1 /* 2 * Copyright (C) 2010, Google Inc. All rights reserved. 3 * 4 * Redistribution and use in source and binary forms, with or without 5 * modification, are permitted provided that the following conditions 6 * are met: 7 * 1. Redistributions of source code must retain the above copyright 8 * notice, this list of conditions and the following disclaimer. 9 * 2. Redistributions in binary form must reproduce the above copyright 10 * notice, this list of conditions and the following disclaimer in the 11 * documentation and/or other materials provided with the distribution. 12 * 13 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY 14 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED 15 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE 16 * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY 17 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES 18 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; 19 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON 20 * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT 21 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS 22 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 23 */ 24 25 #include "config.h" 26 27 #if ENABLE(WEB_AUDIO) 28 29 #include "HRTFPanner.h" 30 31 #include "AudioBus.h" 32 #include "FFTConvolver.h" 33 #include "HRTFDatabase.h" 34 #include "HRTFDatabaseLoader.h" 35 #include <algorithm> 36 #include <wtf/MathExtras.h> 37 #include <wtf/RefPtr.h> 38 39 using namespace std; 40 41 namespace WebCore { 42 43 // The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds). 44 // We ASSERT the delay values used in process() with this value. 45 const double MaxDelayTimeSeconds = 0.002; 46 47 HRTFPanner::HRTFPanner(double sampleRate) 48 : Panner(PanningModelHRTF) 49 , m_sampleRate(sampleRate) 50 , m_isFirstRender(true) 51 , m_azimuthIndex(0) 52 , m_convolverL(fftSizeForSampleRate(sampleRate)) 53 , m_convolverR(fftSizeForSampleRate(sampleRate)) 54 , m_delayLineL(MaxDelayTimeSeconds, sampleRate) 55 , m_delayLineR(MaxDelayTimeSeconds, sampleRate) 56 { 57 } 58 59 HRTFPanner::~HRTFPanner() 60 { 61 } 62 63 size_t HRTFPanner::fftSizeForSampleRate(double sampleRate) 64 { 65 // The HRTF impulse responses (loaded as audio resources) are 512 sample-frames @44.1KHz. 66 // Currently, we truncate the impulse responses to half this size, but an FFT-size of twice impulse response size is needed (for convolution). 67 // So for sample rates around 44.1KHz an FFT size of 512 is good. We double that size for higher sample rates. 68 ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0); 69 return (sampleRate <= 48000.0) ? 512 : 1024; 70 } 71 72 void HRTFPanner::reset() 73 { 74 m_isFirstRender = true; 75 m_convolverL.reset(); 76 m_convolverR.reset(); 77 m_delayLineL.reset(); 78 m_delayLineR.reset(); 79 } 80 81 static bool wrapDistance(int i, int j, int length) 82 { 83 int directDistance = abs(i - j); 84 int indirectDistance = length - directDistance; 85 86 return indirectDistance < directDistance; 87 } 88 89 int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend) 90 { 91 // Convert the azimuth angle from the range -180 -> +180 into the range 0 -> 360. 92 // The azimuth index may then be calculated from this positive value. 93 if (azimuth < 0) 94 azimuth += 360.0; 95 96 HRTFDatabase* database = HRTFDatabaseLoader::defaultHRTFDatabase(); 97 ASSERT(database); 98 99 int numberOfAzimuths = database->numberOfAzimuths(); 100 const double angleBetweenAzimuths = 360.0 / numberOfAzimuths; 101 102 // Calculate the azimuth index and the blend (0 -> 1) for interpolation. 103 double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths; 104 int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat); 105 azimuthBlend = desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex); 106 107 // We don't immediately start using this azimuth index, but instead approach this index from the last index we rendered at. 108 // This minimizes the clicks and graininess for moving sources which occur otherwise. 109 desiredAzimuthIndex = max(0, desiredAzimuthIndex); 110 desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex); 111 return desiredAzimuthIndex; 112 } 113 114 void HRTFPanner::pan(double desiredAzimuth, double elevation, AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess) 115 { 116 unsigned numInputChannels = inputBus ? inputBus->numberOfChannels() : 0; 117 118 bool isInputGood = inputBus && numInputChannels >= 1 && numInputChannels <= 2; 119 ASSERT(isInputGood); 120 121 bool isOutputGood = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length(); 122 ASSERT(isOutputGood); 123 124 if (!isInputGood || !isOutputGood) { 125 if (outputBus) 126 outputBus->zero(); 127 return; 128 } 129 130 // This code only runs as long as the context is alive and after database has been loaded. 