Add stats for frequency offset when converting RTP timestamp to NTP time.

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.
  */
 
@@ skipping 22 matching lines @@
 
   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;
   }
 
   bool new_rtcp_sr = false;
-  if (!UpdateRtcpList(
-      ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
+  if (!UpdateRtcpList(ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp,
+                      &new_rtcp_sr)) {
     return -1;
   }
 
   return 0;
 }
 }  // namespace
 
 RtpStreamsSynchronizer::RtpStreamsSynchronizer(
     vcm::VideoReceiver* video_receiver,
     RtpStreamReceiver* rtp_stream_receiver)
@@ skipping 106 matching lines @@
   }
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
 bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
-    const VideoFrame& frame, int64_t* stream_offset_ms) const {
+    const VideoFrame& frame,
+    int64_t* stream_offset_ms,
+    double* estimated_freq_khz) const {
   rtc::CritScope lock(&crit_);
   if (voe_channel_id_ == -1)
     return false;
 
   uint32_t playout_timestamp = 0;
   if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
                                                playout_timestamp) != 0) {
     return false;
   }
 
   int64_t latest_audio_ntp;
   if (!RtpToNtpMs(playout_timestamp, audio_measurement_.rtcp,
                   &latest_audio_ntp)) {
     return false;
   }
 
   int64_t latest_video_ntp;
   if (!RtpToNtpMs(frame.timestamp(), video_measurement_.rtcp,
                   &latest_video_ntp)) {
     return false;
   }
 
   int64_t time_to_render_ms =
       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