getAllNetworksFromCache => true (stop using old Android API)
requestVPN => true (is default in old api)
They have been enabled using field trial
for more than a year.
Bug: webrtc:13741
Change-Id: I288c4067193e95251f79d51e935dce555f6eb198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361581
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42945}
Disable the checks ensuring we reject mixed-codec simulcast
when the field trial is enabled.
The feature is however not yet implemented.
Bug: webrtc:362277533
Change-Id: Ib1601767c951d61aaa37a3d8767d0a81444d626c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361404
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42942}
Since media/ and pc/ both have to use this, and both
depend on call/, this seems to be the right place to put it.
Also factor out the interface that media will use in a separate
interface class.
Bug: webrtc:360058654
Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42933}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: I03bc27c920787a7305a9775e6341e26904592fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42931}
Will be later used to conditionally enable mixed codec simulcast
with a field trial.
Bug: webrtc:42220378
Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42929}
The fieldtrials can be used to override the static QP threshold
that is used in QualityConvergenceMonitor to determine if an
encoded video stream has reached its target quality.
The fieldtrials do not change the dynamic detection.
Bug: chromium:328598314
Change-Id: I5995860eff461f0c712293e34cf75834ce414bed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42928}
gmock and gtest includes are replaced in the script but this wasn't applied to the 'CHECK_MODE' causing false error report.
nit: Usage is printed when no arguments are provided.
Change-Id: I418a17b998934b0079f5bf19513097481f35aa70
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42927}
We keep information about the PipeWire camera status as a member of the
PipeWire session, but it's never updated and remains in uninitialized
state. Make sure it gets updated once PipeWire is initialized or when it
fails. There is currently no use for this member variable, but there is
a plan to use it so I'm rather keeping it instead of removing it.
Bug: webrtc:42225999
Change-Id: If409761b148be8f0724fd9ab7a1ed4cf0e459503
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360922
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42926}
This is to make it clear that this field indicate whether the upper bits
of the sequence number should be communicated. However, the current
implementation only sets the field if it is a key frame.
Bug: webrtc:358039777
Change-Id: Ic2c8b6d91499e4e5cf25b8ce9591d326d7044fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361402
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42924}
The VideoAdapter is used to configure encoding resolutions based on
requested_resolution in an orientation agnostic way[1]. This means that
if you request 1280x720 and the input frame is 720x1280, there is no
downscale happening.
However in the same file there is one instance of
VideoAdapter::OnSinkWants() where requested_resolution is assumed to be
expressed in landscape mode. This breaks the case where the 720x1280 is
requested but the frame is 1280x720 which causes inconsistent behavior
and breaks symmetry. This would also break simulcast since this code
path is only applied with the top layer's requested resolution while the
lower layers are still scaled in an agnostic way.
A new test is added to verify the fix. Prior to the fix, the first half
of the test was passing, after the fix both parts of the test pass.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/base/video_adapter.h;l=76;drc=02b5b024b66755a851a752b7851b124ba03f6cb6
Bug: webrtc:363019836
Change-Id: I564068e98c93cab89eb38a10b0f8378899438e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361160
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42923}
Remove mention of absl_deps - it is history already.
Rewrite motiviation of banning absl::Span to be up to date with c++20 state.
Remove motivation of banning absl::Mutex as it likely no longer accurate, and that ban might be re-evaluated.
Ensure allow list matches what is in root DEPS
No-Try: True
Bug: b/363943024, webrtc:342905193
Change-Id: I890a87511bafac7c51355d8f49e0237352eee7b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361302
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42914}
which broke since libsrtp included openssl/srtp.h instead of
its own srtp.h due to the order of include directories
BUG=webrtc:42234521
Change-Id: Idc5cba2114febd1e0835d201b6c23424a88e62d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360705
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42913}
Also add a presubmit check to verify we're not reintroducing it.
Bug: webrtc:342905193
Change-Id: Ic7eedb6a7fb257e3fd110b84d3921feb58f799d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42912}
ModuleRtpRtcpImpl and ModuleRtpRtcpImpl2 share certain components, RtcpReceiver in particular.
To always have Environment in RtcpReceiver both legacy and new module need to propagate it.
No-Iwyu: suggests too many changes, better address them separately.
Bug: webrtc:362762208
Change-Id: I2c885f57e24f135229fb7cd9781126d663017b3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361142
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42908}
This reverts commit a4cf34def1d82db263c3fa149afc8cd7e809b356.
Reason for revert: Let's test the bot a bit manually before adding it to the CQ.
Original change's description:
> Enable 'iwyu_verifier' bot.
>
> Change-Id: Idff49157c6a000c1693c3d9f1e3fc085beb36b76
> Bug: b/236227627
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361202
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42902}
Bug: b/236227627
Change-Id: Ifb9365e9e78514325b4333261e79b795e466c488
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361261
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42907}
Delegate control over number of times to encoder using AV1E_SET_AUTO_TILES that was added in https://aomedia-review.googlesource.com/c/aom/+/191102.
Bug: webrtc:351644568
Change-Id: I87ed11734e907c7f6c6508ac7389c84ececf5b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361140
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42903}
