StartTimestamp generated randomly in RtpSender constructor

webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc

 /*
  *  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 225 matching lines @@
     configuration.audio = false;
     configuration.clock = &clock_;
     configuration.outgoing_transport = &test_transport_;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
     rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
                                       nullptr, nullptr, &test_transport_));
     rtcp_sender_->SetSSRC(kSenderSsrc);
     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-    rtcp_sender_->SetStartTimestamp(kStartRtpTimestamp);
+    rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
     rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
   }
 
   void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
     RTPHeader header;
     header.ssrc = remote_ssrc;
     header.sequenceNumber = seq_num;
     header.timestamp = 12345;
     header.headerLength = 12;
     size_t kPacketLength = 100;
@@ skipping 549 matching lines @@
     }
 
     return true;
   }));
 
   // Re-configure rtcp_sender_ with mock_transport_
   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
                                     nullptr, nullptr, &mock_transport));
   rtcp_sender_->SetSSRC(kSenderSsrc);
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
-  rtcp_sender_->SetStartTimestamp(kStartRtpTimestamp);
+  rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
   rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
 
   // Set up XR VoIP metric to be included with BYE
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   RTCPVoIPMetric metric;
   EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
 }
 
 }  // namespace webrtc