Cast:Transport: Dividing A/V Initialization pipeline

media/cast/transport/rtp_sender/rtp_sender.h

 // Copyright 2013 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 file contains the interface to the cast RTP sender.
 
 #ifndef MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
 #define MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
 
 #include <map>
@@ skipping 12 matching lines @@
 #include "media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"
 
 namespace media {
 namespace cast {
 namespace transport {
 
 // This object is only called from the main cast thread.
 // This class handles splitting encoded audio and video frames into packets and
 // add an RTP header to each packet. The sent packets are stored until they are
 // acknowledged by the remote peer or timed out.
-class RtpSender : public base::SupportsWeakPtr<RtpSender>{
+class RtpSender {
  public:
-  RtpSender(base::TickClock* clock,
-            const CastTransportConfig& config,
-            bool is_audio,
-            const scoped_refptr<base::TaskRunner>& transport_task_runner,
-            PacedSender* const transport);
+  RtpSender(
+      base::TickClock* clock,
+      const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
+      PacedSender* const transport);
 
   ~RtpSender();
 
+  // Initialize audio stack. Audio must be initialized prior to sending encoded
+  // audio frames.
+  void InitializeAudio(const CastTransportAudioConfig& config);
+
+  // Initialize video stack. Video must be initialized prior to sending encoded
+  // video frames.
+  void InitializeVideo(const CastTransportVideoConfig& config);
+
   // The video_frame objects ownership is handled by the main cast thread.
   void IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame,
                                  const base::TimeTicks& capture_time);
 
   // The audio_frame objects ownership is handled by the main cast thread.
   void IncomingEncodedAudioFrame(const EncodedAudioFrame* audio_frame,
                                  const base::TimeTicks& recorded_time);
 
   void ResendPackets(const MissingFramesAndPacketsMap& missing_packets);
 
   // Set the callback on which RTP statistics data will be returned. Calling
   // this function would start a timer that would schedule the callback in
   // a constant interval.
   void SubscribeRtpStatsCallback(const CastTransportRtpStatistics& callback);
 
  private:
   void ScheduleNextStatsReport();
   void RtpStatistics();
   void UpdateSequenceNumber(Packet* packet);
 
+  base::TickClock* clock_;  // Not owned by this class.
   RtpPacketizerConfig config_;
   scoped_ptr<RtpPacketizer> packetizer_;
   scoped_ptr<PacketStorage> storage_;
   PacedSender* const transport_;
   CastTransportRtpStatistics stats_callback_;
-  scoped_refptr<base::TaskRunner> transport_task_runner_;
+  scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
+
+  base::WeakPtrFactory<RtpSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpSender);
 };
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_