Add receive bitrate UMA stats.

Add receive bitrate UMA stats.

Committed: https://crrev.com/91d926038f7cebf889ef843f2f087d72bc8c60c2
Cr-Commit-Position: refs/heads/master@{#10605}

[stefan-webrtc, 2015-11-11 13:53:47]

stefan@webrtc.org changed reviewers:
+ pbos@webrtc.org

[stefan-webrtc, 2015-11-11 14:24:38]

add test and only report non-zero values

[stefan-webrtc, 2015-11-11 14:45:41]

stefan@webrtc.org changed reviewers:
+ asapersson@webrtc.org

[pbos-webrtc, 2015-11-11 14:56:58]

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:143: size_t received_video_bytes_;
Can you put notes on how these are synchronized?

[åsapersson, 2015-11-11 15:35:25]

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:229: if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
maybe early return

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:645: received_video_bytes_ += length;
Could media type any be counted twice?

[stefan-webrtc, 2015-11-11 15:49:25]

Switched to using RateTracker per Åsa's request.

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:143: size_t received_video_bytes_;
On 2015/11/11 14:56:57, pbos-webrtc wrote:
> Can you put notes on how these are synchronized?

Done.

[stefan-webrtc, 2015-11-11 15:59:04]

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:229: if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
On 2015/11/11 15:35:24, åsapersson wrote:
> maybe early return

Done.

https://codereview.webrtc.org/1440603002/diff/40001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:645: received_video_bytes_ += length;
On 2015/11/11 15:35:24, åsapersson wrote:
> Could media type any be counted twice?

Fixed.

[stefan-webrtc, 2015-11-11 16:03:35]

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

[commit-bot: I haz the power, 2015-11-11 16:03:37]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1440603002/80001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1440603002/80001

[åsapersson, 2015-11-11 16:13:56]

lgtm

[pbos-webrtc, 2015-11-11 16:24:29]

lgtm

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:149: int64_t first_rtp_packet_received_ms_;
(I assume the corner case where RTCP arrives first isn't very interesting.)

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:641: return DELIVERY_UNKNOWN_SSRC;
Do you wanna track unknown bitrate as well?

[commit-bot: I haz the power, 2015-11-11 17:55:21]

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

[commit-bot: I haz the power, 2015-11-11 17:55:22]

Dry run: This issue passed the CQ dry run.

[stefan-webrtc, 2015-11-11 18:11:40]

The CQ bit was checked by stefan@webrtc.org

[commit-bot: I haz the power, 2015-11-11 18:11:47]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1440603002/80001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1440603002/80001

[commit-bot: I haz the power, 2015-11-11 18:13:05]

Committed patchset #5 (id:80001)

[commit-bot: I haz the power, 2015-11-11 18:13:17]

Patchset 5 (id:??) landed as
https://crrev.com/91d926038f7cebf889ef843f2f087d72bc8c60c2
Cr-Commit-Position: refs/heads/master@{#10605}

[stefan-webrtc, 2015-11-11 18:15:22]

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc
File webrtc/call/call.cc (right):

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:149: int64_t first_rtp_packet_received_ms_;
On 2015/11/11 16:24:29, pbos-webrtc wrote:
> (I assume the corner case where RTCP arrives first isn't very interesting.)

Its a very slim corner case. If rtcp arrives first, and the stream is 20 secs
from that rtcp, we will not record the stats, since we measure from first rtp...

https://codereview.webrtc.org/1440603002/diff/80001/webrtc/call/call.cc#newco...
webrtc/call/call.cc:641: return DELIVERY_UNKNOWN_SSRC;
On 2015/11/11 16:24:29, pbos-webrtc wrote:
> Do you wanna track unknown bitrate as well?

We can add that later if we see a need.