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 39namespace blink { 40 41// Takes the input AudioChannel as an input impulse response and calculates the average group delay. 42// This represents the initial delay before the most energetic part of the impulse response. 43// The sample-frame delay is removed from the impulseP impulse response, and this value is returned. 44// the length of the passed in AudioChannel must be a power of 2. 45static float extractAverageGroupDelay(AudioChannel* channel, size_t analysisFFTSize) 46{ 47 ASSERT(channel); 48 49 float* impulseP = channel->mutableData(); 50 51 bool isSizeGood = channel->length() >= analysisFFTSize; 52 ASSERT(isSizeGood); 53 if (!isSizeGood) 54 return 0; 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 float frameDelay = narrowPrecisionToFloat(estimationFrame.extractAverageGroupDelay()); 63 estimationFrame.doInverseFFT(impulseP); 64 65 return frameDelay; 66} 67 68HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate) 69 : m_frameDelay(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->mutableData(); 78 size_t responseLength = channel->length(); 79 80 // We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution. 81 size_t truncatedResponseLength = std::min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT 82 83 // Quick fade-out (apply window) at truncation point 84 unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate 85 ASSERT(numberOfFadeOutFrames < truncatedResponseLength); 86 if (numberOfFadeOutFrames < truncatedResponseLength) { 87 for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames; i < truncatedResponseLength; ++i) { 88 float x = 1.0f - static_cast<float>(i - (truncatedResponseLength - numberOfFadeOutFrames)) / numberOfFadeOutFrames; 89 impulseResponse[i] *= x; 90 } 91 } 92 93 m_fftFrame = adoptPtr(new FFTFrame(fftSize)); 94 m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength); 95} 96 97PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse() 98{ 99 OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize())); 100 FFTFrame fftFrame(*m_fftFrame); 101 102 // Add leading delay back in. 103 fftFrame.addConstantGroupDelay(m_frameDelay); 104 fftFrame.doInverseFFT(channel->mutableData()); 105 106 return channel.release(); 107} 108 109// Interpolates two kernels with x: 0 -> 1 and returns the result. 110PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x) 111{ 112 ASSERT(kernel1 && kernel2); 113 if (!kernel1 || !kernel2) 114 return nullptr; 115 116 ASSERT(x >= 0.0 && x < 1.0); 117 x = std::min(1.0f, std::max(0.0f, x)); 118 119 float sampleRate1 = kernel1->sampleRate(); 120 float sampleRate2 = kernel2->sampleRate(); 121 ASSERT(sampleRate1 == sampleRate2); 122 if (sampleRate1 != sampleRate2) 123 return nullptr; 124 125 float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay(); 126 127 OwnPtr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x); 128 return HRTFKernel::create(interpolatedFrame.release(), frameDelay, sampleRate1); 129} 130 131} // namespace blink 132 133#endif // ENABLE(WEB_AUDIO) 134