Add m-esr52 at 52.6.0

1 files changed, 243 insertions, 0 deletions

diff --git a/dom/media/webaudio/blink/Reverb.cpp b/dom/media/webaudio/blink/Reverb.cpp
new file mode 100644
index 000000000..4fca0822b
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.cpp
@@ -0,0 +1,243 @@
+/*
+ * Copyright (C) 2010 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 "Reverb.h"
+#include "ReverbConvolverStage.h"
+
+#include <math.h>
+#include "ReverbConvolver.h"
+#include "mozilla/FloatingPoint.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
+const float GainCalibration = -58;
+const float GainCalibrationSampleRate = 44100;
+
+// A minimum power value to when normalizing a silent (or very quiet) impulse response
+const float MinPower = 0.000125f;
+
+static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
+{
+    // Normalize by RMS power
+    size_t numberOfChannels = response->GetChannels();
+
+    float power = 0;
+
+    for (size_t i = 0; i < numberOfChannels; ++i) {
+        float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
+        power += channelPower;
+    }
+
+    power = sqrt(power / (numberOfChannels * aLength));
+
+    // Protect against accidental overload
+    if (!IsFinite(power) || IsNaN(power) || power < MinPower)
+        power = MinPower;
+
+    float scale = 1 / power;
+
+    scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
+
+    // Scale depends on sample-rate.
+    if (sampleRate)
+        scale *= GainCalibrationSampleRate / sampleRate;
+
+    // True-stereo compensation
+    if (response->GetChannels() == 4)
+        scale *= 0.5f;
+
+    return scale;
+}
+
+Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
+{
+    float scale = 1;
+
+    AutoTArray<const float*,4> irChannels;
+    for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
+        irChannels.AppendElement(impulseResponse->GetData(i));
+    }
+    AutoTArray<float,1024> tempBuf;
+
+    if (normalize) {
+        scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
+
+        if (scale) {
+            tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
+            for (uint32_t i = 0; i < irChannels.Length(); ++i) {
+                float* buf = &tempBuf[i*impulseResponseBufferLength];
+                AudioBufferCopyWithScale(irChannels[i], scale, buf,
+                                         impulseResponseBufferLength);
+                irChannels[i] = buf;
+            }
+        }
+    }
+
+    initialize(irChannels, impulseResponseBufferLength,
+               maxFFTSize, numberOfChannels, useBackgroundThreads);
+}
+
+size_t Reverb::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+    amount += m_convolvers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_convolvers.Length(); i++) {
+        if (m_convolvers[i]) {
+            amount += m_convolvers[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    amount += m_tempBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
+    return amount;
+}
+
+
+void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                        size_t impulseResponseBufferLength,
+                        size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+{
+    m_impulseResponseLength = impulseResponseBufferLength;
+
+    // The reverb can handle a mono impulse response and still do stereo processing
+    size_t numResponseChannels = impulseResponseBuffer.Length();
+    m_convolvers.SetCapacity(numberOfChannels);
+
+    int convolverRenderPhase = 0;
+    for (size_t i = 0; i < numResponseChannels; ++i) {
+        const float* channel = impulseResponseBuffer[i];
+        size_t length = impulseResponseBufferLength;
+
+        nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
+        m_convolvers.AppendElement(convolver.forget());
+
+        convolverRenderPhase += WEBAUDIO_BLOCK_SIZE;
+    }
+
+    // For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
+    // It can be bad to allocate memory in a real-time thread.
+    if (numResponseChannels == 4) {
+        m_tempBuffer.AllocateChannels(2);
+        WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
+    }
+}
+
+void Reverb::process(const AudioBlock* sourceBus, AudioBlock* destinationBus)
+{
+    // Do a fairly comprehensive sanity check.
+    // If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
+    bool isSafeToProcess = sourceBus && destinationBus && sourceBus->ChannelCount() > 0 && destinationBus->mChannelData.Length() > 0
+        && WEBAUDIO_BLOCK_SIZE <= MaxFrameSize && WEBAUDIO_BLOCK_SIZE <= size_t(sourceBus->GetDuration()) && WEBAUDIO_BLOCK_SIZE <= size_t(destinationBus->GetDuration());
+
+    MOZ_ASSERT(isSafeToProcess);
+    if (!isSafeToProcess)
+        return;
+
+    // For now only handle mono or stereo output
+    MOZ_ASSERT(destinationBus->ChannelCount() <= 2);
+
+    float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
+    const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
+
+    // Handle input -> output matrixing...
+    size_t numInputChannels = sourceBus->ChannelCount();
+    size_t numOutputChannels = destinationBus->ChannelCount();
+    size_t numReverbChannels = m_convolvers.Length();
+
+    if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
+        // 2 -> 2 -> 2
+        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusR, destinationChannelR);
+    } else  if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
+        // 1 -> 2 -> 2
+        for (int i = 0; i < 2; ++i) {
+            float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
+            m_convolvers[i]->process(sourceBusL, destinationChannel);
+        }
+    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
+        // 1 -> 1 -> 2
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+
+        // simply copy L -> R
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+        bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->GetDuration()) >= WEBAUDIO_BLOCK_SIZE;
+        MOZ_ASSERT(isCopySafe);
+        if (!isCopySafe)
+            return;
+        PodCopy(destinationChannelR, destinationChannelL, WEBAUDIO_BLOCK_SIZE);
+    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
+        // 1 -> 1 -> 1
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+    } else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
+        // 2 -> 4 -> 2 ("True" stereo)
+        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+        // Process left virtual source
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+        // Process right virtual source
+        m_convolvers[2]->process(sourceBusR, tempChannelL);
+        m_convolvers[3]->process(sourceBusR, tempChannelR);
+
+        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
+        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
+    } else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
+        // 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
+        // This is an inefficient use of a four-channel impulse response, but we should handle the case.
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+        // Process left virtual source
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+        // Process right virtual source
+        m_convolvers[2]->process(sourceBusL, tempChannelL);
+        m_convolvers[3]->process(sourceBusL, tempChannelR);
+
+        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
+        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
+    } else {
+        // Handle gracefully any unexpected / unsupported matrixing
+        // FIXME: add code for 5.1 support...
+        destinationBus->SetNull(destinationBus->GetDuration());
+    }
+}
+
+} // namespace WebCore