Source/WebCore/platform/audio/DynamicsCompressor.cpp - WebKit - Git at Google

 /*
  * 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 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 "config.h"

 #if ENABLE(WEB_AUDIO)

 #include "DynamicsCompressor.h"

 #include "AudioBus.h"
 #include "AudioUtilities.h"
 #include <wtf/MathExtras.h>
 #include <wtf/StdLibExtras.h>

 namespace WebCore {

 using namespace AudioUtilities;

 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();
 }

 void DynamicsCompressor::setParameterValue(unsigned parameterID, float value)
 {
     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)
 {
     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 * piFloat);
     float r2 = expf(-f2 * piFloat);

     ASSERT(m_numberOfChannels == m_preFilterPacks.size());

     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 AudioBus* sourceBus, AudioBus* destinationBus, 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 = destinationBus->numberOfChannels();
     unsigned numberOfSourceChannels = sourceBus->numberOfChannels();

     ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);

     if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
         destinationBus->zero();
         return;
     }

     switch (numberOfChannels) {
     case 2: // stereo
         m_sourceChannels[0] = sourceBus->channel(0)->data();

         if (numberOfSourceChannels > 1)
             m_sourceChannels[1] = sourceBus->channel(1)->data();
         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.
         ASSERT_NOT_REACHED();
         destinationBus->zero();
         return;
     }

     for (unsigned i = 0; i < numberOfChannels; ++i)
         m_destinationChannels[i] = destinationBus->channel(i)->mutableData();

     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);
     }

     // 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 = m_sourceChannels[i];
         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.size() == numberOfChannels)
         return;

     m_preFilterPacks.clear();
     m_postFilterPacks.clear();
     for (unsigned i = 0; i < numberOfChannels; ++i) {
         m_preFilterPacks.append(makeUnique<ZeroPoleFilterPack4>());
         m_postFilterPacks.append(makeUnique<ZeroPoleFilterPack4>());
     }

     m_sourceChannels = makeUniqueArray<const float*>(numberOfChannels);
     m_destinationChannels = makeUniqueArray<float*>(numberOfChannels);

     m_compressor.setNumberOfChannels(numberOfChannels);
     m_numberOfChannels = numberOfChannels;
 }

 } // namespace WebCore

 #endif // ENABLE(WEB_AUDIO)
	/*
	* 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 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 "config.h"

	#if ENABLE(WEB_AUDIO)

	#include "DynamicsCompressor.h"

	#include "AudioBus.h"
	#include "AudioUtilities.h"
	#include <wtf/MathExtras.h>
	#include <wtf/StdLibExtras.h>

	namespace WebCore {

	using namespace AudioUtilities;

	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();
	}

	void DynamicsCompressor::setParameterValue(unsigned parameterID, float value)
	{
	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)
	{
	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 * piFloat);
	float r2 = expf(-f2 * piFloat);

	ASSERT(m_numberOfChannels == m_preFilterPacks.size());

	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 AudioBus* sourceBus, AudioBus* destinationBus, 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 = destinationBus->numberOfChannels();
	unsigned numberOfSourceChannels = sourceBus->numberOfChannels();

	ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);

	if (numberOfChannels != m_numberOfChannels \|\| !numberOfSourceChannels) {
	destinationBus->zero();
	return;
	}

	switch (numberOfChannels) {
	case 2: // stereo
	m_sourceChannels[0] = sourceBus->channel(0)->data();

	if (numberOfSourceChannels > 1)
	m_sourceChannels[1] = sourceBus->channel(1)->data();
	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.
	ASSERT_NOT_REACHED();
	destinationBus->zero();
	return;
	}

	for (unsigned i = 0; i < numberOfChannels; ++i)
	m_destinationChannels[i] = destinationBus->channel(i)->mutableData();

	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);
	}

	// 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 = m_sourceChannels[i];
	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.size() == numberOfChannels)
	return;

	m_preFilterPacks.clear();
	m_postFilterPacks.clear();
	for (unsigned i = 0; i < numberOfChannels; ++i) {
	m_preFilterPacks.append(makeUnique<ZeroPoleFilterPack4>());
	m_postFilterPacks.append(makeUnique<ZeroPoleFilterPack4>());
	}

	m_sourceChannels = makeUniqueArray<const float*>(numberOfChannels);
	m_destinationChannels = makeUniqueArray<float*>(numberOfChannels);

	m_compressor.setNumberOfChannels(numberOfChannels);
	m_numberOfChannels = numberOfChannels;
	}

	} // namespace WebCore

	#endif // ENABLE(WEB_AUDIO)