98f53510b2
system_wrappers: rename interface -> include
...
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
86b016027d
Add stats for average QP per frame for VP8 (for received video streams):
...
"WebRTC.Video.Decoded.VP8.Qp"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1340623002
Cr-Commit-Position: refs/heads/master@{#10349}
2015-10-21 06:55:32 +00:00
f839dcc870
Add stats for rendered pixels (sqrt(w*h)) per second:
...
- "WebRTC.Video.RenderSqrtPixelsPerSecond"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1366583002
Cr-Commit-Position: refs/heads/master@{#10208}
2015-10-08 07:42:07 +00:00
13c433c299
Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay):
...
- "WebRTC.Video.OnewayDelayInMs"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1351403008
Cr-Commit-Position: refs/heads/master@{#10180}
2015-10-06 11:08:22 +00:00
91d6edef35
Add RTC_ prefix to (D)CHECKs and related macros.
...
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
6304626268
Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate.
...
BUG=
R=asapersson@webrtc.org , noahric@chromium.org , pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1279433006 .
Cr-Commit-Position: refs/heads/master@{#9933}
2015-09-14 17:38:20 +00:00
f42376c601
Wire up currently-received video codec to stats.
...
BUG=webrtc:1844, webrtc:4808
R=mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1315413002
Cr-Commit-Position: refs/heads/master@{#9810}
2015-08-28 14:35:40 +00:00
6718e97e73
Add encode and decode time to histograms stats:
...
- "WebRTC.Video.EncodeTimeInMs"
- "WebRTC.Video.DecodeTimeInMs"
BUG=chromium:488243
Review URL: https://codereview.webrtc.org/1250203002
Cr-Commit-Position: refs/heads/master@{#9630}
2015-07-24 07:21:02 +00:00
d89920b74a
Add resolution and fps stats to histograms:
...
- "WebRTC.Video.InputWidthInPixels"
- "WebRTC.Video.InputHeightInPixels"
- "WebRTC.Video.SentWidthInPixels"
- "WebRTC.Video.SentHeightInPixels"
- "WebRTC.Video.ReceivedWidthInPixels"
- "WebRTC.Video.ReceivedHeightInPixels"
- "WebRTC.Video.RenderFramesPerSecond"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1228393008
Cr-Commit-Position: refs/heads/master@{#9611}
2015-07-22 13:52:03 +00:00
f2f828374c
Use rtc::CriticalSection in webrtc/video/.
...
Removes heap allocation from CriticalSection creation.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50839004
Cr-Commit-Position: refs/heads/master@{#9126}
2015-05-01 14:25:53 +00:00
3c391cbabb
Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api.
...
Add tests for verifying that video histograms are updated.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44309004
Cr-Commit-Position: refs/heads/master@{#9085}
2015-04-27 08:09:27 +00:00
982cd2a94c
Filter receiver-side DataCountersUpdated on SSRC.
...
BUG=1788,1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44509004
Cr-Commit-Position: refs/heads/master@{#8575}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8575 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 15:57:18 +00:00
09c77b95bb
Add decoder-timing stats to VideoReceiveStream.
...
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:42:45 +00:00
1d0fa5d352
Add RtcpPacketTypeCounter stats to new API.
...
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
5570769210
Remove the last getters from VideoReceiveStream stats.
...
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/32899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:45:03 +00:00
ce4e9a3562
Refactor some receive-side stats.
...
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:50:16 +00:00
98c04b38a8
Get avg_delay_ms from DecoderTiming callback.
...
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:12:52 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
9d453931c5
Change return value for number of discarded packets to be int.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:07:44 +00:00
de1429e9ad
Add thread annotations to Call API.
...
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org , stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 13:00:21 +00:00
09315705b9
Wire up statistics in video receive stream of new API
...
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 12:06:29 +00:00
