DynamicsCompressor.cpp
/*
* Copyright (C) 2011 Google Inc. All rights reserved.
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#include "DynamicsCompressor.h"
#include "AudioBlock.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.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < m_preFilterPacks.Length(); i++) {
if (m_preFilterPacks[i]) {
amount += m_preFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
}
}
amount += m_postFilterPacks.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < m_postFilterPacks.Length(); i++) {
if (m_postFilterPacks[i]) {
amount += m_postFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
}
}
amount += aMallocSizeOf(m_sourceChannels.get());
amount += aMallocSizeOf(m_destinationChannels.get());
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 AudioBlock* sourceChunk, AudioBlock* 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->ChannelCount();
unsigned numberOfSourceChannels = sourceChunk->ChannelCount();
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 = mozilla::MakeUnique<const float* []>(numberOfChannels);
m_destinationChannels = mozilla::MakeUnique<float* []>(numberOfChannels);
m_compressor.setNumberOfChannels(numberOfChannels);
m_numberOfChannels = numberOfChannels;
}
} // namespace WebCore