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