Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 277 insertions, 0 deletions

diff --git a/dom/media/webaudio/blink/Reverb.cpp b/dom/media/webaudio/blink/Reverb.cpp
new file mode 100644
index 0000000000..bd56a5af27
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.cpp
@@ -0,0 +1,277 @@
+/*
+ * 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 = 0.00125f;
+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(const nsTArray<const float*>& response,
+                                         size_t aLength, float sampleRate) {
+  // Normalize by RMS power
+  size_t numberOfChannels = response.Length();
+
+  float power = 0;
+
+  for (size_t i = 0; i < numberOfChannels; ++i) {
+    float channelPower = AudioBufferSumOfSquares(response[i], aLength);
+    power += channelPower;
+  }
+
+  power = sqrt(power / (numberOfChannels * aLength));
+
+  // Protect against accidental overload
+  if (!std::isfinite(power) || std::isnan(power) || power < MinPower)
+    power = MinPower;
+
+  float scale = 1 / power;
+
+  scale *= GainCalibration;  // calibrate to make perceived volume same as
+                             // unprocessed
+
+  // Scale depends on sample-rate.
+  if (sampleRate) scale *= GainCalibrationSampleRate / sampleRate;
+
+  // True-stereo compensation
+  if (numberOfChannels == 4) scale *= 0.5f;
+
+  return scale;
+}
+
+Reverb::Reverb(const AudioChunk& impulseResponse, size_t maxFFTSize,
+               bool useBackgroundThreads, bool normalize, float sampleRate,
+               bool* aAllocationFailure) {
+  MOZ_ASSERT(aAllocationFailure);
+  size_t impulseResponseBufferLength = impulseResponse.mDuration;
+  float scale = impulseResponse.mVolume;
+
+  CopyableAutoTArray<const float*, 4> irChannels(
+      impulseResponse.ChannelData<float>());
+  AutoTArray<float, 1024> tempBuf;
+
+  if (normalize) {
+    scale = calculateNormalizationScale(irChannels, impulseResponseBufferLength,
+                                        sampleRate);
+  }
+
+  if (scale != 1.0f) {
+    bool rv = tempBuf.SetLength(
+        irChannels.Length() * impulseResponseBufferLength, mozilla::fallible);
+    *aAllocationFailure = !rv;
+    if (*aAllocationFailure) {
+      return;
+    }
+
+    for (uint32_t i = 0; i < irChannels.Length(); ++i) {
+      float* buf = &tempBuf[i * impulseResponseBufferLength];
+      AudioBufferCopyWithScale(irChannels[i], scale, buf,
+                               impulseResponseBufferLength);
+      irChannels[i] = buf;
+    }
+  }
+
+  *aAllocationFailure = !initialize(irChannels, impulseResponseBufferLength,
+                                    maxFFTSize, 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;
+}
+
+bool Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                        size_t impulseResponseBufferLength, size_t maxFFTSize,
+                        bool useBackgroundThreads) {
+  m_impulseResponseLength = impulseResponseBufferLength;
+
+  // The reverb can handle a mono impulse response and still do stereo
+  // processing
+  size_t numResponseChannels = impulseResponseBuffer.Length();
+  MOZ_ASSERT(numResponseChannels > 0);
+  // The number of convolvers required is at least the number of audio
+  // channels.  Even if there is initially only one audio channel, another
+  // may be added later, and so a second convolver is created now while the
+  // impulse response is available.
+  size_t numConvolvers = std::max<size_t>(numResponseChannels, 2);
+  m_convolvers.SetCapacity(numConvolvers);
+
+  int convolverRenderPhase = 0;
+  for (size_t i = 0; i < numConvolvers; ++i) {
+    size_t channelIndex = i < numResponseChannels ? i : 0;
+    const float* channel = impulseResponseBuffer[channelIndex];
+    size_t length = impulseResponseBufferLength;
+
+    bool allocationFailure;
+    UniquePtr<ReverbConvolver> convolver(
+        new ReverbConvolver(channel, length, maxFFTSize, convolverRenderPhase,
+                            useBackgroundThreads, &allocationFailure));
+    if (allocationFailure) {
+      return false;
+    }
+    m_convolvers.AppendElement(std::move(convolver));
+
+    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);
+  }
+  return true;
+}
+
+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 && numOutputChannels == 1) {
+    // 1 -> 1 -> 1 (Only one of the convolvers is used.)
+    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 {
+    MOZ_ASSERT_UNREACHABLE("Unexpected Reverb configuration");
+    destinationBus->SetNull(destinationBus->GetDuration());
+  }
+}
+
+}  // namespace WebCore