Video/Screenshare loopback tool.

webrtc/logging/rtc_event_log/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
@@ skipping 75 matching lines @@
                               BandwidthUsage detector_state) override;
   void LogAudioNetworkAdaptation(
       const AudioEncoderRuntimeConfig& config) override;
   void LogProbeClusterCreated(int id,
                               int bitrate_bps,
                               int min_probes,
                               int min_bytes) override;
   void LogProbeResultSuccess(int id, int bitrate_bps) override;
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override;
+  void LogAckedBitrate(uint32_t bitrate_bps) override;
+  void LogAlrState(bool in_alr, uint32_t bitrate_bps) override;
+  void LogPacketQueueTime(uint32_t ssrc, int64_t time_ms) override;
 
  private:
   // Private constructor to ensure that creation is done by RtcEventLog::Create.
   RtcEventLogImpl();
 
   void StoreEvent(std::unique_ptr<rtclog::Event> event);
   void LogProbeResult(int id,
                       rtclog::BweProbeResult::ResultType result,
                       int bitrate_bps);
 
@@ skipping 452 matching lines @@
   event->set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
 
   auto probe_result = event->mutable_probe_result();
   probe_result->set_id(id);
   probe_result->set_result(result);
   if (result == rtclog::BweProbeResult::SUCCESS)
     probe_result->set_bitrate_bps(bitrate_bps);
   StoreEvent(std::move(event));
 }
 
+void RtcEventLogImpl::LogAckedBitrate(uint32_t bitrate_bps) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(rtc::TimeMicros());
+  event->set_type(rtclog::Event::BWE_ACKED_BITRATE_EVENT);
+
+  auto acked_bitrate = event->mutable_acked_bitrate();
+  acked_bitrate->set_bitrate_bps(bitrate_bps);
+  StoreEvent(std::move(event));
+}
+
+void RtcEventLogImpl::LogAlrState(bool in_alr, uint32_t usage_bps) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(rtc::TimeMicros());
+  event->set_type(rtclog::Event::ALR_STATE_EVENT);
+
+  auto alr_state = event->mutable_alr_state();
+  alr_state->set_in_alr(in_alr);
+  alr_state->set_usage_bps(usage_bps);
+  StoreEvent(std::move(event));
+}
+
+void RtcEventLogImpl::LogPacketQueueTime(uint32_t ssrc, int64_t time_ms) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(rtc::TimeMicros());
+  event->set_type(rtclog::Event::PACKET_QUEUE_TIME);
+
+  auto queue_time = event->mutable_packet_queue_time();
+  queue_time->set_queue_time_ms(time_ms);
+  queue_time->set_ssrc(ssrc);
+  StoreEvent(std::move(event));
+}
+
 void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event> event) {
   RTC_DCHECK(event.get() != nullptr);
   if (!event_queue_.Insert(&event)) {
     LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
   }
   helper_thread_.SignalNewEvent();
 }
 
 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
                                    rtclog::EventStream* result) {
@@ skipping 28 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc