42089 Commits

Author SHA1 Message Date
webrtc-version-updater
c7eacf81f2 Update WebRTC code version (2024-08-01T04:03:51).
Bug: None
Change-Id: I3b5d56ddce83c51017bf59e5537056fcfea16058
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358201
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42707}
2024-08-01 06:22:01 +00:00
Jan Grulich
9e755f0e19 PipeWire camera: Annotate functions with PipeWire calls to avoid CFI
Similar to PipeWire implementation of desktop capture, we have to avoid
CFI check for calls of dlopened PipeWire library. This avoid crashing
PipeWire camera backend when "is_official_build=true" option is used as
this turns on "is_cfi=true" enabling control flow integrity.

Bug: chromium:354776214
Change-Id: I7a9fc1c2d77c4ee0e8fe0586369b7246e0bb9180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358103
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42706}
2024-07-31 18:11:40 +00:00
Xinyu Ma
954fdb0c31 Pass Environment into RtpVideoSender
To make it available for FEC to use field trials in follow ups

Bug: webrtc:355577231
Change-Id: Id176d1320ef1c8b9a7243ebafb6986bd436d32d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357842
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42705}
2024-07-31 17:36:08 +00:00
Björn Terelius
8d7642a9f7 Remove unused QpFastFilterLow method
Bug: None
Change-Id: I63665a3fc9afd57aec8f0f7d2a2a2e631452f6c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358080
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42704}
2024-07-31 10:40:42 +00:00
webrtc-version-updater
3c177dd268 Update WebRTC code version (2024-07-31T04:06:12).
Bug: None
Change-Id: I7223fcbb3f4af52edde834b78c543059f929ff72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358121
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42703}
2024-07-31 06:02:24 +00:00
Philipp Hancke
b5ff55adee Reland "Enable TLS Client Hello extension permutation by default"
This is a reland of commit e13945bf0761d34b902ecbd4e1cc6deb1788a2c9
with additional backward compability defaulting to the new value.

Original change's description:
> Enable TLS Client Hello extension permutation by default
>
> similar to the previous change for DTLS. This affects native TURN/TLS
> connections which are already using this in Chromium.
>
> BUG=webrtc:422225803
>
> Change-Id: I605f106371f2dbe23b1ad5f8385e0e01abe7c48f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357903
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42688}

Bug: webrtc:422225803
Change-Id: Ic194e4f763029e65c1a15a6bbaabcfbcd2866eac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358120
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42702}
2024-07-31 06:01:22 +00:00
Gavin Mak
c51a5c00c4 Replace FindSrcDirPath
If the webrtc/src repo is checked out in a directory that isn't named
"src", FindSrcDirPath will loop forever. Instead of trying to find the
repo root, just use the location of the scripts and work out the
root with os.pardir.

Bug: b/333744051
Change-Id: Ifccdb85d3f9c7cb27ca57cc0b7bb96adf783660d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357980
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Gavin Mak <gavinmak@google.com>
Cr-Commit-Position: refs/heads/main@{#42701}
2024-07-30 21:11:33 +00:00
Gavin Mak
bde30d393b Make update_version.py and build_helpers(_test).py pylint compliant
Trying to change any one of these files will make presubmits
complain that the file isn't properly formatted. Format and rename
variables to be PEP-8 and pylint compliant.

Bug: b/333744051
Change-Id: I8dd4f7f05e52777a62b49659a3c264fe28926539
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42700}
2024-07-30 21:08:59 +00:00
Philipp Hancke
4ddd931023 video_replay: add pcapng support
as documented on
  https://pcapng.com/

Nanosec resolution for timestamps is assumed as described in
  https://gitlab.com/wireshark/wireshark/-/blob/master/writecap/pcapio.c

BUG=webrtc:351327754

Change-Id: Ieec601a33c131908e30e7f7e41ddc89ddc1c36b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42699}
2024-07-30 20:14:06 +00:00
Dan Tan
96c1b9c5ea Add variables to lend unused audio bits to video
This CL only adds variables necessary for the feature, which will be
implemented in later CLs.

