Move common code and variables from audio/video sender to frame sender.

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 56 matching lines @@
   RtpTimestampHelper rtp_timestamp_helper_;
 
   // 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();
+
   const base::TimeDelta rtcp_interval_;
 
   // The total amount of time between a frame's capture/recording on the sender
   // and its playback on the receiver (i.e., shown to a user).  This is fixed as
   // a value large enough to give the system sufficient time to encode,
   // transmit/retransmit, receive, decode, and render; given its run-time
   // environment (sender/receiver hardware performance, network conditions,
   // etc.).
   base::TimeDelta target_playout_delay_;
 
   // If true, we transmit the target playout delay to the receiver.
   bool send_target_playout_delay_;
 
   // Max encoded frames generated per second.
   double max_frame_rate_;
 
   // Maximum number of outstanding frames before the encoding and sending of
   // new frames shall halt.
   int max_unacked_frames_;
 
+  // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
+  // frame) at the start of the session.  Once a threshold is reached, RTCP
+  // reports are instead sent at the configured interval + random drift.
+  int num_aggressive_rtcp_reports_sent_;
+
+  // This is "null" until the first frame is sent.  Thereafter, this tracks the
+  // last time any frame was sent or re-sent.
+  base::TimeTicks last_send_time_;
+
+  // The ID of the last frame sent.  Logic throughout AudioSender assumes this
+  // can safely wrap-around.  This member is invalid until
+  // |!last_send_time_.is_null()|.
+  uint32 last_sent_frame_id_;
+
+  // The ID of the latest (not necessarily the last) frame that has been
+  // acknowledged.  Logic throughout AudioSender assumes this can safely
+  // wrap-around.  This member is invalid until |!last_send_time_.is_null()|.
+  uint32 latest_acked_frame_id_;
+
+  // Counts the number of duplicate ACK that are being received.  When this
+  // number reaches a threshold, the sender will take this as a sign that the
+  // receiver hasn't yet received the first packet of the next frame.  In this
+  // case, VideoSender will trigger a re-send of the next frame.
+  int duplicate_ack_counter_;
+
+  // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED or
+  // STATUS_VIDEO_INITIALIZED.
+  CastInitializationStatus cast_initialization_status_;
+
+  // This is a "good enough" mapping for finding the RTP timestamp associated
+  // with a video frame. The key is the lowest 8 bits of frame id (which is
+  // what is sent via RTCP). This map is used for logging purposes.
+  RtpTimestamp frame_id_to_rtp_timestamp_[256];
+
  private:
   // 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_