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

media/cast/sender/audio_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.
 
 #ifndef MEDIA_CAST_SENDER_AUDIO_SENDER_H_
 #define MEDIA_CAST_SENDER_AUDIO_SENDER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
@@ skipping 37 matching lines @@
   // Note: It is invalid to call this method if InitializationResult() returns
   // anything but STATUS_AUDIO_INITIALIZED.
   void InsertAudio(scoped_ptr<AudioBus> audio_bus,
                    const base::TimeTicks& recorded_time);
 
  protected:
   // Protected for testability.
   void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
 
  private:
-  // 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();
-
   // Returns true if there are too many frames in flight, as defined by the
   // configured target playout delay plus simple logic.  When this is true,
   // InsertAudio() will silenty drop frames instead of sending them to the audio
   // encoder.
   bool AreTooManyFramesInFlight() const;
 
   // Called by the |audio_encoder_| with the next EncodedFrame to send.
   void SendEncodedAudioFrame(scoped_ptr<EncodedFrame> audio_frame);
 
   // Encodes AudioBuses into EncodedFrames.
   scoped_ptr<AudioEncoder> audio_encoder_;
   const int configured_encoder_bitrate_;
 
-  // 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, AudioSender 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.
-  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];
-
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<AudioSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioSender);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_SENDER_AUDIO_SENDER_H_