Simplify interface for media/cast: CastTransportSenderImpl

media/cast/net/cast_transport_sender_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 send 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 60 matching lines @@
   //   "send_buffer_min_size": int
   //        - Specifies the minimum socket send buffer size.
   //   "disable_wifi_scan" (value ignored)
   //        - Disable wifi scans while streaming.
   //   "media_streaming_mode" (value ignored)
   //        - Turn media streaming mode on.
   // Note, these options may be ignored on some platforms.
   // TODO(hubbe): Too many callbacks, replace with an interface.
   // http://crbug.com/557477
   CastTransportSenderImpl(
-      net::NetLog* net_log,
+      base::WeakPtr<CastTransportSenderInfo> cast_transport_sender_info,
       base::TickClock* clock,
-      const net::IPEndPoint& local_end_point,
-      const net::IPEndPoint& remote_end_point,
-      scoped_ptr<base::DictionaryValue> options,
-      const CastTransportStatusCallback& status_callback,
-      const BulkRawEventsCallback& raw_events_callback,
-      base::TimeDelta raw_events_callback_interval,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
-      const PacketReceiverCallback& packet_callback,
       PacketSender* external_transport);
 
   ~CastTransportSenderImpl() final;
 
   // CastTransportSender implementation.
   void InitializeAudio(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
                        const RtcpRttCallback& rtt_cb) final;
   void InitializeVideo(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
@@ skipping 48 matching lines @@
 
   // 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 ssrc,
                              const RtcpCastMessageCallback& cast_message_cb,
                              const RtcpCastMessage& cast_message);
 
+  base::WeakPtr<CastTransportSenderInfo> cast_transport_sender_info_;
   base::TickClock* clock_;  // Not owned by this class.
-  CastTransportStatusCallback status_callback_;
   scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
 
   // FrameEvents and PacketEvents pending delivery via |raw_events_callback_|.
   // Do not add elements to these when |raw_events_callback_.is_null()|.
   std::vector<FrameEvent> recent_frame_events_;
   std::vector<PacketEvent> recent_packet_events_;
 
   // Interface to a UDP socket.
   scoped_ptr<UdpTransport> transport_;
 
   // Packet sender that performs pacing.
   PacedSender pacer_;
 
   // Packetizer for audio and video frames.
   scoped_ptr<RtpSender> audio_sender_;
   scoped_ptr<RtpSender> video_sender_;
 
   // Maintains RTCP session for audio and video.
   scoped_ptr<Rtcp> audio_rtcp_session_;
   scoped_ptr<Rtcp> video_rtcp_session_;
 
   // 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_;
 
-  BulkRawEventsCallback raw_events_callback_;
   base::TimeDelta raw_events_callback_interval_;
 
   // 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 last_byte_acked_for_audio_;
 
   // Packets that don't match these ssrcs are ignored.
   std::set<uint32> valid_ssrcs_;
 
-  // Called with incoming packets. (Unless they match the
-  // channels created by Initialize{Audio,Video}.
-  PacketReceiverCallback packet_callback_;
-
   scoped_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
 
   base::WeakPtrFactory<CastTransportSenderImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportSenderImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_