Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.

webrtc/video/stream_synchronization_unittest.cc

Index: webrtc/video/stream_synchronization_unittest.cc
diff --git a/webrtc/video/stream_synchronization_unittest.cc b/webrtc/video/stream_synchronization_unittest.cc
index d328d6af437ff5c8761f73ecbc46e425efa9beb8..770bbf6eebd56aced50f89d394bea406c8718cb1 100644
--- a/webrtc/video/stream_synchronization_unittest.cc
+++ b/webrtc/video/stream_synchronization_unittest.cc
@@ -36,19 +36,20 @@ class Time {
 
   RtcpMeasurement GenerateRtcp(int frequency, uint32_t offset) const {
     RtcpMeasurement rtcp;
-    NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
-    rtcp.rtp_timestamp = NowRtp(frequency, offset);
+    rtcp.ntp_time = GetNowNtp();
+    rtcp.rtp_timestamp = GetNowRtp(frequency, offset);
     return rtcp;
   }
 
-  void NowNtp(uint32_t* ntp_secs, uint32_t* ntp_frac) const {
-    *ntp_secs = time_now_ms_ / 1000 + kNtpJan1970;
+  NtpTime GetNowNtp() const {
+    uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970;
     int64_t remainder_ms = time_now_ms_ % 1000;
-    *ntp_frac = static_cast<uint32_t>(
+    uint32_t ntp_frac = static_cast<uint32_t>(
         static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5);
+    return NtpTime(ntp_secs, ntp_frac);
   }
 
-  uint32_t NowRtp(int frequency, uint32_t offset) const {
+  uint32_t GetNowRtp(int frequency, uint32_t offset) const {
     return frequency * time_now_ms_ / 1000 + offset;
   }
 
@@ -105,31 +106,36 @@ class StreamSynchronizationTest : public ::testing::Test {
     // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP.
     RtcpMeasurement rtcp =
         send_time_->GenerateRtcp(audio_frequency, audio_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
+                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
                                &audio.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
     rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
+                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
                                &video.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(900);
     receive_time_->IncreaseTimeMs(900);
     rtcp = send_time_->GenerateRtcp(audio_frequency, audio_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
+                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
                                &audio.rtcp, &new_sr));
     send_time_->IncreaseTimeMs(100);
     receive_time_->IncreaseTimeMs(100);
     rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
-    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+    EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(),
+                               rtcp.ntp_time.fractions(), rtcp.rtp_timestamp,
                                &video.rtcp, &new_sr));
+
     send_time_->IncreaseTimeMs(900);
     receive_time_->IncreaseTimeMs(900);
 
     // Capture an audio and a video frame at the same time.
-    audio.latest_timestamp = send_time_->NowRtp(audio_frequency,
-                                                audio_offset);
-    video.latest_timestamp = send_time_->NowRtp(video_frequency,
-                                                video_offset);
+    audio.latest_timestamp =
+        send_time_->GetNowRtp(audio_frequency, audio_offset);
+    video.latest_timestamp =
+        send_time_->GetNowRtp(video_frequency, video_offset);
 
     if (audio_delay_ms > video_delay_ms) {
       // Audio later than video.