| 53805324c0fa904d796cc0b333868c591f2c5f2c |
|
21-Dec-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| b7d9a97ce41022e984348efb5f28bf6dd6c6b779 |
|
18-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 2557b86e7648ffebc5781df9f093ca5a84efc219 |
|
18-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| ad13d2f8178af5efbe516184995af02a171ec66a |
|
11-Nov-2015 |
Tim Psiaki <tpsiaki@google.com> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 86b016027d2d27c62fedd108354a2b1274118ae3 |
|
21-Oct-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| f839dcc8709243d1d81d5a23eb15aa43fc0466f6 |
|
08-Oct-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 13c433c299822c6f2aefef46fa6aca52ec46ea49 |
|
06-Oct-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 6304626268238a074051910d201e9a77aae677e0 |
|
14-Sep-2015 |
Tim Psiaki <tpsiaki@google.com> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| f42376c60111edba6f29060bf3dd79e75d8dbb97 |
|
28-Aug-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 6718e97e730dfeb0c4290128b5682e123dd75866 |
|
24-Jul-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| d89920b74a173b7bf80c6760908a382c095a66cc |
|
22-Jul-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| f2f828374c3ee1e1834c72bb27eaae88ef67bb40 |
|
01-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 3c391cbabb5416ca7275d3f6e6cc4fb49c4cf523 |
|
27-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 982cd2a94ce21bbb2ad50c1f32bc6a9a09cc3ab3 |
|
03-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 09c77b95bb62566be64da662f0b3b6a838ec6553 |
|
25-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 55707692105a4765f8f321ec7c30a1034d03d23a |
|
19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| ce4e9a356200170abcdd44ff2af95f87a6781b8e |
|
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 98c04b38a8e4d6afb9b90f728ea465d979a5e18b |
|
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
|
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 9d453931c51d43b768c63bc8f4b0eef241b8aa35 |
|
04-Sep-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| de1429e9ad9a3a207ca191e1d748aa7271066860 |
|
28-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|
| 09315705b9caf3bff455e3515b9bd99492a7c3e3 |
|
07-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/receive_statistics_proxy.cc
|