Bug: webrtc:350555527
Change-Id: I71e56666e629f56168d316bf693150c0df0e2ecf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42698}
2024-07-30 18:42:16 +00:00
Danil Chapovalov
79518639d1 Propagate simulcast config field trials in VideoCodecTester
Instead of passing them through the global field trials string

Bug: webrtc:42220378
Change-Id: I75e406a9fb8bbee8de47f20ff8c68a1b49dfbf5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358141
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42697}
2024-07-30 18:17:27 +00:00
Mirko Bonadei
6b0de3f05b Revert "Replace instead of queueing render updates."
This reverts commit 76960dfdb638b5bee2f08f41236ea2a25a6aab08.

Reason for revert: Speculative rollback (performance).

Original change's description:
> Replace instead of queueing render updates.
>
> Bug: webrtc:351858995
> Change-Id: I6c07d71afeae886ff6a20509bca5b5c65f131e41
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356800
> Reviewed-by: Fabian Bergmark <fabianbergmark@google.com>
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Commit-Queue: Ranveer Aggarwal‎ <ranvr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42617}

Bug: webrtc:351858995, b/355009708
Change-Id: I0552238732a940fcb06543960fc563b9bd7ca6f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358140
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42696}
2024-07-30 16:49:57 +00:00
Florent Castelli
0012bfa128 Change DataChannelInit::priority to integer and forward to SCTP transport
The new type PriorityValue is a strong 16-bit integer matching RFC 8831
requirements that can be built from a Priority enum.
The value is now propagated and used by the SCTP transport, but enabling
the feature still requires a field trial for now.

Bug: webrtc:42225365
Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-30 15:07:25 +00:00
Gavin Mak
fa934326db Use input_api.change.RepositoryRoot instead of finding src/ dir
input_api provides a method to find the repo root so we don't have
to find it ourselves.

Bug: b/333744051
Change-Id: I95eaffba8b65de8ae3a13f6cd4874879ebd0a464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357902
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Gavin Mak <gavinmak@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42694}
2024-07-30 14:17:45 +00:00
Danil Chapovalov
cbb13bba86 Delete deprecated CreateAudioEncoderFactory with unused field trials parameter
Field trials are passed during AudioEncoder construction through Environment parameter
All known users were migrated to the same named function without parameters.

Bug: webrtc:343086059
Change-Id: I79e2edae22ab43f98a386430da82b41d1c71e426
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358061
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42693}
2024-07-30 12:32:24 +00:00
Danil Chapovalov
05309c5236 Delete AudioEncoderOpus constructor that doesn't provide Environment
Bug: webrtc:343086059
Change-Id: I55573eff8a13c504c7e14f370398bba1a6eae906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358060
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42692}
2024-07-30 11:40:34 +00:00
Danil Chapovalov
c2160b14b1 Delete expired field trial Audio-OpusAvoidNoisePumpingDuringDtx
Bug: webrtc:42222522, chromium:40174928
Change-Id: I2391b3078e5fff93edca3c3e6e568560b2a1c1cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357742
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42691}
2024-07-30 09:43:52 +00:00
Danil Chapovalov
1932b44aa2 Provide Environment for AudioEncoderOpus in tests when created using the trait
To allow delete old signature of the AudioEncoderOpus::MakeAudioEncoder function and thus guarantee Opus AudioEncoder always has an Environment

Bug: webrtc:343086059
Change-Id: Ib660678aeb5a549dddd1dffa3d8c28b2ec6b9d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356981
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42690}
2024-07-30 09:29:11 +00:00
Mirko Bonadei
dfa7b2b425 Revert "Enable TLS Client Hello extension permutation by default"
This reverts commit e13945bf0761d34b902ecbd4e1cc6deb1788a2c9.

