Add logging of delay-based bandwidth estimate.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 122 matching lines @@
 
   void LogBwePacketLossEvent(int32_t bitrate,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
     }
   }
 
+  void LogBwePacketDelayEvent(int32_t bitrate,
+                              BandwidthUsage detector_state) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogBwePacketDelayEvent(bitrate, detector_state);
+    }
+  }
+
   void LogAudioNetworkAdaptation(
       const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioNetworkAdaptation(config);
     }
   }
 
   void SetEventLog(RtcEventLog* event_log) {
     rtc::CritScope lock(&crit_);
@@ skipping 3171 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc