Refactoring: CastTransport InitializeAudio/InitializeVideo.

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 47 matching lines @@
   CastTransportImpl(
       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,
-                       std::unique_ptr<RtcpObserver> rtcp_observer) final;
-  void InitializeVideo(const CastTransportRtpConfig& config,
-                       std::unique_ptr<RtcpObserver> rtcp_observer) final;
+  void InitializeStream(const CastTransportRtpConfig& config,
+                        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 27 matching lines @@
   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;
 
+  struct RtpStreamSession;
+
   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,
@@ skipping 24 matching lines @@
 
   // 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.
   int64_t last_byte_acked_for_audio_;
 
   // Packets that don't match these sender ssrcs are ignored.
   std::set<uint32_t> valid_sender_ssrcs_;
 
   // While non-null, global WiFi behavior modifications are in effect. This is
   // used, for example, to turn off WiFi scanning that tends to interfere with
   // the reliability of UDP packet transmission.
   std::unique_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
 
   // Do not initialize the |rtcp_builder_at_rtp_receiver_| if the RTP receiver
   // SSRC does not match these ssrcs. Only RTP receiver needs to register its
   // SSRC in this set.
   std::set<uint32_t> valid_rtp_receiver_ssrcs_;
 
   std::unique_ptr<RtcpBuilder> rtcp_builder_at_rtp_receiver_;
 
+  // Records the initialized stream sessions on RTP sender.

[miu, 2016/07/06 21:45:42]

Looks like the key is the SSRC. Can you mention this in the comment?

[xjz, 2016/07/13 20:31:17]

Done.

+  using SessionMap = std::map<uint32_t, std::unique_ptr<RtpStreamSession>>;
+  SessionMap sessions_;
+
   base::WeakPtrFactory<CastTransportImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_