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-rw-r--r--third_party/libwebrtc/rtc_tools/unpack_aecdump/OWNERS3
-rw-r--r--third_party/libwebrtc/rtc_tools/unpack_aecdump/unpack.cc621
2 files changed, 624 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_tools/unpack_aecdump/OWNERS b/third_party/libwebrtc/rtc_tools/unpack_aecdump/OWNERS
new file mode 100644
index 0000000000..3ef412054e
--- /dev/null
+++ b/third_party/libwebrtc/rtc_tools/unpack_aecdump/OWNERS
@@ -0,0 +1,3 @@
+alessiob@webrtc.org
+peah@webrtc.org
+saza@webrtc.org
diff --git a/third_party/libwebrtc/rtc_tools/unpack_aecdump/unpack.cc b/third_party/libwebrtc/rtc_tools/unpack_aecdump/unpack.cc
new file mode 100644
index 0000000000..a43fe75b36
--- /dev/null
+++ b/third_party/libwebrtc/rtc_tools/unpack_aecdump/unpack.cc
@@ -0,0 +1,621 @@
+/*
+ * 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/include/audio_util.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.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 applied input volume 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.");
+ABSL_FLAG(bool,
+ use_init_suffix,
+ false,
+ "Use init index instead of capture frame count as file name suffix.");
+
+#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 {
+class RawFile final {
+ public:
+ explicit RawFile(const std::string& filename)
+ : file_handle_(fopen(filename.c_str(), "wb")) {}
+ ~RawFile() { fclose(file_handle_); }
+
+ RawFile(const RawFile&) = delete;
+ RawFile& operator=(const RawFile&) = delete;
+
+ void WriteSamples(const int16_t* samples, size_t num_samples) {
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert samples to little-endian when writing to PCM file"
+#endif
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+ }
+
+ void WriteSamples(const float* samples, size_t num_samples) {
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+ }
+
+ private:
+ FILE* file_handle_;
+};
+
+void WriteIntData(const int16_t* data,
+ size_t length,
+ WavWriter* wav_file,
+ RawFile* raw_file) {
+ if (wav_file) {
+ wav_file->WriteSamples(data, length);
+ }
+ if (raw_file) {
+ raw_file->WriteSamples(data, length);
+ }
+}
+
+void WriteFloatData(const float* const* data,
+ size_t samples_per_channel,
+ size_t num_channels,
+ WavWriter* wav_file,
+ RawFile* raw_file) {
+ size_t length = num_channels * samples_per_channel;
+ std::unique_ptr<float[]> buffer(new float[length]);
+ Interleave(data, samples_per_channel, num_channels, buffer.get());
+ if (raw_file) {
+ raw_file->WriteSamples(buffer.get(), length);
+ }
+ // TODO(aluebs): Use ScaleToInt16Range() from audio_util
+ for (size_t i = 0; i < length; ++i) {
+ buffer[i] = buffer[i] > 0
+ ? buffer[i] * std::numeric_limits<int16_t>::max()
+ : -buffer[i] * std::numeric_limits<int16_t>::min();
+ }
+ if (wav_file) {
+ wav_file->WriteSamples(buffer.get(), length);
+ }
+}
+
+// Exits on failure; do not use in unit tests.
+FILE* OpenFile(const std::string& filename, const char* mode) {
+ FILE* file = fopen(filename.c_str(), mode);
+ RTC_CHECK(file) << "Unable to open file " << filename;
+ return file;
+}
+
+void WriteData(const void* data,
+ size_t size,
+ FILE* file,
+ const std::string& filename) {
+ RTC_CHECK_EQ(fwrite(data, size, 1, file), 1)
+ << "Error when writing to " << filename.c_str();
+}
+
+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());
+ })};
+}
+
+std::string GetWavFileIndex(int init_index, int frame_count) {
+ rtc::StringBuilder suffix;
+ if (absl::GetFlag(FLAGS_use_init_suffix)) {
+ suffix << "_" << init_index;
+ } else {
+ suffix << frame_count;
+ }
+ return suffix.str();
+}
+
+} // 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;
+ int init_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_applied_input_volume()) {
+ static FILE* level_file =
+ OpenFile(absl::GetFlag(FLAGS_level_file), "wb");
+ int32_t level = msg.applied_input_volume();
+ 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;
+ }
+
+ ++init_count;
+ const Init msg = event_msg.init();
+ // These should print out zeros if they're missing.
+ fprintf(settings_file, "Init #%d at frame: %d\n", init_count,
+ 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: %zu\n", num_input_channels);
+ num_output_channels = msg.num_output_channels();
+ fprintf(settings_file, " Output channels: %zu\n", num_output_channels);
+ num_reverse_channels = msg.num_reverse_channels();
+ fprintf(settings_file, " Reverse channels: %zu\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.
+
+ std::string suffix = GetWavFileIndex(init_count, frame_count);
+ rtc::StringBuilder reverse_name;
+ reverse_name << absl::GetFlag(FLAGS_reverse_file) << suffix << ".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) << suffix << ".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) << suffix << ".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) << suffix
+ << ".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);
+}