saza 382f21cd9c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ )
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession

A Windows build fails with a mysterious compile error.

Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> 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
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237a

TBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
2017-07-04 08:11:49 +00:00
..
2017-06-29 13:21:20 +00:00
2017-07-03 07:47:44 +00:00
2017-06-30 10:04:59 +00:00
2017-06-30 10:04:59 +00:00

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.