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 "HRTFKernel.h" 34 35 #include "AudioChannel.h" 36 #include "Biquad.h" 37 #include "FFTFrame.h" 38 #include <wtf/MathExtras.h> 39 40 using namespace std; 41 42 namespace WebCore { 43 44 // Takes the input AudioChannel as an input impulse response and calculates the average group delay. 45 // This represents the initial delay before the most energetic part of the impulse response. 46 // The sample-frame delay is removed from the impulseP impulse response, and this value is returned. 47 // the length of the passed in AudioChannel must be a power of 2. 48 static double extractAverageGroupDelay(AudioChannel* channel, size_t analysisFFTSize) 49 { 50 ASSERT(channel); 51 52 float* impulseP = channel->data(); 53 54 ASSERT(channel->length() >= analysisFFTSize); 55 56 // Check for power-of-2. 57 ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) == analysisFFTSize); 58 59 FFTFrame estimationFrame(analysisFFTSize); 60 estimationFrame.doFFT(impulseP); 61 62 double frameDelay = estimationFrame.extractAverageGroupDelay(); 63 estimationFrame.doInverseFFT(impulseP); 64 65 return frameDelay; 66 } 67 68 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost) 69 : m_frameDelay(0.0) 70 , m_sampleRate(sampleRate) 71 { 72 ASSERT(channel); 73 74 // Determine the leading delay (average group delay) for the response. 75 m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2); 76 77 float* impulseResponse = channel->data(); 78 size_t responseLength = channel->length(); 79 80 if (bassBoost) { 81 // Run through some post-processing to boost the bass a little -- the HRTF's seem to be a little bass-deficient. 82 // FIXME: this post-processing should have already been applied to the HRTF file resources. Once the files are put into this form, 83 // then this code path can be removed along with the bassBoost parameter. 84 Biquad filter; 85 filter.setLowShelfParams(700.0 / nyquist(), 6.0); // boost 6dB at 700Hz 86 filter.process(impulseResponse, impulseResponse, responseLength); 87 } 88 89 // We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution. 90 size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT 91 92 // Quick fade-out (apply window) at truncation point 93 unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate 94 ASSERT(numberOfFadeOutFrames < truncatedResponseLength); 95 if (numberOfFadeOutFrames < truncatedResponseLength) { 96 for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames; i < truncatedResponseLength; ++i) { 97 float x = 1.0f - static_cast<float>(i - (truncatedResponseLength - numberOfFadeOutFrames)) / numberOfFadeOutFrames; 98 impulseResponse[i] *= x; 99 } 100 } 101 102 m_fftFrame = adoptPtr(new FFTFrame(fftSize)); 103 m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength); 104 } 105 106 PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse() 107 { 108 OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize())); 109 FFTFrame fftFrame(*m_fftFrame); 110 111 // Add leading delay back in. 112 fftFrame.addConstantGroupDelay(m_frameDelay); 113 fftFrame.doInverseFFT(channel->data()); 114 115 return channel.release(); 116 } 117 118 // Interpolates two kernels with x: 0 -> 1 and returns the result. 119 PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, double x) 120 { 121 ASSERT(kernel1 && kernel2); 122 if (!kernel1 || !kernel2) 123 return 0; 124 125 ASSERT(x >= 0.0 && x < 1.0); 126 x = min(1.0, max(0.0, x)); 127 128 double sampleRate1 = kernel1->sampleRate(); 129 double sampleRate2 = kernel2->sampleRate(); 130 ASSERT(sampleRate1 == sampleRate2); 131 if (sampleRate1 != sampleRate2) 132 return 0; 133 134 double frameDelay = (1.0 - x) * kernel1->frameDelay() + x * kernel2->frameDelay(); 135 136 OwnPtr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x); 137 return HRTFKernel::create(interpolatedFrame.release(), frameDelay, sampleRate1); 138 } 139 140 } // namespace WebCore 141 142 #endif // ENABLE(WEB_AUDIO) 143
