| /* | |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| // Commandline tool to unpack audioproc debug files. | |
| // | |
| // The debug files are dumped as protobuf blobs. For analysis, it's necessary | |
| // to unpack the file into its component parts: audio and other data. | |
| #include <stdio.h> | |
| #include "gflags/gflags.h" | |
| #include "webrtc/audio_processing/debug.pb.h" | |
| #include "webrtc/base/scoped_ptr.h" | |
| #include "webrtc/modules/audio_processing/test/protobuf_utils.h" | |
| #include "webrtc/modules/audio_processing/test/test_utils.h" | |
| #include "webrtc/typedefs.h" | |
| // TODO(andrew): unpack more of the data. | |
| DEFINE_string(input_file, "input", "The name of the input stream file."); | |
| DEFINE_string(output_file, "ref_out", | |
| "The name of the reference output stream file."); | |
| DEFINE_string(reverse_file, "reverse", | |
| "The name of the reverse input stream file."); | |
| DEFINE_string(delay_file, "delay.int32", "The name of the delay file."); | |
| DEFINE_string(drift_file, "drift.int32", "The name of the drift file."); | |
| DEFINE_string(level_file, "level.int32", "The name of the level file."); | |
| DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file."); | |
| DEFINE_string(settings_file, "settings.txt", "The name of the settings file."); | |
| DEFINE_bool(full, false, | |
| "Unpack the full set of files (normally not needed)."); | |
| DEFINE_bool(raw, false, "Write raw data instead of a WAV file."); | |
| DEFINE_bool(text, | |
| false, | |
| "Write non-audio files as text files instead of binary files."); | |
| namespace webrtc { | |
| using audioproc::Event; | |
| using audioproc::ReverseStream; | |
| using audioproc::Stream; | |
| using audioproc::Init; | |
| void WriteData(const void* data, size_t size, FILE* file, | |
| const std::string& filename) { | |
| if (fwrite(data, size, 1, file) != 1) { | |
| printf("Error when writing to %s\n", filename.c_str()); | |
| exit(1); | |
| } | |
| } | |
| int do_main(int argc, char* argv[]) { | |
| std::string program_name = argv[0]; | |
| std::string usage = "Commandline tool to unpack audioproc debug files.\n" | |
| "Example usage:\n" + program_name + " debug_dump.pb\n"; | |
| google::SetUsageMessage(usage); | |
| google::ParseCommandLineFlags(&argc, &argv, true); | |
| if (argc < 2) { | |
| printf("%s", google::ProgramUsage()); | |
| return 1; | |
| } | |
| FILE* debug_file = OpenFile(argv[1], "rb"); | |
| Event event_msg; | |
| int frame_count = 0; | |
| int reverse_samples_per_channel = 0; | |
| int input_samples_per_channel = 0; | |
| int output_samples_per_channel = 0; | |
| int num_reverse_channels = 0; | |
| int num_input_channels = 0; | |
| int num_output_channels = 0; | |
| rtc::scoped_ptr<WavWriter> reverse_wav_file; | |
| rtc::scoped_ptr<WavWriter> input_wav_file; | |
| rtc::scoped_ptr<WavWriter> output_wav_file; | |
| rtc::scoped_ptr<RawFile> reverse_raw_file; | |
| rtc::scoped_ptr<RawFile> input_raw_file; | |
| rtc::scoped_ptr<RawFile> output_raw_file; | |
| while (ReadMessageFromFile(debug_file, &event_msg)) { | |
| if (event_msg.type() == Event::REVERSE_STREAM) { | |
| if (!event_msg.has_reverse_stream()) { | |
| printf("Corrupt input file: ReverseStream missing.\n"); | |
| return 1; | |
| } | |
| const ReverseStream msg = event_msg.reverse_stream(); | |
| if (msg.has_data()) { | |
| if (FLAGS_raw && !reverse_raw_file) { | |
| reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm")); | |
| } | |
| // TODO(aluebs): Replace "num_reverse_channels * | |
| // reverse_samples_per_channel" with "msg.data().size() / | |
| // sizeof(int16_t)" and so on when this fix in audio_processing has made | |
| // it into stable: https://webrtc-codereview.appspot.com/15299004/ | |
| WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()), | |
| num_reverse_channels * reverse_samples_per_channel, | |
| reverse_wav_file.get(), | |
| reverse_raw_file.get()); | |
| } else if (msg.channel_size() > 0) { | |
| if (FLAGS_raw && !reverse_raw_file) { | |
| reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float")); | |
| } | |
| rtc::scoped_ptr<const float* []> data( | |
| new const float* [num_reverse_channels]); | |
| for (int i = 0; i < num_reverse_channels; ++i) { | |
| data[i] = reinterpret_cast<const float*>(msg.channel(i).data()); | |
| } | |
| WriteFloatData(data.get(), | |
| reverse_samples_per_channel, | |
| num_reverse_channels, | |
| reverse_wav_file.get(), | |
| reverse_raw_file.get()); | |
| } | |
| } else if (event_msg.type() == Event::STREAM) { | |
| frame_count++; | |
| if (!event_msg.has_stream()) { | |
| printf("Corrupt input file: Stream missing.\n"); | |
| return 1; | |
| } | |
| const Stream msg = event_msg.stream(); | |
| if (msg.has_input_data()) { | |
| if (FLAGS_raw && !input_raw_file) { | |
| input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm")); | |
| } | |
| WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()), | |
| num_input_channels * input_samples_per_channel, | |
| input_wav_file.get(), | |