Reason for revert: Breaks downstream project

Original change's description:
> Enable TLS Client Hello extension permutation by default
>
> similar to the previous change for DTLS. This affects native TURN/TLS
> connections which are already using this in Chromium.
>
> BUG=webrtc:422225803
>
> Change-Id: I605f106371f2dbe23b1ad5f8385e0e01abe7c48f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357903
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42688}

Bug: webrtc:422225803
Change-Id: I8020e420e270c0f47cb8e26a210c801e94f8de7d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357883
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42689}
2024-07-30 07:56:29 +00:00
Philipp Hancke
e13945bf07 Enable TLS Client Hello extension permutation by default
similar to the previous change for DTLS. This affects native TURN/TLS
connections which are already using this in Chromium.

BUG=webrtc:422225803

Change-Id: I605f106371f2dbe23b1ad5f8385e0e01abe7c48f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357903
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42688}
2024-07-30 06:01:19 +00:00
webrtc-version-updater
e8a3380ca0 Update WebRTC code version (2024-07-30T04:05:48).
Bug: None
Change-Id: I31f32bfeaaf03b9ffc2be86faa158755632ab2cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358040
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42687}
2024-07-30 06:00:17 +00:00
Danil Chapovalov
e2f02c2df0 Delete AudioEncoderFactory::MakeAudioEncoder
Make AudioEncoderFactory::Create pure virtual.

To finalize migrating AudioEncoderFactory to new interface for creating AudioEncoder and thus guarantee AudioEncoders always have an Environment at construction.

Bug: webrtc:343086059
Change-Id: I1d607082437c15201c8a75dd7a3925fe0f75b70f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42686}
2024-07-29 16:00:28 +00:00
Jeremy Leconte
c81f07b95d Add doc on how to handle python presubmit failures.
Change-Id: I346b622e6b9934090c0a6b5fd9d81596e957a14e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357882
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42685}
2024-07-29 14:17:35 +00:00
Artem Titov
e02a200f5e [numpy] Fix users of NumPy APIs that are removed in NumPy 2.0.
This change migrates users of APIs removed in NumPy 2.0 to their
recommended replacements
(https://numpy.org/devdocs/numpy_2_0_migration_guide.html).

Bug: None
Change-Id: I5c275ed3f39863d42b5c34df0723933f7a8b94a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358020
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42684}
2024-07-29 12:46:53 +00:00
Jeremy Leconte
2a8cca6a5d Run tests on Mac-14 machines.
All the builder machines have been upgraded to Mac-14.

Change-Id: Ia6fe055e21bcf483d08debc85109b36dedf18c5b
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357864
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42683}
2024-07-29 12:13:13 +00:00
Jeremy Leconte
d04efa32d6 Update builder xcode from 14 to 15.
Change-Id: If958fa2bf7e9fc1ecc7fbacf2316f11d3fefe9d1
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357881
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42682}
2024-07-29 11:42:43 +00:00
Florent Castelli
5b9d4adfc8 Move rtp_packet_sender.h to api/
Old copy of the header and some previous usage is kept around
for compatibility with downstream projects for now.

Bug: chromium:345101934
Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42681}
2024-07-29 11:40:45 +00:00
Jeremy Leconte
9b81d2c954 Increase iOS deployment target from 12 to 14.
Change-Id: I9e2eccc245ff7f168152fc628ac12f3517b16501
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357741
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42680}
2024-07-29 09:14:42 +00:00
Tony Herre
5079e8a30a Allow supplying a custom NetworkControllerInterfaceFactory per-Call in PeerConnectionDependencies
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.

Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
2024-07-29 07:17:14 +00:00
webrtc-version-updater
d74d085dc5 Update WebRTC code version (2024-07-27T04:04:57).
Bug: None
Change-Id: I98a58f2ca209fd153dc0ee3b09d7952ba232f2fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358000
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42678}
2024-07-27 05:54:56 +00:00
Gavin Mak
be6bda7f64 Flush NewContents cache in CheckPatchFormatted
Prior to https://crrev.com/c/5740609, NewContents never flushed cache
so the second NewContents() would always produce the same contents
post-yapf as as pre-yapf. Flush cache on second NewContents() call to
get updated file contents. Also fix the formatting a bit.

