Update libwebrtc to 133
Categories
(Core :: WebRTC, task)
Tracking
()
Tracking | Status | |
---|---|---|
firefox137 | --- | fixed |
People
(Reporter: mjf, Assigned: mjf)
References
(Blocks 1 open bug)
Details
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(153 files)
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Our update to v132 has landed in Nightly. It is time to update to v133.
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Comment 3•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/2031ab5c788e338954d7b37da13039eff0a7c354
Update fuzzer documentation.
Change-Id: Id9dbaf42881a3ed3f377142b116151aebfd21192
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368020
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43388}
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Comment 4•18 days ago
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Essentially a no-op since we're going to see this change
reverted when we vendor in 54ed3ad524.
Upstream commit: https://webrtc.googlesource.com/src/+/775639e930f14a619974944594b40c633cc574a3
Set default scalability mode for H.265 to L1T1.
H.265 does not have software fallback, and it may have issue supporting
more than 1 temporal layers on some devices. Set default to L1T1 when
scalability is not configured, or if a scalability mode is reported as
not supported by encoder.
Bug: chromium:41480904
Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43389}
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Comment 5•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/c3021aece37dd6c07ec9460b1400d80e12c6757b
Update WebRTC code version (2024-11-13T04:06:47).
Bug: None
Change-Id: Icf9312dd8536646896c74d61cc0e003f4cc39175
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368101
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@{#43390}
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Comment 6•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/019bca9590c5440ee3553ec0ce83147c15f7778f
Remove deprecated CreateFromIvfFileFrameGenerator.
Change-Id: Ic33c1fa0a61a8e4f35f951f0334df71f34cb212b
Bug: b/378855419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368161
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43391}
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Comment 7•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/b7f5e7fb2902f9267c8d262430b3d76e83e3fb7f
Fix video renderer slowdown by wrong RenderTime
This commit fixes the issue of video playback in slow motion caused by VCMTiming being unable to provide the correct rendering time in
scenarios of continuous network packet loss
WANT_LGTM=mbonadei
Bug: webrtc:376183208
Change-Id: I63617068506e536c4b812215ea084eec18e8ee06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367000
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43392}
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Comment 8•18 days ago
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We already cherry-picked this when we vendored 463ce639a6.
Upstream commit: https://webrtc.googlesource.com/src/+/ebb11c4c87e4c0288d1ac18f38b4efa4254b3c6a
With stereo decoding and mono packets produce mono after CN/PLC
The workaround in https://webrtc-review.googlesource.com/c/src/+/367740
is incomplete because it does not fix the issue for the first decoded
mono packet after CN/PLC. This CL extends the workaround to such a case
and adds a unit test for it.
Note: it was verified that the 2nd packet after CN/PLC is trivial
stereo.
Credits: jakobi@webrtc.org for raising the concern
Bug: webrtc:376493209
Change-Id: Ide27e411781693f14629cf9db8b6c0c0fc762a17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368160
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43393}
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Comment 9•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/4c171e84c35d3e6342cf8a4d1bc7a77f429918d2
Prevent upscaling when calculating sample values.
Bug: webrtc:358039777
Change-Id: I33edc12f312d0d37eac0c39a913313a1aa6f1de5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366942
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43394}
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Comment 10•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/90da0650b53d338c19de0dfee9ca3d9b382d7b5c
Allow to specify a 'fps_hint' when creating a IvfVideoFrameGenerator.
Change-Id: Id75694f9dccfa6523f383e03dd90067fb6894b37
Bug: b/378855419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368162
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43395}
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Comment 11•18 days ago
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We already cherry-picked this when we vendored 775639e930.
Upstream commit: https://webrtc.googlesource.com/src/+/54ed3ad524caf45cb043d7dbb59914a1167290f2
Revert "Set default scalability mode for H.265 to L1T1."
This reverts commit 775639e930f14a619974944594b40c633cc574a3.
Reason for revert: Breaks internal tests.
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
No-Try: true
Change-Id: I5485b1abfd5f586ec187cc57817940aa2efd72af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368200
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43396}
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Comment 12•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/d4a3002b9bd0947a19ecb1bcb6df45ce884b7be6
srtp: remove deprecated non-span versions of key setters
BUG=webrtc:357776213
Change-Id: Idca7defe99b6d3dafb538a8a7599fe7edf2bff43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363141
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43397}
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Comment 13•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/0c6d31919e17393e302eb1c00c5efa0e3b225ec5
Enable RFC 8888 feedback if negotiated
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.
This should eliminate the need for the "force" flag in the field trial.
Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
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Comment 15•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/2578802038c383c6dcd7e7071d52cf884967cb0b
Add third_party/zstd to the DEPS file.
Dependency is required for Chromium roll in WebRTC.
