1 /*
2 * Copyright (c) 2011 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 // Commandline tool to unpack audioproc debug files.
12 //
13 // The debug files are dumped as protobuf blobs. For analysis, it's necessary
14 // to unpack the file into its component parts: audio and other data.
15
16 #include <stdio.h>
17
18 #include "gflags/gflags.h"
19 #include "webrtc/audio_processing/debug.pb.h"
20 #include "webrtc/base/format_macros.h"
21 #include "webrtc/base/scoped_ptr.h"
22 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
23 #include "webrtc/modules/audio_processing/test/test_utils.h"
24 #include "webrtc/typedefs.h"
25
26 // TODO(andrew): unpack more of the data.
27 DEFINE_string(input_file, "input", "The name of the input stream file.");
28 DEFINE_string(output_file, "ref_out",
29 "The name of the reference output stream file.");
30 DEFINE_string(reverse_file, "reverse",
31 "The name of the reverse input stream file.");
32 DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
33 DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
34 DEFINE_string(level_file, "level.int32", "The name of the level file.");
35 DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
36 DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
37 DEFINE_bool(full, false,
38 "Unpack the full set of files (normally not needed).");
39 DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
40 DEFINE_bool(text,
41 false,
42 "Write non-audio files as text files instead of binary files.");
43
44 #define PRINT_CONFIG(field_name) \
45 if (msg.has_##field_name()) { \
46 fprintf(settings_file, " " #field_name ": %d\n", msg.field_name()); \
47 }
48
49 namespace webrtc {
50
51 using audioproc::Event;
52 using audioproc::ReverseStream;
53 using audioproc::Stream;
54 using audioproc::Init;
55
WriteData(const void * data,size_t size,FILE * file,const std::string & filename)56 void WriteData(const void* data, size_t size, FILE* file,
57 const std::string& filename) {
58 if (fwrite(data, size, 1, file) != 1) {
59 printf("Error when writing to %s\n", filename.c_str());
60 exit(1);
61 }
62 }
63
do_main(int argc,char * argv[])64 int do_main(int argc, char* argv[]) {
65 std::string program_name = argv[0];
66 std::string usage = "Commandline tool to unpack audioproc debug files.\n"
67 "Example usage:\n" + program_name + " debug_dump.pb\n";
68 google::SetUsageMessage(usage);
69 google::ParseCommandLineFlags(&argc, &argv, true);
70
71 if (argc < 2) {
72 printf("%s", google::ProgramUsage());
73 return 1;
74 }
75
76 FILE* debug_file = OpenFile(argv[1], "rb");
77
78 Event event_msg;
79 int frame_count = 0;
80 size_t reverse_samples_per_channel = 0;
81 size_t input_samples_per_channel = 0;
82 size_t output_samples_per_channel = 0;
83 size_t num_reverse_channels = 0;
84 size_t num_input_channels = 0;
85 size_t num_output_channels = 0;
86 rtc::scoped_ptr<WavWriter> reverse_wav_file;
87 rtc::scoped_ptr<WavWriter> input_wav_file;
88 rtc::scoped_ptr<WavWriter> output_wav_file;
89 rtc::scoped_ptr<RawFile> reverse_raw_file;
90 rtc::scoped_ptr<RawFile> input_raw_file;
91 rtc::scoped_ptr<RawFile> output_raw_file;
92
93 FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
94
95 while (ReadMessageFromFile(debug_file, &event_msg)) {
96 if (event_msg.type() == Event::REVERSE_STREAM) {
97 if (!event_msg.has_reverse_stream()) {
98 printf("Corrupt input file: ReverseStream missing.\n");
99 return 1;
100 }
101
102 const ReverseStream msg = event_msg.reverse_stream();
103 if (msg.has_data()) {
104 if (FLAGS_raw && !reverse_raw_file) {
105 reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
106 }
107 // TODO(aluebs): Replace "num_reverse_channels *
108 // reverse_samples_per_channel" with "msg.data().size() /
109 // sizeof(int16_t)" and so on when this fix in audio_processing has made
110 // it into stable: https://webrtc-codereview.appspot.com/15299004/
111 WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
112 num_reverse_channels * reverse_samples_per_channel,
113 reverse_wav_file.get(),
114 reverse_raw_file.get());
115 } else if (msg.channel_size() > 0) {
116 if (FLAGS_raw && !reverse_raw_file) {
117 reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
118 }
119 rtc::scoped_ptr<const float* []> data(
120 new const float* [num_reverse_channels]);
121 for (size_t i = 0; i < num_reverse_channels; ++i) {
122 data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
123 }
124 WriteFloatData(data.get(),
125 reverse_samples_per_channel,
126 num_reverse_channels,
127 reverse_wav_file.get(),
128 reverse_raw_file.get());
129 }
130 } else if (event_msg.type() == Event::STREAM) {
131 frame_count++;
132 if (!event_msg.has_stream()) {
133 printf("Corrupt input file: Stream missing.\n");
134 return 1;
135 }
136
137 const Stream msg = event_msg.stream();
138 if (msg.has_input_data()) {
139 if (FLAGS_raw && !input_raw_file) {
140 input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
141 }
142 WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
143 num_input_channels * input_samples_per_channel,
144 input_wav_file.get(),
145 input_raw_file.get());
146 } else if (msg.input_channel_size() > 0) {
147 if (FLAGS_raw && !input_raw_file) {
148 input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
149 }
150 rtc::scoped_ptr<const float* []> data(
151 new const float* [num_input_channels]);
152 for (size_t i = 0; i < num_input_channels; ++i) {