Bug: b/333744051
Change-Id: Ic627dd72675d7d3694b1978635ae047b38f06596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357960
Auto-Submit: Gavin Mak <gavinmak@google.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42677}
2024-07-26 20:41:42 +00:00
Philipp Hancke
76430c0bf1 TLS: enable TLS client hello permutation by default
this is flipping
  WebRTC-PermuteTlsClientHello
to a killswitch in the SSLStreamAdapter used for DTLS.

BUG=webrtc:42225803

Change-Id: I942851c474ec5e723c5b6c9f6206e7eafbe80ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42676}
2024-07-26 17:19:40 +00:00
Sergey Silkin
12f9d5ce60 Revert "Update support for missing HIGH profiles and 1080p"
This reverts commit 46b43e007296737751aea10685f92ddf4df63e0d.

Reason for revert: chromium:354143228

Original change's description:
> Update support for missing HIGH profiles and 1080p
>
> The High and ConstrainedHigh profiles are missing from the decoder capabilities. Also level 3.1 doesn't allow 1080p
>
> Bug: webrtc:347724928
> Change-Id: I3f33468327d2aaf352fc80f69d2ee31481bafcb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355001
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42528}

Bug: webrtc:347724928
Change-Id: I4d55b2982aca2e94ec983473336c4fa2a72d842f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357861
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42675}
2024-07-26 09:32:40 +00:00
Gavin Mak
d7d21337d1 Support infra/specs/PRESUBMIT.py on cog
cog workspaces don't have a git directory and can't run "git diff".
Replace it with python's difflib instead.

Bug: b/333744051
Change-Id: I5bd8ccd873a0db55f0bbadf165180b3f2aa42903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357900
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42674}
2024-07-26 08:48:02 +00:00
webrtc-version-updater
e9d066d3b7 Update WebRTC code version (2024-07-26T04:03:34).
Bug: None
Change-Id: Ic9535697379751a522a125a131340e97c90fdbfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357841
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42673}
2024-07-26 06:05:27 +00:00
Jeremy Leconte
6ed136c6ae Fix TestExtendedReportsCanSignalZeroTargetBitrate flakiness on Mac.
The test is flaky on Mac on it seems related to a timeout.
https://luci-analysis.appspot.com/p/webrtc/clusters/testname-v4/82d0b764552f0811b37cc651c0962399?tab=recent-failures

../../video/end_to_end_tests/extended_reports_tests.cc:196: Failure
Value of: Wait()
  Actual: false
Expected: true
Timed out while waiting for RTCP SR/RR packets to be sent.

Change-Id: I9b19d3952a761415ab65d15f188ae3336e43e97e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357820
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42672}
2024-07-25 09:16:44 +00:00
Danil Chapovalov
161956b89d Cleanup deprecated accessors in VideoFrame
Bug: None
Change-Id: I3f8f428f04e86c38d5cf6d481709b7bcdfbd495c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357781
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42671}
2024-07-24 13:49:19 +00:00
Jeremy Leconte
f9ddf7fed6 Replace test frame capturer wanted_fps_ by target_capture_fps_.
wanted_fps_ seems redundant with target_capture_fps_.
The problem with wanted_fps_ is that it lowers the capture fps but does not decimate frames so that a 30 fps stream played at 5 fps is played slowly instead of played at the normal speed with dropped frames.

Change-Id: I1440953f9909ad1d4a102a0671fe933d95498a1f
Bug: b/355120692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42670}
2024-07-24 09:30:22 +00:00
Björn Terelius
8089959877 Remove private SRTP include
Bug: chromium:40272799
Change-Id: I42a63497aa8321475bd3e2604376c1514ecd623e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357543
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42669}
2024-07-23 17:23:45 +00:00
Danil Chapovalov
f065ff85e2 Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch
Bug: chromium:40108588
Change-Id: Ifc334819dd486ac791b5d04faa6d6bd77a481dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349644
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42668}
2024-07-23 13:23:26 +00:00
Abby Yeh
35f10a083d Add listener to detect mute speech event, and callback function to handle the event
Bug: webrtc:343347289
Change-Id: I56b1433b0dd8220f95d7d72fb04b4f92fe4a905e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355761
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Abby Yeh <abbyyeh@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42667}
2024-07-23 13:01:39 +00:00
memetao
7fe62f25d1 Reland "Fix 'Image will be cropped if WindowCapturerWinGdi used'"
This is a reland of commit 844225a76a98aa3be5aca09c19ab72a5e7b6c38a