Change-Id: I284c55f97bae3eee638d7a9f9fb5319fa1ae24e8
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43399}
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Comment 16•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/3b2402bd238289ea5539fc464435ce3c217d8f55
Fix rust DEPS.
* This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/367921.
* Also add 'enable_rust' gn arg when running build_aar because it is failing otherwise (https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket/8731293654930007281/+/u/build_android_archive/stdout).
Change-Id: I676ca47255e9b33f04487624625b0078dcb137a7
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368300
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43400}
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Comment 17•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/8d8fc3222a05b34015e3a8aa27230599df06c17f
Cleanup WebRTC-LegacyTlsProtocols field trial from field trial list
which is already gone from the code.
BUG=webrtc:40644300
Change-Id: Ic4a53d7895fa49d8417f11778d128740cecaee49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43401}
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Comment 18•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/0b333f2a40762680a1d6026319fb075c8c483e73
Update WebRTC code version (2024-11-15T04:04:53).
Bug: None
Change-Id: Icb6f055fbaafb5e4c0f52258bb4c2250f191628b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368262
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@{#43402}
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Comment 19•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/7ef136048559ec436d9fc733eb76cc3c6f4d0ce9
Fix issue that all macros not defined in rtc_pc_unittests
The gn target for rtc_pc_unittests cleared the "configs" that is by
default set for rtc_test. Restore it back so we get RTC_ENABLE_H265
macro when rtc_use_h265 is configured.
BUG: chromium:41480904
Change-Id: If172482776e5be2ad99d976db12dcfa556ee8a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368183
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43403}
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Comment 20•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/f54707cd715e030c8162c476ab70691437fffbf9
Reuse VP9 simulcast stream limits for H.265.
H.265 should have limits probably between VP9 and AV1, instead of using
VP8 tables. For now we reuse VP9 tables but eventually we may create
table for H.265.
Bug: chromium:41480904
Change-Id: I6dc2ec629142ee06f1c82a2df30d753ec1353496
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368240
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43404}
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Comment 21•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b2fc13d0941677ca67fa0d91e0736a3c49d70278
fix stun prober return fail in windows
stun_prober will fail on Windows and return RESOLVE_FAILED
Bug: webrtc:378688998
Change-Id: I3b957f6b2adf6658a0f6b83c8ff427ffd9779f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43405}
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Comment 22•18 days ago
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||
Upstream commit: https://webrtc.googlesource.com/src/+/dd8d2ab8908b79ccc50dee2cae21a82eaa915f70
Allow union initiliazation for webrtc::webrtc_pc_e2e::AudioConfig.
Change-Id: If7f4ac960528099111dd4e195f5934084bde564a
Bug: b/379255467
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368340
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43406}
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Comment 23•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/9ab2ac126dfee8270dd9221ad181a0661cfada89
Update WebRTC code version (2024-11-16T04:04:32).
Bug: None
Change-Id: I9c348c3b2bab12b969ed1ee209928ea4b7b90b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368360
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43407}
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Comment 24•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1dcf202ffe88a6de007f1331808680576fcc955d
Fixes a linear interpolation bad access
Bug: webrtc:353425611
Change-Id: I9c38428d2463d9d76047ad5be84ed57cba9ebb72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367981
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43408}
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Comment 25•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/ab0b1888a5ee11739659ff7bd13f7e020d225e3b
Update WebRTC code version (2024-11-18T04:04:38).
Bug: None
Change-Id: Ia97424876545632ec904d87a4466105650745297
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368420
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@{#43409}
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Comment 26•18 days ago
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||
Upstream commit: https://webrtc.googlesource.com/src/+/79c380c5b7b5d4626518810b058cd9fe5358984b
Always compile rtc_base/trace_categories.{h,cc}.
Instead use the preprocessor to avoid compiling Perfetto related code
when RTC_USE_PERFETTO is not defined.
Bug: None
Change-Id: I85b37cb0287327035ac2e8feb3caf9505486a1e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43410}
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Comment 28•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/752235261edeffeed54fdd179a214387305cfd82
Remove all references to codec-level transport-cc functions and flags.
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
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Comment 29•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c79be57b479d5bd4a696730f476c457032e6a804
Reland "Set default scalability mode for H.265 to L1T1."
This is a reland of commit 775639e930f14a619974944594b40c633cc574a3
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43412}
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Comment 30•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/bda11ca6da9a420a92cb2d5d22ea28d29dd85251
Add histogram WebRTC.Video.EstimatedClockDrift_ppm
TimestampExtrapolator maps RTP timestamps of received video frames
to local timestamps. As part of this mapping, the clock drift
between the local and remote clock is estimated.
Add the histogram WebRTC.Video.EstimatedClockDrift_ppm to log the
relative clock drift in points per million.
