Add time line for acked bitrate.

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 151 matching lines @@
                  size_t length,
                  int probe_cluster_id) override {
     RTC_DCHECK(pacer_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     if (feedback_observer_)
       feedback_observer_->AddPacket(sequence_number, length, probe_cluster_id);
   }
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
     RTC_DCHECK(network_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
-    if (feedback_observer_)
-      feedback_observer_->OnTransportFeedback(feedback);
+    feedback_observer_->OnTransportFeedback(feedback);
+  }
+  std::vector<PacketInfo> GetTransportFeedbackVector() const override {
+    RTC_NOTREACHED();
+    return std::vector<PacketInfo>();
   }
 
  private:
   rtc::CriticalSection crit_;
   rtc::ThreadChecker thread_checker_;
   rtc::ThreadChecker pacer_thread_;
   rtc::ThreadChecker network_thread_;
   TransportFeedbackObserver* feedback_observer_ GUARDED_BY(&crit_);
 };
 
@@ skipping 3302 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