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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 |
11 #include <stdio.h> | 11 #include <stdio.h> |
| 12 #include <iostream> |
12 #include <sstream> | 13 #include <sstream> |
13 #include <string> | 14 #include <string> |
14 | 15 |
15 #include "gflags/gflags.h" | 16 #include "gflags/gflags.h" |
16 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
18 #include "webrtc/common_audio/channel_buffer.h" | 19 #include "webrtc/common_audio/channel_buffer.h" |
19 #include "webrtc/common_audio/wav_file.h" | 20 #include "webrtc/common_audio/wav_file.h" |
20 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 21 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 22 #include "webrtc/modules/audio_processing/test/audio_file_processor.h" |
21 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" | 23 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
22 #include "webrtc/modules/audio_processing/test/test_utils.h" | 24 #include "webrtc/modules/audio_processing/test/test_utils.h" |
23 #include "webrtc/system_wrappers/include/tick_util.h" | 25 #include "webrtc/system_wrappers/include/tick_util.h" |
24 #include "webrtc/test/testsupport/trace_to_stderr.h" | 26 #include "webrtc/test/testsupport/trace_to_stderr.h" |
25 | 27 |
26 DEFINE_string(dump, "", "The name of the debug dump file to read from."); | 28 DEFINE_string(dump, "", "Name of the aecdump debug file to read from."); |
27 DEFINE_string(i, "", "The name of the input file to read from."); | 29 DEFINE_string(i, "", "Name of the capture input stream file to read from."); |
28 DEFINE_string(i_rev, "", "The name of the reverse input file to read from."); | 30 DEFINE_string( |
29 DEFINE_string(o, "out.wav", "Name of the output file to write to."); | 31 o, |
30 DEFINE_string(o_rev, | 32 "out.wav", |
31 "out_rev.wav", | 33 "Name of the output file to write the processed capture stream to."); |
32 "Name of the reverse output file to write to."); | 34 DEFINE_int32(out_channels, 1, "Number of output channels."); |
33 DEFINE_int32(out_channels, 0, "Number of output channels. Defaults to input."); | 35 DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz."); |
34 DEFINE_int32(out_sample_rate, 0, | |
35 "Output sample rate in Hz. Defaults to input."); | |
36 DEFINE_string(mic_positions, "", | 36 DEFINE_string(mic_positions, "", |
37 "Space delimited cartesian coordinates of microphones in meters. " | 37 "Space delimited cartesian coordinates of microphones in meters. " |
38 "The coordinates of each point are contiguous. " | 38 "The coordinates of each point are contiguous. " |
39 "For a two element array: \"x1 y1 z1 x2 y2 z2\""); | 39 "For a two element array: \"x1 y1 z1 x2 y2 z2\""); |
40 DEFINE_double(target_angle_degrees, 90, "The azimuth of the target in radians"); | 40 DEFINE_double( |
| 41 target_angle_degrees, |
| 42 90, |
| 43 "The azimuth of the target in degrees. Only applies to beamforming."); |
41 | 44 |
42 DEFINE_bool(aec, false, "Enable echo cancellation."); | 45 DEFINE_bool(aec, false, "Enable echo cancellation."); |
43 DEFINE_bool(agc, false, "Enable automatic gain control."); | 46 DEFINE_bool(agc, false, "Enable automatic gain control."); |
44 DEFINE_bool(hpf, false, "Enable high-pass filtering."); | 47 DEFINE_bool(hpf, false, "Enable high-pass filtering."); |
45 DEFINE_bool(ns, false, "Enable noise suppression."); | 48 DEFINE_bool(ns, false, "Enable noise suppression."); |
46 DEFINE_bool(ts, false, "Enable transient suppression."); | 49 DEFINE_bool(ts, false, "Enable transient suppression."); |
47 DEFINE_bool(bf, false, "Enable beamforming."); | 50 DEFINE_bool(bf, false, "Enable beamforming."); |
48 DEFINE_bool(ie, false, "Enable intelligibility enhancer."); | 51 DEFINE_bool(ie, false, "Enable intelligibility enhancer."); |
49 DEFINE_bool(all, false, "Enable all components."); | 52 DEFINE_bool(all, false, "Enable all components."); |
50 | 53 |
51 DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3]."); | 54 DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3]."); |
52 | 55 |
53 DEFINE_bool(perf, false, "Enable performance tests."); | 56 DEFINE_bool(perf, false, "Enable performance tests."); |
54 | 57 |
55 namespace webrtc { | 58 namespace webrtc { |
56 namespace { | 59 namespace { |
57 | 60 |
58 const int kChunksPerSecond = 100; | 61 const int kChunksPerSecond = 100; |
59 const char kUsage[] = | 62 const char kUsage[] = |
60 "Command-line tool to run audio processing on WAV files. Accepts either\n" | 63 "Command-line tool to run audio processing on WAV files. Accepts either\n" |
61 "an input capture WAV file or protobuf debug dump and writes to an output\n" | 64 "an input capture WAV file or protobuf debug dump and writes to an output\n" |
62 "WAV file.\n" | 65 "WAV file.\n" |
63 "\n" | 66 "\n" |
64 "All components are disabled by default. If any bi-directional components\n" | 67 "All components are disabled by default. If any bi-directional components\n" |
65 "are enabled, only debug dump files are permitted."; | 68 "are enabled, only debug dump files are permitted."; |
66 | 69 |
67 // Returns a StreamConfig corresponding to wav_file if it's non-nullptr. | |
68 // Otherwise returns a default initialized StreamConfig. | |
69 StreamConfig MakeStreamConfig(const WavFile* wav_file) { | |
70 if (wav_file) { | |
71 return {wav_file->sample_rate(), wav_file->num_channels()}; | |
72 } | |
73 return {}; | |
74 } | |
75 | |
76 } // namespace | 70 } // namespace |
77 | 71 |
78 int main(int argc, char* argv[]) { | 72 int main(int argc, char* argv[]) { |
79 google::SetUsageMessage(kUsage); | 73 google::SetUsageMessage(kUsage); |
80 google::ParseCommandLineFlags(&argc, &argv, true); | 74 google::ParseCommandLineFlags(&argc, &argv, true); |
81 | 75 |
82 if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) { | 76 if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) { |
83 fprintf(stderr, | 77 fprintf(stderr, |
84 "An input file must be specified with either -i or -dump.\n"); | 78 "An input file must be specified with either -i or -dump.\n"); |
85 return 1; | 79 return 1; |
86 } | 80 } |
87 if (!FLAGS_dump.empty()) { | 81 if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) { |
88 fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n"); | 82 fprintf(stderr, "-aec and -ie require a -dump file.\n"); |
| 83 return 1; |
| 84 } |
| 85 if (FLAGS_ie) { |
| 86 fprintf(stderr, |
| 87 "FIXME(ajm): The intelligibility enhancer output is not dumped.\n"); |
89 return 1; | 88 return 1; |
90 } | 89 } |
91 | 90 |
92 test::TraceToStderr trace_to_stderr(true); | 91 test::TraceToStderr trace_to_stderr(true); |
93 WavReader in_file(FLAGS_i); | |
94 // If the output format is uninitialized, use the input format. | |
95 const int out_channels = | |
96 FLAGS_out_channels ? FLAGS_out_channels : in_file.num_channels(); | |
97 const int out_sample_rate = | |
98 FLAGS_out_sample_rate ? FLAGS_out_sample_rate : in_file.sample_rate(); | |
99 WavWriter out_file(FLAGS_o, out_sample_rate, out_channels); | |
100 | |
101 Config config; | 92 Config config; |
| 93 if (FLAGS_bf || FLAGS_all) { |
| 94 if (FLAGS_mic_positions.empty()) { |
| 95 fprintf(stderr, "-mic_positions must be specified when -bf is used.\n"); |
| 96 return 1; |
| 97 } |
| 98 config.Set<Beamforming>(new Beamforming( |
| 99 true, ParseArrayGeometry(FLAGS_mic_positions), |
| 100 SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f, |
| 101 1.f))); |
| 102 } |
102 config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all)); | 103 config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all)); |
103 config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all)); | 104 config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all)); |
104 | 105 |
105 if (FLAGS_bf || FLAGS_all) { | |
106 const size_t num_mics = in_file.num_channels(); | |
107 const std::vector<Point> array_geometry = | |
108 ParseArrayGeometry(FLAGS_mic_positions, num_mics); | |
109 RTC_CHECK_EQ(array_geometry.size(), num_mics); | |
110 | |
