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7f070cfa1c
This code was copied from Blink SVN revision 153183. --HG-- extra : rebase_source : 968f69846f5c532eda3580c8ce2176e2e6b5fc63
315 lines
14 KiB
C++
315 lines
14 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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* 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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*
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* THIS SOFTWARE IS PROVIDED BY APPLE INC. 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 INC. 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 ON
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* 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 THIS
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* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "config.h"
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#if ENABLE(WEB_AUDIO)
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#include "core/platform/audio/HRTFPanner.h"
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#include <algorithm>
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#include "core/platform/audio/AudioBus.h"
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#include "core/platform/audio/FFTConvolver.h"
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#include "core/platform/audio/HRTFDatabase.h"
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#include <wtf/MathExtras.h>
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#include <wtf/RefPtr.h>
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using namespace std;
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namespace WebCore {
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// The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds).
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// We ASSERT the delay values used in process() with this value.
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const double MaxDelayTimeSeconds = 0.002;
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const int UninitializedAzimuth = -1;
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const unsigned RenderingQuantum = 128;
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HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader)
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: Panner(PanningModelHRTF)
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, m_databaseLoader(databaseLoader)
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, m_sampleRate(sampleRate)
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, m_crossfadeSelection(CrossfadeSelection1)
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, m_azimuthIndex1(UninitializedAzimuth)
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, m_elevation1(0)
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, m_azimuthIndex2(UninitializedAzimuth)
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, m_elevation2(0)
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, m_crossfadeX(0)
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, m_crossfadeIncr(0)
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, m_convolverL1(fftSizeForSampleRate(sampleRate))
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, m_convolverR1(fftSizeForSampleRate(sampleRate))
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, m_convolverL2(fftSizeForSampleRate(sampleRate))
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, m_convolverR2(fftSizeForSampleRate(sampleRate))
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, m_delayLineL(MaxDelayTimeSeconds, sampleRate)
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, m_delayLineR(MaxDelayTimeSeconds, sampleRate)
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, m_tempL1(RenderingQuantum)
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, m_tempR1(RenderingQuantum)
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, m_tempL2(RenderingQuantum)
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, m_tempR2(RenderingQuantum)
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{
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ASSERT(databaseLoader);
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}
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HRTFPanner::~HRTFPanner()
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{
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}
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size_t HRTFPanner::fftSizeForSampleRate(float sampleRate)
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{
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// The HRTF impulse responses (loaded as audio resources) are 512 sample-frames @44.1KHz.
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// Currently, we truncate the impulse responses to half this size, but an FFT-size of twice impulse response size is needed (for convolution).
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// So for sample rates around 44.1KHz an FFT size of 512 is good. We double the FFT-size only for sample rates at least double this.
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ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0);
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return (sampleRate < 88200.0) ? 512 : 1024;
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}
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void HRTFPanner::reset()
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{
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m_convolverL1.reset();
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m_convolverR1.reset();
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m_convolverL2.reset();
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m_convolverR2.reset();
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m_delayLineL.reset();
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m_delayLineR.reset();
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}
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int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend)
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{
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// Convert the azimuth angle from the range -180 -> +180 into the range 0 -> 360.
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// The azimuth index may then be calculated from this positive value.
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if (azimuth < 0)
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azimuth += 360.0;
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HRTFDatabase* database = m_databaseLoader->database();
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ASSERT(database);
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int numberOfAzimuths = database->numberOfAzimuths();
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const double angleBetweenAzimuths = 360.0 / numberOfAzimuths;
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// Calculate the azimuth index and the blend (0 -> 1) for interpolation.
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double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths;
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int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat);
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azimuthBlend = desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex);
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// We don't immediately start using this azimuth index, but instead approach this index from the last index we rendered at.
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// This minimizes the clicks and graininess for moving sources which occur otherwise.
