Upstream commit: https://webrtc.googlesource.com/src/+/0012bfa128c0cd6911a526471f92cd1b0587f35e
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}
Upstream commit: https://webrtc.googlesource.com/src/+/fa934326db5b39b7b2fdde9df943a36a7a469098
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}
Upstream commit: https://webrtc.googlesource.com/src/+/cbb13bba868989475c2059e92f6588dc07727642
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}
Upstream commit: https://webrtc.googlesource.com/src/+/1932b44aa22b3b5291b41db9e29f6ebbbfe4fb62
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}
Essentially a no-op since we're going to see this change
reverted when we vendor in dfa7b2b425.
Upstream commit: https://webrtc.googlesource.com/src/+/e13945bf0761d34b902ecbd4e1cc6deb1788a2c9
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}
Upstream commit: https://webrtc.googlesource.com/src/+/e2f02c2df07e00d3fd3d0ddab294dcddf49aaca0
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}
Upstream commit: https://webrtc.googlesource.com/src/+/5b9d4adfc88b7df0aea6f2f4d4b884d2eabcd657
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}
Upstream commit: https://webrtc.googlesource.com/src/+/5079e8a30aa082983379391a5d5ad64826d57286
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}
Upstream commit: https://webrtc.googlesource.com/src/+/be6bda7f64dcbdce0568c85bedb6a68fc6490706
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}
Upstream commit: https://webrtc.googlesource.com/src/+/76430c0bf1085a07fbc1b80b4f8e4970193c87ab
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}
We already cherry-picked this when we vendored bef5d63112.
Upstream commit: https://webrtc.googlesource.com/src/+/12f9d5ce601a13cd45d56f40ed9ed94f3a90d91e
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}
Upstream commit: https://webrtc.googlesource.com/src/+/d7d21337d1d027740a57dfa4e7f12e722a415547
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}
Upstream commit: https://webrtc.googlesource.com/src/+/f9ddf7fed6617c21f8be6f32737953751fb5701b
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}
Upstream commit: https://webrtc.googlesource.com/src/+/7fe62f25d14aef7dc17c4ba7742f8065c22fc968
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}
Allows Wasm DebuggerFrame to be suspended/resumed.
Disables ASAN for StackPointerInfo for better testing.
Depends on D214001
Differential Revision: https://phabricator.services.mozilla.com/D218246
Instead of dropping the frame's memory chunk when a frame is replaced, place the chunks into a global pool. This mostly helps when running stress tests like motionmark's html suite that cause intense allocation spikes (around 90 chunks per frame).
We should revisit whether this is needed if the cost of deallocating large regions of memory in mozjemalloc improves in the future.
Differential Revision: https://phabricator.services.mozilla.com/D223924
Rewrite `BigInt.asIntN(cst, BigInt(IntPtr-or-Int64))` to perform the sign-extension
in inline assembly using either `MSignExtendIntPtr` or `MSignExtendInt64`. (Or use
shift instructions when native sign extension isn't possible.)
See bug 1685708 for why optimising `BigInt.asIntN` is useful.
Differential Revision: https://phabricator.services.mozilla.com/D223337
This instruction is similar to `MSignExtendInt32`, but instead of just
extending its input to 32-bit, it allows extending to pointer size.
Used in part 9.
Differential Revision: https://phabricator.services.mozilla.com/D223336