Bug: b/363166487
Change-Id: I0c2e628ef72c05a93e1f3138c8f71c77467130b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368342
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43413}
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Comment 31•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/69cc6956999204539e68aa33d190391fe6fa61f8
Comment unused variables in implemented functions 14\n
Bug: webrtc:370878648
Change-Id: I7c48313e64fafb8f23121e9bae1d50c3d32f7d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366983
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43414}
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Comment 33•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/3ffe94314a3f18402b159e6c7146088c48137de5
Fix lint warnings in TaskQueueStdlib
Bug: None
Change-Id: I4fd89dac39c0585793601d7adb5181a6ac15a64f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368460
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43415}
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Comment 34•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/17554c1c4ce06f4aec3d4d49b11efc23bbb7f3ab
Add graph for ecn packet count in incoming/outgoing CCFB
Also add a plot group l4s.
Usage: event_log_visualizer --plot=l4s filename |python3
Bug: webrtc:42225697
Change-Id: I5e1ee7028b9fb0707d5cabfe6d6f27c348e70a22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367199
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43416}
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Comment 35•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/7f775bc94c0b5b4329b4dd5ec2d9ea66161ce789
Ensure accurate FPS calculation for low frame rates
When receiving streams with frame rates around 1 fps, the decode and
render fps were incorrectly reported as 0, even though frames were being
decoded successfully.
This commit addresses the issue by adjusting the calculation in
RateStatistics to better handle streams with frame intervals that are
close to the window size.
1 fps streams are an important special case that occur frequently in
in screen share scenarios.
Fixed: webrtc:354625675
Change-Id: I1362768229a3abab5929220ba4bbd5ccb06a33d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368080
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43417}
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Comment 36•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/331ca306353bd8308d5a1f05098584543a99a6f3
Remove py_quality_assessment
and old TODOs in conversational_speech
Bug: webrtc:379542219
Change-Id: I7a6c087ce42f854d9b440da018248323b2435b55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368500
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43418}
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Comment 37•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/56085ea0d1b84e4390ff9e08e78a3888b8e631d2
AGC2 test: add missing include
Bug: webrtc:42232605
Change-Id: I8fcb66cf8ee27bf630433cdfee4a3386138cd7a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365521
Owners-Override: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43419}
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Comment 38•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/4c9dbd508d37254975790ab40def1d2c024d8074
Remove/update TODOs assigned to alessiob
Bug: webrtc:379542219
Change-Id: I1da54a9a13187d9e7d836dd4e1a85e49b685d971
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368540
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43420}
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Comment 39•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/aae790e3fe29c9c5b977a4bf925586f633265049
Update WebRTC code version (2024-11-19T04:07:36).
Bug: None
Change-Id: I9d956188867acee11f3342c93ca34fa5bdd1723b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368600
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@{#43421}
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Comment 40•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/999f02bd5fa39c2c51dc146ad2112aa57b60059d
Implement playout stats for ios AudioDeviceModule
Bug: webrtc:378966976
Change-Id: I30169b43f7fc8aba4832a77043566129d5b087a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368320
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43422}
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Comment 41•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/8337c966d469339e63a1b71eb95d675656e3e0ef
Use default probe duration if target higher than networkstate estimate
Use default probe duration and probe delta if probe target higher than
network state estimate.
Bug: webrtc:42224658, b/379234056
Change-Id: I1e6283681d005111fce5fc90e468b1ce2ce4b81f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368620
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43423}
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Comment 42•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/5f163fcaa0d3190f25a5186da3d7ec033128b405
Align Int16FrameData test class with AudioFrame
This updates test code that tests interleaved audio frames to use
some of the same properties and types as AudioFrame (rather than copy).
The CL also moves code from audio_processing_unittest.cc that modifies
the buffer owned by Int16FrameData, into Int16FrameData.
Bug: none
Change-Id: Iab37227deb302bf4fc832633d312262e5249caad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355960
Reviewed-by: Per Ă…hgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43424}
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Comment 43•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/2d47c9395b393aff95a93f3ed60d96c266a008f0
Correct H.265 level-id in fmtp line for offer/answer.
On a sendrecv m-line, the offered level-id represents the maximum that
can be both sent and received; on a sendonly m-line, the offered
level-id represents the maximum that can be sent; on a recvonly m-line,
the offered level-id represents the maximum that can be received.
Also according to RFC 7798 section 5, the highest level indicated by the
answer is either equal to or lower than that in the offer
Bug: chromium:41480904
Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43425}
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Comment 44•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/fb62f907066102a82ba503fa4b712a2cae9be516
Verify that transport-cc is used when RFC8888 field trial is off.
This is preparatory to ensuring that transport-cc gets turned off when
RFC8888 ccfb is negotiated.
