1. It moves calculation of the needed padding to VideoSendStream instead of ViEEncoder and only does it once per send Stream instead of every time the network estimate changes.
2. The maximum amount of padding sent was prior to this cl calculated and updated based on network estimate changes. However, it can only change based on encoder configuration changes and if send streams are added or removed. This cl change the VideoSendStream/VieEncoder to notify the BitrateAllocator of changes to the needed padding bitrate and for BitrateAllocator to notify Call of these changes.
3. Fixed an issue in the SendPacer where it could send a padding packet before sending a real packet. This caused the test EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx to fail with these refactorings since the pacer suddenly could send a padding packet before the encoder had produced its first frame.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1993113003
Cr-Commit-Position: refs/heads/master@{#13149}
Reason for revert:
Reverting while we track down the issue on the Win10 bot.
Original issue's description:
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
>
> Further work done:
>
> * I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
>
> * I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=
>
> Committed: https://crrev.com/1c7eef652b0aa22d8ebb0bfe2b547094a794be22
> Cr-Commit-Position: refs/heads/master@{#13129}
TBR=mflodman@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/2061363002
Cr-Commit-Position: refs/heads/master@{#13146}
EncodedFrameCallbackAdapter was used VideoSendStream and
VideoReceiveStream, but there is no reason to have it as these classes
can call EncodedFrameObserver directly.
Review-Url: https://codereview.webrtc.org/2068463004
Cr-Commit-Position: refs/heads/master@{#13145}
This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
Further work done:
* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
* I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=
Review-Url: https://codereview.webrtc.org/2035173002
Cr-Commit-Position: refs/heads/master@{#13129}
To avoid the case where a single data point or too short window is used,
causing bad behavior due to bad stats, update RateStatistics to return
an Optional rather than a plain rate.
There was also a strange off by one bug where the rate was slightly
overestimated (N + 1 buckets, N ms time window).
These changes requires updates to a number of places, and may very well
cause seeming perf regressions (but the stats were probablty more wrong
previously).
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2029593002 .
Cr-Commit-Position: refs/heads/master@{#13103}
Instead of the default copy constructor, the Copy() method has to be used. In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream. Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case). Most importantly, creating copies is made harder and the interface encourages ownership transfers.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2042603002 .
Cr-Commit-Position: refs/heads/master@{#13102}
This CL implements auto pausing video streams per stream with logic to
avoid toggling state too often.
Also re-enabling tests disabled for Mac, with the assumption the new
logic removes flakiness.
BUG=webrtc:5868,webrtc:5407
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2035383002 .
Cr-Commit-Position: refs/heads/master@{#13092}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This shouldn't be needed, but because the receiver assumes RTX packets
contain RED if configured to receive them (due to an incompatibility
issue), we also have to make sure we send them for now.
BUG=webrtc:5675
Review-Url: https://codereview.webrtc.org/2033763002
Cr-Commit-Position: refs/heads/master@{#13024}
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder
Note that the logic of how FEC and the needed bitrates is not changed.
BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972083002 .
Cr-Commit-Position: refs/heads/master@{#13018}
The fake network pipe will still only drop packets at an average rate of
|loss_percent| but in bursts at an average length specified by
|avg_burst_loss_length|.
Also added the flag -avg_burst_loss_length to video loopback.
BUG=
Review-Url: https://codereview.webrtc.org/1995683003
Cr-Commit-Position: refs/heads/master@{#12969}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
Also set the Configuration parameters in CreateRtpRtcpModules in the same order as the members are declared.
BUG=webrtc:5917
Review-Url: https://codereview.webrtc.org/2011433002
Cr-Commit-Position: refs/heads/master@{#12905}
During a period (e.g. 2 seconds), different metrics can be computed, e.g.:
- average of samples
- percentage/permille of samples
- units per second
Each periodic metric can be either:
- reported to an observer each period
- aggregated during the call (e.g. min, max, average)
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/1640053003
Cr-Commit-Position: refs/heads/master@{#12847}
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}
Reason for revert:
Should work after cl https://codereview.webrtc.org/1985693002/ is landed, which initializes the frames used by FakeWebRtcVideoCaptureModule. So intend to reland after that, with no changes.
Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
>
> Reason for revert:
> Speculative revert to see if failures on the DrMemory bot are related to this cl. See e.g. here:
> https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
>
> UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
> # 0 CopyRow_AVX
> # 1 CopyPlane
> # 2 I420Copy
> # 3 webrtc::ExtractBuffer
> # 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
> # 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
> # 6 FakeWebRtcVideoCaptureModule::SendFrame
> # 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
> # 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
>
> Original issue's description:
> > Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
> >
> > Reason for revert:
> > I plan to reland this change in a week or two, after downstream users are updated.
