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

webrtc/video/rtp_streams_synchronizer.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.
  */
 
 #include "webrtc/video/rtp_streams_synchronizer.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 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
 
 RtpStreamsSynchronizer::RtpStreamsSynchronizer(
     vcm::VideoReceiver* video_receiver,
     RtpStreamReceiver* rtp_stream_receiver)
     : clock_(Clock::GetRealTimeClock()),
       video_receiver_(video_receiver),
       video_rtp_receiver_(rtp_stream_receiver->GetRtpReceiver()),
       video_rtp_rtcp_(rtp_stream_receiver->rtp_rtcp()),
@@ skipping 58 matching lines @@
   int playout_buffer_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms) != 0) {
     return;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       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;
   }
 
   if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
     // No new video packet has been received since last update.
     return;
   }
 
   int relative_delay_ms;
   // Calculate how much later or earlier the audio stream is compared to video.
@@ skipping 53 matching lines @@
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   *estimated_freq_khz = video_measurement_.rtcp.params.frequency_khz;
   return true;
 }
 
 }  // namespace webrtc