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+/*
+ * 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 "AlignmentUtils.h"
+#include "AudioBlock.h"
+
+#include <cmath>
+#include "AudioNodeEngine.h"
+#include "nsDebug.h"
+
+using mozilla::AudioBlockCopyChannelWithScale;
+using mozilla::WEBAUDIO_BLOCK_SIZE;
+
+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;
+ case 1:
+ m_sourceChannels[0] =
+ static_cast<const float*>(sourceChunk->mChannelData[0]);
+ break;
+ default:
+ MOZ_CRASH("not supported.");
+ }
+
+ 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 + 4];
+ float* alignedSourceWithVolume = ALIGNED16(sourceWithVolume);
+ ASSERT_ALIGNED16(alignedSourceWithVolume);
+
+ // 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,
+ alignedSourceWithVolume);
+ sourceData = alignedSourceWithVolume;
+ }
+
+ 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