[Cast] RTT clean-up to the max!

media/cast/sender/frame_sender.h

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 //
 // This is the base class for an object that send frames to a receiver.
 // TODO(hclam): Refactor such that there is no separate AudioSender vs.
 // VideoSender, and the functionality of both is rolled into this class.
 
 #ifndef MEDIA_CAST_SENDER_FRAME_SENDER_H_
 #define MEDIA_CAST_SENDER_FRAME_SENDER_H_
@@ skipping 25 matching lines @@
 
   base::TimeDelta GetTargetPlayoutDelay() const {
     return target_playout_delay_;
   }
 
  protected:
   // Schedule and execute periodic sending of RTCP report.
   void ScheduleNextRtcpReport();
   void SendRtcpReport(bool schedule_future_reports);
 
-  void OnReceivedRtt(base::TimeDelta rtt,
-                     base::TimeDelta avg_rtt,
-                     base::TimeDelta min_rtt,
-                     base::TimeDelta max_rtt);
+  void OnMeasuredRoundTripTime(base::TimeDelta rtt);
 
-  bool is_rtt_available() const { return rtt_available_; }
+  bool is_rtt_available() const {
+    return current_round_trip_time_ > base::TimeDelta();
+  }
+
+  base::TimeDelta current_round_trip_time() const {
+    return current_round_trip_time_;
+  }
 
   const scoped_refptr<CastEnvironment> cast_environment_;
 
   // Sends encoded frames over the configured transport (e.g., UDP).  In
   // Chromium, this could be a proxy that first sends the frames from a renderer
   // process to the browser process over IPC, with the browser process being
   // responsible for "packetizing" the frames and pushing packets into the
   // network layer.
   CastTransportSender* const transport_sender_;
 
   const uint32 ssrc_;
 
-  // RTT information from RTCP.
-  bool rtt_available_;
-  base::TimeDelta rtt_;
-  base::TimeDelta avg_rtt_;
-  base::TimeDelta min_rtt_;
-  base::TimeDelta max_rtt_;
 
  protected:
   // Schedule and execute periodic checks for re-sending packets.  If no
   // acknowledgements have been received for "too long," AudioSender will
   // speculatively re-send certain packets of an unacked frame to kick-start
   // re-transmission.  This is a last resort tactic to prevent the session from
   // getting stuck after a long outage.
   void ScheduleNextResendCheck();
   void ResendCheck();
   void ResendForKickstart();
@@ skipping 60 matching lines @@
  private:
   // RTP timestamp increment representing one second.
   const int rtp_timebase_;
 
   // Ring buffers to keep track of recent frame timestamps (both in terms of
   // local reference time and RTP media time).  These should only be accessed
   // through the Record/GetXXX() methods.
   base::TimeTicks frame_reference_times_[256];
   RtpTimestamp frame_rtp_timestamps_[256];
 
+  // The most recently measured round trip time.
+  base::TimeDelta current_round_trip_time_;
+
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<FrameSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(FrameSender);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_SENDER_FRAME_SENDER_H_