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desiredAzimuthIndex = max(0, desiredAzimuthIndex);
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desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex);
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return desiredAzimuthIndex;
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}
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void HRTFPanner::pan(double desiredAzimuth, double elevation, const AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess)
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{
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unsigned numInputChannels = inputBus ? inputBus->numberOfChannels() : 0;
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bool isInputGood = inputBus && numInputChannels >= 1 && numInputChannels <= 2;
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ASSERT(isInputGood);
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bool isOutputGood = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length();
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ASSERT(isOutputGood);
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if (!isInputGood || !isOutputGood) {
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if (outputBus)
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outputBus->zero();
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return;
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}
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HRTFDatabase* database = m_databaseLoader->database();
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ASSERT(database);
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if (!database) {
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outputBus->zero();
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return;
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}
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// IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth.
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double azimuth = -desiredAzimuth;
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bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0;
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ASSERT(isAzimuthGood);
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if (!isAzimuthGood) {
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outputBus->zero();
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return;
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}
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// Normally, we'll just be dealing with mono sources.
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// If we have a stereo input, implement stereo panning with left source processed by left HRTF, and right source by right HRTF.
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const AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft);
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const AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0;
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// Get source and destination pointers.
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const float* sourceL = inputChannelL->data();
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const float* sourceR = numInputChannels > 1 ? inputChannelR->data() : sourceL;
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float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->mutableData();
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float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->mutableData();
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double azimuthBlend;
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int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
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// Initially snap azimuth and elevation values to first values encountered.
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if (m_azimuthIndex1 == UninitializedAzimuth) {
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m_azimuthIndex1 = desiredAzimuthIndex;
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m_elevation1 = elevation;
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}
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if (m_azimuthIndex2 == UninitializedAzimuth) {
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m_azimuthIndex2 = desiredAzimuthIndex;
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m_elevation2 = elevation;
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}
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// Cross-fade / transition over a period of around 45 milliseconds.
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// This is an empirical value tuned to be a reasonable trade-off between
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// smoothness and speed.
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const double fadeFrames = sampleRate() <= 48000 ? 2048 : 4096;
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// Check for azimuth and elevation changes, initiating a cross-fade if needed.
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if (!m_crossfadeX && m_crossfadeSelection == CrossfadeSelection1) {
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if (desiredAzimuthIndex != m_azimuthIndex1 || elevation != m_elevation1) {
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// Cross-fade from 1 -> 2
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m_crossfadeIncr = 1 / fadeFrames;
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m_azimuthIndex2 = desiredAzimuthIndex;
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m_elevation2 = elevation;
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}
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}
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if (m_crossfadeX == 1 && m_crossfadeSelection == CrossfadeSelection2) {
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if (desiredAzimuthIndex != m_azimuthIndex2 || elevation != m_elevation2) {
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// Cross-fade from 2 -> 1
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m_crossfadeIncr = -1 / fadeFrames;
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m_azimuthIndex1 = desiredAzimuthIndex;
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m_elevation1 = elevation;
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}
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}
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// This algorithm currently requires that we process in power-of-two size chunks at least RenderingQuantum.
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ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
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ASSERT(framesToProcess >= RenderingQuantum);
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const unsigned framesPerSegment = RenderingQuantum;
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const unsigned numberOfSegments = framesToProcess / framesPerSegment;
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for (unsigned segment = 0; segment < numberOfSegments; ++segment) {
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// Get the HRTFKernels and interpolated delays.
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HRTFKernel* kernelL1;
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HRTFKernel* kernelR1;
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HRTFKernel* kernelL2;
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HRTFKernel* kernelR2;
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double frameDelayL1;
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double frameDelayR1;
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double frameDelayL2;
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double frameDelayR2;
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database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex1, m_elevation1, kernelL1, kernelR1, frameDelayL1, frameDelayR1);
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database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex2, m_elevation2, kernelL2, kernelR2, frameDelayL2, frameDelayR2);
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bool areKernelsGood = kernelL1 && kernelR1 && kernelL2 && kernelR2;
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ASSERT(areKernelsGood);
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if (!areKernelsGood) {
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outputBus->zero();
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return;
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}
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ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds && frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
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ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds && frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
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// Crossfade inter-aural delays based on transitions.
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double frameDelayL = (1 - m_crossfadeX) * frameDelayL1 + m_crossfadeX * frameDelayL2;
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double frameDelayR = (1 - m_crossfadeX) * frameDelayR1 + m_crossfadeX * frameDelayR2;
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// Calculate the source and destination pointers for the current segment.
