Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp
class RtpToNtpEstimator
- UpdateMeasurements
- Estimate
List with rtcp measurements is now private.
BUG=none
Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}
diff --git a/webrtc/video/rtp_streams_synchronizer.cc b/webrtc/video/rtp_streams_synchronizer.cc
index d7fd949..0d026b3 100644
--- a/webrtc/video/rtp_streams_synchronizer.cc
+++ b/webrtc/video/rtp_streams_synchronizer.cc
@@ -41,8 +41,8 @@
}
bool new_rtcp_sr = false;
- if (!UpdateRtcpList(ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp,
- &new_rtcp_sr)) {
+ if (!stream->rtp_to_ntp.UpdateMeasurements(ntp_secs, ntp_frac, rtp_timestamp,
+ &new_rtcp_sr)) {
return false;
}
@@ -183,14 +183,14 @@
}
int64_t latest_audio_ntp;
- if (!RtpToNtpMs(playout_timestamp, audio_measurement_.rtcp,
- &latest_audio_ntp)) {
+ if (!audio_measurement_.rtp_to_ntp.Estimate(playout_timestamp,
+ &latest_audio_ntp)) {
return false;
}
int64_t latest_video_ntp;
- if (!RtpToNtpMs(frame.timestamp(), video_measurement_.rtcp,
- &latest_video_ntp)) {
+ if (!video_measurement_.rtp_to_ntp.Estimate(frame.timestamp(),
+ &latest_video_ntp)) {
return false;
}
@@ -200,7 +200,7 @@
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;
+ *estimated_freq_khz = video_measurement_.rtp_to_ntp.params().frequency_khz;
return true;
}