Refactor: simplify interface of SenderRtcpSession and CastTransport.

media/cast/net/cast_transport_impl.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 class maintains a transport for audio and video in a Cast Streaming
 // session.
 // Audio, video frames and RTCP messages are submitted to this object
 // and then packetized and paced to the underlying UDP socket.
 //
 // The hierarchy of send transport in a Cast Streaming session:
@@ skipping 48 matching lines @@
       base::TickClock* clock,  // Owned by the caller.
       base::TimeDelta logging_flush_interval,
       std::unique_ptr<Client> client,
       std::unique_ptr<PacketTransport> transport,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
 
   ~CastTransportImpl() final;
 
   // CastTransport implementation for sending.
   void InitializeAudio(const CastTransportRtpConfig& config,
-                       const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb,
-                       const RtcpPliCallback& pli_cb) final;
+                       std::unique_ptr<RtcpObserver> rtcp_observer) final;
   void InitializeVideo(const CastTransportRtpConfig& config,
-                       const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb,
-                       const RtcpPliCallback& pli_cb) final;
+                       std::unique_ptr<RtcpObserver> rtcp_observer) final;
   void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) final;
 
   void SendSenderReport(uint32_t ssrc,
                         base::TimeTicks current_time,
                         RtpTimeTicks current_time_as_rtp_timestamp) final;
 
   void CancelSendingFrames(uint32_t ssrc,
                            const std::vector<FrameId>& frame_ids) final;
 
   void ResendFrameForKickstart(uint32_t ssrc, FrameId frame_id) final;
@@ skipping 24 matching lines @@
   void AddCastFeedback(const RtcpCastMessage& cast_message,
                        base::TimeDelta target_delay) final;
   void AddPli(const RtcpPliMessage& pli_message) final;
   void AddRtcpEvents(
       const ReceiverRtcpEventSubscriber::RtcpEvents& rtcp_events) final;
   void AddRtpReceiverReport(
       const RtcpReportBlock& rtp_receiver_report_block) final;
   void SendRtcpFromRtpReceiver() final;
 
  private:
+  // Handle received RTCP messages on RTP sender.
+  class RtcpClient;
+
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, NacksCancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, CancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, Kickstart);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, DedupRetransmissionWithAudio);
 
   // Resend packets for the stream identified by |ssrc|.
   // If |cancel_rtx_if_not_in_list| is true then transmission of packets for the
   // frames but not in the list will be dropped.
   // See PacedSender::ResendPackets() to see how |dedup_info| works.
   void ResendPackets(uint32_t ssrc,
                      const MissingFramesAndPacketsMap& missing_packets,
                      bool cancel_rtx_if_not_in_list,
                      const DedupInfo& dedup_info);
 
   // If |logging_flush_interval| is set, this is called at approximate periodic
   // intervals.
   void SendRawEvents();
 
   // Called when a packet is received.
   bool OnReceivedPacket(std::unique_ptr<Packet> packet);
 
   // Called when a log message is received.
   void OnReceivedLogMessage(EventMediaType media_type,
                             const RtcpReceiverLogMessage& log);
 
   // Called when a RTCP Cast message is received.
   void OnReceivedCastMessage(uint32_t ssrc,
-                             const RtcpCastMessageCallback& cast_message_cb,
                              const RtcpCastMessage& cast_message);
 
   base::TickClock* const clock_;  // Not owned by this class.
   const base::TimeDelta logging_flush_interval_;
   const std::unique_ptr<Client> transport_client_;
   const std::unique_ptr<PacketTransport> transport_;
   const scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
 
   // FrameEvents and PacketEvents pending delivery via raw events callback.
   // Do not add elements to these when |logging_flush_interval| is
   // |base::TimeDelta()|.
   std::vector<FrameEvent> recent_frame_events_;
   std::vector<PacketEvent> recent_packet_events_;
 
   // Packet sender that performs pacing.
   PacedSender pacer_;
 
   // Packetizer for audio and video frames.
   std::unique_ptr<RtpSender> audio_sender_;
   std::unique_ptr<RtpSender> video_sender_;
 
   // Maintains RTCP session for audio and video.
   std::unique_ptr<SenderRtcpSession> audio_rtcp_session_;
   std::unique_ptr<SenderRtcpSession> video_rtcp_session_;
 
+  // RTCP observer for SenderRtcpSession.
+  std::unique_ptr<RtcpObserver> audio_rtcp_observer_;
+  std::unique_ptr<RtcpObserver> video_rtcp_observer_;
+
   // Encrypts data in EncodedFrames before they are sent.  Note that it's
   // important for the encryption to happen here, in code that would execute in
   // the main browser process, for security reasons.  This helps to mitigate
   // the damage that could be caused by a compromised renderer process.
   TransportEncryptionHandler audio_encryptor_;
   TransportEncryptionHandler video_encryptor_;
 
   // Right after a frame is sent we record the number of bytes sent to the
   // socket. We record the corresponding bytes sent for the most recent ACKed
   // audio packet.
@@ skipping 16 matching lines @@
 
   base::WeakPtrFactory<CastTransportImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_