153 data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
154 }
155 WriteFloatData(data.get(),
156 input_samples_per_channel,
157 num_input_channels,
158 input_wav_file.get(),
159 input_raw_file.get());
160 }
161
162 if (msg.has_output_data()) {
163 if (FLAGS_raw && !output_raw_file) {
164 output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
165 }
166 WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
167 num_output_channels * output_samples_per_channel,
168 output_wav_file.get(),
169 output_raw_file.get());
170 } else if (msg.output_channel_size() > 0) {
171 if (FLAGS_raw && !output_raw_file) {
172 output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
173 }
174 rtc::scoped_ptr<const float* []> data(
175 new const float* [num_output_channels]);
176 for (size_t i = 0; i < num_output_channels; ++i) {
177 data[i] =
178 reinterpret_cast<const float*>(msg.output_channel(i).data());
179 }
180 WriteFloatData(data.get(),
181 output_samples_per_channel,
182 num_output_channels,
183 output_wav_file.get(),
184 output_raw_file.get());
185 }
186
187 if (FLAGS_full) {
188 if (msg.has_delay()) {
189 static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
190 int32_t delay = msg.delay();
191 if (FLAGS_text) {
192 fprintf(delay_file, "%d\n", delay);
193 } else {
194 WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
195 }
196 }
197
198 if (msg.has_drift()) {
199 static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
200 int32_t drift = msg.drift();
201 if (FLAGS_text) {
202 fprintf(drift_file, "%d\n", drift);
203 } else {
204 WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
205 }
206 }
207
208 if (msg.has_level()) {
209 static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
210 int32_t level = msg.level();
211 if (FLAGS_text) {
212 fprintf(level_file, "%d\n", level);
213 } else {
214 WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
215 }
216 }
217
218 if (msg.has_keypress()) {
219 static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
220 bool keypress = msg.keypress();
221 if (FLAGS_text) {
222 fprintf(keypress_file, "%d\n", keypress);
223 } else {
224 WriteData(&keypress, sizeof(keypress), keypress_file,
225 FLAGS_keypress_file);
226 }
227 }
228 }
229 } else if (event_msg.type() == Event::CONFIG) {
230 if (!event_msg.has_config()) {
231 printf("Corrupt input file: Config missing.\n");
232 return 1;
233 }
234 const audioproc::Config msg = event_msg.config();
235
236 fprintf(settings_file, "APM re-config at frame: %d\n", frame_count);
237
238 PRINT_CONFIG(aec_enabled);
239 PRINT_CONFIG(aec_delay_agnostic_enabled);
240 PRINT_CONFIG(aec_drift_compensation_enabled);
241 PRINT_CONFIG(aec_extended_filter_enabled);
242 PRINT_CONFIG(aec_suppression_level);
243 PRINT_CONFIG(aecm_enabled);
244 PRINT_CONFIG(aecm_comfort_noise_enabled);
245 PRINT_CONFIG(aecm_routing_mode);
246 PRINT_CONFIG(agc_enabled);
247 PRINT_CONFIG(agc_mode);
248 PRINT_CONFIG(agc_limiter_enabled);
249 PRINT_CONFIG(noise_robust_agc_enabled);
250 PRINT_CONFIG(hpf_enabled);
251 PRINT_CONFIG(ns_enabled);
252 PRINT_CONFIG(ns_level);
253 PRINT_CONFIG(transient_suppression_enabled);
254 } else if (event_msg.type() == Event::INIT) {
255 if (!event_msg.has_init()) {
256 printf("Corrupt input file: Init missing.\n");
257 return 1;
258 }
259
260 const Init msg = event_msg.init();
261 // These should print out zeros if they're missing.
262 fprintf(settings_file, "Init at frame: %d\n", frame_count);
263 int input_sample_rate = msg.sample_rate();
264 fprintf(settings_file, " Input sample rate: %d\n", input_sample_rate);
265 int output_sample_rate = msg.output_sample_rate();
266 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate);
267 int reverse_sample_rate = msg.reverse_sample_rate();
268 fprintf(settings_file,
269 " Reverse sample rate: %d\n",
270 reverse_sample_rate);
271 num_input_channels = msg.num_input_channels();
272 fprintf(settings_file, " Input channels: %" PRIuS "\n",
273 num_input_channels);
274 num_output_channels = msg.num_output_channels();
275 fprintf(settings_file, " Output channels: %" PRIuS "\n",
276 num_output_channels);
277 num_reverse_channels = msg.num_reverse_channels();
278 fprintf(settings_file, " Reverse channels: %" PRIuS "\n",
279 num_reverse_channels);
280
281 fprintf(settings_file, "\n");
282
283 if (reverse_sample_rate == 0) {
284 reverse_sample_rate = input_sample_rate;
285 }
286 if (output_sample_rate == 0) {
287 output_sample_rate = input_sample_rate;
288 }
289
290 reverse_samples_per_channel =
291 static_cast<size_t>(reverse_sample_rate / 100);
292 input_samples_per_channel =
293 static_cast<size_t>(input_sample_rate / 100);
294 output_samples_per_channel =
295 static_cast<size_t>(output_sample_rate / 100);
296
297 if (!FLAGS_raw) {
298 // The WAV files need to be reset every time, because they cant change
299 // their sample rate or number of channels.
300 reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav",
301 reverse_sample_rate,
302 num_reverse_channels));
303 input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav",
304 input_sample_rate,
305 num_input_channels));
306 output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav",
307 output_sample_rate,
308 num_output_channels));
309 }
310 }
311 }
312
313 return 0;
314 }
315
316 } // namespace webrtc
317
main(int argc,char * argv[])318 int main(int argc, char* argv[]) {
319 return webrtc::do_main(argc, argv);
320 }
321