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

webrtc/video/rtp_streams_synchronizer.cc

Index: webrtc/video/rtp_streams_synchronizer.cc
diff --git a/webrtc/video/rtp_streams_synchronizer.cc b/webrtc/video/rtp_streams_synchronizer.cc
index 3bc208fe99e5bf67b4f6ba4f25f402afdb539295..d7fd949fff7634cf5fc0b202990279274c9c1f4e 100644
--- a/webrtc/video/rtp_streams_synchronizer.cc
+++ b/webrtc/video/rtp_streams_synchronizer.cc
@@ -24,28 +24,29 @@
 
 namespace webrtc {
 namespace {
-int UpdateMeasurements(StreamSynchronization::Measurements* stream,
-                       RtpRtcp* rtp_rtcp, RtpReceiver* receiver) {
+bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
+                        RtpRtcp* rtp_rtcp,
+                        RtpReceiver* receiver) {
   if (!receiver->Timestamp(&stream->latest_timestamp))
-    return -1;
+    return false;
   if (!receiver->LastReceivedTimeMs(&stream->latest_receive_time_ms))
-    return -1;
+    return false;
 
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;
   if (rtp_rtcp->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
                           &rtp_timestamp) != 0) {
-    return -1;
+    return false;
   }
 
   bool new_rtcp_sr = false;
   if (!UpdateRtcpList(ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp,
                       &new_rtcp_sr)) {
-    return -1;
+    return false;
   }
 
-  return 0;
+  return true;
 }
 }  // namespace
 
@@ -124,13 +125,13 @@ void RtpStreamsSynchronizer::Process() {
       playout_buffer_delay_ms;
 
   int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
-  if (UpdateMeasurements(&video_measurement_, video_rtp_rtcp_,
-                         video_rtp_receiver_) != 0) {
+  if (!UpdateMeasurements(&video_measurement_, video_rtp_rtcp_,
+                          video_rtp_receiver_)) {
     return;
   }
 
-  if (UpdateMeasurements(&audio_measurement_, audio_rtp_rtcp_,
-                         audio_rtp_receiver_) != 0) {
+  if (!UpdateMeasurements(&audio_measurement_, audio_rtp_rtcp_,
+                          audio_rtp_receiver_)) {
     return;
   }