| /* |
| * 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 <inttypes.h> |
| #include <stdint.h> |
| #include <stdio.h> |
| #include <stdlib.h> |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/flags/flag.h" |
| #include "absl/flags/parse.h" |
| #include "api/function_view.h" |
| #include "common_audio/wav_file.h" |
| #include "modules/audio_processing/test/protobuf_utils.h" |
| #include "modules/audio_processing/test/test_utils.h" |
| #include "rtc_base/format_macros.h" |
| #include "rtc_base/ignore_wundef.h" |
| #include "rtc_base/strings/string_builder.h" |
| |
| RTC_PUSH_IGNORING_WUNDEF() |
| #include "modules/audio_processing/debug.pb.h" |
| RTC_POP_IGNORING_WUNDEF() |
| |
| ABSL_FLAG(std::string, |
| input_file, |
| "input", |
| "The name of the input stream file."); |
| ABSL_FLAG(std::string, |
| output_file, |
| "ref_out", |
| "The name of the reference output stream file."); |
| ABSL_FLAG(std::string, |
| reverse_file, |
| "reverse", |
| "The name of the reverse input stream file."); |
| ABSL_FLAG(std::string, |
| delay_file, |
| "delay.int32", |
| "The name of the delay file."); |
| ABSL_FLAG(std::string, |
| drift_file, |
| "drift.int32", |
| "The name of the drift file."); |
| ABSL_FLAG(std::string, |
| level_file, |
| "level.int32", |
| "The name of the level file."); |
| ABSL_FLAG(std::string, |
| keypress_file, |
| "keypress.bool", |
| "The name of the keypress file."); |
| ABSL_FLAG(std::string, |
| callorder_file, |
| "callorder", |
| "The name of the render/capture call order file."); |
| ABSL_FLAG(std::string, |
| settings_file, |
| "settings.txt", |
| "The name of the settings file."); |
| ABSL_FLAG(bool, |
| full, |
| false, |
| "Unpack the full set of files (normally not needed)."); |
| ABSL_FLAG(bool, raw, false, "Write raw data instead of a WAV file."); |
| ABSL_FLAG(bool, |
| text, |
| false, |
| "Write non-audio files as text files instead of binary files."); |
| |
| #define PRINT_CONFIG(field_name) \ |
| if (msg.has_##field_name()) { \ |
| fprintf(settings_file, " " #field_name ": %d\n", msg.field_name()); \ |
| } |
| |
| #define PRINT_CONFIG_FLOAT(field_name) \ |
| if (msg.has_##field_name()) { \ |
| fprintf(settings_file, " " #field_name ": %f\n", msg.field_name()); \ |
| } |
| |
| namespace webrtc { |
| |
| using audioproc::Event; |
| using audioproc::Init; |
| using audioproc::ReverseStream; |
| using audioproc::Stream; |
| |
| namespace { |
| |
| 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); |
| } |
| } |
| |
| void WriteCallOrderData(const bool render_call, |
| FILE* file, |
| const std::string& filename) { |
| const char call_type = render_call ? 'r' : 'c'; |
| WriteData(&call_type, sizeof(call_type), file, filename.c_str()); |
| } |
| |
| bool WritingCallOrderFile() { |
| return absl::GetFlag(FLAGS_full); |
| } |
| |
| bool WritingRuntimeSettingFiles() { |
| return absl::GetFlag(FLAGS_full); |
| } |
| |
| // Exports RuntimeSetting AEC dump events to Audacity-readable files. |
| // This class is not RAII compliant. |
| class RuntimeSettingWriter { |
| public: |
| RuntimeSettingWriter( |
| std::string name, |
| rtc::FunctionView<bool(const Event)> is_exporter_for, |
| rtc::FunctionView<std::string(const Event)> get_timeline_label) |
| : setting_name_(std::move(name)), |
| is_exporter_for_(is_exporter_for), |
| get_timeline_label_(get_timeline_label) {} |
| ~RuntimeSettingWriter() { Flush(); } |
| |
| bool IsExporterFor(const Event& event) const { |
| return is_exporter_for_(event); |
| } |
| |
| // Writes to file the payload of |event| using |frame_count| to calculate |
| // timestamp. |
