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..2ebd184d425968871b1b7a1334a46f7cf5f0f960 100644
--- a/webrtc/video/stream_synchronization_unittest.cc
+++ b/webrtc/video/stream_synchronization_unittest.cc
@@ -36,16 +36,17 @@ class Time {
 
   RtcpMeasurement GenerateRtcp(int frequency, uint32_t offset) const {
     RtcpMeasurement rtcp;
-    NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
+    NowNtp(&rtcp.ntp_time);
     rtcp.rtp_timestamp = NowRtp(frequency, offset);
     return rtcp;
   }
 
-  void NowNtp(uint32_t* ntp_secs, uint32_t* ntp_frac) const {
-    *ntp_secs = time_now_ms_ / 1000 + kNtpJan1970;
+  void NowNtp(NtpTime* ntp_time) const {

[stefan-webrtc, 2016/11/13 13:24:59]

Change to NtpTime GetNowNtp() const?

[åsapersson, 2016/11/14 16:44:40]

Done.

+    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);
+    ntp_time->Set(ntp_secs, ntp_frac);
   }
 
   uint32_t NowRtp(int frequency, uint32_t offset) const {
@@ -105,23 +106,28 @@ 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);