saza 0d7f04daa0 Reland of Add received audio/video call duration metrics based on packets.
Original issue:
https://codereview.webrtc.org/2957073002/

Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.

Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
2017-07-04 11:05:06 +00:00
..
2016-11-17 14:48:56 +00:00
2016-11-16 19:11:38 +00:00