> >
> > Original issue's description:
> > > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> > >
> > > Reason for revert:
> > > Breaks chrome FYI bots.
> > >
> > > Original issue's description:
> > > > Delete webrtc::VideoFrame methods buffer and stride.
> > > >
> > > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > > to not imply an AddRef.
> > > >
> > > > BUG=webrtc:5682
> > >
> > > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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:5682
> > >
> > > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > > Cr-Commit-Position: refs/heads/master@{#12558}
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> > Cr-Commit-Position: refs/heads/master@{#12721}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d49c30cd2fe442f2b5b4ecec8d5cbaa430464725
> Cr-Commit-Position: refs/heads/master@{#12745}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1979193003
Cr-Commit-Position: refs/heads/master@{#12773}
Reason for revert:
Speculative revert to see if failures on the DrMemory bot are related to this cl. See e.g. here:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
# 0 CopyRow_AVX
# 1 CopyPlane
# 2 I420Copy
# 3 webrtc::ExtractBuffer
# 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
# 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
# 6 FakeWebRtcVideoCaptureModule::SendFrame
# 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
# 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
Original issue's description:
> Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
>
> Reason for revert:
> I plan to reland this change in a week or two, after downstream users are updated.
>
> Original issue's description:
> > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> >
> > Reason for revert:
> > Breaks chrome FYI bots.
> >
> > Original issue's description:
> > > Delete webrtc::VideoFrame methods buffer and stride.
> > >
> > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > to not imply an AddRef.
> > >
> > > BUG=webrtc:5682
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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:5682
> >
> > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > Cr-Commit-Position: refs/heads/master@{#12558}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> Cr-Commit-Position: refs/heads/master@{#12721}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1983583002
Cr-Commit-Position: refs/heads/master@{#12745}
Remove ViEEncoder::SetNetworkStatus.
Original cl description:
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.
BUG=webrtc:5884
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1970343002 .
Cr-Commit-Position: refs/heads/master@{#12741}
Reason for revert:
I plan to reland this change in a week or two, after downstream users are updated.
Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
>
> Reason for revert:
> Breaks chrome FYI bots.
>
> Original issue's description:
> > Delete webrtc::VideoFrame methods buffer and stride.
> >
> > To make the HasOneRef/IsMutable hack work, also had to change the
> > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > to not imply an AddRef.
> >
> > BUG=webrtc:5682
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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:5682
>
> Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> Cr-Commit-Position: refs/heads/master@{#12558}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1963413004
Cr-Commit-Position: refs/heads/master@{#12721}
This CL contains a few minor changes to names, function signatures and
merges two structs into one.
BUG=5868
Review-Url: https://codereview.webrtc.org/1952923005
Cr-Commit-Position: refs/heads/master@{#12716}
Reason for revert:
Breaks Chrome FYI using H264.
Need to investigate.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4170
Original issue's description:
> Remove ViEEncoder::SetNetworkStatus
>
> This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
>
> BUG=webrtc:5687
> NOTRY=True
>
> Committed: https://crrev.com/50b5c3be844ef571a28b2681c549443a26735d72
> Cr-Commit-Position: refs/heads/master@{#12699}
TBR=stefan@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/1978783002
Cr-Commit-Position: refs/heads/master@{#12715}
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
BUG=webrtc:5687
NOTRY=True
Review-Url: https://codereview.webrtc.org/1932683002
Cr-Commit-Position: refs/heads/master@{#12699}
This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)
Original reverted cl is in patch set #1.
Changes in following patch sets.
The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()
It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True // Due to bug in android_x86 cq builder....
Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
There are still a few places in VideoReceiveStream where the RTP module
is explicitly used, e.g. setting up a/v sync, but it's a bigger task to
change and that will be done in a follow up instead of in this CL.
BUG=webrtc:5838
Review-Url: https://codereview.webrtc.org/1947913002
Cr-Commit-Position: refs/heads/master@{#12642}
SSRC knowledge is contained withing VideoSendStream. That also means that debug recording is moved to VideoSendStream.
I think that make sence since that allows debug recording with external encoder implementations one day.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1936503002
Cr-Commit-Position: refs/heads/master@{#12632}
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.
This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/
patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1947873002 .
Cr-Commit-Position: refs/heads/master@{#12630}
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1917793002
Cr-Commit-Position: refs/heads/master@{#12620}
ViEChannel is now called VideoStreamReceiver.
There will be a follow up CL removing all rtp references from VideoReceiveStream, but that made this CL to big and it will be done separately.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/1929313002
Cr-Commit-Position: refs/heads/master@{#12619}
