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

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc

 /*
  *  Copyright (c) 2013 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 291 matching lines @@
   clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
 
   // Verify RTT.
   int64_t rtt;
   int64_t avg_rtt;
   int64_t min_rtt;
   int64_t max_rtt;
   EXPECT_EQ(0,
       sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, rtt);
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, avg_rtt);
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, min_rtt);
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, max_rtt);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, avg_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, min_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, max_rtt, 1);
 
   // No RTT from other ssrc.
   EXPECT_EQ(-1,
       sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
 
   // Verify RTT from rtt_stats config.
   EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt());
   EXPECT_EQ(0, sender_.impl_->rtt_ms());
   sender_.impl_->Process();
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.rtt_stats_.LastProcessedRtt());
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms());
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.rtt_stats_.LastProcessedRtt(),
+              1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms(), 1);
 }
 
 TEST_F(RtpRtcpImplTest, SetRtcpXrRrtrStatus) {
   EXPECT_FALSE(receiver_.impl_->RtcpXrRrtrStatus());
   receiver_.impl_->SetRtcpXrRrtrStatus(true);
   EXPECT_TRUE(receiver_.impl_->RtcpXrRrtrStatus());
 }
 
 TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   receiver_.impl_->SetRtcpXrRrtrStatus(true);
 
   // Receiver module should send a Receiver time reference report (RTRR).
   EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
 
   // Sender module should send a response to the last received RTRR (DLRR).
   clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport));
 
   // Verify RTT.
   EXPECT_EQ(0, receiver_.rtt_stats_.LastProcessedRtt());
   EXPECT_EQ(0, receiver_.impl_->rtt_ms());
   receiver_.impl_->Process();
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.rtt_stats_.LastProcessedRtt());
-  EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs,
+              receiver_.rtt_stats_.LastProcessedRtt(), 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms(), 1);
 }
 
 TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
   // Ignore fake transport delays in this test.
   sender_.transport_.SimulateNetworkDelay(0, &clock_);
   receiver_.transport_.SimulateNetworkDelay(0, &clock_);
 
   sender_.impl_->Process();
   EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
 
@@ skipping 180 matching lines @@
   EXPECT_EQ(6U, receiver_.RtcpSent().unique_nack_requests);
   EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(11, 18, 20, 21));
 
   // Send module receives the request.
   EXPECT_EQ(2U, sender_.RtcpReceived().nack_packets);
   EXPECT_EQ(8U, sender_.RtcpReceived().nack_requests);
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
 }  // namespace webrtc