Commit Graph

36 Commits

Author SHA1 Message Date
35fdb2a914 Log WebRTC.Video.AVSyncOffsetInMs.
BUG=
R=asapersson@webrtc.org

Review-Url: https://codereview.webrtc.org/1941993002
Cr-Commit-Position: refs/heads/master@{#12605}
2016-05-03 10:32:16 +00:00
8688a4e2b5 Add histogram stats for jitter buffer delay and current/target delay for received video streams:
- "WebRTC.Video.JitterBufferDelayInMs"
- "WebRTC.Video.TargetDelayInMs" (jitter delay + decode time + render delay)
- "WebRTC.Video.CurrentDelayInMs"

BUG=

Review-Url: https://codereview.webrtc.org/1885393002
Cr-Commit-Position: refs/heads/master@{#12539}
2016-04-28 06:42:42 +00:00
a186288fd0 Revert of Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay. (patchset #4 id:60001 of https://codereview.webrtc.org/1688143003/ )
Reason for revert:
The delay stats are high.

Original issue's description:
> Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay.
> Previous logged delay was: network delay (rtt/2) + jitter delay + decode time + render delay.
>
> Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
>
> BUG=
>
> Committed: https://crrev.com/5249599a9b69ad9c2d513210d694719f1011f977
> Cr-Commit-Position: refs/heads/master@{#11901}

TBR=stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:603838

Review URL: https://codereview.webrtc.org/1893543003

Cr-Commit-Position: refs/heads/master@{#12400}
2016-04-18 07:41:09 +00:00
74f6e9ea23 Replace NULL with nullptr in webrtc/video.
BUG=
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/1855433002 .

Cr-Commit-Position: refs/heads/master@{#12218}
2016-04-04 15:56:22 +00:00
58d992e025 Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats

BUG=

Review URL: https://codereview.webrtc.org/1788783002

Cr-Commit-Position: refs/heads/master@{#12133}
2016-03-29 09:15:11 +00:00
f8cdd184d5 Add histogram stats for AV sync stream offset:
"WebRTC.Video.AVSyncOffsetInMs"

The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.

Updated sync tests in call_perf_tests.cc to use this implementation.

BUG=webrtc:5493

Review URL: https://codereview.webrtc.org/1756193005

Cr-Commit-Position: refs/heads/master@{#11993}
2016-03-15 08:00:54 +00:00
5249599a9b Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay.
Previous logged delay was: network delay (rtt/2) + jitter delay + decode time + render delay.

Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).

BUG=

Review URL: https://codereview.webrtc.org/1688143003

Cr-Commit-Position: refs/heads/master@{#11901}
2016-03-08 10:10:24 +00:00
07fb9be37f Move RTCP histograms from vie_channel to video channel stats proxies.
Also slice those histograms on content type.

BUG=

Review URL: https://codereview.webrtc.org/1720883002

Cr-Commit-Position: refs/heads/master@{#11748}
2016-02-24 15:55:06 +00:00
0ab8e81e12 Move histograms for rtp receive counters to ReceiveStatisticsProxy
BUG=

Review URL: https://codereview.webrtc.org/1726503003

Cr-Commit-Position: refs/heads/master@{#11735}
2016-02-24 09:35:45 +00:00
28ba92731d Switch to use new implementation in metrics.h.
Sparse macro replaced for all video histograms that have a constant name.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1616153005

Cr-Commit-Position: refs/heads/master@{#11368}
2016-01-25 13:58:27 +00:00
97888bd95a Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.
While doing this, I made a couple of minor changes:
* Removed unused variables (one lock and one video frame variable)
* Switched over to a scoped lock in remb.cc and removed an if() in a function where we can just return the expression being checked.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1613053003 .

Cr-Commit-Position: refs/heads/master@{#11349}
2016-01-21 22:25:12 +00:00
53805324c0 Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
This implementation will be replaced by a faster one and sparse will be removed.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1530913002

Cr-Commit-Position: refs/heads/master@{#11099}
2015-12-21 09:46:25 +00:00
b7d9a97ce4 Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.

BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1406903002 .

Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18 15:01:23 +00:00
2557b86e76 modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.

To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).

Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417283007 .

Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 21:00:33 +00:00
ad13d2f817 Round Rate computations from RateTracker.
BUG=534221
R=asapersson@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1410533004 .

Cr-Commit-Position: refs/heads/master@{#10592}
2015-11-11 00:34:58 +00:00
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