/*

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 *     from this software without specific prior written permission.

 *

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#include "DynamicsCompressor.h"

#include "AlignmentUtils.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 + 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