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 * 8 * 1. Redistributions of source code must retain the above copyright 9 * notice, this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright 11 * notice, this list of conditions and the following disclaimer in the 12 * documentation and/or other materials provided with the distribution. 13 * 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of 14 * its contributors may be used to endorse or promote products derived 15 * from this software without specific prior written permission. 16 * 17 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY 18 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED 19 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE 20 * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY 21 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES 22 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; 23 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND 24 * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT 25 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF 26 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 27 */ 28 29 #include "config.h" 30 31 #if ENABLE(WEB_AUDIO) 32 33 #include "platform/audio/HRTFKernel.h" 34 35 #include "platform/audio/AudioChannel.h" 36 #include "platform/FloatConversion.h" 37 #include "wtf/MathExtras.h" 38 39 using namespace std; 40 41 namespace WebCore { 42 43 // Takes the input AudioChannel as an input impulse response and calculates the average group delay. 44 // This represents the initial delay before the most energetic part of the impulse response. 45 // The sample-frame delay is removed from the impulseP impulse response, and this value is returned. 46 // the length of the passed in AudioChannel must be a power of 2. 47 static float extractAverageGroupDelay(AudioChannel* channel, size_t analysisFFTSize) 48 { 49 ASSERT(channel); 50 51 float* impulseP = channel->mutableData(); 52 53 bool isSizeGood = channel->length() >= analysisFFTSize; 54 ASSERT(isSizeGood); 55 if (!isSizeGood) 56 return 0; 57 58 // Check for power-of-2. 59 ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) == analysisFFTSize); 60 61 FFTFrame estimationFrame(analysisFFTSize); 62 estimationFrame.doFFT(impulseP); 63 64 float frameDelay = narrowPrecisionToFloat(estimationFrame.extractAverageGroupDelay()); 65 estimationFrame.doInverseFFT(impulseP); 66 67 return frameDelay; 68 } 69 70 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate) 71 : m_frameDelay(0) 72 , m_sampleRate(sampleRate) 73 { 74 ASSERT(channel); 75 76 // Determine the leading delay (average group delay) for the response. 77 m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2); 78 79 float* impulseResponse = channel->mutableData(); 80 size_t responseLength = channel->length(); 81 82 // We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution. 83 size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT 84 85 // Quick fade-out (apply window) at truncation point 86 unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate 87 ASSERT(numberOfFadeOutFrames < truncatedResponseLength); 88 if (numberOfFadeOutFrames < truncatedResponseLength) { 89 for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames; i < truncatedResponseLength; ++i) { 90 float x = 1.0f - static_cast<float>(i - (truncatedResponseLength - numberOfFadeOutFrames)) / numberOfFadeOutFrames; 91 impulseResponse[i] *= x; 92 } 93 } 94 95 m_fftFrame = adoptPtr(new FFTFrame(fftSize)); 96 m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength); 97 } 98 99 PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse() 100 { 101 OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize())); 102 FFTFrame fftFrame(*m_fftFrame); 103 104 // Add leading delay back in. 105 fftFrame.addConstantGroupDelay(m_frameDelay); 106 fftFrame.doInverseFFT(channel->mutableData()); 107 108 return channel.release(); 109 } 110 111 // Interpolates two kernels with x: 0 -> 1 and returns the result. 112 PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x) 113 { 114 ASSERT(kernel1 && kernel2); 115 if (!kernel1 || !kernel2) 116 return 0; 117 118 ASSERT(x >= 0.0 && x < 1.0); 119 x = min(1.0f, max(0.0f, x)); 120 121 float sampleRate1 = kernel1->sampleRate(); 122 float sampleRate2 = kernel2->sampleRate(); 123 ASSERT(sampleRate1 == sampleRate2); 124 if (sampleRate1 != sampleRate2) 125 return 0; 126 127 float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay(); 128 129 OwnPtr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x); 130 return HRTFKernel::create(interpolatedFrame.release(), frameDelay, sampleRate1); 131 } 132 133 } // namespace WebCore 134 135 #endif // ENABLE(WEB_AUDIO) 136