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.
extractAverageGroupDelay(AudioChannel * channel,size_t analysisFFTSize)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
HRTFKernel(AudioChannel * channel,size_t fftSize,double sampleRate,bool bassBoost)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
createImpulseResponse()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.
createInterpolatedKernel(HRTFKernel * kernel1,HRTFKernel * kernel2,double x)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