Delete method RTPPayloadRegistry::ulpfec_payload_type().
RtpVideoStreamReceiver can check its own config to know what the
payload type is.
Bug: webrtc:8995
Change-Id: Idc2bc7d747d77127f2b2261ff50610422e5686a6
Reviewed-on: https://webrtc-review.googlesource.com/61501
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22414}
This CL adds methods to the SendSideCongestionController (SSCC)
interface for configuring pacing factor and allocation based data rate limits.
This means that old SSCC implement the same interface as the new, task
queue based SSCC. This also allows merging the max total allocated
bit rate into SetAllocatedSendBitrateLimits.
This is done in preparation for an upcoming CL where the SSCC version
is controlled by a field trial.
Bug: webrtc:8415
Change-Id: I4d5446a3bedd5b0c725dbd009fb75815fd661eff
Reviewed-on: https://webrtc-review.googlesource.com/61320
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22408}
Replacing Module based mechanism for processing with posting tasks.
This prepares for allowing the interval to be changed at runtime and
for removing the dependency on Module threads.
Bug: webrtc:8415
Change-Id: Iaad50466bec695be4ba26d8bd670a1981f2e0df4
Reviewed-on: https://webrtc-review.googlesource.com/60862
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22406}
Adding configuration of new GoogCcNetworkController to initializer, this
makes sure that it is properly initialized from the start. To achieve
this SendSideCongestionController waits until it has received the
necessary information to construct the object. This information should
be provided in the constructor for SendSideCongestionController in the
future.
Bug: webrtc:8415
Change-Id: Icc09b8b246bae9f9704b80855fc4caa3450b34fc
Reviewed-on: https://webrtc-review.googlesource.com/58099
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22404}
Cleanup after moving test/fake_audio_device to
modules/audio_device/include/test_audio_device.
Hide implementation of test audio device module in the anonymous namespace.
Bug: webrtc:8946
Change-Id: I2d49c3ec5d43eeb5f155d38de95f69ed3c537805
Reviewed-on: https://webrtc-review.googlesource.com/61426
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22401}
Removing the Synchronous call AvailableBandwidth from the
RtpTransportControllerSend interface. The bandwidth estimate is
provided trough a new interface that communicates with a struct
making it easier to add parameters in the future.
This prepares for removing locking behavior in
SendSideCongestionController that exists just to support this feature.
To keep backwards compatibility with the old
SendSideCongestionController, the struct TargetTransferRate
is constructed in RtpTransportControllerSend. This step can be
removed in the future when the old SendSideCongestionController
is deprecated.
Bug: webrtc:8415
Change-Id: I06f64a89848157de412901c989650d1ecf35246b
Reviewed-on: https://webrtc-review.googlesource.com/60800
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22387}
This CL corrects some errors that were included in the CL for reading
the AEC3 options in the audioproc_f tool
Bug: webrtc:8671
Change-Id: Iecaee0ebf08f8a8f75aba1d395dd467a41b876f3
Reviewed-on: https://webrtc-review.googlesource.com/60870
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22384}
For the buffering of |input_frames_|, we assume that frames
are ordered per simulcast layer but we make no assumptions
between layers.
For SVC, we still assume ordering of encode callbacks for
the spatial layers. If we ever add async codecs that support SVC,
they should still obey this assumption.
Bug: webrtc:8448
Change-Id: I4ebb0c1e1d0eef41d850ed5b92aacc79d0a11137
Reviewed-on: https://webrtc-review.googlesource.com/60801
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22380}
The stereo-ness depends on the input Capturer and Renderer:
1) The stereo-ness of playout equals to Renderer;
2) The stereo-ness of recording equals to Capturer.
Bug: webrtc:8978
Change-Id: Ib41b8294c30ef6db54fdaf9d1890de0135a976d1
Reviewed-on: https://webrtc-review.googlesource.com/60100
Commit-Queue: Pengyu Liao <pengyul@google.com>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22374}
This refactoring makes it easier to experiment with injectable components.
Bug: webrtc:8732
Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217
Reviewed-on: https://webrtc-review.googlesource.com/60863
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22372}
This adds the ability to the pacer to apply a congestion window by
tracking sent data. This makes it more reliable when the congestion
window is small enough to be filled at a high rate as there are less
thread context switches that might affect the timing and performance.
