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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 namespace 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.
extractAverageGroupDelay(AudioChannel * channel,size_t analysisFFTSize)45 static 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 
HRTFKernel(AudioChannel * channel,size_t fftSize,float sampleRate)68 HRTFKernel::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 
createImpulseResponse()97 PassOwnPtr<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.
createInterpolatedKernel(HRTFKernel * kernel1,HRTFKernel * kernel2,float x)110 PassRefPtr<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