Cast: Simplify code path for RTCP sender report

media/cast/transport/transport_audio_sender.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.
 
 #ifndef MEDIA_CAST_TRANSPORT_TRANSPORT_AUDIO_SENDER_H_
 #define MEDIA_CAST_TRANSPORT_TRANSPORT_AUDIO_SENDER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
@@ skipping 21 matching lines @@
 
   // Handles the encoded audio frames to be processed.
   // Frames will be encrypted, packetized and transmitted to the network.
   void InsertCodedAudioFrame(const EncodedAudioFrame* audio_frame,
                              const base::TimeTicks& recorded_time);
 
   // Retransmision request.
   void ResendPackets(
       const MissingFramesAndPacketsMap& missing_frames_and_packets);
 
+  size_t send_packet_count() const { return rtp_sender_.send_packet_count(); }
+  size_t send_octet_count() const { return rtp_sender_.send_octet_count(); }
+  uint32 ssrc() const { return rtp_sender_.ssrc(); }
   bool initialized() const { return initialized_; }
 
-  // Subscribe callback to get RTP Audio stats.
-  void SubscribeAudioRtpStatsCallback(
-      const CastTransportRtpStatistics& callback);
-
  private:
   friend class LocalRtcpAudioSenderFeedback;
 
   // Caller must allocate the destination |encrypted_frame|. The data member
   // will be resized to hold the encrypted size.
   bool EncryptAudioFrame(const EncodedAudioFrame& audio_frame,
                          EncodedAudioFrame* encrypted_frame);
 
   RtpSender rtp_sender_;
   TransportEncryptionHandler encryptor_;
   bool initialized_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(TransportAudioSender);
 };
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_TRANSPORT_TRANSPORT_AUDIO_SENDER_H_