| 59bac1a4c557bb4ba4e5f9f2f6db6040ae4b41a7 |
|
08-Jan-2016 |
asapersson <asapersson@webrtc.org> |
Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest.
BUG=
Review URL: https://codereview.webrtc.org/1543933004
Cr-Commit-Position: refs/heads/master@{#11180}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 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/send_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/send_statistics_proxy.cc
|
| 17821db19702aca15d0d93cb60515ca70823fad7 |
|
14-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Wire up bandwidth limitation info to GetStats and adapt_reason.
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.
BUG=webrtc:4112
Review URL: https://codereview.webrtc.org/1502173002
Cr-Commit-Position: refs/heads/master@{#11006}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 1aa420b6aa70bd97cbe33e396e5dc0346aeb6415 |
|
07-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.
BUG=
Review URL: https://codereview.webrtc.org/1278383002
Cr-Commit-Position: refs/heads/master@{#10911}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| b4a1ae5299fd57be66c7cbb7a982179bb1ecfb90 |
|
03-Dec-2015 |
sprang <sprang@webrtc.org> |
Add separate send-side UMA stats for screenshare and video.
This CL duplicates all the histograms in SendStatisticsProxy. Might be
overkill, but we don't know which stats will be interesting and it makes
the change easier.
BUG=
Review URL: https://codereview.webrtc.org/1433393002
Cr-Commit-Position: refs/heads/master@{#10885}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 6f14be8df8b67feb480f55b3a41e2b8cd06a836d |
|
16-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Add limit for minimum number of required samples before recording input and sent framerate stats.
BUG=
Review URL: https://codereview.webrtc.org/1446443002
Cr-Commit-Position: refs/heads/master@{#10644}
/external/webrtc/webrtc/video/send_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/send_statistics_proxy.cc
|
| f040b2367d894345677192f924590309be8c72fc |
|
04-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Add histograms for send-side delay stats for a sent video stream:
- "WebRTC.Video.SendSideDelayInMs"
- "WebRTC.Video.SendSideDelayMaxInMs"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1405023014
Cr-Commit-Position: refs/heads/master@{#10502}
/external/webrtc/webrtc/video/send_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/send_statistics_proxy.cc
|
| 2a0a2a410f36b85fdc184d53b6992f4ba1a43b60 |
|
27-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Add stats for used video codec type for a sent video stream:
- "WebRTC.Video.Encoder.CodecType"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1426673002
Cr-Commit-Position: refs/heads/master@{#10423}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 415d2cd7454d93b3727fce9147090a24e4c3ccba |
|
26-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for video.
BUG=webrtc:5118
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1415413004 .
Cr-Commit-Position: refs/heads/master@{#10403}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 49e196af4060624d620297a6bc017699daa33550 |
|
23-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoFrameType aliases for FrameType.
No longer used in Chromium, so these can now be removed.
BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1415693002 .
Cr-Commit-Position: refs/heads/master@{#10390}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| da535c405597864b8396b2029dec70ab9fb76e8b |
|
20-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"
If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"
BUG=
Review URL: https://codereview.webrtc.org/1311533012
Cr-Commit-Position: refs/heads/master@{#10333}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 4306fc70d778887d8a2ea71b6f4bc6a12d1d9447 |
|
19-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Add histogram for percentage of sent frames that are limited in resolution due to quality:
- "WebRTC.Video.QualityLimitedResolutionInPercent"
and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"
BUG=
Review URL: https://codereview.webrtc.org/1325153009
Cr-Commit-Position: refs/heads/master@{#10319}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 5d0379da2cbdcce6f8494209c7ab559cd6de076e |
|
06-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove kSkipFrame usage.
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| dec5ebf10614115d6f2561c65f0cce3fd80ecfd2 |
|
05-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Move sent key frame stats to send_statistics_proxy class.
BUG=
Review URL: https://codereview.webrtc.org/1374673003
Cr-Commit-Position: refs/heads/master@{#10166}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 7083e119e8f39d2ec9e504461c1bb6e0bc6be5ff |
|
22-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove callback_cs_ in ViEEncoder.
