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}
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}
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}
Small changes to the unit tests for send side congestion controller.
Mostly adding some extra waits and expectation verifications. This
prepares for an upcoming CL.
Bug: webrtc:8415
Change-Id: Id3086a485eda99732d01192cac9a91141158ab45
Reviewed-on: https://webrtc-review.googlesource.com/59223
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22270}
Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.
Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
DeRegisterNetworkObserver is not used, since
RtpTransportControllerSend owns the thread on which
SendSideCongestionController runs it would not be safe to allow it to be
called from outside. Deregistration should be done by destroying
RtpTransportControllerSend.
In the future, the RegisterObserver functions should be removed as well,
in favor of providing the observer in the constructor. This requires
breaking a circular dependency between RtpTransportControllerSend and
Call.
Bug: webrtc:8415
Change-Id: Ifeb4c5d4a41e4d8419994b3146980bdaaf9cd6a9
Reviewed-on: https://webrtc-review.googlesource.com/58098
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22249}
This is a reland of e6cefdf9c572cdce55ff0497ad6e516c76132ee8.
Original change's description:
> Moved congestion controller to goog_cc folder.
>
> Bug: webrtc:8415
> Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
> Reviewed-on: https://webrtc-review.googlesource.com/43841
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21928}
Bug: webrtc:8415
Change-Id: Ib5cf8641466655d64ac80f720561817f4cab49a9
Reviewed-on: https://webrtc-review.googlesource.com/53062
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22244}
The functionality in network_control is by design not rtp specific and
should live outside the rtp directory in congestion_controller.
Bug: webrtc:8415
Change-Id: I68778f1a630c8e93950040bc56ee864c201b5990
Reviewed-on: https://webrtc-review.googlesource.com/55383
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22240}
Makes the new task queue based congestion controller implement the
SendSideCongestionControllerInterface.
Bug: webrtc:8415
Change-Id: I3dfe11c2eb200bc8d85c83edf78d7fdd0129bbff
Reviewed-on: https://webrtc-review.googlesource.com/56781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22235}
This CL activates unit tests for the task queue based send side
congestion controller that will replace the current one in the future.
To be able to have the tests run side by side with the tests of the old
congestion controller, the old tests have been prefixed with "Legacy".
This CL also contains some minor fixes to the new congestion controller
code.
Bug: webrtc:8415
Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695
Reviewed-on: https://webrtc-review.googlesource.com/55381
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22229}
SetTransportOverhead was used by send streams to signal the packet
overhead that they received from Call. However, call receives the value
from OnNetworkRouteChanged in WebRtcVideoChannel and
WebRtcVoiceMediaChannel which is already propagated to
RtpTransportControllerSend. By skipping the round trip, the interface on
the rtp transport controller can be reduced.
Bug: None
Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7
Reviewed-on: https://webrtc-review.googlesource.com/57182
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22226}
This makes it easier to have the new and the old send side congestion
controller side by side. This namespace is only temporary. As soon the
new task queue based congestion controller is fully functional, the old
will be deprecated and removed together with the temporary namespace.
Bug: webrtc:8415
Change-Id: Ie817511345c91cab2ebca68f038075875c7e6529
Reviewed-on: https://webrtc-review.googlesource.com/56720
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22221}
To be able to safely move over to the new code, the revised
code is added alongside the old code. Most of the files added
in this CL are more or less direct copies of the previous code.
This new version of send side congestion controller will be
activated under a field trial in a followup CL.
Bug: webrtc:8415
Change-Id: I034e583cf891a8f6357119739a1517cc0a4abe88
Reviewed-on: https://webrtc-review.googlesource.com/53322
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22212}
This prepares for a later CL providing two implementations of
SendSideCongestionController.
Bug: webrtc:8415
Change-Id: I890dbe4b88bf609921558e03aac66b42629857c8
Reviewed-on: https://webrtc-review.googlesource.com/56700
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22151}
The fake rtp transport controller is only used by CallBitrateTest, but
the functionality tested in CallBitrateTest is now tested in
RtpBitrateConfiguratorTest. Removing the fake rtp transport controller
send reduces the complexity of refactoring the rtp transport controller
send interface.
