Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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 143 matching lines @@
   uint32_t remoteSSRC;  // SSRC of sender of this report.
   uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
   uint8_t fractionLost;
   uint32_t cumulativeLost;  // 24 bits valid.
   uint32_t extendedHighSeqNum;
   uint32_t jitter;
   uint32_t lastSR;
   uint32_t delaySinceLastSR;
 };
 
-struct RtcpReceiveTimeInfo {
-  // Fields as described by RFC 3611 4.5.
-  uint32_t sourceSSRC;
-  uint32_t lastRR;
-  uint32_t delaySinceLastRR;
-};
-
 typedef std::list<RTCPReportBlock> ReportBlockList;
 
 struct RtpState {
   RtpState()
       : sequence_number(0),
         start_timestamp(0),
         timestamp(0),
         capture_time_ms(-1),
         last_timestamp_time_ms(-1),
         media_has_been_sent(false) {}
@@ skipping 215 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_