content/media/webaudio/blink/Reverb.cpp

changeset 0
6474c204b198
     1.1 --- /dev/null	Thu Jan 01 00:00:00 1970 +0000
     1.2 +++ b/content/media/webaudio/blink/Reverb.cpp	Wed Dec 31 06:09:35 2014 +0100
     1.3 @@ -0,0 +1,254 @@
     1.4 +/*
     1.5 + * Copyright (C) 2010 Google Inc. All rights reserved.
     1.6 + *
     1.7 + * Redistribution and use in source and binary forms, with or without
     1.8 + * modification, are permitted provided that the following conditions
     1.9 + * are met:
    1.10 + *
    1.11 + * 1.  Redistributions of source code must retain the above copyright
    1.12 + *     notice, this list of conditions and the following disclaimer.
    1.13 + * 2.  Redistributions in binary form must reproduce the above copyright
    1.14 + *     notice, this list of conditions and the following disclaimer in the
    1.15 + *     documentation and/or other materials provided with the distribution.
    1.16 + * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
    1.17 + *     its contributors may be used to endorse or promote products derived
    1.18 + *     from this software without specific prior written permission.
    1.19 + *
    1.20 + * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
    1.21 + * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
    1.22 + * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
    1.23 + * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
    1.24 + * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
    1.25 + * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
    1.26 + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
    1.27 + * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
    1.28 + * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
    1.29 + * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
    1.30 + */
    1.31 +
    1.32 +#include "Reverb.h"
    1.33 +#include "ReverbConvolverStage.h"
    1.34 +
    1.35 +#include <math.h>
    1.36 +#include "ReverbConvolver.h"
    1.37 +#include "mozilla/FloatingPoint.h"
    1.38 +
    1.39 +using namespace mozilla;
    1.40 +
    1.41 +namespace WebCore {
    1.42 +
    1.43 +// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
    1.44 +const float GainCalibration = -58;
    1.45 +const float GainCalibrationSampleRate = 44100;
    1.46 +
    1.47 +// A minimum power value to when normalizing a silent (or very quiet) impulse response
    1.48 +const float MinPower = 0.000125f;
    1.49 +
    1.50 +static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
    1.51 +{
    1.52 +    // Normalize by RMS power
    1.53 +    size_t numberOfChannels = response->GetChannels();
    1.54 +
    1.55 +    float power = 0;
    1.56 +
    1.57 +    for (size_t i = 0; i < numberOfChannels; ++i) {
    1.58 +        float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
    1.59 +        power += channelPower;
    1.60 +    }
    1.61 +
    1.62 +    power = sqrt(power / (numberOfChannels * aLength));
    1.63 +
    1.64 +    // Protect against accidental overload
    1.65 +    if (!IsFinite(power) || IsNaN(power) || power < MinPower)
    1.66 +        power = MinPower;
    1.67 +
    1.68 +    float scale = 1 / power;
    1.69 +
    1.70 +    scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
    1.71 +
    1.72 +    // Scale depends on sample-rate.
    1.73 +    if (sampleRate)
    1.74 +        scale *= GainCalibrationSampleRate / sampleRate;
    1.75 +
    1.76 +    // True-stereo compensation
    1.77 +    if (response->GetChannels() == 4)
    1.78 +        scale *= 0.5f;
    1.79 +
    1.80 +    return scale;
    1.81 +}
    1.82 +
    1.83 +Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
    1.84 +{
    1.85 +    float scale = 1;
    1.86 +
    1.87 +    nsAutoTArray<const float*,4> irChannels;
    1.88 +    for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
    1.89 +        irChannels.AppendElement(impulseResponse->GetData(i));
    1.90 +    }
    1.91 +    nsAutoTArray<float,1024> tempBuf;
    1.92 +
    1.93 +    if (normalize) {
    1.94 +        scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
    1.95 +
    1.96 +        if (scale) {
    1.97 +            tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
    1.98 +            for (uint32_t i = 0; i < irChannels.Length(); ++i) {
    1.99 +                float* buf = &tempBuf[i*impulseResponseBufferLength];
   1.100 +                AudioBufferCopyWithScale(irChannels[i], scale, buf,
   1.101 +                                         impulseResponseBufferLength);
   1.102 +                irChannels[i] = buf;
   1.103 +            }
   1.104 +        }
   1.105 +    }
   1.106 +
   1.107 +    initialize(irChannels, impulseResponseBufferLength, renderSliceSize,
   1.108 +               maxFFTSize, numberOfChannels, useBackgroundThreads);
   1.109 +}
   1.110 +
   1.111 +size_t Reverb::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
   1.112 +{
   1.113 +    size_t amount = aMallocSizeOf(this);
   1.114 +    amount += m_convolvers.SizeOfExcludingThis(aMallocSizeOf);
   1.115 +    for (size_t i = 0; i < m_convolvers.Length(); i++) {
   1.116 +        if (m_convolvers[i]) {
   1.117 +            amount += m_convolvers[i]->sizeOfIncludingThis(aMallocSizeOf);
   1.118 +        }
   1.119 +    }
   1.120 +
   1.121 +    amount += m_tempBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
   1.122 +    return amount;
   1.123 +}
   1.124 +
   1.125 +
   1.126 +void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
   1.127 +                        size_t impulseResponseBufferLength, size_t renderSliceSize,
   1.128 +                        size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
   1.129 +{
   1.130 +    m_impulseResponseLength = impulseResponseBufferLength;
   1.131 +
   1.132 +    // The reverb can handle a mono impulse response and still do stereo processing
   1.133 +    size_t numResponseChannels = impulseResponseBuffer.Length();
   1.134 +    m_convolvers.SetCapacity(numberOfChannels);
   1.135 +
   1.136 +    int convolverRenderPhase = 0;
   1.137 +    for (size_t i = 0; i < numResponseChannels; ++i) {
   1.138 +        const float* channel = impulseResponseBuffer[i];
   1.139 +        size_t length = impulseResponseBufferLength;
   1.140 +
   1.141 +        nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
   1.142 +        m_convolvers.AppendElement(convolver.forget());
   1.143 +
   1.144 +        convolverRenderPhase += renderSliceSize;
   1.145 +    }
   1.146 +
   1.147 +    // For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
   1.148 +    // It can be bad to allocate memory in a real-time thread.
   1.149 +    if (numResponseChannels == 4) {
   1.150 +        AllocateAudioBlock(2, &m_tempBuffer);
   1.151 +        WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
   1.152 +    }
   1.153 +}
   1.154 +
   1.155 +void Reverb::process(const AudioChunk* sourceBus, AudioChunk* destinationBus, size_t framesToProcess)
   1.156 +{
   1.157 +    // Do a fairly comprehensive sanity check.
   1.158 +    // If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
   1.159 +    bool isSafeToProcess = sourceBus && destinationBus && sourceBus->mChannelData.Length() > 0 && destinationBus->mChannelData.Length() > 0
   1.160 +        && framesToProcess <= MaxFrameSize && framesToProcess <= size_t(sourceBus->mDuration) && framesToProcess <= size_t(destinationBus->mDuration);
   1.161 +
   1.162 +    MOZ_ASSERT(isSafeToProcess);
   1.163 +    if (!isSafeToProcess)
   1.164 +        return;
   1.165 +
   1.166 +    // For now only handle mono or stereo output
   1.167 +    MOZ_ASSERT(destinationBus->mChannelData.Length() <= 2);
   1.168 +
   1.169 +    float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
   1.170 +    const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
   1.171 +
   1.172 +    // Handle input -> output matrixing...
   1.173 +    size_t numInputChannels = sourceBus->mChannelData.Length();
   1.174 +    size_t numOutputChannels = destinationBus->mChannelData.Length();
   1.175 +    size_t numReverbChannels = m_convolvers.Length();
   1.176 +
   1.177 +    if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
   1.178 +        // 2 -> 2 -> 2
   1.179 +        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
   1.180 +        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
   1.181 +        m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
   1.182 +        m_convolvers[1]->process(sourceBusR, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
   1.183 +    } else  if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
   1.184 +        // 1 -> 2 -> 2
   1.185 +        for (int i = 0; i < 2; ++i) {
   1.186 +            float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
   1.187 +            m_convolvers[i]->process(sourceBusL, sourceBus->mDuration, destinationChannel, destinationBus->mDuration, framesToProcess);
   1.188 +        }
   1.189 +    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
   1.190 +        // 1 -> 1 -> 2
   1.191 +        m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
   1.192 +
   1.193 +        // simply copy L -> R
   1.194 +        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
   1.195 +        bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->mDuration) >= framesToProcess && size_t(destinationBus->mDuration) >= framesToProcess;
   1.196 +        MOZ_ASSERT(isCopySafe);
   1.197 +        if (!isCopySafe)
   1.198 +            return;
   1.199 +        PodCopy(destinationChannelR, destinationChannelL, framesToProcess);
   1.200 +    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
   1.201 +        // 1 -> 1 -> 1
   1.202 +        m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
   1.203 +    } else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
   1.204 +        // 2 -> 4 -> 2 ("True" stereo)
   1.205 +        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
   1.206 +        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
   1.207 +
   1.208 +        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
   1.209 +        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
   1.210 +
   1.211 +        // Process left virtual source
   1.212 +        m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
   1.213 +        m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
   1.214 +
   1.215 +        // Process right virtual source
   1.216 +        m_convolvers[2]->process(sourceBusR, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
   1.217 +        m_convolvers[3]->process(sourceBusR, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
   1.218 +
   1.219 +        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
   1.220 +        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
   1.221 +    } else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
   1.222 +        // 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
   1.223 +        // This is an inefficient use of a four-channel impulse response, but we should handle the case.
   1.224 +        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
   1.225 +
   1.226 +        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
   1.227 +        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
   1.228 +
   1.229 +        // Process left virtual source
   1.230 +        m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
   1.231 +        m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
   1.232 +
   1.233 +        // Process right virtual source
   1.234 +        m_convolvers[2]->process(sourceBusL, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
   1.235 +        m_convolvers[3]->process(sourceBusL, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
   1.236 +
   1.237 +        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
   1.238 +        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
   1.239 +    } else {
   1.240 +        // Handle gracefully any unexpected / unsupported matrixing
   1.241 +        // FIXME: add code for 5.1 support...
   1.242 +        destinationBus->SetNull(destinationBus->mDuration);
   1.243 +    }
   1.244 +}
   1.245 +
   1.246 +void Reverb::reset()
   1.247 +{
   1.248 +    for (size_t i = 0; i < m_convolvers.Length(); ++i)
   1.249 +        m_convolvers[i]->reset();
   1.250 +}
   1.251 +
   1.252 +size_t Reverb::latencyFrames() const
   1.253 +{
   1.254 +    return !m_convolvers.IsEmpty() ? m_convolvers[0]->latencyFrames() : 0;
   1.255 +}
   1.256 +
   1.257 +} // namespace WebCore

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