Bug: webrtc:8415
Change-Id: I4673daea4e68521e7e14293514830d6e704219bc
Reviewed-on: https://webrtc-review.googlesource.com/54480
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22125}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Reland "Moved congestion controller to task queue.""
>
> This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
>
> Reason for revert: Cause increased congestion and deadlocks in downstream project
>
> Original change's description:
> > Reland "Moved congestion controller to task queue."
> >
> > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
> >
> > Original change's description:
> > > Moved congestion controller to task queue.
> > >
> > > The goal of this work is to make it easier to experiment with the
> > > bandwidth estimation implementation. For this reason network control
> > > functionality is moved from SendSideCongestionController(SSCC),
> > > PacedSender and BitrateController to the newly created
> > > GoogCcNetworkController which implements the newly created
> > > NetworkControllerInterface. This allows the implementation to be
> > > replaced at runtime in the future.
> > >
> > > This is the first part of a split of a larger CL, see:
> > > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > > For further explanations.
> > >
> > > Bug: webrtc:8415
> > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21868}
> >
> > Bug: webrtc:8415
> > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> > Reviewed-on: https://webrtc-review.googlesource.com/48000
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21899}
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8415
> Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
> Reviewed-on: https://webrtc-review.googlesource.com/52980
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22017}
TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/53262
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22023}
This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
Reason for revert: Cause increased congestion and deadlocks in downstream project
Original change's description:
> Reland "Moved congestion controller to task queue."
>
> This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
>
> Original change's description:
> > Moved congestion controller to task queue.
> >
> > The goal of this work is to make it easier to experiment with the
> > bandwidth estimation implementation. For this reason network control
> > functionality is moved from SendSideCongestionController(SSCC),
> > PacedSender and BitrateController to the newly created
> > GoogCcNetworkController which implements the newly created
> > NetworkControllerInterface. This allows the implementation to be
> > replaced at runtime in the future.
> >
> > This is the first part of a split of a larger CL, see:
> > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > For further explanations.
> >
> > Bug: webrtc:8415
> > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21868}
>
> Bug: webrtc:8415
> Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> Reviewed-on: https://webrtc-review.googlesource.com/48000
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21899}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
Reviewed-on: https://webrtc-review.googlesource.com/52980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22017}
This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
Bug: webrtc:8415
Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
Reviewed-on: https://webrtc-review.googlesource.com/48000
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21899}
This reverts commit 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
Reason for revert: Major regressions on perf bots.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
The goal of this work is to make it easier to experiment with the
bandwidth estimation implementation. For this reason network control
functionality is moved from SendSideCongestionController(SSCC),
PacedSender and BitrateController to the newly created
GoogCcNetworkController which implements the newly created
NetworkControllerInterface. This allows the implementation to be
replaced at runtime in the future.
This is the first part of a split of a larger CL, see:
https://webrtc-review.googlesource.com/c/src/+/39788/8
For further explanations.
Bug: webrtc:8415
Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
Reviewed-on: https://webrtc-review.googlesource.com/43840
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21868}
Merge OveruseDetector into the TrendlineEstimator (send side BWE only) and remove the OveruseDetector from DelayBasedBwe. The behavior should be the same as before. One expection is that if no packets were received for kStreamTimeOutMs (2 seconds), it would previously reset the trendline estimator but not the detector. Since they have been merged, it now resets both.
Create an interface that the estimators will implement to facilitate experimentation with different estimators/detectors.
Bug: webrtc:8729
Change-Id: I5c3d2161a0d0dcb2e8a140c0fd887f0286d70fd4
Reviewed-on: https://webrtc-review.googlesource.com/38781
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21852}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.
Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
Diff patch set 1 and 2 to see actual differences to the last
patch.
Bug: webrtc:6828
Change-Id: Ie0c85d41df47c2a2505bc71b20fdb3834bdeaf12
Reviewed-on: https://webrtc-review.googlesource.com/36920
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21492}