| void WriteEvent(const Event& event, int frame_count) { |
| RTC_DCHECK(is_exporter_for_(event)); |
| if (file_ == nullptr) { |
| rtc::StringBuilder file_name; |
| file_name << setting_name_ << frame_offset_ << ".txt"; |
| file_ = OpenFile(file_name.str(), "wb"); |
| } |
| |
| // Time in the current WAV file, in seconds. |
| double time = (frame_count - frame_offset_) / 100.0; |
| std::string label = get_timeline_label_(event); |
| // In Audacity, all annotations are encoded as intervals. |
| fprintf(file_, "%.6f\t%.6f\t%s \n", time, time, label.c_str()); |
| } |
| |
| // Handles an AEC dump initialization event, occurring at frame |
| // |frame_offset|. |
| void HandleInitEvent(int frame_offset) { |
| Flush(); |
| frame_offset_ = frame_offset; |
| } |
| |
| private: |
| void Flush() { |
| if (file_ != nullptr) { |
| fclose(file_); |
| file_ = nullptr; |
| } |
| } |
| |
| FILE* file_ = nullptr; |
| int frame_offset_ = 0; |
| const std::string setting_name_; |
| const rtc::FunctionView<bool(Event)> is_exporter_for_; |
| const rtc::FunctionView<std::string(Event)> get_timeline_label_; |
| }; |
| |
| // Returns RuntimeSetting exporters for runtime setting types defined in |
| // debug.proto. |
| std::vector<RuntimeSettingWriter> RuntimeSettingWriters() { |
| return { |
| RuntimeSettingWriter( |
| "CapturePreGain", |
| [](const Event& event) -> bool { |
| return event.runtime_setting().has_capture_pre_gain(); |
| }, |
| [](const Event& event) -> std::string { |
| return std::to_string(event.runtime_setting().capture_pre_gain()); |
| }), |
| RuntimeSettingWriter( |
| "CustomRenderProcessingRuntimeSetting", |
| [](const Event& event) -> bool { |
| return event.runtime_setting() |
| .has_custom_render_processing_setting(); |
| }, |
| [](const Event& event) -> std::string { |
| return std::to_string( |
| event.runtime_setting().custom_render_processing_setting()); |
| }), |
| RuntimeSettingWriter( |
| "CaptureFixedPostGain", |
| [](const Event& event) -> bool { |
| return event.runtime_setting().has_capture_fixed_post_gain(); |
| }, |
| [](const Event& event) -> std::string { |
| return std::to_string( |
| event.runtime_setting().capture_fixed_post_gain()); |
| }), |
| RuntimeSettingWriter( |
| "PlayoutVolumeChange", |
| [](const Event& event) -> bool { |
| return event.runtime_setting().has_playout_volume_change(); |
| }, |
| [](const Event& event) -> std::string { |
| return std::to_string( |
| event.runtime_setting().playout_volume_change()); |
| })}; |
| } |
| |
| } // namespace |
| |
| int do_main(int argc, char* argv[]) { |
| std::vector<char*> args = absl::ParseCommandLine(argc, argv); |
| std::string program_name = args[0]; |
| std::string usage = |
| "Commandline tool to unpack audioproc debug files.\n" |
| "Example usage:\n" + |
| program_name + " debug_dump.pb\n"; |
| |
| if (args.size() < 2) { |
| printf("%s", usage.c_str()); |
| return 1; |
| } |
| |
| FILE* debug_file = OpenFile(args[1], "rb"); |
| |
| Event event_msg; |
| int frame_count = 0; |
| size_t reverse_samples_per_channel = 0; |
| size_t input_samples_per_channel = 0; |
| size_t output_samples_per_channel = 0; |
| size_t num_reverse_channels = 0; |
| size_t num_input_channels = 0; |
| size_t num_output_channels = 0; |
| std::unique_ptr<WavWriter> reverse_wav_file; |
| std::unique_ptr<WavWriter> input_wav_file; |
| std::unique_ptr<WavWriter> output_wav_file; |
| std::unique_ptr<RawFile> reverse_raw_file; |
| std::unique_ptr<RawFile> input_raw_file; |
| std::unique_ptr<RawFile> output_raw_file; |
| |
| rtc::StringBuilder callorder_raw_name; |
| callorder_raw_name << absl::GetFlag(FLAGS_callorder_file) << ".char"; |