| input_raw_file.get()); | |
| } else if (msg.input_channel_size() > 0) { | |
| if (FLAGS_raw && !input_raw_file) { | |
| input_raw_file.reset(new RawFile(FLAGS_input_file + ".float")); | |
| } | |
| rtc::scoped_ptr<const float* []> data( | |
| new const float* [num_input_channels]); | |
| for (int i = 0; i < num_input_channels; ++i) { | |
| data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data()); | |
| } | |
| WriteFloatData(data.get(), | |
| input_samples_per_channel, | |
| num_input_channels, | |
| input_wav_file.get(), | |
| input_raw_file.get()); | |
| } | |
| if (msg.has_output_data()) { | |
| if (FLAGS_raw && !output_raw_file) { | |
| output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm")); | |
| } | |
| WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()), | |
| num_output_channels * output_samples_per_channel, | |
| output_wav_file.get(), | |
| output_raw_file.get()); | |
| } else if (msg.output_channel_size() > 0) { | |
| if (FLAGS_raw && !output_raw_file) { | |
| output_raw_file.reset(new RawFile(FLAGS_output_file + ".float")); | |
| } | |
| rtc::scoped_ptr<const float* []> data( | |
| new const float* [num_output_channels]); | |
| for (int i = 0; i < num_output_channels; ++i) { | |
| data[i] = | |
| reinterpret_cast<const float*>(msg.output_channel(i).data()); | |
| } | |
| WriteFloatData(data.get(), | |
| output_samples_per_channel, | |
| num_output_channels, | |
| output_wav_file.get(), | |
| output_raw_file.get()); | |
| } | |
| if (FLAGS_full) { | |
| if (msg.has_delay()) { | |
| static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb"); | |
| int32_t delay = msg.delay(); | |
| if (FLAGS_text) { | |
| fprintf(delay_file, "%d\n", delay); | |
| } else { | |
| WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file); | |
| } | |
| } | |
| if (msg.has_drift()) { | |
| static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb"); | |
| int32_t drift = msg.drift(); | |
| if (FLAGS_text) { | |
| fprintf(drift_file, "%d\n", drift); | |
| } else { | |
| WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file); | |
| } | |
| } | |
| if (msg.has_level()) { | |
| static FILE* level_file = OpenFile(FLAGS_level_file, "wb"); | |
| int32_t level = msg.level(); | |
| if (FLAGS_text) { | |
| fprintf(level_file, "%d\n", level); | |
| } else { | |
| WriteData(&level, sizeof(level), level_file, FLAGS_level_file); | |
| } | |
| } | |
| if (msg.has_keypress()) { | |
| static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb"); | |
| bool keypress = msg.keypress(); | |
| if (FLAGS_text) { | |
| fprintf(keypress_file, "%d\n", keypress); | |
| } else { | |
| WriteData(&keypress, sizeof(keypress), keypress_file, | |
| FLAGS_keypress_file); | |
| } | |
| } | |
| } | |
| } else if (event_msg.type() == Event::INIT) { | |
| if (!event_msg.has_init()) { | |
| printf("Corrupt input file: Init missing.\n"); | |
| return 1; | |
| } | |
| static FILE* settings_file = OpenFile(FLAGS_settings_file, "wb"); | |
| const Init msg = event_msg.init(); | |
| // These should print out zeros if they're missing. | |
| fprintf(settings_file, "Init at frame: %d\n", frame_count); | |
| int input_sample_rate = msg.sample_rate(); | |
| fprintf(settings_file, " Input sample rate: %d\n", input_sample_rate); | |
| int output_sample_rate = msg.output_sample_rate(); | |
| fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate); | |
| int reverse_sample_rate = msg.reverse_sample_rate(); | |
| fprintf(settings_file, | |
| " Reverse sample rate: %d\n", | |
| reverse_sample_rate); | |
| num_input_channels = msg.num_input_channels(); | |
| fprintf(settings_file, " Input channels: %d\n", num_input_channels); | |
| num_output_channels = msg.num_output_channels(); | |
| fprintf(settings_file, " Output channels: %d\n", num_output_channels); | |
| num_reverse_channels = msg.num_reverse_channels(); | |
| fprintf(settings_file, " Reverse channels: %d\n", num_reverse_channels); | |
| fprintf(settings_file, "\n"); | |
| if (reverse_sample_rate == 0) { | |
| reverse_sample_rate = input_sample_rate; | |
| } | |
| if (output_sample_rate == 0) { | |
| output_sample_rate = input_sample_rate; | |
| } | |
| reverse_samples_per_channel = reverse_sample_rate / 100; | |
| input_samples_per_channel = input_sample_rate / 100; | |
| output_samples_per_channel = output_sample_rate / 100; | |
| if (!FLAGS_raw) { | |
| // The WAV files need to be reset every time, because they cant change | |
| // their sample rate or number of channels. | |
| reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav", | |
| reverse_sample_rate, | |
| num_reverse_channels)); | |
| input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav", | |
| input_sample_rate, | |
| num_input_channels)); | |
| output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav", | |
| output_sample_rate, | |
| num_output_channels)); | |
| } | |
| } | |
| } | |
| return 0; | |
| } | |
| } // namespace webrtc | |
| int main(int argc, char* argv[]) { | |
| return webrtc::do_main(argc, argv); | |
| } |