Cast: Simplify code path for RTCP sender report

media/cast/transport/cast_transport_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 is the main interface for the cast transport sender. The cast sender
 // handles the cast pipeline from encoded frames (both audio and video), to
 // encryption, packetization and transport.
 
 // Construction of the Cast Sender and the Cast Transport Sender should be done
 // in the following order:
@@ skipping 27 matching lines @@
 
 namespace media {
 namespace cast {
 namespace transport {
 
 // Following the initialization of either audio or video an initialization
 // status will be sent via this callback.
 typedef base::Callback<void(CastTransportStatus status)>
     CastTransportStatusCallback;
 
-typedef base::Callback<void(const RtcpSenderInfo& sender_info,
-                            base::TimeTicks time_sent,
-                            uint32 rtp_timestamp)> CastTransportRtpStatistics;
-
 typedef base::Callback<void(const std::vector<PacketEvent>&)>
     BulkRawEventsCallback;
 
 // The application should only trigger this class from the transport thread.
 class CastTransportSender : public base::NonThreadSafe {
  public:
   static scoped_ptr<CastTransportSender> Create(
       net::NetLog* net_log,
       base::TickClock* clock,
       const net::IPEndPoint& remote_end_point,
@@ skipping 19 matching lines @@
   // The following two functions handle the encoded media frames (audio and
   // video) to be processed.
   // Frames will be encrypted, packetized and transmitted to the network.
   virtual void InsertCodedAudioFrame(const EncodedAudioFrame* audio_frame,
                                      const base::TimeTicks& recorded_time) = 0;
 
   virtual void InsertCodedVideoFrame(const EncodedVideoFrame* video_frame,
                                      const base::TimeTicks& capture_time) = 0;
 
   // Builds an RTCP packet and sends it to the network.
+  // |ntp_seconds|, |ntp_fraction| and |rtp_timestamp| are used in the
+  // RTCP Sender Report.
   virtual void SendRtcpFromRtpSender(uint32 packet_type_flags,
-                                     const RtcpSenderInfo& sender_info,
+                                     uint32 ntp_seconds,
+                                     uint32 ntp_fraction,
+                                     uint32 rtp_timestamp,
                                      const RtcpDlrrReportBlock& dlrr,
                                      uint32 sending_ssrc,
                                      const std::string& c_name) = 0;
 
   // Retransmission request.
   virtual void ResendPackets(
       bool is_audio,
       const MissingFramesAndPacketsMap& missing_packets) = 0;
-
-  // RTP statistics will be returned on a regular interval on the designated
-  // callback.
-  // Must be called after initialization of the corresponding A/V pipeline.
-  virtual void SubscribeAudioRtpStatsCallback(
-      const CastTransportRtpStatistics& callback) = 0;
-
-  virtual void SubscribeVideoRtpStatsCallback(
-      const CastTransportRtpStatistics& callback) = 0;
 };
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_