Delay based logging.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 472 matching lines @@
   RTC_CHECK(loss_event.has_fraction_loss());
   if (fraction_loss != nullptr) {
     *fraction_loss = loss_event.fraction_loss();
   }
   RTC_CHECK(loss_event.has_total_packets());
   if (total_packets != nullptr) {
     *total_packets = loss_event.total_packets();
   }
 }
 
-void ParsedRtcEventLog::GetDelayBasedBweUpdate(
-    size_t index,
-    int32_t* bitrate_bps,
-    BandwidthUsage* detector_state) const {
+ParsedRtcEventLog::BweDelayBasedUpdate
+ParsedRtcEventLog::GetDelayBasedBweUpdate(size_t index) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
   RTC_CHECK(event.has_delay_based_bwe_update());
   const rtclog::DelayBasedBweUpdate& delay_event =
       event.delay_based_bwe_update();
+
+  BweDelayBasedUpdate res;
+  res.timestamp = GetTimestamp(index);
   RTC_CHECK(delay_event.has_bitrate_bps());
-  if (bitrate_bps != nullptr) {
-    *bitrate_bps = delay_event.bitrate_bps();
-  }
+  res.bitrate_bps = delay_event.bitrate_bps();
   RTC_CHECK(delay_event.has_detector_state());
-  if (detector_state != nullptr) {
-    *detector_state = GetRuntimeDetectorState(delay_event.detector_state());
-  }
+  res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
+  return res;
 }
 
 void ParsedRtcEventLog::GetAudioNetworkAdaptation(
     size_t index,
     AudioEncoderRuntimeConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
   RTC_CHECK(event.has_audio_network_adaptation());
@@ skipping 62 matching lines @@
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
 }  // namespace webrtc