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Video jitter stats fix: Convert RTP timestamp

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stats.rtp_stats.jitter is a RTP timestamp so we needed to convert it back to regular timestamps

See https://bugs.chromium.org/p/webrtc/issues/detail?id=12980#c7

Bug: webrtc:12980
Change-Id: I0d94a22e043ac6ecec4926d950abbdcf787b7168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227100
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Di Wu <meetwudi@gmail.com>
Cr-Commit-Position: refs/heads/master@{#34590}
This commit is contained in:
Di Wu
2021-07-27 22:21:22 -07:00
committed by WebRTC LUCI CQ
parent c662c95605
commit 61bd5d9c1c
1 changed files with 1 additions and 1 deletions
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2
media/engine/webrtc_video_engine.cc
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@ -3161,7 +3161,7 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
stats.rtp_stats.packet_counter.padding_bytes;
info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
info.packets_lost = stats.rtp_stats.packets_lost;
info.jitter_ms = stats.rtp_stats.jitter;
info.jitter_ms = stats.rtp_stats.jitter / (kVideoCodecClockrate / 1000);
info.framerate_rcvd = stats.network_frame_rate;
info.framerate_decoded = stats.decode_frame_rate;