111 config.Set<Beamforming>(new Beamforming( | |
112 true, array_geometry, | |
113 SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f, | |
114 1.f))); | |
115 } | |
116 | |
117 rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config)); | 106 rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config)); |
118 if (!FLAGS_dump.empty()) { | 107 RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all)); |
119 RTC_CHECK_EQ(kNoErr, | |
120 ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all)); | |
121 } else if (FLAGS_aec) { | |
122 fprintf(stderr, "-aec requires a -dump file.\n"); | |
123 return -1; | |
124 } | |
125 bool process_reverse = !FLAGS_i_rev.empty(); | |
126 RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all)); | 108 RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all)); |
127 RTC_CHECK_EQ(kNoErr, | |
128 ap->gain_control()->set_mode(GainControl::kFixedDigital)); | |
129 RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all)); | 109 RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all)); |
130 RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all)); | 110 RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all)); |
131 if (FLAGS_ns_level != -1) | 111 if (FLAGS_ns_level != -1) { |
132 RTC_CHECK_EQ(kNoErr, | 112 RTC_CHECK_EQ(kNoErr, |
133 ap->noise_suppression()->set_level( | 113 ap->noise_suppression()->set_level( |
134 static_cast<NoiseSuppression::Level>(FLAGS_ns_level))); | 114 static_cast<NoiseSuppression::Level>(FLAGS_ns_level))); |
135 | |
136 printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n", | |
137 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate()); | |
138 printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n", | |
139 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate()); | |
140 | |
141 ChannelBuffer<float> in_buf( | |
142 rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond), | |
143 in_file.num_channels()); | |
144 ChannelBuffer<float> out_buf( | |
145 rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond), | |
146 out_file.num_channels()); | |
147 | |
148 std::vector<float> in_interleaved(in_buf.size()); | |
149 std::vector<float> out_interleaved(out_buf.size()); | |
150 | |
151 rtc::scoped_ptr<WavReader> in_rev_file; | |
152 rtc::scoped_ptr<WavWriter> out_rev_file; | |
153 rtc::scoped_ptr<ChannelBuffer<float>> in_rev_buf; | |
154 rtc::scoped_ptr<ChannelBuffer<float>> out_rev_buf; | |
155 std::vector<float> in_rev_interleaved; | |
156 std::vector<float> out_rev_interleaved; | |
157 if (process_reverse) { | |
158 in_rev_file.reset(new WavReader(FLAGS_i_rev)); | |
159 out_rev_file.reset(new WavWriter(FLAGS_o_rev, in_rev_file->sample_rate(), | |
160 in_rev_file->num_channels())); | |
161 printf("In rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n", | |
162 FLAGS_i_rev.c_str(), in_rev_file->num_channels(), | |
163 in_rev_file->sample_rate()); | |
164 printf("Out rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n", | |
165 FLAGS_o_rev.c_str(), out_rev_file->num_channels(), | |
166 out_rev_file->sample_rate()); | |
167 in_rev_buf.reset(new ChannelBuffer<float>( | |
168 rtc::CheckedDivExact(in_rev_file->sample_rate(), kChunksPerSecond), | |
169 in_rev_file->num_channels())); | |
170 in_rev_interleaved.resize(in_rev_buf->size()); | |
171 out_rev_buf.reset(new ChannelBuffer<float>( | |
172 rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond), | |
173 out_rev_file->num_channels())); | |
174 out_rev_interleaved.resize(out_rev_buf->size()); | |
175 } | 115 } |
176 | 116 |
177 TickTime processing_start_time; | 117 rtc::scoped_ptr<AudioFileProcessor> processor; |
178 TickInterval accumulated_time; | 118 auto out_file = rtc_make_scoped_ptr( |
| 119 new WavWriter(FLAGS_o, FLAGS_out_sample_rate, FLAGS_out_channels)); |
| 120 std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl; |
| 121 if (FLAGS_dump.empty()) { |
| 122 auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i)); |
| 123 std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl; |