Bug: webrtc:378698658
Change-Id: Ie76677bd6aa046701562bbd93d8489858488f863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368543
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43426}
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Comment 46•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/d63aacb4604b0ea8ef49b8747f047beef25f98ae
Use 'checkout_linux' instead of 'checkout_fuzzer' to checkout libFuzzer.
Using 'checkout_fuzzer' breaks the autoroller. Another fix would be to add 'checkout_fuzzer' to True here:
https://source.chromium.org/chromium/infra/infra_superproject/+/main:build/recipes/recipes/webrtc/auto_roll_webrtc_deps.py;l=30;drc=61d198818ce21c9a9721a9880b806ff35b61d322
Change-Id: I0003a1bab58947e733dbe11dfa2fb349a95fda0c
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368660
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43427}
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Comment 47•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/4c5e72e3e0c8a80c2c0704bb1ab32718d63c7a04
Update WebRTC code version (2024-11-20T04:03:13).
Bug: None
Change-Id: I5e38b728b8f2915b82d898561c2e4c50f4f42a36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368701
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@{#43428}
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Comment 48•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/da7b7ca1c12621f30136f0c59e651e04806e7ba5
Comment unused variables in implemented functions 15\n
Bug: webrtc:370878648
Change-Id: I4529c17f54c653864cca27097e44c843210b9c52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368061
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43429}
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Comment 49•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/2a69ddbe9e8ec9a30618876ef8e2516f4f02b311
Remove an unused conversion function.
Followup to https://webrtc-review.googlesource.com/c/src/+/366943
Bug: None
Change-Id: I3a1fa2307300f7ea4f03a73b9c162d8b98d4c02f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43430}
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Comment 50•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c18143277289858115304db5bae3bb04bf0746da
Add debug logging in WavWriterTest.LargeFile
Also CHECK in OutputPathWithRandomDirectory. This function is used in tests that need a unique folder to avoid interaction with other tests that may run in parallel. Continuing with a non-unique folder if the creation fails, is likely to cause surprising errors later on.
Bug: webrtc:379973428
Change-Id: I6a30ef9034be8132e2362eff5e46e3b99b30acd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Ă…hgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43431}
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Comment 51•18 days ago
|
||
We already cherry-picked this when we vendored 18db2cac3f.
Upstream commit: https://webrtc.googlesource.com/src/+/9aeed0c5f4518763ad6e8aa9547223301612805b
Avoid potential deadlock due to queue in corruption detection.
In particular, some platforms have a limited pool of frames in the
capturer stack, so we need to avoid stashing raw frames in the frame
instrumentation generator that may be dropped by limiting the size of
the queue and avoid putting anything in there until we know we will
send it to the encoder.
Bug: webrtc:358039777
Change-Id: I054ae53dd5e6ac6a22da39c5049f47788561e77a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368641
Reviewed-by: Ă…sa Persson <asapersson@webrtc.org>
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43432}
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Comment 52•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/00e86b3cb06b5f8d89e2f1d679ec313036afca3a
Update WebRTC code version (2024-11-21T04:06:03).
Bug: None
Change-Id: I092b703fcd557f2536fef3237858cd66a4ac2573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368787
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@{#43433}
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Comment 53•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/5ad1daeed92b65b5b136cb5a1df549521783fbfb
setParameters should not throw when only level mismatch.
According to latest requirement, when the level reported by
RtpSender.getCapabilities() for H.265 is different from that was
negotiated, we should not throw when setParameters() is called with
level-id set to that reported by RtpSender.getCapabilities().
Underlingly negotiated codec level should remain unchanged.
Bug: chromium:41480904
Change-Id: I28bbdb5f0a0ab0d98315f56c80004601afc91a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43434}
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Comment 54•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/2e7e049bb4a7b5ca6d2e5e0763a4d147d2bd6e37
Don't use transport-cc if RFC8888 feedback is negotiated.
Bug: webrtc:378698658
Change-Id: I06536445d32577b7b4d24ae7ca529d9b270b34d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368863
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43435}
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Comment 55•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/8da15c43dd0b841ee518a43b612597277ebfc6a0
Avoid depending on codec info for audio jitter stat.
The clock rate is already known by the RTP statistician.
Also included some minor code cleanup.
Bug: b/331602608
Change-Id: I335fa2a1cfd7dcceb286706d295a175a92f6797c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368920
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43436}
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Comment 56•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/253b8464ffe06426e5d603636039d75602eba6d1
Fix AV1 encoder do't set end_of_picture when the top layer is dropped
Bug: webrtc:357721007
Change-Id: I4e318618192aa9d58a2ef6338f7b1e2ee5140254
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366100
Reviewed-by: Ă…sa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43437}
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Comment 57•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/319892c4d90f9c54ca075fc32816e52e9fc9074b
Update WebRTC code version (2024-11-22T04:05:20).
