Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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);
+}