| 124 processor.reset( |
| 125 new WavFileProcessor(ap.Pass(), in_file.Pass(), out_file.Pass())); |
| 126 |
| 127 } else { |
| 128 processor.reset(new AecDumpFileProcessor( |
| 129 ap.Pass(), fopen(FLAGS_dump.c_str(), "rb"), out_file.Pass())); |
| 130 } |
| 131 |
179 int num_chunks = 0; | 132 int num_chunks = 0; |
| 133 while (processor->ProcessChunk()) { |
| 134 trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond); |
| 135 ++num_chunks; |
| 136 } |
180 | 137 |
181 const auto input_config = MakeStreamConfig(&in_file); | 138 if (FLAGS_perf) { |
182 const auto output_config = MakeStreamConfig(&out_file); | 139 const auto& proc_time = processor->proc_time(); |
183 const auto reverse_input_config = MakeStreamConfig(in_rev_file.get()); | 140 int64_t exec_time_us = proc_time.sum.Microseconds(); |
184 const auto reverse_output_config = MakeStreamConfig(out_rev_file.get()); | 141 printf( |
| 142 "\nExecution time: %.3f s, File time: %.2f s\n" |
| 143 "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n", |
| 144 exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond, |
| 145 exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(), |
| 146 1.f * proc_time.min.Microseconds()); |
| 147 } |
185 | 148 |
186 while (in_file.ReadSamples(in_interleaved.size(), | |
187 &in_interleaved[0]) == in_interleaved.size()) { | |
188 // Have logs display the file time rather than wallclock time. | |
189 trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond); | |
190 FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(), | |
191 &in_interleaved[0]); | |
192 Deinterleave(&in_interleaved[0], in_buf.num_frames(), | |
193 in_buf.num_channels(), in_buf.channels()); | |
194 if (process_reverse) { | |
195 in_rev_file->ReadSamples(in_rev_interleaved.size(), | |
196 in_rev_interleaved.data()); | |
197 FloatS16ToFloat(in_rev_interleaved.data(), in_rev_interleaved.size(), | |
198 in_rev_interleaved.data()); | |
199 Deinterleave(in_rev_interleaved.data(), in_rev_buf->num_frames(), | |
200 in_rev_buf->num_channels(), in_rev_buf->channels()); | |
201 } | |
202 | |
203 if (FLAGS_perf) { | |
204 processing_start_time = TickTime::Now(); | |
205 } | |
206 RTC_CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config, | |
207 output_config, out_buf.channels())); | |
208 if (process_reverse) { | |
209 RTC_CHECK_EQ(kNoErr, ap->ProcessReverseStream( | |
210 in_rev_buf->channels(), reverse_input_config, | |
211 reverse_output_config, out_rev_buf->channels())); | |
212 } | |
213 if (FLAGS_perf) { | |
214 accumulated_time += TickTime::Now() - processing_start_time; | |
215 } | |
216 | |
217 Interleave(out_buf.channels(), out_buf.num_frames(), | |
218 out_buf.num_channels(), &out_interleaved[0]); | |
219 FloatToFloatS16(&out_interleaved[0], out_interleaved.size(), | |
220 &out_interleaved[0]); | |
221 out_file.WriteSamples(&out_interleaved[0], out_interleaved.size()); | |
222 if (process_reverse) { | |
223 Interleave(out_rev_buf->channels(), out_rev_buf->num_frames(), | |
224 out_rev_buf->num_channels(), out_rev_interleaved.data()); | |
225 FloatToFloatS16(out_rev_interleaved.data(), out_rev_interleaved.size(), | |
226 out_rev_interleaved.data()); | |
227 out_rev_file->WriteSamples(out_rev_interleaved.data(), | |
228 out_rev_interleaved.size()); | |
229 } | |
230 num_chunks++; | |
231 } | |
232 if (FLAGS_perf) { | |
233 int64_t execution_time_ms = accumulated_time.Milliseconds(); | |
234 printf("\nExecution time: %.3f s\nFile time: %.2f s\n" | |
235 "Time per chunk: %.3f ms\n", | |
236 execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond, | |
237 execution_time_ms * 1.f / num_chunks); | |
238 } | |
239 return 0; | 149 return 0; |
240 } | 150 } |
241 | 151 |
242 } // namespace webrtc | 152 } // namespace webrtc |
243 | 153 |
244 int main(int argc, char* argv[]) { | 154 int main(int argc, char* argv[]) { |
245 return webrtc::main(argc, argv); | 155 return webrtc::main(argc, argv); |
246 } | 156 } |
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