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+/*
+ * 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 "config.h"
+
+#if ENABLE(WEB_AUDIO)
+
+#include "Reverb.h"
+
+#include "AudioBus.h"
+#include "AudioFileReader.h"
+#include "ReverbConvolver.h"
+#include <math.h>
+#include <wtf/MathExtras.h>
+#include <wtf/OwnPtr.h>
+#include <wtf/PassOwnPtr.h>
+
+#if OS(DARWIN)
+using namespace std;
+#endif
+
+namespace WebCore {
+
+// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
+const double GainCalibration = -58.0;
+
+// A minimum power value to when normalizing a silent (or very quiet) impulse response
+const double MinPower = 0.000125;
+
+static double calculateNormalizationScale(AudioBus* response)
+{
+ // Normalize by RMS power
+ size_t numberOfChannels = response->numberOfChannels();
+ size_t length = response->length();
+
+ double power = 0.0;
+
+ for (size_t i = 0; i < numberOfChannels; ++i) {
+ int n = length;
+ float* p = response->channel(i)->data();
+
+ while (n--) {
+ float sample = *p++;
+ power += sample * sample;
+ }
+ }
+
+ power = sqrt(power / (numberOfChannels * length));
+
+ // Protect against accidental overload
+ if (isinf(power) || isnan(power) || power < MinPower)
+ power = MinPower;
+
+ double scale = 1.0 / power;
+
+ scale *= pow(10.0, GainCalibration * 0.05); // calibrate to make perceived volume same as unprocessed
+
+ // True-stereo compensation
+ if (response->numberOfChannels() == 4)
+ scale *= 0.5;
+
+ return scale;
+}
+
+Reverb::Reverb(AudioBus* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+{
+ double scale = calculateNormalizationScale(impulseResponse);
+ if (scale)
+ impulseResponse->scale(scale);
+
+ initialize(impulseResponse, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
+
+ // Undo scaling since this shouldn't be a destructive operation on impulseResponse
+ if (scale)
+ impulseResponse->scale(1.0 / scale);
+}
+
+void Reverb::initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+{
+ m_impulseResponseLength = impulseResponseBuffer->length();
+
+ // The reverb can handle a mono impulse response and still do stereo processing
+ size_t numResponseChannels = impulseResponseBuffer->numberOfChannels();
+ m_convolvers.reserveCapacity(numberOfChannels);
+
+ int convolverRenderPhase = 0;
+ for (size_t i = 0; i < numResponseChannels; ++i) {
+ AudioChannel* channel = impulseResponseBuffer->channel(i);
+
+ OwnPtr<ReverbConvolver> convolver = adoptPtr(new ReverbConvolver(channel, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
+ m_convolvers.append(convolver.release());
+
+ convolverRenderPhase += renderSliceSize;
+ }
+
+ // 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 = new AudioBus(2, MaxFrameSize);
+}
+
+void Reverb::process(AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess)
+{
+ // 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->numberOfChannels() > 0 && destinationBus->numberOfChannels() > 0
+ && framesToProcess <= MaxFrameSize && framesToProcess <= sourceBus->length() && framesToProcess <= destinationBus->length();
+
+ ASSERT(isSafeToProcess);
+ if (!isSafeToProcess)
+ return;
+
+ // For now only handle mono or stereo output
+ if (destinationBus->numberOfChannels() > 2) {
+ destinationBus->zero();
+ return;
+ }
+
+ AudioChannel* destinationChannelL = destinationBus->channel(0);
+ AudioChannel* sourceChannelL = sourceBus->channel(0);
+
+ // Handle input -> output matrixing...
+ size_t numInputChannels = sourceBus->numberOfChannels();
+ size_t numOutputChannels = destinationBus->numberOfChannels();
+ size_t numReverbChannels = m_convolvers.size();
+
+ if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
+ // 2 -> 2 -> 2
+ AudioChannel* sourceChannelR = sourceBus->channel(1);
+ AudioChannel* destinationChannelR = destinationBus->channel(1);
+ m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
+ m_convolvers[1]->process(sourceChannelR, destinationChannelR, framesToProcess);
+ } else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
+ // 1 -> 2 -> 2
+ for (int i = 0; i < 2; ++i) {
+ AudioChannel* destinationChannel = destinationBus->channel(i);
+ m_convolvers[i]->process(sourceChannelL, destinationChannel, framesToProcess);
+ }
+ } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
+ // 1 -> 1 -> 2
+ m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
+
+ // simply copy L -> R
+ AudioChannel* destinationChannelR = destinationBus->channel(1);
+ bool isCopySafe = destinationChannelL->data() && destinationChannelR->data() && destinationChannelL->length() >= framesToProcess && destinationChannelR->length() >= framesToProcess;
+ ASSERT(isCopySafe);
+ if (!isCopySafe)
+ return;
+ memcpy(destinationChannelR->data(), destinationChannelL->data(), sizeof(float) * framesToProcess);
+ } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
+ // 1 -> 1 -> 1
+ m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
+ } else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
+ // 2 -> 4 -> 2 ("True" stereo)
+ AudioChannel* sourceChannelR = sourceBus->channel(1);
+ AudioChannel* destinationChannelR = destinationBus->channel(1);
+
+ AudioChannel* tempChannelL = m_tempBuffer->channel(0);
+ AudioChannel* tempChannelR = m_tempBuffer->channel(1);
+
+ // Process left virtual source
+ m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
+ m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
+
+ // Process right virtual source
+ m_convolvers[2]->process(sourceChannelR, tempChannelL, framesToProcess);
+ m_convolvers[3]->process(sourceChannelR, tempChannelR, framesToProcess);
+
+ destinationBus->sumFrom(*m_tempBuffer);
+ } 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.
+ AudioChannel* destinationChannelR = destinationBus->channel(1);
+
+ AudioChannel* tempChannelL = m_tempBuffer->channel(0);
+ AudioChannel* tempChannelR = m_tempBuffer->channel(1);
+
+ // Process left virtual source
+ m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
+ m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
+
+ // Process right virtual source
+ m_convolvers[2]->process(sourceChannelL, tempChannelL, framesToProcess);
+ m_convolvers[3]->process(sourceChannelL, tempChannelR, framesToProcess);
+
+ destinationBus->sumFrom(*m_tempBuffer);
+ } else {
+ // Handle gracefully any unexpected / unsupported matrixing
+ // FIXME: add code for 5.1 support...
+ destinationBus->zero();
+ }
+}
+
+void Reverb::reset()
+{
+ for (size_t i = 0; i < m_convolvers.size(); ++i)
+ m_convolvers[i]->reset();
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)