Make class of static functions in rtp_to_ntp.h:

webrtc/video/stream_synchronization.cc

 /*
  *  Copyright (c) 2012 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 22 matching lines @@
       base_target_delay_ms_(0),
       avg_diff_ms_(0) {
 }
 
 bool StreamSynchronization::ComputeRelativeDelay(
     const Measurements& audio_measurement,
     const Measurements& video_measurement,
     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) {
     return false;
   }
   // Positive diff means that video_measurement is behind audio_measurement.
   *relative_delay_ms = video_measurement.latest_receive_time_ms -
       audio_measurement.latest_receive_time_ms -
       (video_last_capture_time_ms - audio_last_capture_time_ms);
   if (*relative_delay_ms > kMaxDeltaDelayMs ||
@@ skipping 129 matching lines @@
       target_delay_ms - base_target_delay_ms_;
 
   channel_delay_.extra_video_delay_ms +=
       target_delay_ms - base_target_delay_ms_;
 
   // Video is already delayed by the desired amount.
   base_target_delay_ms_ = target_delay_ms;
 }
 
 }  // namespace webrtc