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

Issue 1207353002: Add new variance update option and unittests for intelligibility (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Merge Created 5 years, 5 months ago
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1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 //
12 // Command line tool for speech intelligibility enhancement. Provides for
13 // running and testing intelligibility_enhancer as an independent process.
14 // Use --help for options.
15 //
16
17 #include <stdint.h>
18 #include <stdlib.h>
19 #include <string>
20 #include <sys/stat.h>
21 #include <sys/types.h>
22
23 #include "gflags/gflags.h"
24 #include "testing/gtest/include/gtest/gtest.h"
25 #include "webrtc/base/checks.h"
26 #include "webrtc/common_audio/real_fourier.h"
27 #include "webrtc/common_audio/wav_file.h"
28 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhanc er.h"
29 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils. h"
30 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
31 #include "webrtc/test/testsupport/fileutils.h"
32
33 using std::complex;
34
35 namespace webrtc {
36
37 using webrtc::RealFourier;
38 using webrtc::IntelligibilityEnhancer;
39
40 DEFINE_int32(clear_type,
41 webrtc::intelligibility::VarianceArray::kStepInfinite,
42 "Variance algorithm for clear data.");
43 DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
44 DEFINE_int32(clear_window,
45 475,
46 "Window size for windowed variance for clear data.");
47 DEFINE_int32(sample_rate,
48 16000,
49 "Audio sample rate used in the input and output files.");
50 DEFINE_int32(ana_rate,
51 800,
52 "Analysis rate; gains recalculated every N blocks.");
53 DEFINE_int32(
54 var_rate,
55 2,
56 "Variance clear rate; history is forgotten every N gain recalculations.");
57 DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
58
59 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
60 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
61 DEFINE_string(out_file,
62 "proc_enhanced.wav",
63 "Enhanced output. Use '-' to "
64 "play through aplay immediately.");
65
66 // Constant IntelligibilityEnhancer constructor parameters.
67 const int kErbResolution = 2;
68 const int kNumChannels = 1;
69
70 // void function for gtest
71 void void_main(int argc, char* argv[]) {
72 google::SetUsageMessage(
73 "\n\nVariance algorithm types are:\n"
74 " 0 - infinite/normal,\n"
75 " 1 - exponentially decaying,\n"
76 " 2 - rolling window.\n"
77 "\nInput files must be little-endian 16-bit signed raw PCM.\n");
78 google::ParseCommandLineFlags(&argc, &argv, true);
79
80 size_t samples; // Number of samples in input PCM file
81 size_t fragment_size; // Number of samples to process at a time
82 // to simulate APM stream processing
83
84 // Load settings and wav input.
85
86 fragment_size = FLAGS_sample_rate / 100; // Mirror real time APM chunk size.
87 // Duplicates chunk_length_ in
88 // IntelligibilityEnhancer.
89
90 struct stat in_stat, noise_stat;
91 ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
92 << "Empty speech file.";
93 ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
94 << "Empty noise file.";
95
96 samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
97
98 WavReader in_file(FLAGS_clear_file);
99 std::vector<float> in_fpcm(samples);
100 in_file.ReadSamples(samples, &in_fpcm[0]);
101
102 WavReader noise_file(FLAGS_noise_file);
103 std::vector<float> noise_fpcm(samples);
104 noise_file.ReadSamples(samples, &noise_fpcm[0]);
105
106 // Run intelligibility enhancement.
107
108 IntelligibilityEnhancer enh(
109 kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type,
110 static_cast<float>(FLAGS_clear_alpha), FLAGS_clear_window, FLAGS_ana_rate,
111 FLAGS_var_rate, FLAGS_gain_limit);
112
113 // Slice the input into smaller chunks, as the APM would do, and feed them
114 // through the enhancer.
115 float* clear_cursor = &in_fpcm[0];
116 float* noise_cursor = &noise_fpcm[0];
117
118 for (size_t i = 0; i < samples; i += fragment_size) {
119 enh.ProcessCaptureAudio(&noise_cursor);
120 enh.ProcessRenderAudio(&clear_cursor);
121 clear_cursor += fragment_size;
122 noise_cursor += fragment_size;
123 }
124
125 if (FLAGS_out_file.compare("-") == 0) {
126 const std::string temp_out_filename =
127 test::TempFilename(test::WorkingDir(), "temp_wav_file");
128 {
129 WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
130 out_file.WriteSamples(&in_fpcm[0], samples);
131 }
132 system(("aplay " + temp_out_filename).c_str());
133 system(("rm " + temp_out_filename).c_str());
134 } else {
135 WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
136 out_file.WriteSamples(&in_fpcm[0], samples);
137 }
138 }
139
140 } // namespace webrtc
141
142 int main(int argc, char* argv[]) {
143 webrtc::void_main(argc, argv);
144 return 0;
145 }
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