Make class of static functions in rtp_to_ntp.h:

webrtc/video/stream_synchronization.cc

Index: webrtc/video/stream_synchronization.cc
diff --git a/webrtc/video/stream_synchronization.cc b/webrtc/video/stream_synchronization.cc
index ff389736f4bd09883ac89d188a3d334e27c3545b..145922c3a081ef2c976fadf5c353d2b0474ad97c 100644
--- a/webrtc/video/stream_synchronization.cc
+++ b/webrtc/video/stream_synchronization.cc
@@ -40,15 +40,13 @@ bool StreamSynchronization::ComputeRelativeDelay(
     int* relative_delay_ms) {
   assert(relative_delay_ms);
   int64_t audio_last_capture_time_ms;
-  if (!RtpToNtpMs(audio_measurement.latest_timestamp,
-                  audio_measurement.rtcp,
-                  &audio_last_capture_time_ms)) {
+  if (!audio_measurement.rtp_to_ntp.Estimate(audio_measurement.latest_timestamp,
+                                             &audio_last_capture_time_ms)) {
     return false;
   }
   int64_t video_last_capture_time_ms;
-  if (!RtpToNtpMs(video_measurement.latest_timestamp,
-                  video_measurement.rtcp,
-                  &video_last_capture_time_ms)) {
+  if (!video_measurement.rtp_to_ntp.Estimate(video_measurement.latest_timestamp,
+                                             &video_last_capture_time_ms)) {
     return false;
   }
   if (video_last_capture_time_ms < 0) {