Bug: None
Change-Id: I6e509615a6eaf2d78c051aaafb42911b4f1a53b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368890
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@{#43438}
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Comment 58•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1b0371a54e601476e660c3d668561782eed16ffc
Reduce the moving median window size in Remote ntp time estimator.
Too big median window will cause errors with large clock drifts, since we'll end up using old values for estimated clock drift.
If the window is too small, the remote clock offset estimation could be noisy or we could even end up using outliers as the offset estimation.
I will not claim that I choose the correct value, and I'm not sure how to measure the quality of the remote clock offset estimations.
Bug: webrtc:379809147
Change-Id: Ib317548d3eec74105d468ef53830e12eb114df7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368580
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43439}
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Comment 59•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/cd013b1d59eb6ac301b9bd872b1b85b12def7ae9
Opus decoder: stereo decoding by default (behind field trial)
- Add `WebRTC-Audio-OpusDecodeStereoByDefault` field trial
- Behind that field trial, `AudioDecoderOpus::SdpToConfig` uses 2
instead of 1 as default number of channels when the `stereo` codec
param is unspecified
- Instead of wiring up `FieldTrialsView` to `SdpToConfig`, which
requires API changes that break downstream projects, a change in
`AudioDecoderOpus::Config` is made to signal when the number of
channels is forced via SDP config
Bug: webrtc:379996136
Change-Id: If70eb19bc7e3bc74dd0423610cb04ae33ea602fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368860
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43440}
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Comment 60•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/394da76a9c9d8959049a202f5db0c209c176d8c6
Propagate ECN information through Network Emulation
Bug: webrtc:42225697
Change-Id: Idbd1ded3b5401c86d9afc6fd74f6da58e47bf5cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368862
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43441}
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Comment 61•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/24992e95183d4c5a853b6e791485851a0cf55cbd
Report all usage patterns to UKM
This stores usage for all cases, making it easier to discover
abusive usages on unexpected patterns.
Bug: None
Change-Id: I62c9b07498e811ac04c221f57cfbc02312aaaacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43442}
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Comment 62•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/e5f6f1fab452de0e6b519290fe7d93c9946f1263
Add optional corruption filter settings to EncodedImage.
This is a prerequisite for enabling implementation-specific filter
settings for automatic corruption detection.
Bug: webrtc:358039777
Change-Id: I363c592aa35164f690dd4ad1204e90afc0277d8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368940
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43443}
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Comment 64•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c63e43f27d2fecb57b8f288736f238662753c83f
Deprecate PeerConnectionFactoryDependencies::audio_processing
Bug: webrtc:369904700
Change-Id: Ic0982abcff2097e4e52e55a4b9c90ec25ae33b90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367961
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43444}
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Comment 65•18 days ago
|
||
Essentially a no-op since we're going to see this change
reverted when we vendor in a08189b948.
Upstream commit: https://webrtc.googlesource.com/src/+/38ddea5ee3320bf3441aeb3654e099b3695c9789
Add InsertPacket method that takes RtpPacketInfo.
The version which only passes receive_time will be removed (once migrated).
Keeping the version that only passes header and payload for convenience.
This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
Bug: webrtc:42223109
Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43445}
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Comment 66•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/89432bc2258729c29f22d1f84e008014c3817ea3
Update WebRTC code version (2024-11-24T04:08:23).
Bug: None
Change-Id: Ifdd62847941018c9e0431a2fce7d12f0de3b0df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369085
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@{#43446}
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Comment 67•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/924dc088dc886cacdbd29fe87e0ec895f57938f2
Use 16bit unsigned for channel id for TURN
Bug: webrtc:345518625
Change-Id: I0ee879e9a35cd9831e035a661d54201dc6defac9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353901
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43447}
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Comment 68•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/06723eaab880ceb1e5bac2f5cb69ba859b52b4aa
Default max limit probe to 2x current bwe
If max allocated bitrate change, default max limit probe to 2x current
BWE.
Bug: webrtc:369044000, b/370883514
Change-Id: Ibaf79fff94157186002728828d6574bea21afd24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368820
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43448}
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Comment 69•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b40c5598580478b44b5e14ebd25d74ae3f5f1217
Set voice RTCP mode based on the RemoteContent and not based on the LocalContent.
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).
Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
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Comment 70•18 days ago
|
||
We already cherry-picked this when we vendored 38ddea5ee3.
Upstream commit: https://webrtc.googlesource.com/src/+/a08189b948fb1e820e8bfa772bf70d8c598b87e5
Revert "Add InsertPacket method that takes RtpPacketInfo."
This reverts commit 38ddea5ee3320bf3441aeb3654e099b3695c9789.
Reason for revert: not backwards compatible
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: Ie7cf397cfbe5dedca009f16e5e9e3af40adbe99b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369200
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43450}
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Comment 71•18 days ago
|
||
We already cherry-picked this when we vendored 711ecf44bb.
