Reland of Add received audio/video call duration metrics based on packets.

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}
Committed: https://chromium.googlesource.com/external/webrtc/+/0d7f04daa0ddc621a7e3d6a0373b3ae458511030

[saza WebRTC, 2017-07-04 09:32:23]

Description was changed from

==========
Reapply packet based call duration metric changes.

BUG=
==========

to

==========
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.

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
==========

[saza WebRTC, 2017-07-04 10:22:10]

Description was changed from

==========
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.

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
==========

to

==========
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
==========

[saza WebRTC, 2017-07-04 10:22:32]

The CQ bit was checked by saza@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2017-07-04 10:22:35]

Dry run: CQ is trying da patch.

Follow status at:
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2017-07-04 10:58:26]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2017-07-04 10:58:27]

Dry run: This issue passed the CQ dry run.

[saza WebRTC, 2017-07-04 11:02:41]

The CQ bit was checked by saza@webrtc.org

[commit-bot: I haz the power, 2017-07-04 11:02:45]

CQ is trying da patch.

Follow status at:
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2017-07-04 11:05:06]

CQ is committing da patch.

Bot data: {"patchset_id": 1, "attempt_start_ts": 1499166161543970, "parent_rev":
"38fecafa48ff6abfdf0279ce3c04df7e26401e08", "commit_rev":
"0d7f04daa0ddc621a7e3d6a0373b3ae458511030"}

[commit-bot: I haz the power, 2017-07-04 11:05:11]

Description was changed from

==========
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
==========

to

==========
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}
Committed:
https://chromium.googlesource.com/external/webrtc/+/0d7f04daa0ddc621a7e3d6a03...
==========

[commit-bot: I haz the power, 2017-07-04 11:05:12]

Committed patchset #1 (id:1) as
https://chromium.googlesource.com/external/webrtc/+/0d7f04daa0ddc621a7e3d6a03...