Outstanding data is not reduced by the pacer as it has no information
about acknowledged packet feedback. This is by design as the pacer would
also need to keep track of on which connection packets were sent or
received, requiring a larger, more complex, change to the pacer.
Bug: webrtc:8415
Change-Id: I4ecd303e835552ced042cd21186da910288a8258
Reviewed-on: https://webrtc-review.googlesource.com/51764
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22371}
This ensures that SendSideCongestionController is always initialized
with starting bitrate and bitrate limits.
Bug: webrtc:8415
Change-Id: If3b75e935dda755f9e0f40af1021f97ff150c9e9
Reviewed-on: https://webrtc-review.googlesource.com/59224
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22370}
If 10ms delayed report is scheduled at 1.9ms (truncated by TaskQueue clock to 1ms)
it may run at 11.1ms (truncated to 11ms, i.e. first time it look like 10ms passed).
But (test) clock with different time offset may see passed time as 9ms
which result in a test failure for a wrong reason.
Relaxing period expectation by 1ms should mitigate the issue
Bug: webrtc:8945
Change-Id: I902d8af436fc74d4a3a0ad8ffdb5a6d3565adb7d
Reviewed-on: https://webrtc-review.googlesource.com/58095
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22361}
* Removed video files which were not used by any tests.
* Removed ConferenceMotion_1280_720_50.yuv for mobile builds.
Bug: webrtc:8936
Change-Id: I0539e9fce20470fcc2f0af84bd297faffc4b587a
Reviewed-on: https://webrtc-review.googlesource.com/60942
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22359}
This reverts commit 7bb37b884b197ea22e2830b043c09018c186bad5.
Reason for revert: Breaks downstream projects
Original change's description:
> Reland "Rework rtp packet history"
>
> This is a reland of 6328d7cbbc8a72fdc81a766c0bf4039e1e2e7887
>
> Original change's description:
> > Rework rtp packet history
> >
> > This CL rewrites the history from the ground up, but keeps the logic
> > (mostly) intact. It does however lay the groundwork for adding a new
> > mode where TransportFeedback messages can be used to remove packets
> > from the history as we know the remote end has received them.
> >
> > This should both reduce memory usage and make the payload based padding
> > a little more likely to be useful.
> >
> > My tests show a reduction of ca 500-800kB reduction in memory usage per
> > rtp module. So with simulcast and/or fec this will increase. Lossy
> > links and long RTT will use more memory.
> >
> > I've also slightly update the interface to make usage with/without
> > pacer less unintuitive, and avoid making a copy of the entire RTP
> > packet just to find the ssrc and sequence number to put into the pacer.
> >
> > The more aggressive culling is not enabled by default. I will
> > wire that up in a follow-up CL, as there's some interface refactoring
> > required.
> >
> > Bug: webrtc:8975
> > Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> > Reviewed-on: https://webrtc-review.googlesource.com/59441
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22347}
>
> Bug: webrtc:8975
> Change-Id: Ibbdbcc3c13bd58d994ad66f789a95ef9bd9bc19b
> Reviewed-on: https://webrtc-review.googlesource.com/60900
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22356}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: Id698f5dbba6f9f871f37501d056e2b8463ebae50
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/61020
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22358}
This is a reland of 6328d7cbbc8a72fdc81a766c0bf4039e1e2e7887
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
Bug: webrtc:8975
Change-Id: Ibbdbcc3c13bd58d994ad66f789a95ef9bd9bc19b
Reviewed-on: https://webrtc-review.googlesource.com/60900
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22356}
This is a reland of 6f37ed78d99daa36e964ff0a65b205f0916d9949
CQ dry run OK except for missing iOS swarming bots.
NOTRY=True
Original change's description:
> Deprecate the adaptive level controller
>
> Level control handled by default-on AGC.
>
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}
Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
This reverts commit 6328d7cbbc8a72fdc81a766c0bf4039e1e2e7887.
Reason for revert: Breaks downstream build, due to use of std::pair constructor that some compilers appear to not support yet. See comment.