| FILE* callorder_char_file = WritingCallOrderFile() |
| ? OpenFile(callorder_raw_name.str(), "wb") |
| : nullptr; |
| FILE* settings_file = OpenFile(absl::GetFlag(FLAGS_settings_file), "wb"); |
| |
| std::vector<RuntimeSettingWriter> runtime_setting_writers = |
| RuntimeSettingWriters(); |
| |
| 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 (absl::GetFlag(FLAGS_raw) && !reverse_raw_file) { |
| reverse_raw_file.reset( |
| new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !reverse_raw_file) { |
| reverse_raw_file.reset( |
| new RawFile(absl::GetFlag(FLAGS_reverse_file) + ".float")); |
| } |
| std::unique_ptr<const float*[]> data( |
| new const float*[num_reverse_channels]); |
| for (size_t 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()); |
| } |
| if (absl::GetFlag(FLAGS_full)) { |
| if (WritingCallOrderFile()) { |
| WriteCallOrderData(true /* render_call */, callorder_char_file, |
| absl::GetFlag(FLAGS_callorder_file)); |
| } |
| } |
| } 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 (absl::GetFlag(FLAGS_raw) && !input_raw_file) { |
| input_raw_file.reset( |
| new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !input_raw_file) { |
| input_raw_file.reset( |
| new RawFile(absl::GetFlag(FLAGS_input_file) + ".float")); |
| } |
| std::unique_ptr<const float*[]> data( |
| new const float*[num_input_channels]); |
| for (size_t 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 (absl::GetFlag(FLAGS_raw) && !output_raw_file) { |
| output_raw_file.reset( |
| new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !output_raw_file) { |
| output_raw_file.reset( |
| new RawFile(absl::GetFlag(FLAGS_output_file) + ".float")); |
| } |
| std::unique_ptr<const float*[]> data( |
| new const float*[num_output_channels]); |
| for (size_t 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 (absl::GetFlag(FLAGS_full)) { |
| if (WritingCallOrderFile()) { |
| WriteCallOrderData(false /* render_call */, callorder_char_file, |
| absl::GetFlag(FLAGS_callorder_file)); |
| } |
| if (msg.has_delay()) { |
| static FILE* delay_file = |
| OpenFile(absl::GetFlag(FLAGS_delay_file), "wb"); |
| int32_t delay = msg.delay(); |
| if (absl::GetFlag(FLAGS_text)) { |
| fprintf(delay_file, "%d\n", delay); |
| } else { |
| WriteData(&delay, sizeof(delay), delay_file, |
| absl::GetFlag(FLAGS_delay_file)); |
| } |
| } |
| |
| if (msg.has_drift()) { |
| static FILE* drift_file = |
| OpenFile(absl::GetFlag(FLAGS_drift_file), "wb"); |
| int32_t drift = msg.drift(); |
| if (absl::GetFlag(FLAGS_text)) { |
| fprintf(drift_file, "%d\n", drift); |
| } else { |
| WriteData(&drift, sizeof(drift), drift_file, |
| absl::GetFlag(FLAGS_drift_file)); |
| } |
| } |
| |
| if (msg.has_level()) { |
| static FILE* level_file = |
| OpenFile(absl::GetFlag(FLAGS_level_file), "wb"); |
| int32_t level = msg.level(); |
| if (absl::GetFlag(FLAGS_text)) { |
| fprintf(level_file, "%d\n", level); |
| } else { |
| WriteData(&level, sizeof(level), level_file, |
| absl::GetFlag(FLAGS_level_file)); |
| } |
| } |
| |
| if (msg.has_keypress()) { |
| static FILE* keypress_file = |
| OpenFile(absl::GetFlag(FLAGS_keypress_file), "wb"); |
| bool keypress = msg.keypress(); |
| if (absl::GetFlag(FLAGS_text)) { |
| fprintf(keypress_file, "%d\n", keypress); |
| } else { |
| WriteData(&keypress, sizeof(keypress), keypress_file, |
| absl::GetFlag(FLAGS_keypress_file)); |
| } |
| } |
| } |
| } else if (event_msg.type() == Event::CONFIG) { |
| if (!event_msg.has_config()) { |
| printf("Corrupt input file: Config missing.\n"); |
| return 1; |
| } |
| const audioproc::Config msg = event_msg.config(); |
