:     float releaseZone4 = parameterValue(ParamReleaseZone4);
michael@0: 
michael@0:     // Apply compression to the pre-filtered signal.
michael@0:     // The processing is performed in place.
michael@0:     m_compressor.process(m_destinationChannels.get(),
michael@0:                          m_destinationChannels.get(),
michael@0:                          numberOfChannels,
michael@0:                          framesToProcess,
michael@0: 
michael@0:                          dbThreshold,
michael@0:                          dbKnee,
michael@0:                          ratio,
michael@0:                          attackTime,
michael@0:                          releaseTime,
michael@0:                          preDelayTime,
michael@0:                          dbPostGain,
michael@0:                          effectBlend,
michael@0: 
michael@0:                          releaseZone1,
michael@0:                          releaseZone2,
michael@0:                          releaseZone3,
michael@0:                          releaseZone4
michael@0:                          );
michael@0:                          
michael@0:     // Update the compression amount.                     
michael@0:     setParameterValue(ParamReduction, m_compressor.meteringGain());
michael@0: 
michael@0:     // Apply de-emphasis filter.
michael@0:     for (unsigned i = 0; i < numberOfChannels; ++i) {
michael@0:         float* destinationData = m_destinationChannels[i];
michael@0:         ZeroPole* postFilters = m_postFilterPacks[i]->filters;
michael@0: 
michael@0:         postFilters[0].process(destinationData, destinationData, framesToProcess);
michael@0:         postFilters[1].process(destinationData, destinationData, framesToProcess);
michael@0:         postFilters[2].process(destinationData, destinationData, framesToProcess);
michael@0:         postFilters[3].process(destinationData, destinationData, framesToProcess);
michael@0:     }
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::reset()
michael@0: {
michael@0:     m_lastFilterStageRatio = -1; // for recalc
michael@0:     m_lastAnchor = -1;
michael@0:     m_lastFilterStageGain = -1;
michael@0: 
michael@0:     for (unsigned channel = 0; channel < m_numberOfChannels; ++channel) {
michael@0:         for (unsigned stageIndex = 0; stageIndex < 4; ++stageIndex) {
michael@0:             m_preFilterPacks[channel]->filters[stageIndex].reset();
michael@0:             m_postFilterPacks[channel]->filters[stageIndex].reset();
michael@0:         }
michael@0:     }
michael@0: 
michael@0:     m_compressor.reset();
michael@0: }
michael@0: 
michael@0: void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels)
michael@0: {
michael@0:     if (m_preFilterPacks.Length() == numberOfChannels)
michael@0:         return;
michael@0: 
michael@0:     m_preFilterPacks.Clear();
michael@0:     m_postFilterPacks.Clear();
michael@0:     for (unsigned i = 0; i < numberOfChannels; ++i) {
michael@0:         m_preFilterPacks.AppendElement(new ZeroPoleFilterPack4());
michael@0:         m_postFilterPacks.AppendElement(new ZeroPoleFilterPack4());
michael@0:     }
michael@0: 
michael@0:     m_sourceChannels = new const float* [numberOfChannels];
michael@0:     m_destinationChannels = new float* [numberOfChannels];
michael@0: 
michael@0:     m_compressor.setNumberOfChannels(numberOfChannels);
michael@0:     m_numberOfChannels = numberOfChannels;
michael@0: }
michael@0: 
michael@0: } // namespace WebCore