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Side by Side Diff: third_party/WebKit/Source/platform/audio/HRTFKernel.cpp

Issue 2803733002: Convert ASSERT(foo) to DCHECK(foo) in platform/audio (Closed)
Patch Set: Mechanical change from ASSERT(foo) to DCHECK(foo) Created 3 years, 8 months ago
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1 /* 1 /*
2 * Copyright (C) 2010 Google Inc. All rights reserved. 2 * Copyright (C) 2010 Google Inc. All rights reserved.
3 * 3 *
4 * Redistribution and use in source and binary forms, with or without 4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted provided that the following conditions 5 * modification, are permitted provided that the following conditions
6 * are met: 6 * are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright 8 * 1. Redistributions of source code must retain the above copyright
9 * notice, this list of conditions and the following disclaimer. 9 * notice, this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright 10 * 2. Redistributions in binary form must reproduce the above copyright
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36 36
37 namespace blink { 37 namespace blink {
38 38
39 // Takes the input AudioChannel as an input impulse response and calculates the 39 // Takes the input AudioChannel as an input impulse response and calculates the
40 // average group delay. This represents the initial delay before the most 40 // average group delay. This represents the initial delay before the most
41 // energetic part of the impulse response. The sample-frame delay is removed 41 // energetic part of the impulse response. The sample-frame delay is removed
42 // from the impulseP impulse response, and this value is returned. The length 42 // from the impulseP impulse response, and this value is returned. The length
43 // of the passed in AudioChannel must be a power of 2. 43 // of the passed in AudioChannel must be a power of 2.
44 static float extractAverageGroupDelay(AudioChannel* channel, 44 static float extractAverageGroupDelay(AudioChannel* channel,
45 size_t analysisFFTSize) { 45 size_t analysisFFTSize) {
46 ASSERT(channel); 46 DCHECK(channel);
47 47
48 float* impulseP = channel->mutableData(); 48 float* impulseP = channel->mutableData();
49 49
50 bool isSizeGood = channel->length() >= analysisFFTSize; 50 bool isSizeGood = channel->length() >= analysisFFTSize;
51 ASSERT(isSizeGood); 51 DCHECK(isSizeGood);
52 if (!isSizeGood) 52 if (!isSizeGood)
53 return 0; 53 return 0;
54 54
55 // Check for power-of-2. 55 // Check for power-of-2.
56 ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) == 56 ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) ==
57 analysisFFTSize); 57 analysisFFTSize);
58 58
59 FFTFrame estimationFrame(analysisFFTSize); 59 FFTFrame estimationFrame(analysisFFTSize);
60 estimationFrame.doFFT(impulseP); 60 estimationFrame.doFFT(impulseP);
61 61
62 float frameDelay = clampTo<float>(estimationFrame.extractAverageGroupDelay()); 62 float frameDelay = clampTo<float>(estimationFrame.extractAverageGroupDelay());
63 estimationFrame.doInverseFFT(impulseP); 63 estimationFrame.doInverseFFT(impulseP);
64 64
65 return frameDelay; 65 return frameDelay;
66 } 66 }
67 67
68 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate) 68 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
69 : m_frameDelay(0), m_sampleRate(sampleRate) { 69 : m_frameDelay(0), m_sampleRate(sampleRate) {
70 ASSERT(channel); 70 DCHECK(channel);
71 71
72 // Determine the leading delay (average group delay) for the response. 72 // Determine the leading delay (average group delay) for the response.
73 m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2); 73 m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2);
74 74
75 float* impulseResponse = channel->mutableData(); 75 float* impulseResponse = channel->mutableData();
76 size_t responseLength = channel->length(); 76 size_t responseLength = channel->length();
77 77
78 // We need to truncate to fit into 1/2 the FFT size (with zero padding) in 78 // We need to truncate to fit into 1/2 the FFT size (with zero padding) in
79 // order to do proper convolution. 79 // order to do proper convolution.
80 // Truncate if necessary to max impulse response length allowed by FFT. 80 // Truncate if necessary to max impulse response length allowed by FFT.
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133 (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay(); 133 (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();
134 134
135 std::unique_ptr<FFTFrame> interpolatedFrame = 135 std::unique_ptr<FFTFrame> interpolatedFrame =
136 FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(), 136 FFTFrame::createInterpolatedFrame(*kernel1->fftFrame(),
137 *kernel2->fftFrame(), x); 137 *kernel2->fftFrame(), x);
138 return HRTFKernel::create(std::move(interpolatedFrame), frameDelay, 138 return HRTFKernel::create(std::move(interpolatedFrame), frameDelay,
139 sampleRate1); 139 sampleRate1);
140 } 140 }
141 141
142 } // namespace blink 142 } // namespace blink
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