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unsigned offset = segment * framesPerSegment;
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const float* segmentSourceL = sourceL + offset;
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const float* segmentSourceR = sourceR + offset;
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float* segmentDestinationL = destinationL + offset;
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float* segmentDestinationR = destinationR + offset;
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// First run through delay lines for inter-aural time difference.
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m_delayLineL.setDelayFrames(frameDelayL);
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m_delayLineR.setDelayFrames(frameDelayR);
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m_delayLineL.process(segmentSourceL, segmentDestinationL, framesPerSegment);
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m_delayLineR.process(segmentSourceR, segmentDestinationR, framesPerSegment);
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bool needsCrossfading = m_crossfadeIncr;
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// Have the convolvers render directly to the final destination if we're not cross-fading.
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float* convolutionDestinationL1 = needsCrossfading ? m_tempL1.data() : segmentDestinationL;
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float* convolutionDestinationR1 = needsCrossfading ? m_tempR1.data() : segmentDestinationR;
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float* convolutionDestinationL2 = needsCrossfading ? m_tempL2.data() : segmentDestinationL;
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float* convolutionDestinationR2 = needsCrossfading ? m_tempR2.data() : segmentDestinationR;
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// Now do the convolutions.
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// Note that we avoid doing convolutions on both sets of convolvers if we're not currently cross-fading.
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if (m_crossfadeSelection == CrossfadeSelection1 || needsCrossfading) {
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m_convolverL1.process(kernelL1->fftFrame(), segmentDestinationL, convolutionDestinationL1, framesPerSegment);
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m_convolverR1.process(kernelR1->fftFrame(), segmentDestinationR, convolutionDestinationR1, framesPerSegment);
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}
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if (m_crossfadeSelection == CrossfadeSelection2 || needsCrossfading) {
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m_convolverL2.process(kernelL2->fftFrame(), segmentDestinationL, convolutionDestinationL2, framesPerSegment);
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m_convolverR2.process(kernelR2->fftFrame(), segmentDestinationR, convolutionDestinationR2, framesPerSegment);
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}
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if (needsCrossfading) {
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// Apply linear cross-fade.
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float x = m_crossfadeX;
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float incr = m_crossfadeIncr;
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for (unsigned i = 0; i < framesPerSegment; ++i) {
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segmentDestinationL[i] = (1 - x) * convolutionDestinationL1[i] + x * convolutionDestinationL2[i];
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segmentDestinationR[i] = (1 - x) * convolutionDestinationR1[i] + x * convolutionDestinationR2[i];
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x += incr;
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}
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// Update cross-fade value from local.
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m_crossfadeX = x;
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if (m_crossfadeIncr > 0 && fabs(m_crossfadeX - 1) < m_crossfadeIncr) {
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// We've fully made the crossfade transition from 1 -> 2.
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m_crossfadeSelection = CrossfadeSelection2;
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m_crossfadeX = 1;
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m_crossfadeIncr = 0;
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} else if (m_crossfadeIncr < 0 && fabs(m_crossfadeX) < -m_crossfadeIncr) {
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// We've fully made the crossfade transition from 2 -> 1.
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m_crossfadeSelection = CrossfadeSelection1;
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m_crossfadeX = 0;
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m_crossfadeIncr = 0;
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}
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}
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}
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}
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double HRTFPanner::tailTime() const
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{
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// Because HRTFPanner is implemented with a DelayKernel and a FFTConvolver, the tailTime of the HRTFPanner
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// is the sum of the tailTime of the DelayKernel and the tailTime of the FFTConvolver, which is MaxDelayTimeSeconds
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// and fftSize() / 2, respectively.
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return MaxDelayTimeSeconds + (fftSize() / 2) / static_cast<double>(sampleRate());
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}
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double HRTFPanner::latencyTime() const
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{
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// The latency of a FFTConvolver is also fftSize() / 2, and is in addition to its tailTime of the
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// same value.
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return (fftSize() / 2) / static_cast<double>(sampleRate());
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}
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} // namespace WebCore
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#endif // ENABLE(WEB_AUDIO)
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