| |
| fprintf(settings_file, "APM re-config at frame: %d\n", frame_count); |
| |
| PRINT_CONFIG(aec_enabled); |
| PRINT_CONFIG(aec_delay_agnostic_enabled); |
| PRINT_CONFIG(aec_drift_compensation_enabled); |
| PRINT_CONFIG(aec_extended_filter_enabled); |
| PRINT_CONFIG(aec_suppression_level); |
| PRINT_CONFIG(aecm_enabled); |
| PRINT_CONFIG(aecm_comfort_noise_enabled); |
| PRINT_CONFIG(aecm_routing_mode); |
| PRINT_CONFIG(agc_enabled); |
| PRINT_CONFIG(agc_mode); |
| PRINT_CONFIG(agc_limiter_enabled); |
| PRINT_CONFIG(noise_robust_agc_enabled); |
| PRINT_CONFIG(hpf_enabled); |
| PRINT_CONFIG(ns_enabled); |
| PRINT_CONFIG(ns_level); |
| PRINT_CONFIG(transient_suppression_enabled); |
| PRINT_CONFIG(pre_amplifier_enabled); |
| PRINT_CONFIG_FLOAT(pre_amplifier_fixed_gain_factor); |
| |
| if (msg.has_experiments_description()) { |
| fprintf(settings_file, " experiments_description: %s\n", |
| msg.experiments_description().c_str()); |
| } |
| } else if (event_msg.type() == Event::INIT) { |
| if (!event_msg.has_init()) { |
| printf("Corrupt input file: Init missing.\n"); |
| return 1; |
| } |
| |
| 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: %" RTC_PRIuS "\n", |
| num_input_channels); |
| num_output_channels = msg.num_output_channels(); |
| fprintf(settings_file, " Output channels: %" RTC_PRIuS "\n", |
| num_output_channels); |
| num_reverse_channels = msg.num_reverse_channels(); |
| fprintf(settings_file, " Reverse channels: %" RTC_PRIuS "\n", |
| num_reverse_channels); |
| if (msg.has_timestamp_ms()) { |
| const int64_t timestamp = msg.timestamp_ms(); |
| fprintf(settings_file, " Timestamp in millisecond: %" PRId64 "\n", |
| timestamp); |
| } |
| |
| 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 = |
| static_cast<size_t>(reverse_sample_rate / 100); |
| input_samples_per_channel = static_cast<size_t>(input_sample_rate / 100); |
| output_samples_per_channel = |
| static_cast<size_t>(output_sample_rate / 100); |
| |
| if (!absl::GetFlag(FLAGS_raw)) { |
| // The WAV files need to be reset every time, because they cant change |
| // their sample rate or number of channels. |
| rtc::StringBuilder reverse_name; |
| reverse_name << absl::GetFlag(FLAGS_reverse_file) << frame_count |
| << ".wav"; |
| reverse_wav_file.reset(new WavWriter( |
| reverse_name.str(), reverse_sample_rate, num_reverse_channels)); |
| rtc::StringBuilder input_name; |
| input_name << absl::GetFlag(FLAGS_input_file) << frame_count << ".wav"; |
| input_wav_file.reset(new WavWriter(input_name.str(), input_sample_rate, |
| num_input_channels)); |
| rtc::StringBuilder output_name; |
| output_name << absl::GetFlag(FLAGS_output_file) << frame_count |
| << ".wav"; |
| output_wav_file.reset(new WavWriter( |
| output_name.str(), output_sample_rate, num_output_channels)); |
| |
| if (WritingCallOrderFile()) { |
| rtc::StringBuilder callorder_name; |
| callorder_name << absl::GetFlag(FLAGS_callorder_file) << frame_count |
| << ".char"; |
| callorder_char_file = OpenFile(callorder_name.str(), "wb"); |
| } |
| |
| if (WritingRuntimeSettingFiles()) { |
| for (RuntimeSettingWriter& writer : runtime_setting_writers) { |
| writer.HandleInitEvent(frame_count); |
| } |
| } |
| } |
| } else if (event_msg.type() == Event::RUNTIME_SETTING) { |
| if (WritingRuntimeSettingFiles()) { |
| for (RuntimeSettingWriter& writer : runtime_setting_writers) { |
| if (writer.IsExporterFor(event_msg)) { |
| writer.WriteEvent(event_msg, frame_count); |
| } |
| } |
| } |
| } |
| } |
| |
| return 0; |
| } |
| |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| return webrtc::do_main(argc, argv); |
| } |
