Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/video/vie_channel.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 23 matching lines @@
 class ChannelStatsObserver;
 class Config;
 class EncodedImageCallback;
 class I420FrameCallback;
 class IncomingVideoStream;
 class PacedSender;
 class PacketRouter;
 class PayloadRouter;
 class ProcessThread;
 class ReceiveStatisticsProxy;
-class ReportBlockStats;
 class RtcpRttStats;
 class ViEChannelProtectionCallback;
 class ViERTPObserver;
 class VideoCodingModule;
 class VideoRenderCallback;
 class VoEVideoSync;
 
 enum StreamType {
   kViEStreamTypeNormal = 0,  // Normal media stream
   kViEStreamTypeRtx = 1      // Retransmission media stream
@@ skipping 54 matching lines @@
 
   void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
   RtpState GetRtpStateForSsrc(uint32_t ssrc) const;
 
   // Sets the CName for the outgoing stream on the channel.
   int32_t SetRTCPCName(const char* rtcp_cname);
 
   // Gets the CName of the incoming stream.
   int32_t GetRemoteRTCPCName(char rtcp_cname[]);
 
-  // Returns statistics reported by the remote client in an RTCP packet.
-  // TODO(pbos): Remove this along with VideoSendStream::GetRtt().
-  int32_t GetSendRtcpStatistics(uint16_t* fraction_lost,
-                                uint32_t* cumulative_lost,
-                                uint32_t* extended_max,
-                                uint32_t* jitter_samples,
-                                int64_t* rtt_ms) const;
-
   // Called on receipt of RTCP report block from remote side.
   void RegisterSendChannelRtcpStatisticsCallback(
       RtcpStatisticsCallback* callback);
 
   // Gets send statistics for the rtp and rtx stream.
   void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
                                  StreamDataCounters* rtx_counters) const;
 
   // Gets received stream data counters.
   void GetReceiveStreamDataCounters(StreamDataCounters* rtp_counters,
@@ skipping 231 matching lines @@
   PacedSender* const paced_sender_;
   PacketRouter* const packet_router_;
 
   const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
   TransportFeedbackObserver* const transport_feedback_observer_;
 
   int nack_history_size_sender_;
   int max_nack_reordering_threshold_;
   I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);
 
-  const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
-
-  int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
-  int64_t rtt_sum_ms_ GUARDED_BY(crit_);
   int64_t last_rtt_ms_ GUARDED_BY(crit_);
-  size_t num_rtts_ GUARDED_BY(crit_);
 
   // RtpRtcp modules, declared last as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
   size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_