131 HRTFDatabase* database = HRTFDatabaseLoader::defaultHRTFDatabase(); 132 ASSERT(database); 133 if (!database) { 134 outputBus->zero(); 135 return; 136 } 137 138 // IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth. 139 double azimuth = -desiredAzimuth; 140 141 bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0; 142 ASSERT(isAzimuthGood); 143 if (!isAzimuthGood) { 144 outputBus->zero(); 145 return; 146 } 147 148 // Normally, we'll just be dealing with mono sources. 149 // If we have a stereo input, implement stereo panning with left source processed by left HRTF, and right source by right HRTF. 150 AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft); 151 AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0; 152 153 // Get source and destination pointers. 154 float* sourceL = inputChannelL->data(); 155 float* sourceR = numInputChannels > 1 ? inputChannelR->data() : sourceL; 156 float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->data(); 157 float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->data(); 158 159 double azimuthBlend; 160 int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend); 161 162 // This algorithm currently requires that we process in power-of-two size chunks at least 128. 163 ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess); 164 ASSERT(framesToProcess >= 128); 165 166 const unsigned framesPerSegment = 128; 167 const unsigned numberOfSegments = framesToProcess / framesPerSegment; 168 169 for (unsigned segment = 0; segment < numberOfSegments; ++segment) { 170 if (m_isFirstRender) { 171 // Snap exactly to desired position (first time and after reset()). 172 m_azimuthIndex = desiredAzimuthIndex; 173 m_isFirstRender = false; 174 } else { 175 // Each segment renders with an azimuth index closer by one to the desired azimuth index. 176 // Because inter-aural time delay is mostly a factor of azimuth and the delay is where the clicks and graininess come from, 177 // we don't bother smoothing the elevations. 178 int numberOfAzimuths = database->numberOfAzimuths(); 179 bool wrap = wrapDistance(m_azimuthIndex, desiredAzimuthIndex, numberOfAzimuths); 180 if (wrap) { 181 if (m_azimuthIndex < desiredAzimuthIndex) 182 m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths; 183 else if (m_azimuthIndex > desiredAzimuthIndex) 184 m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths; 185 } else { 186 if (m_azimuthIndex < desiredAzimuthIndex) 187 m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths; 188 else if (m_azimuthIndex > desiredAzimuthIndex) 189 m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths; 190 } 191 } 192 193 // Get the HRTFKernels and interpolated delays. 194 HRTFKernel* kernelL; 195 HRTFKernel* kernelR; 196 double frameDelayL; 197 double frameDelayR; 198 database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex, elevation, kernelL, kernelR, frameDelayL, frameDelayR); 199 200 ASSERT(kernelL && kernelR); 201 if (!kernelL || !kernelR) { 202 outputBus->zero(); 203 return; 204 } 205 206 ASSERT(frameDelayL / sampleRate() < MaxDelayTimeSeconds && frameDelayR / sampleRate() < MaxDelayTimeSeconds); 207 208 // Calculate the source and destination pointers for the current segment. 209 unsigned offset = segment * framesPerSegment; 210 float* segmentSourceL = sourceL + offset; 211 float* segmentSourceR = sourceR + offset; 212 float* segmentDestinationL = destinationL + offset; 213 float* segmentDestinationR = destinationR + offset; 214 215 // First run through delay lines for inter-aural time difference. 216 m_delayLineL.setDelayFrames(frameDelayL); 217 m_delayLineR.setDelayFrames(frameDelayR); 218 m_delayLineL.process(segmentSourceL, segmentDestinationL, framesPerSegment); 219 m_delayLineR.process(segmentSourceR, segmentDestinationR, framesPerSegment); 220 221 // Now do the convolutions in-place. 222 m_convolverL.process(kernelL->fftFrame(), segmentDestinationL, segmentDestinationL, framesPerSegment); 223 m_convolverR.process(kernelR->fftFrame(), segmentDestinationR, segmentDestinationR, framesPerSegment); 224 } 225 } 226 227 } // namespace WebCore 228 229 #endif // ENABLE(WEB_AUDIO) 230