Upstream commit: https://webrtc.googlesource.com/src/+/b7cb8fe75a0b0f19a003e4f78f83fb7bd8fa84ef
h264: skip empty NAL units, do not reject them
BUG=webrtc:380291923
Change-Id: If05268bde2ac0c600dcef479c88ca54dce708dcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368893
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43451}
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Comment 72•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/d171832b6cd2fef6daf7442f1ccf029bdda3453e
Set default simulcast temporal layer to 1 if not configured.
For H.265 when scalability mode is not configured for simulcast layers,
the default mode of L1T1 should be assumed instead of L1T3, as that is
the most commonly supported temporal scalability on all devices for
H.265.
Bug: chromium:41480904
Change-Id: Ia9bc91729eb393850dfe5e8fb04280b4f784560d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369080
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43452}
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Comment 73•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/98b3588974aa7bb7f8ea0c2aae7e41f93a5d3b31
Make CreateSendChannel and CreateReceiveChannel methods pure virtual
These methods previously had a default implementation that triggered
a crash. All implementations must now return a valid object, which
simplifies the code that calls them.
Bug: webrtc:13931
Change-Id: I877fbc929b58c6b83767c6ac5a81c8aa942e3fef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369021
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43453}
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Comment 74•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/ff88950833a3d2abbf0db292119fe8bf381d0b94
Reland "Add InsertPacket method that takes RtpPacketInfo."
This is a reland of commit 38ddea5ee3320bf3441aeb3654e099b3695c9789
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: I97d1d3d390e6d3de8bf9355b895ec336339d079f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369260
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43454}
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Comment 75•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/0a69daf38bdc706145b0e502c2a341c6fe083b13
Add counter of ECN marking to EmulatedNetwork stats
Bug: webrtc:42225697
Change-Id: I99c68afafe20fcdbc785d489a8b484cec3b3987d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368941
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43455}
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Comment 76•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/d257b4a054024b434ea98e074cc3a3fa1c10251e
Minor ClearChannel() and SetChannel() simplifications
Bug: none
Change-Id: I3ee302429b1412143fecf3036766c89a5226f8e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324302
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43456}
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Comment 77•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/05cf9c72358addf88f9ff2b90801bf70290fb62c
Clean up temp files in WavWriterTest.LargeFile
Bug: webrtc:379973428
Change-Id: Ide7d8b3d348a25270d8c99a602bec475fcafddc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368861
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Ă…hgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43457}
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Comment 78•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/e9193d7031cc1735bd833ffe85a068dd706ca330
Add histograms for Abs-Capture-Timestamp
Bug: webrtc:380712819
Change-Id: I5f56caffe33a257432551321f7c097c852b134dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368903
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43458}
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Comment 79•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/72b5769bb8a87e098ed9956b5f0e34faa98adf6f
Test both WriteSamples overloads in WavWriterTest.LargeFile
Bug: webrtc:379973428
Change-Id: Id856e76dc521027bfd59521e20e23523526678eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368900
Reviewed-by: Per Ă…hgren <peah@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43459}
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Comment 80•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/e0a524b5e02f293a0c13fd687334f7e6eedb971d
Add default constructor to relative units types
0 is natural default value for types that can be accumulated
Having default constructor simplify usage of these types in templated code.
Bug: None
Change-Id: If005c69018a2a11011bc789502fdbc600cad3278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43460}
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Comment 81•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/f4ee1a1ef30ebbcf77e606294d0d693c9db5bf66
Make PercentileFilter usable with DataRate and other types
Return default value T() if no values have been added to the filter.
Together with
https://webrtc-review.googlesource.com/c/src/+/369440, DataRate etc can be used by the filter.
Bug: None
Change-Id: I3d0e1a3e698a91a6197bf434ace2ff8246dc393e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43461}
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Comment 82•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/4060745995b8d054584969c7f55a6ef57da0a54d
spanify SSLStreamAdapter::SetPeerCertificateDigest
BUG=webrtc:357776213
Change-Id: Ie6189ac21b9f76f7ce5ddb3e4208c08793df73ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43462}
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Comment 83•18 days ago
|
||
We cherry-picked this in bug 1210175
Upstream commit: https://webrtc.googlesource.com/src/+/4a6a7465d03f0b581122f11385aca87294eebd5d
In ParseNonParameterSetNalu check BitstreamReader::Ok before returning early
~BitstreamReader() DCHECKs that the last read has been verified, so all
paths where we may leave the slice_reader instance's scope early must be
guarded by an Ok().
