[rtp_rtcp] RTT intermidiate calculation use ntp time instead of milliseconds.

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 625 matching lines @@
 
 TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
   rtcp_sender_->SendRtcpXrReceiverReferenceTime(false);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
   EXPECT_EQ(0, parser()->xr_header()->num_packets());
   EXPECT_EQ(0, parser()->rrtr()->num_packets());
 }
 
-TEST_F(RtcpSenderTest, TestSendTimeOfXrRrtr) {
-  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false));
-  rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
-  uint32_t ntp_sec;
-  uint32_t ntp_frac;
-  clock_.CurrentNtp(ntp_sec, ntp_frac);
-  uint32_t initial_mid_ntp = RTCPUtility::MidNtp(ntp_sec, ntp_frac);
-
-  // No packet sent.
-  int64_t time_ms;
-  EXPECT_FALSE(rtcp_sender_->SendTimeOfXrRrReport(initial_mid_ntp, &time_ms));
-
-  // Send XR RR packets.
-  for (int i = 0; i <= RTCP_NUMBER_OF_SR; ++i) {
-    EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
-    EXPECT_EQ(i + 1, test_transport_.parser_.rrtr()->num_packets());
-    clock_.CurrentNtp(ntp_sec, ntp_frac);
-    uint32_t mid_ntp = RTCPUtility::MidNtp(ntp_sec, ntp_frac);
-    EXPECT_TRUE(rtcp_sender_->SendTimeOfXrRrReport(mid_ntp, &time_ms));
-    EXPECT_EQ(clock_.CurrentNtpInMilliseconds(), time_ms);
-    clock_.AdvanceTimeMilliseconds(1000);
-  }
-  // The first report should no longer be stored.
-  EXPECT_FALSE(rtcp_sender_->SendTimeOfXrRrReport(initial_mid_ntp, &time_ms));
-}
-
 TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
   RtcpPacketTypeCounterObserverImpl observer;
   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
                                     &observer, nullptr, &test_transport_));
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
   EXPECT_EQ(1, parser()->pli()->num_packets());
   EXPECT_EQ(kRemoteSsrc, observer.ssrc_);
   EXPECT_EQ(1U, observer.counter_.pli_packets);
@@ skipping 115 matching lines @@
   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
 
   // 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