Add m-esr52 at 52.6.0

1 files changed, 321 insertions, 0 deletions

diff --git a/dom/media/webaudio/blink/DynamicsCompressor.cpp b/dom/media/webaudio/blink/DynamicsCompressor.cpp
new file mode 100644
index 000000000..8f18913c0
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressor.cpp
@@ -0,0 +1,321 @@
+/*
+ * 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::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