Bug: None
Change-Id: Ic67f87c04d1f042392c1dd6a066fdccf26e19003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43463}
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Comment 84•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1bda6a6a58abd4e98d91f714811a490bc2b03e66
Make SSLStreamAdapter::SetPeerCertificateDigest use of const uint8_t
This allows it to accept rtc::CopyOnWriteBuffer.
Bug: webrtc:357776213
Change-Id: I8c9eeb5577e8de902db144aff5ad8eee87e5a530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369640
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43464}
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Comment 85•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/56c5507ae3dea512e9c2cf740a26c6eea27999f9
Fix delta computation in abs-capture statistics
Previous computation assumed that local clock is UTC. It isn't.
Adding integration test for abs-capture stats.
Bug: webrtc:380712819
Change-Id: I054d61984cbd017b7ad04ab13e5a687eab89db69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43465}
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Comment 86•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/caa3eff65fd629cd3c9c2f666c9084ee08ba8a69
Update WebRTC code version (2024-11-28T04:11:10).
Bug: None
Change-Id: I0110c5a07e31d51fa69c6d4a871103da7cbdebc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369605
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@{#43466}
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Comment 87•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c75fbe24e67baca458c283b5985eeb3bb8b7344b
Clean up legacy variant of DTLS-SRTP key exporter
BUG=webrtc:357776213
Change-Id: Id383c3a2a8627e3d0aceb80da30db14ea689ac93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368181
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43467}
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Comment 88•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b4d09df6c20da1cb9ad7cdccf7299daddd60e6a0
Fix bug that can cause invalid reset of corruption detection state.
`VideoStreamEncoder` should not recreate the
`FrameInstrumentationGenerator` instace unless the encoder is actually
released. Otherwise it will restart and expect a keyframe the encoder
will likely not produce for a while.
Bug: webrtc:358039777
Change-Id: I111149d5e9b632df9eeb88bbbe8a07969c3e3f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369740
Auto-Submit: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43468}
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Comment 89•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/58562a8229948d7f76b1eb77fe1da1edac60f133
dcsctp: Add handover state for received MIDs
The next expected MID to use (which applies to both ordered and
unordered streams, in contrast to SSNs) was properly handed over for
streams this socket sends on, but not for streams this socket receives
on.
Adding handover state first.
Bug: webrtc:41481008
Change-Id: Ib3941f0ee1a34c24605792d9f0b658bb6a9ade4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369821
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43469}
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Comment 90•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/a79c709ed3b1d7aab8afc6a1708ac682987defdf
dcsctp: Rename test module
A few tests in dcsctp_socket_test was named DcSctpSocketResendInitTest
instead of DcSctpSocketTest. There is no reason for that.
Bug: None
Change-Id: I845eb0ab6150c4e5d457307e12f056486f86e468
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43470}
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Comment 91•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/23438deabba66b55ef9d780d75791eca2565b5af
Allow enums that have AbslStringify to be logged as text
When an enum has AbslStringify, we log the text coming from stringifying it, not the numeric value.
Drive by changes,
1. Changed the tests to use string matchers rather than
std::string::find.
2. Fixed test includes.
3. Fix spelling.
Bug: webrtc:381502973
Change-Id: I6bab0afda1b20d72c02629e80ff2ac567650183a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369861
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43471}
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Comment 92•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/934c9834041a4b183125d23ad2027c3531ada18c
Deflake VP9_SimulcastDeactiveActiveLayer_StandardSvc
Two fixes to deflake,
1. Increase the ramp up time for all layers - short time was flaky for
720p.
2. Wait for both the scalability mode AND implementation name to update.
Sometimes the implementation name would change before the scalability
mode did due to a race, so some OutboundRtpStats would have the wrong
values.
To achieve #2 (and #1 with some debugging) a new utility
WaitForCondition was added in order to apply matchers to a condition.
This is used instead of EXPECT_WAIT_EQ and similar because it gives
clear feedback on failure.
I have made 500 runs without a further flake.
Bug: webrtc:381216372
Change-Id: I0132377774e379857664e9a0c20f432bc9dc9fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369742
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43472}
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Comment 93•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/8d085422ed9d8ce684c4dc4fc1f95edc18743356
Tolerate very large deltas in abs-capture-timestamp
Cases above 100 ms have been observed on mac; use 60 seconds as
an offset.
Bug: webrtc:380712819
Change-Id: I52a085cb196472188bb5493276a1b32524717c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369881
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43473}
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Comment 94•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/19b5b3abaf33c7132f4645503c9a10e72d518178
Update WebRTC code version (2024-11-30T04:03:38).
Bug: None
Change-Id: Idee4a24780c5042f32185a74fd2ec33950cf5509
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369847
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@{#43474}
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Comment 95•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/a62e5b80be2331f9da0e4c466ededc137bfd0c50
Update WebRTC code version (2024-12-01T04:04:25).
