OLD | NEW |
| (Empty) |
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 } | |
OLD | NEW |