Refactor rtcp statistics: Rtcp module take narrow interface

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 59 matching lines @@
 
     bool has_last_xr_rr;
     rtcp::ReceiveTimeInfo last_xr_rr;
 
     // Used when generating TMMBR.
     ModuleRtpRtcpImpl* module;
   };
 
   RTCPSender(bool audio,
              Clock* clock,
-             ReceiveStatistics* receive_statistics,
+             ReceiveStatisticsReporter* receive_statistics,
              RtcpPacketTypeCounterObserver* packet_type_counter_observer,
              RtcEventLog* event_log,
              Transport* outgoing_transport);
   virtual ~RTCPSender();
 
   RtcpMode Status() const;
   void SetRTCPStatus(RtcpMode method);
 
   bool Sending() const;
   int32_t SetSendingStatus(const FeedbackState& feedback_state,
@@ skipping 59 matching lines @@
   void SetVideoBitrateAllocation(const BitrateAllocation& bitrate);
   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
 
  private:
   class RtcpContext;
 
   // Determine which RTCP messages should be sent and setup flags.
   void PrepareReport(const FeedbackState& feedback_state)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
-  bool AddReportBlock(const FeedbackState& feedback_state,
-                      uint32_t ssrc,
-                      StreamStatistician* statistician)
-      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
-
   std::unique_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
@@ skipping 30 matching lines @@
   int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
 
   uint32_t timestamp_offset_ GUARDED_BY(critical_section_rtcp_sender_);
   uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
   int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
   uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
   // SSRC that we receive on our RTP channel
   uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
   std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
 
-  ReceiveStatistics* receive_statistics_
+  ReceiveStatisticsReporter* receive_statistics_
       GUARDED_BY(critical_section_rtcp_sender_);
-  std::map<uint32_t, rtcp::ReportBlock> report_blocks_
+  std::vector<rtcp::ReportBlock> report_blocks_
       GUARDED_BY(critical_section_rtcp_sender_);
   std::map<uint32_t, std::string> csrc_cnames_
       GUARDED_BY(critical_section_rtcp_sender_);
 
   // send CSRCs
   std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_);
 
   // Full intra request
   uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_);
 
@@ skipping 55 matching lines @@
   typedef std::unique_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
       const RtcpContext&);
   // Map from RTCPPacketType to builder.
   std::map<uint32_t, BuilderFunc> builders_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_