Cast: Log sender side packet events.

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>
 #include <set>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
 #include "base/memory/weak_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/transport/cast_transport_defines.h"
 #include "media/cast/transport/cast_transport_sender.h"
 #include "media/cast/transport/pacing/paced_sender.h"
 #include "media/cast/transport/rtp_sender/packet_storage/packet_storage.h"
 #include "media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"
 
 namespace media {
 namespace cast {
+
+class LoggingImpl;
+
 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:
   RtpSender(
       base::TickClock* clock,
+      LoggingImpl* logging,
       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
@@ skipping 14 matching lines @@
   // 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.
+  LoggingImpl* logging_;    // Not owned by this class.
   RtpPacketizerConfig config_;
   scoped_ptr<RtpPacketizer> packetizer_;
   scoped_ptr<PacketStorage> storage_;
   PacedSender* const transport_;
   CastTransportRtpStatistics stats_callback_;
   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_