Cast: Refactor RTCP handling

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"
 #include "base/memory/weak_ptr.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/cast_environment.h"
-#include "media/cast/logging/logging_defines.h"
-#include "media/cast/net/rtcp/rtcp.h"
-#include "media/cast/sender/rtp_timestamp_helper.h"
+#include "media/cast/sender/frame_sender.h"
 
 namespace media {
 namespace cast {
 
 class AudioEncoder;
 
 // Not thread safe. Only called from the main cast thread.
 // This class owns all objects related to sending audio, objects that create RTP
 // packets, congestion control, audio encoder, parsing and sending of
 // RTCP packets.
 // Additionally it posts a bunch of delayed tasks to the main thread for various
 // timeouts.
-class AudioSender : public RtcpSenderFeedback,
+class AudioSender : public FrameSender,
                     public base::NonThreadSafe,
                     public base::SupportsWeakPtr<AudioSender> {
  public:
   AudioSender(scoped_refptr<CastEnvironment> cast_environment,
               const AudioSenderConfig& audio_config,
               CastTransportSender* const transport_sender);
 
   virtual ~AudioSender();
 
   CastInitializationStatus InitializationResult() const {
     return cast_initialization_status_;
   }
 
   // Note: It is not guaranteed that |audio_frame| will actually be encoded and
   // sent, if AudioSender detects too many frames in flight.  Therefore, clients
   // should be careful about the rate at which this method is called.
   //
   // 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);
 
-  // Only called from the main cast thread.
-  void IncomingRtcpPacket(scoped_ptr<Packet> packet);
-
  protected:
   // Protected for testability.
-  virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
-      OVERRIDE;
+  void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
 
  private:
-  // Schedule and execute periodic sending of RTCP report.
-  void ScheduleNextRtcpReport();
-  void SendRtcpReport(bool schedule_future_reports);
-
   // 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);
 
-  const scoped_refptr<CastEnvironment> cast_environment_;
-
   // 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.).
   const base::TimeDelta target_playout_delay_;
 
-  // 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_;
-
   // Maximum number of outstanding frames before the encoding and sending of
   // new frames shall halt.
   const int max_unacked_frames_;
 
   // Encodes AudioBuses into EncodedFrames.
   scoped_ptr<AudioEncoder> audio_encoder_;
   const int configured_encoder_bitrate_;
 
-  // Manages sending/receiving of RTCP packets, including sender/receiver
-  // reports.
-  Rtcp rtcp_;
-
-  // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and
-  // extrapolates this mapping to any other point in time.
-  RtpTimestampHelper rtp_timestamp_helper_;
-
   // 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
@@ skipping 23 matching lines @@
   // 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_