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 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 "FloatConversion.h" |
39 | #include <wtf/MathExtras.h> |
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 (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 = std::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 = std::make_unique<FFTFrame>(fftSize); |
96 | m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength); |
97 | } |
98 | |
99 | size_t HRTFKernel::fftSize() const |
100 | { |
101 | return m_fftFrame->fftSize(); |
102 | } |
103 | |
104 | std::unique_ptr<AudioChannel> HRTFKernel::createImpulseResponse() |
105 | { |
106 | auto channel = std::make_unique<AudioChannel>(fftSize()); |
107 | FFTFrame fftFrame(*m_fftFrame); |
108 | |
109 | // Add leading delay back in. |
110 | fftFrame.addConstantGroupDelay(m_frameDelay); |
111 | fftFrame.doInverseFFT(channel->mutableData()); |
112 | |
113 | return channel; |
114 | } |
115 | |
116 | // Interpolates two kernels with x: 0 -> 1 and returns the result. |
117 | RefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x) |
118 | { |
119 | ASSERT(kernel1 && kernel2); |
120 | if (!kernel1 || !kernel2) |
121 | return nullptr; |
122 | |
123 | ASSERT(x >= 0.0 && x < 1.0); |
124 | x = std::min(1.0f, std::max(0.0f, x)); |
125 | |
126 | float sampleRate1 = kernel1->sampleRate(); |
127 | float sampleRate2 = kernel2->sampleRate(); |
128 | ASSERT(sampleRate1 == sampleRate2); |
129 | if (sampleRate1 != sampleRate2) |
130 | return nullptr; |
131 | |
132 | float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay(); |
133 | |
134 | std::unique_ptr<FFTFrame> interpolatedFrame = FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), *kernel2->fftFrame(), x); |
135 | return HRTFKernel::create(WTFMove(interpolatedFrame), frameDelay, sampleRate1); |
136 | } |
137 | |
138 | } // namespace WebCore |
139 | |
140 | #endif // ENABLE(WEB_AUDIO) |
141 | |