third_party/WebKit/Source/platform/audio/DynamicsCompressor.cpp

/*
 * Copyright (C) 2011 Google Inc. All rights reserved.
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 *     from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
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#include "platform/audio/AudioBus.h"
#include "platform/audio/AudioUtilities.h"
#include "platform/audio/DynamicsCompressor.h"
#include "wtf/MathExtras.h"
#include "wtf/PtrUtil.h"

namespace blink {

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::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.
      NOTREACHED();
      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;
  }

  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 source signal.
  m_compressor.process(m_sourceChannels.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());
}

void DynamicsCompressor::reset() {
  m_lastFilterStageRatio = -1;  // for recalc
  m_lastAnchor = -1;
  m_lastFilterStageGain = -1;

  m_compressor.reset();
}

void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels) {
  m_sourceChannels = wrapArrayUnique(new const float*[numberOfChannels]);
  m_destinationChannels = wrapArrayUnique(new float*[numberOfChannels]);

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

}  // namespace blink