WebRTC.Video.QualityLimitedResolutionInPercent is reported as zero for calls when the quality scaler is off (and the degradation preference allows scaling).
Update SetResolutionRestrictionStats to specify if quality scaler is enabled.
BUG=webrtc:7432
Review-Url: https://codereview.webrtc.org/2783213002
Cr-Commit-Position: refs/heads/master@{#17487}
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.BitrateSentInKbps"
"WebRTC.Video.MediaBitrateSentInKbps"
"WebRTC.Video.PaddingBitrateSentInKbps"
"WebRTC.Video.RetransmittedBitrateSentInKbps"
"WebRTC.Video.RtxBitrateSentInKbps"
"WebRTC.Video.FecBitrateSentInKbps"
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2536613002
Cr-Commit-Position: refs/heads/master@{#16447}
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
"WebRTC.Call.NumberOfPauseEvents" -> "WebRTC.Video.NumberOfPauseEvents"
Recorded if a certain time has passed (10 sec) since the first media packet was sent.
Moved to per stream to know when media has started and to prevent logging stats for calls that was never in use.
Add histogram for percentage of paused video time for sent video streams:
"WebRTC.Video.PausedTimeInPercent"
BUG=b/32659204
Review-Url: https://codereview.webrtc.org/2530393003
Cr-Commit-Position: refs/heads/master@{#15681}
Reason for revert:
Bug affecting perf tests has been fixed. The issue was that I had accidentally disabled cpu overuse adaptation based on the encoders ScalingSettings, not just quality-based scaling.
Original issue's description:
> Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
>
> Reason for revert:
> Breaks perf tests
>
> Original issue's description:
> > Properly report number of quality downscales in stats.
> >
> > A regression was introduced in 876222f that caused these stats to
> > be reported incorrectly. This used to be only implemented for VP8
> > but should now be available for all codecs.
> >
> > BUG=webrtc:6860
> >
> > Review-Url: https://codereview.webrtc.org/2564373002
> > Cr-Commit-Position: refs/heads/master@{#15673}
> > Committed: 0c8c538835
>
> TBR=asapersson@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2586783003
> Cr-Commit-Position: refs/heads/master@{#15678}
> Committed: fe04bd43ccTBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860
Review-Url: https://codereview.webrtc.org/2588743002
Cr-Commit-Position: refs/heads/master@{#15680}
Reason for revert:
Breaks perf tests
Original issue's description:
> Properly report number of quality downscales in stats.
>
> A regression was introduced in 876222f that caused these stats to
> be reported incorrectly. This used to be only implemented for VP8
> but should now be available for all codecs.
>
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2564373002
> Cr-Commit-Position: refs/heads/master@{#15673}
> Committed: 0c8c538835TBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860
Review-Url: https://codereview.webrtc.org/2586783003
Cr-Commit-Position: refs/heads/master@{#15678}
A regression was introduced in 876222f that caused these stats to
be reported incorrectly. This used to be only implemented for VP8
but should now be available for all codecs.
BUG=webrtc:6860
Review-Url: https://codereview.webrtc.org/2564373002
Cr-Commit-Position: refs/heads/master@{#15673}
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
but has a callback to ViEEncoder that it uses to express it's desire
for lower resolution.
BUG=webrtc:6495
Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
WebRTC.Video.BitrateSentInKbps
WebRTC.Video.MediaBitrateSentInKbps
WebRTC.Video.PaddingBitrateSentInKbps
WebRTC.Video.RetransmittedBitrateSentInKbps
WebRTC.Video.FecBitrateSentInKbps
RtpSender has two StreamDataCounters: for the non-RTX and the RTX stream.
The same counter (for the non-RTX stream) is reported for both the media SSRC and the FlexFEC SSRC.
Bitrate stats are summed for all SSRCs, thus the counter for the non-RTX stream is counted twice.
Do not store the counter for the FlexFEC SSRC.
Do not include info from FlexFEC substreams in VideoSendStream::Stats::ToString (periodically logged during a call).
BUG=webrtc:6774
Review-Url: https://codereview.webrtc.org/2525293002
Cr-Commit-Position: refs/heads/master@{#15238}
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.
To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
Also rename some related minor methods. No functional changes
are intended/expected.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2391963002
Cr-Commit-Position: refs/heads/master@{#14513}
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.
BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/2368223002 .
Cr-Commit-Position: refs/heads/master@{#14431}
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
Reason for revert:
It keeps breaking upstream.
Original issue's description:
> Reland Issue 2061423003: Refactor NACK bitrate allocation
>
> This is a reland of https://codereview.webrtc.org/2061423003/
> Which was reverted in https://codereview.webrtc.org/2131913003/
>
> The reason for the revert was that some upstream code used
> RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
> it's been brought up to date.
>
> TBR=tommi@webrtc.org
>
> Committed: 05ce4ae31fTBR=tommi@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2130423002
Cr-Commit-Position: refs/heads/master@{#13419}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
Updated tests to use the default implementation and removed the test implementation (webrtc/test/histograms.h).
BUG=
Review-Url: https://codereview.webrtc.org/1915523002
Cr-Commit-Position: refs/heads/master@{#12829}
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
And move encoder name cb to VCMSendStatisticsCallback.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
Reason for revert:
A fix is being prepared downstream so this can now go in.
Original issue's description:
> Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
>
> Reason for revert:
> API changes broke downstream.
>
> Original issue's description:
> > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> > EncodedImageCallback can of course be cleaned up in the future.
> >
> > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
> >
> > BUG=webrtc::5687
> >
> > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> > Cr-Commit-Position: refs/heads/master@{#12436}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5687
>
> Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4
> Cr-Commit-Position: refs/heads/master@{#12441}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1905583002
Cr-Commit-Position: refs/heads/master@{#12442}
Reason for revert:
API changes broke downstream.
Original issue's description:
> Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> EncodedImageCallback can of course be cleaned up in the future.
>
> This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
>
> BUG=webrtc::5687
>
> Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> Cr-Commit-Position: refs/heads/master@{#12436}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc::5687
Review URL: https://codereview.webrtc.org/1903193002
Cr-Commit-Position: refs/heads/master@{#12441}
EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
EncodedImageCallback can of course be cleaned up in the future.
This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
BUG=webrtc::5687
Review URL: https://codereview.webrtc.org/1897233002
Cr-Commit-Position: refs/heads/master@{#12436}
Permits measuring encoding time even when performed on another thread,
typically for hardware encoding, instead of assuming that encoding is
blocking the calling thread.
Permitted encoding time is increased for hardware encoders since they
can be timed to keep 30fps, for instance, without indicating overload.
Merges EncodingTimeObserver into EncodedFrameObserver to have one post-encode
callback.
BUG=webrtc:5042, webrtc:5132
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1569853002 .
Cr-Commit-Position: refs/heads/master@{#11499}
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}
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}
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
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
