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}