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242 lines
12 KiB
C++
242 lines
12 KiB
C++
/*
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* Copyright (C) 2010 Google Inc. All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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*
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
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* its contributors may be used to endorse or promote products derived
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* from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
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* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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* DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
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* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
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* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
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* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "Reverb.h"
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#include <math.h>
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#include "ReverbConvolver.h"
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#include "mozilla/FloatingPoint.h"
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using namespace mozilla;
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namespace WebCore {
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// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
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const float GainCalibration = -58;
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const float GainCalibrationSampleRate = 44100;
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// A minimum power value to when normalizing a silent (or very quiet) impulse response
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const float MinPower = 0.000125f;
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static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
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{
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// Normalize by RMS power
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size_t numberOfChannels = response->GetChannels();
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float power = 0;
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for (size_t i = 0; i < numberOfChannels; ++i) {
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float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
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power += channelPower;
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}
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power = sqrt(power / (numberOfChannels * aLength));
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// Protect against accidental overload
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if (!IsFinite(power) || IsNaN(power) || power < MinPower)
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power = MinPower;
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float scale = 1 / power;
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scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
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// Scale depends on sample-rate.
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if (sampleRate)
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scale *= GainCalibrationSampleRate / sampleRate;
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// True-stereo compensation
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if (response->GetChannels() == 4)
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scale *= 0.5f;
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return scale;
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}
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Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
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{
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float scale = 1;
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nsAutoTArray<const float*,4> irChannels;
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for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
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irChannels.AppendElement(impulseResponse->GetData(i));
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}
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nsAutoTArray<float,1024> tempBuf;
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if (normalize) {
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scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
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if (scale) {
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tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
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for (uint32_t i = 0; i < irChannels.Length(); ++i) {
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float* buf = &tempBuf[i*impulseResponseBufferLength];
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AudioBufferCopyWithScale(irChannels[i], scale, buf,
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impulseResponseBufferLength);
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irChannels[i] = buf;
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}
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}
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}
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initialize(irChannels, impulseResponseBufferLength, renderSliceSize,
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maxFFTSize, numberOfChannels, useBackgroundThreads);
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}
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void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
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size_t impulseResponseBufferLength, size_t renderSliceSize,
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size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
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{
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m_impulseResponseLength = impulseResponseBufferLength;
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// The reverb can handle a mono impulse response and still do stereo processing
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size_t numResponseChannels = impulseResponseBuffer.Length();
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m_convolvers.SetCapacity(numberOfChannels);
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int convolverRenderPhase = 0;
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for (size_t i = 0; i < numResponseChannels; ++i) {
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const float* channel = impulseResponseBuffer[i];
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size_t length = impulseResponseBufferLength;
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nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
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m_convolvers.AppendElement(convolver.forget());
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convolverRenderPhase += renderSliceSize;
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}
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// For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
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// It can be bad to allocate memory in a real-time thread.
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if (numResponseChannels == 4) {
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AllocateAudioBlock(2, &m_tempBuffer);
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WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
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}
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}
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void Reverb::process(const AudioChunk* sourceBus, AudioChunk* destinationBus, size_t framesToProcess)
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{
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// Do a fairly comprehensive sanity check.
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// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
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bool isSafeToProcess = sourceBus && destinationBus && sourceBus->mChannelData.Length() > 0 && destinationBus->mChannelData.Length() > 0
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&& framesToProcess <= MaxFrameSize && framesToProcess <= size_t(sourceBus->mDuration) && framesToProcess <= size_t(destinationBus->mDuration);
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MOZ_ASSERT(isSafeToProcess);
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if (!isSafeToProcess)
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return;
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// For now only handle mono or stereo output
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if (destinationBus->mChannelData.Length() > 2) {
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destinationBus->SetNull(destinationBus->mDuration);
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return;
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}
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float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
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const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
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// Handle input -> output matrixing...
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size_t numInputChannels = sourceBus->mChannelData.Length();
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size_t numOutputChannels = destinationBus->mChannelData.Length();
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size_t numReverbChannels = m_convolvers.Length();
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if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
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// 2 -> 2 -> 2
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const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
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float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
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m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
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m_convolvers[1]->process(sourceBusR, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
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} else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
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// 1 -> 2 -> 2
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for (int i = 0; i < 2; ++i) {
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float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
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m_convolvers[i]->process(sourceBusL, sourceBus->mDuration, destinationChannel, destinationBus->mDuration, framesToProcess);
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}
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} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
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// 1 -> 1 -> 2
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m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
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// simply copy L -> R
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float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
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bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->mDuration) >= framesToProcess && size_t(destinationBus->mDuration) >= framesToProcess;
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MOZ_ASSERT(isCopySafe);
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if (!isCopySafe)
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return;
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PodCopy(destinationChannelR, destinationChannelL, framesToProcess);
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} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
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// 1 -> 1 -> 1
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m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
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} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
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// 2 -> 4 -> 2 ("True" stereo)
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const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
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float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
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float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
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float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
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// Process left virtual source
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m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
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m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
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// Process right virtual source
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m_convolvers[2]->process(sourceBusR, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
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m_convolvers[3]->process(sourceBusR, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
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AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
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AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
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} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
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// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
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// This is an inefficient use of a four-channel impulse response, but we should handle the case.
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float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
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float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
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float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
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// Process left virtual source
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m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
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m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
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// Process right virtual source
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m_convolvers[2]->process(sourceBusL, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
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m_convolvers[3]->process(sourceBusL, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
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AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
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AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
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} else {
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// Handle gracefully any unexpected / unsupported matrixing
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// FIXME: add code for 5.1 support...
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destinationBus->SetNull(destinationBus->mDuration);
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}
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}
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void Reverb::reset()
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{
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for (size_t i = 0; i < m_convolvers.Length(); ++i)
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m_convolvers[i]->reset();
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}
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size_t Reverb::latencyFrames() const
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{
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return !m_convolvers.IsEmpty() ? m_convolvers[0]->latencyFrames() : 0;
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}
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} // namespace WebCore
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