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Populate jitter stats for video RTP streams

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Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!

Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
This commit is contained in:
Di Wu (RP Room Eng)
2021-02-19 17:34:22 -08:00
committed by Commit Bot
parent 373bb7bec4
commit 8af6b4928a
7 changed files with 8 additions and 3 deletions
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2
api/stats/rtcstats_objects.h
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@ -417,8 +417,6 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint64_t> header_bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;
// TODO(hbos): Collect and populate this value for both "audio" and "video",
// currently not collected for "video". https://bugs.webrtc.org/7065
RTCStatsMember<double> jitter;
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;