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Side by Side Diff: webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc

Issue 1227213002: Update audio code to use size_t more correctly, webrtc/modules/audio_processing/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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26 #include "webrtc/common_audio/real_fourier.h" 26 #include "webrtc/common_audio/real_fourier.h"
27 #include "webrtc/common_audio/wav_file.h" 27 #include "webrtc/common_audio/wav_file.h"
28 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhanc er.h" 28 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhanc er.h"
29 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils. h" 29 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils. h"
30 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 30 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
31 #include "webrtc/test/testsupport/fileutils.h" 31 #include "webrtc/test/testsupport/fileutils.h"
32 32
33 using std::complex; 33 using std::complex;
34 34
35 namespace webrtc { 35 namespace webrtc {
36 namespace {
36 37
37 using webrtc::RealFourier; 38 bool ValidateClearWindow(const char* flagname, int32_t value) {
38 using webrtc::IntelligibilityEnhancer; 39 return value >= 0;
40 }
39 41
40 DEFINE_int32(clear_type, 42 DEFINE_int32(clear_type,
41 webrtc::intelligibility::VarianceArray::kStepInfinite, 43 webrtc::intelligibility::VarianceArray::kStepInfinite,
42 "Variance algorithm for clear data."); 44 "Variance algorithm for clear data.");
43 DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data."); 45 DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
44 DEFINE_int32(clear_window, 46 DEFINE_int32(clear_window,
45 475, 47 475,
46 "Window size for windowed variance for clear data."); 48 "Window size for windowed variance for clear data.");
49 const bool clear_window_dummy =
50 google::RegisterFlagValidator(&FLAGS_clear_window, &ValidateClearWindow);
47 DEFINE_int32(sample_rate, 51 DEFINE_int32(sample_rate,
48 16000, 52 16000,
49 "Audio sample rate used in the input and output files."); 53 "Audio sample rate used in the input and output files.");
50 DEFINE_int32(ana_rate, 54 DEFINE_int32(ana_rate,
51 800, 55 800,
52 "Analysis rate; gains recalculated every N blocks."); 56 "Analysis rate; gains recalculated every N blocks.");
53 DEFINE_int32( 57 DEFINE_int32(
54 var_rate, 58 var_rate,
55 2, 59 2,
56 "Variance clear rate; history is forgotten every N gain recalculations."); 60 "Variance clear rate; history is forgotten every N gain recalculations.");
57 DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block."); 61 DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
58 62
59 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech."); 63 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
60 DEFINE_string(noise_file, "noise.wav", "Input file with noise data."); 64 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
61 DEFINE_string(out_file, 65 DEFINE_string(out_file,
62 "proc_enhanced.wav", 66 "proc_enhanced.wav",
63 "Enhanced output. Use '-' to " 67 "Enhanced output. Use '-' to "
64 "play through aplay immediately."); 68 "play through aplay immediately.");
65 69
66 // Constant IntelligibilityEnhancer constructor parameters. 70 // Constant IntelligibilityEnhancer constructor parameters.
67 const int kErbResolution = 2; 71 const size_t kErbResolution = 2;
68 const int kNumChannels = 1; 72 const int kNumChannels = 1;
69 73
70 // void function for gtest 74 // void function for gtest
71 void void_main(int argc, char* argv[]) { 75 void void_main(int argc, char* argv[]) {
72 google::SetUsageMessage( 76 google::SetUsageMessage(
73 "\n\nVariance algorithm types are:\n" 77 "\n\nVariance algorithm types are:\n"
74 " 0 - infinite/normal,\n" 78 " 0 - infinite/normal,\n"
75 " 1 - exponentially decaying,\n" 79 " 1 - exponentially decaying,\n"
76 " 2 - rolling window.\n" 80 " 2 - rolling window.\n"
77 "\nInput files must be little-endian 16-bit signed raw PCM.\n"); 81 "\nInput files must be little-endian 16-bit signed raw PCM.\n");
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100 in_file.ReadSamples(samples, &in_fpcm[0]); 104 in_file.ReadSamples(samples, &in_fpcm[0]);
101 105
102 WavReader noise_file(FLAGS_noise_file); 106 WavReader noise_file(FLAGS_noise_file);
103 std::vector<float> noise_fpcm(samples); 107 std::vector<float> noise_fpcm(samples);
104 noise_file.ReadSamples(samples, &noise_fpcm[0]); 108 noise_file.ReadSamples(samples, &noise_fpcm[0]);
105 109
106 // Run intelligibility enhancement. 110 // Run intelligibility enhancement.
107 111
108 IntelligibilityEnhancer enh( 112 IntelligibilityEnhancer enh(
109 kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type, 113 kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type,
110 static_cast<float>(FLAGS_clear_alpha), FLAGS_clear_window, FLAGS_ana_rate, 114 static_cast<float>(FLAGS_clear_alpha),
111 FLAGS_var_rate, FLAGS_gain_limit); 115 static_cast<size_t>(FLAGS_clear_window), FLAGS_ana_rate, FLAGS_var_rate,
116 FLAGS_gain_limit);
112 117
113 // Slice the input into smaller chunks, as the APM would do, and feed them 118 // Slice the input into smaller chunks, as the APM would do, and feed them
114 // through the enhancer. 119 // through the enhancer.
115 float* clear_cursor = &in_fpcm[0]; 120 float* clear_cursor = &in_fpcm[0];
116 float* noise_cursor = &noise_fpcm[0]; 121 float* noise_cursor = &noise_fpcm[0];
117 122
118 for (size_t i = 0; i < samples; i += fragment_size) { 123 for (size_t i = 0; i < samples; i += fragment_size) {
119 enh.ProcessCaptureAudio(&noise_cursor); 124 enh.ProcessCaptureAudio(&noise_cursor);
120 enh.ProcessRenderAudio(&clear_cursor); 125 enh.ProcessRenderAudio(&clear_cursor);
121 clear_cursor += fragment_size; 126 clear_cursor += fragment_size;
122 noise_cursor += fragment_size; 127 noise_cursor += fragment_size;
123 } 128 }
124 129
125 if (FLAGS_out_file.compare("-") == 0) { 130 if (FLAGS_out_file.compare("-") == 0) {
126 const std::string temp_out_filename = 131 const std::string temp_out_filename =
127 test::TempFilename(test::WorkingDir(), "temp_wav_file"); 132 test::TempFilename(test::WorkingDir(), "temp_wav_file");
128 { 133 {
129 WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels); 134 WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
130 out_file.WriteSamples(&in_fpcm[0], samples); 135 out_file.WriteSamples(&in_fpcm[0], samples);
131 } 136 }
132 system(("aplay " + temp_out_filename).c_str()); 137 system(("aplay " + temp_out_filename).c_str());
133 system(("rm " + temp_out_filename).c_str()); 138 system(("rm " + temp_out_filename).c_str());
134 } else { 139 } else {
135 WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels); 140 WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
136 out_file.WriteSamples(&in_fpcm[0], samples); 141 out_file.WriteSamples(&in_fpcm[0], samples);
137 } 142 }
138 } 143 }
139 144
145 } // namespace
140 } // namespace webrtc 146 } // namespace webrtc
141 147
142 int main(int argc, char* argv[]) { 148 int main(int argc, char* argv[]) {
143 webrtc::void_main(argc, argv); 149 webrtc::void_main(argc, argv);
144 return 0; 150 return 0;
145 } 151 }
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