Original change's description:
> Fix 'Image will be cropped if WindowCapturerWinGdi used'
>
> Bug: webrtc:15719
> Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#41503}

Bug: webrtc:15719
Change-Id: Idbb2f4dcc8811d3b2b763a49adc7a57535b3d1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334380
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42666}
2024-07-23 10:28:10 +00:00
Danil Chapovalov
f90a3ad3b3 Reenable disabled passing tests
Libvpx was adjusted to support scenarios test verifies, but WebRTC tests were forgotten.

Bug: webrtc:42223649
Change-Id: I19a10c939d844d00dd564bc0a16fe21844cc7cfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42665}
2024-07-23 07:14:13 +00:00
Jeremy Leconte
1ac162ee20 Revert "Remove iOS Debug (simulator) from LKGR bots."
This reverts commit 4bded9601bfb38f2ef67574554c12370dca4708f.

Reason for revert: Fixed with https://webrtc-review.googlesource.com/c/src/+/357640

Original change's description:
> Remove iOS Debug (simulator) from LKGR bots.
>
> Temporarily skip while the bot gets fixed.
>
> Bug: chromium:353975341
> Change-Id: Ib42c18e929547c7abc58f2878c79f00f87001cae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357540
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42656}

Bug: chromium:353975341
Change-Id: I184c9e597a28fff3ae052a07d8e6f17cc2251188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42664}
2024-07-22 21:41:02 +00:00
Philipp Hancke
7b61b84ab1 Cleanup SSLStreamAdapter unit tests
BUG=None

Change-Id: I71fa442f6f9b95bad63a3d7d797433d95bf5c298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42663}
2024-07-22 17:21:33 +00:00
Danil Chapovalov
ac15a137ac In RtpVideoStreamReceiver do not rely on RTP sequence number unwrap to be stable
Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same.
That might not be true when several packets were inserted in between these two calls and unwrapper changed its state

This CL propose instead to unwrap once, and save the result in the intermediate struct.
To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number

Bug: webrtc:353565743
Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42662}
2024-07-22 15:42:12 +00:00
Jeremy Leconte
36b548b31a Revert "Upgrade iOS buildbot xcode to 15f31d."
This reverts commit 74c761384fdeb6b4acbef2d06cf610c0a1b6482e.

Reason for revert: breaks iOS compilation

Original change's description:
> Upgrade iOS buildbot xcode to 15f31d.
>
> Change-Id: I42a4b07668fe03191d9528fed73eed6500568890
> Bug: None
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357542
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#42660}

Bug: None
Change-Id: I2616433602f7bb5d3e9febc33f2008ffbeeb2065
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357544
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42661}
2024-07-22 11:36:00 +00:00
Jeremy Leconte
74c761384f Upgrade iOS buildbot xcode to 15f31d.
Change-Id: I42a4b07668fe03191d9528fed73eed6500568890
Bug: None
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42660}
2024-07-22 09:51:39 +00:00
Jeremy Leconte
175e0c95e3 iOS simulator upgrade iOS version to run the tests.
iOS 15.5 is not tested anymore and we start to test on iOS 18.0.

Change-Id: Ia7340d25f6cf8480763ea689db267c0c9a843319
Bug: b/353975341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42659}
2024-07-22 08:54:06 +00:00
Jeremy Leconte
69720eda3d Build bot for iOS simulator don't need a xcode version.
Skip CQ because it is currently broken and changes to config.star are not picked up by CQ anyway.

Bug: None
Change-Id: I3fb6c1fd8db6466b6f058f10d1232cc1624e0472
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42658}
2024-07-22 08:21:54 +00:00