summaryrefslogtreecommitdiffstats
path: root/dom/media/webaudio/blink/Reverb.cpp
diff options
context:
space:
mode:
Diffstat (limited to 'dom/media/webaudio/blink/Reverb.cpp')
-rw-r--r--dom/media/webaudio/blink/Reverb.cpp277
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
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-12 05:52:15 +0000