Add stats for frequency offset when converting RTP timestamp to NTP time.
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"
The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.
Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.
- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).
BUG=webrtc:6579
Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
diff --git a/webrtc/video/stream_synchronization_unittest.cc b/webrtc/video/stream_synchronization_unittest.cc
index f09bee4..d328d6a 100644
--- a/webrtc/video/stream_synchronization_unittest.cc
+++ b/webrtc/video/stream_synchronization_unittest.cc
@@ -98,24 +98,30 @@
int audio_offset = 0;
int video_frequency = static_cast<int>(kDefaultVideoFrequency *
video_clock_drift_ + 0.5);
+ bool new_sr;
int video_offset = 0;
StreamSynchronization::Measurements audio;
StreamSynchronization::Measurements video;
// Generate NTP/RTP timestamp pair for both streams corresponding to RTCP.
- audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
- audio_offset));
+ RtcpMeasurement rtcp =
+ send_time_->GenerateRtcp(audio_frequency, audio_offset);
+ EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+ &audio.rtcp, &new_sr));
send_time_->IncreaseTimeMs(100);
receive_time_->IncreaseTimeMs(100);
- video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
- video_offset));
+ rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
+ EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+ &video.rtcp, &new_sr));
send_time_->IncreaseTimeMs(900);
receive_time_->IncreaseTimeMs(900);
- audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
- audio_offset));
+ rtcp = send_time_->GenerateRtcp(audio_frequency, audio_offset);
+ EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+ &audio.rtcp, &new_sr));
send_time_->IncreaseTimeMs(100);
receive_time_->IncreaseTimeMs(100);
- video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
- video_offset));
+ rtcp = send_time_->GenerateRtcp(video_frequency, video_offset);
+ EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_secs, rtcp.ntp_frac, rtcp.rtp_timestamp,
+ &video.rtcp, &new_sr));
send_time_->IncreaseTimeMs(900);
receive_time_->IncreaseTimeMs(900);