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I2fa7efc7d008c56f7a8f77bc9958c19119f69de8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/60880
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22350}
The GetTransportFeedbackVector() method is only used in tests, and they
can access the class directly anyway. Keeping it is adding code bloat
and is also making upcoming refactoring more difficult.
Bug: webrtc:8975
Change-Id: I8323addb3c1461dd73b30353c8d9fe9410471c15
Reviewed-on: https://webrtc-review.googlesource.com/60860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22349}
This CL rewrites the history from the ground up, but keeps the logic
(mostly) intact. It does however lay the groundwork for adding a new
mode where TransportFeedback messages can be used to remove packets
from the history as we know the remote end has received them.
This should both reduce memory usage and make the payload based padding
a little more likely to be useful.
My tests show a reduction of ca 500-800kB reduction in memory usage per
rtp module. So with simulcast and/or fec this will increase. Lossy
links and long RTT will use more memory.
I've also slightly update the interface to make usage with/without
pacer less unintuitive, and avoid making a copy of the entire RTP
packet just to find the ssrc and sequence number to put into the pacer.
The more aggressive culling is not enabled by default. I will
wire that up in a follow-up CL, as there's some interface refactoring
required.
Bug: webrtc:8975
Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
Reviewed-on: https://webrtc-review.googlesource.com/59441
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22347}
* Add support for SimulcastEncoderAdapter wrapping of encoder.
* Store input frame timestamps out-of-band, so we don't need to keep
a raw VideoFrame around just for it's timestamp.
* Store current frame rate in |framerate_fps_|, instead of in
codec settings struct.
* Add some comments and reorder some data members.
* Explicitly include VideoBitrateAllocator.
* Change type of |input_frames_|, to avoid one layer of indirection.
* Move VideoProcessor::CalculateFrameQuality to anonymous namespace.
This change should have no functional implications.
Bug: webrtc:8448
Change-Id: I10c140eeda750d9bd37bfb6cb1e8acb401fb91d3
Reviewed-on: https://webrtc-review.googlesource.com/60520
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22346}
This CL extends the options for the audioproc_f tool to match the options
for AEC3.
Bug: webrtc:8671
Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6
Reviewed-on: https://webrtc-review.googlesource.com/43120
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22344}
Previously, the only user of this code was the
VideoProcessorIntegrationTest. We have now changed that
test to directly calculate image quality metrics using libyuv,
similar to how the full stack tests and browser tests work.
Bug: webrtc:8448
Change-Id: Ia7a607d7ddc37741fba76d56aa7297851ffa1c6b
Reviewed-on: https://webrtc-review.googlesource.com/43760
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22341}
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.
Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
SSCC was accessing the pacer just to report values back to
RtpTransportControllerSend which already owns the pacer.
This CL moves those access methods.
To make RtpTransportControllerSend simpler, Call is made
responsible to keep track of network status used only as a
condition for report the pacer queuing delay.
Bug: webrtc:8415
Change-Id: I306bc9fcd3d8dcc7a637d51f2629ececebd48cad
Reviewed-on: https://webrtc-review.googlesource.com/60483
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22331}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. It will be done in 3 steps:
1. Test/fake_audio_device will be moved to production part of WebRTC
source code and renamed to test_audio_device_module. Old header is
replaced with alias to the new one.
2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice.
3. test/fake_audio_device will be removed.
This CL implements 1st step.
Bug: webrtc:8946
Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f
Reviewed-on: https://webrtc-review.googlesource.com/59740
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22325}
This CL introduces a different rampup behavir for the call startup and after resets
that may occur due to delay changes, clock-drift and audio path glitches.
Bug: chromium:819111, webrtc:8979
Change-Id: Ied1d7896be7f0c69aa6deb61475117021ca6ab09
Reviewed-on: https://webrtc-review.googlesource.com/60002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22312}
Just a null ID for now, but future CLs will fix that.
Bug: webrtc:8941
Change-Id: I393af0fef752ca3711421bdaf4b2e41cbe286bcf
Reviewed-on: https://webrtc-review.googlesource.com/58093
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22296}
This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.
Reason for revert: breaks downstream project
Original change's description:
> Separate test/fake_audio_device on API and implementation.
>
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
>
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
>
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}
TBR=kwiberg@webrtc.org,titovartem@webrtc.org
Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
