2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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.
11 // Commandline tool to unpack audioproc debug files.
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.
18 #include "gflags/gflags.h"
19 #include "webrtc/audio_processing/debug.pb.h"
20 #include "webrtc/modules/audio_processing/test/test_utils.h"
21 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
22 #include "webrtc/typedefs.h"
24 // TODO(andrew): unpack more of the data.
25 DEFINE_string(input_file, "input", "The name of the input stream file.");
26 DEFINE_string(output_file, "ref_out",
27 "The name of the reference output stream file.");
28 DEFINE_string(reverse_file, "reverse",
29 "The name of the reverse input stream file.");
30 DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
31 DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
32 DEFINE_string(level_file, "level.int32", "The name of the level file.");
33 DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
34 DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
35 DEFINE_bool(full, false,
36 "Unpack the full set of files (normally not needed).");
37 DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
41 using audioproc::Event;
42 using audioproc::ReverseStream;
43 using audioproc::Stream;
44 using audioproc::Init;
46 void WriteData(const void* data, size_t size, FILE* file,
47 const std::string& filename) {
48 if (fwrite(data, size, 1, file) != 1) {
49 printf("Error when writing to %s\n", filename.c_str());
54 int do_main(int argc, char* argv[]) {
55 std::string program_name = argv[0];
56 std::string usage = "Commandline tool to unpack audioproc debug files.\n"
57 "Example usage:\n" + program_name + " debug_dump.pb\n";
58 google::SetUsageMessage(usage);
59 google::ParseCommandLineFlags(&argc, &argv, true);
62 printf("%s", google::ProgramUsage());
66 FILE* debug_file = OpenFile(argv[1], "rb");
70 int reverse_samples_per_channel = 0;
71 int input_samples_per_channel = 0;
72 int output_samples_per_channel = 0;
73 int num_reverse_channels = 0;
74 int num_input_channels = 0;
75 int num_output_channels = 0;
76 scoped_ptr<WavWriter> reverse_wav_file;
77 scoped_ptr<WavWriter> input_wav_file;
78 scoped_ptr<WavWriter> output_wav_file;
79 scoped_ptr<RawFile> reverse_raw_file;
80 scoped_ptr<RawFile> input_raw_file;
81 scoped_ptr<RawFile> output_raw_file;
82 while (ReadMessageFromFile(debug_file, &event_msg)) {
83 if (event_msg.type() == Event::REVERSE_STREAM) {
84 if (!event_msg.has_reverse_stream()) {
85 printf("Corrupt input file: ReverseStream missing.\n");
89 const ReverseStream msg = event_msg.reverse_stream();
91 if (FLAGS_raw && !reverse_raw_file) {
92 reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
94 // TODO(aluebs): Replace "num_reverse_channels *
95 // reverse_samples_per_channel" with "msg.data().size() /
96 // sizeof(int16_t)" and so on when this fix in audio_processing has made
97 // it into stable: https://webrtc-codereview.appspot.com/15299004/
98 WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
99 num_reverse_channels * reverse_samples_per_channel,
100 reverse_wav_file.get(),
101 reverse_raw_file.get());
102 } else if (msg.channel_size() > 0) {
103 if (FLAGS_raw && !reverse_raw_file) {
104 reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
106 scoped_ptr<const float*[]> data(new const float*[num_reverse_channels]);
107 for (int i = 0; i < num_reverse_channels; ++i) {
108 data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
110 WriteFloatData(data.get(),
111 reverse_samples_per_channel,
112 num_reverse_channels,
113 reverse_wav_file.get(),
114 reverse_raw_file.get());
116 } else if (event_msg.type() == Event::STREAM) {
118 if (!event_msg.has_stream()) {
119 printf("Corrupt input file: Stream missing.\n");
123 const Stream msg = event_msg.stream();
124 if (msg.has_input_data()) {
125 if (FLAGS_raw && !input_raw_file) {
126 input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
128 WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
129 num_input_channels * input_samples_per_channel,
130 input_wav_file.get(),
131 input_raw_file.get());
132 } else if (msg.input_channel_size() > 0) {
133 if (FLAGS_raw && !input_raw_file) {
134 input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
136 scoped_ptr<const float*[]> data(new const float*[num_input_channels]);
137 for (int i = 0; i < num_input_channels; ++i) {
138 data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
140 WriteFloatData(data.get(),
141 input_samples_per_channel,
143 input_wav_file.get(),
144 input_raw_file.get());
147 if (msg.has_output_data()) {
148 if (FLAGS_raw && !output_raw_file) {
149 output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
151 WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
152 num_output_channels * output_samples_per_channel,
153 output_wav_file.get(),
154 output_raw_file.get());
155 } else if (msg.output_channel_size() > 0) {
156 if (FLAGS_raw && !output_raw_file) {
157 output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
159 scoped_ptr<const float*[]> data(new const float*[num_output_channels]);
160 for (int i = 0; i < num_output_channels; ++i) {
162 reinterpret_cast<const float*>(msg.output_channel(i).data());
164 WriteFloatData(data.get(),
165 output_samples_per_channel,
167 output_wav_file.get(),
168 output_raw_file.get());
172 if (msg.has_delay()) {
173 static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
174 int32_t delay = msg.delay();
175 WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
178 if (msg.has_drift()) {
179 static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
180 int32_t drift = msg.drift();
181 WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
184 if (msg.has_level()) {
185 static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
186 int32_t level = msg.level();
187 WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
190 if (msg.has_keypress()) {
191 static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
192 bool keypress = msg.keypress();
193 WriteData(&keypress, sizeof(keypress), keypress_file,
194 FLAGS_keypress_file);
197 } else if (event_msg.type() == Event::INIT) {
198 if (!event_msg.has_init()) {
199 printf("Corrupt input file: Init missing.\n");
203 static FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
204 const Init msg = event_msg.init();
205 // These should print out zeros if they're missing.
206 fprintf(settings_file, "Init at frame: %d\n", frame_count);
207 int input_sample_rate = msg.sample_rate();
208 fprintf(settings_file, " Input sample rate: %d\n", input_sample_rate);
209 int output_sample_rate = msg.output_sample_rate();
210 fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate);
211 int reverse_sample_rate = msg.reverse_sample_rate();
212 fprintf(settings_file,
213 " Reverse sample rate: %d\n",
214 reverse_sample_rate);
215 num_input_channels = msg.num_input_channels();
216 fprintf(settings_file, " Input channels: %d\n", num_input_channels);
217 num_output_channels = msg.num_output_channels();
218 fprintf(settings_file, " Output channels: %d\n", num_output_channels);
219 num_reverse_channels = msg.num_reverse_channels();
220 fprintf(settings_file, " Reverse channels: %d\n", num_reverse_channels);
222 fprintf(settings_file, "\n");
224 if (reverse_sample_rate == 0) {
225 reverse_sample_rate = input_sample_rate;
227 if (output_sample_rate == 0) {
228 output_sample_rate = input_sample_rate;
231 reverse_samples_per_channel = reverse_sample_rate / 100;
232 input_samples_per_channel = input_sample_rate / 100;
233 output_samples_per_channel = output_sample_rate / 100;
236 // The WAV files need to be reset every time, because they cant change
237 // their sample rate or number of channels.
238 reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav",
240 num_reverse_channels));
241 input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav",
243 num_input_channels));
244 output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav",
246 num_output_channels));
254 } // namespace webrtc
256 int main(int argc, char* argv[]) {
257 return webrtc::do_main(argc, argv);