Instead make callbacks const and set on construction.
BUG=webrtc:1695
R=philipel@webrtc.org
Review URL: https://codereview.webrtc.org/1354143004 .
Cr-Commit-Position: refs/heads/master@{#10017}
/external/webrtc/webrtc/video/send_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/send_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/send_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/send_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/send_statistics_proxy.cc
|
| 24b4eda6f4fdfd33d2c3e82df1390bad55953f5d |
|
16-Jun-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add sent framerates to histogram stats:
"WebRTC.Video.InputFramesPerSecond",
"WebRTC.Video.SentFramesPerSecond".
BUG=488243
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1169543005.
Cr-Commit-Position: refs/heads/master@{#9446}
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 20f3f942a05a4b37d39891ff28be67d984c345f7 |
|
15-May-2015 |
Peter Boström <pbos@webrtc.org> |
Clear bitrate stats for unused SSRCs.
Prevents bug where transmitted bitrate was reported as higher than what
was actually sent, since unused RTP modules weren't updated to say that
they sent zero.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49979004
Cr-Commit-Position: refs/heads/master@{#9192}
/external/webrtc/webrtc/video/send_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/send_statistics_proxy.cc
|
| 2b4ce3a501b8d679f84c1ad10317dea5c78fa595 |
|
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.
BUG=1756
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42879004
Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| af612d5e0769571544952cbe55e675748afa9bdd |
|
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| d7452a016812ab1de69c3d7a53caca5b06c64990 |
|
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| bcead305a2f27c30c72c6a3824fdf12f4b83c2eb |
|
06-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 891d48393e5ccd2f5e03d509c544c00a3d88cbbc |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up target_media_bitrate in VideoSendStream.
Also wires up target_enc_bitrate in WebRtcVideoEngine2.
BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42479004
Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|
| 3e6e271ec3253e78ae0eb72156e5236d43f8731d |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement CpuOveruseMetrics as callbacks.
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.
BUG=1667, 1788
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42429004
Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_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/send_statistics_proxy.cc
|
| 49096de442f6131e90925daff6dc9888d085de00 |
|
24-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
DCHECK send DataCountersUpdated for valid SSRCs.
Also updates RTPSender to not update RTX stats when RTX is disabled.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42399004
Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_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/send_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/send_statistics_proxy.cc
|
| 273a414b0ec2e58fdf3b817ad8b1a02f4ce15287 |
|
01-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report encoded frame size in VideoSendStream.
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_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/send_statistics_proxy.cc
|
| 58e2d262fc6a67d069f6c5b8c5043748570521f9 |
|
14-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.
Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| 168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
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11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| 4ef438e2defd6c46404f6b367287364cde66b7fb |
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11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| de1429e9ad9a3a207ca191e1d748aa7271066860 |
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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/send_statistics_proxy.cc
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| b10363f3b63222b0f6ec7e916ef4ccac15d7205b |
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13-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
This was originally committed as r5687, but reverted due to a flaky
test.
BUG=3040
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5695 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| be3947020382cc9733a9b53dff064f1353375bb5 |
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11-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Routing SuspendChange to VideoSendStream::Stats"
The test VideoSendStreamTest.SuspendBelowMinBitrate seems flaky.
Reverting and investigating.
BUG=3040
Review URL: https://webrtc-codereview.appspot.com/9799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5681 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| 1598b80f52bde9346f3eee20b08f51bcf5cfa245 |
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11-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Routing SuspendChange to VideoSendStream::Stats
Also checking that the statistics are properly updated in
VideoSendStreamTest.SuspendBelowMinBitrate.
Adding a test to SendStatisticsProxyTest.
Checking callback status in rampup test, too.
BUG=2457
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
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| 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/send_statistics_proxy.cc
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| ccd42840bcee8db145be91b3308912a24f710a6f |
|
07-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up statistics in video send stream of new video engine api
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5559006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/send_statistics_proxy.cc
|