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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 "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
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