Bug: None
Change-Id: I5bac2d6da5e7539268c5a5805f72963d491e266e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369925
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@{#43475}
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Comment 96•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/230e5a211cca3abc2b83fc69b3c70cf8bd01f35c
Update WebRTC code version (2024-12-02T04:05:21).
Bug: None
Change-Id: I51a365eeeb3d0b4ca81a64d1609f0a48f2c02eeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369945
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@{#43476}
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Comment 97•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c596dd5eb660b3a3263d9fafb0d21c11cce08d96
Fix setCodecPreference issue with asymmetrical send/recv level
For an offer in sendrecv direction, if for example it can send H.265
level 5.2 while receiving 6.0, setCodecPreferences on offerer's transceiver will currently remove H.265 from the offer SDP, since currently we do a precise level match on send_recv_codecs with the codecs from setCodecPreferences.
Update the matching logic to ignore the level when matching.
Bug: chromium:41480904
Change-Id: Id0f89cbf117ce62249a99257dcce18b35f407cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369960
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43477}
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Comment 98•18 days ago
|
||
We already cherry-picked this when we vendored f632fe9be9.
Upstream commit: https://webrtc.googlesource.com/src/+/5fc7489aa0d99fa97a431a459613e75040c25d64
Fix corruption score not being calculated on higher spatial layers.
This is a re-upload of
https://webrtc-review.googlesource.com/c/src/+/369020
Bug: webrtc:358039777
Change-Id: I7456940965084d0ce55b29b3b9bc98162cfff948
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369862
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43478}
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Comment 99•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b0be928a50d1fa05a0a65eebbafd801abe2f4e31
Cleanup H264 packetization unit tests
improve consistency, formatting and style
BUG=None
Change-Id: Iad382d9a7194b0606c1aa9c7d264dfacf03cde1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369462
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43479}
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Comment 100•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/5a4a69f95cb9808084b43768952dd6c145e53514
doc: add example how to test deprecated functions
by temporarily disabling -Wdeprecated-declarations
BUG=None
No-Try: true
Change-Id: I79433693f12c08ed37a5e5369e6e70a3e4e482bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369500
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43480}
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Comment 101•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/10c7d7368847a2bbed1c5a22cfc6391522e2d520
Fix sign error in UMA for AbsCapture.Delta
Bug: webrtc:380712819
Change-Id: Icfb42f0455718058a54391e5a586f409cd28728d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43481}
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Comment 102•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/7401d5531cd3ae979544726e31ac91353968e86b
Update WebRTC code version (2024-12-03T04:03:43).
Bug: None
Change-Id: I284305b7bee1f0cb3a5827a0587e3e813b1cd896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370042
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@{#43482}
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Comment 103•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/00ec2afc80565f308b638b9fc89aa7d3001b8119
Deflake VideoSendStreamTest::TestNackRetransmission.
Rewrites some of the logic to better takine account RTX padding and
potential acking from transport cc. This should make it both more
robust and a bit faster.
Bug: webrtc:381216373
Change-Id: I1a395c6bd86445ba3e63d79bdc766c7ff582e2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370060
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43483}
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Comment 104•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/07d7ca03522609390b2a1babf7689ed0f48fd147
Mark sigslot version as N/A
and include explanation of source access.
Bug: chromium:362397798
Change-Id: I7af673ffe060507b0e9dea95d650ffb0a681727c
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43484}
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Comment 105•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/ad63489c5871fd6e73d2b30a4b26d624e1765dbd
Remove orphis from OWNERS files
also fix a few TODOs
Bug: None
Change-Id: I2d287ed1a58f71ef32d5dc5624879ae8c63348b5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370122
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43485}
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Comment 106•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/ae1ad04077c819ca3c3fff4a3432d1e1b7f6ff6f
Add support for receiving congestion control messages to rtcp transceiver
Congestion control feedback messages follow RFC 8888.
BUG: webrtc:42225697
Change-Id: If7e55249ac479636c0bab5cbcf96e70c1976a51d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370161
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43486}
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Comment 107•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/94f2b91f1158908e8de6a8c44fd9f7be600ea238
Fix maybe incorrect spatial id when reading corruption detection message
In addition, avoid empty conversion when no message is present.
Bug: chromium:379326016
Change-Id: I855069fa89a157ba862b5162c56858825ebc1a40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Auto-Submit: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43487}
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Comment 108•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/45f58d7fccd2660f8489923e36db1a277a683063
Fixed counting of index 'send_codec_position'
For-loop has a 'continue' statement that skips increment of the index.
Added such an increment before 'continue' for the index to keep up with
the for-loop.
I guess current implementation will bug on codecs sequence like:
'red, unknown, opus'
since the subsequent for-loop (the 'red_codec' one) will not be able to
find 'opus'.
Seems like adding second increment statement is the easiest way to fix it.
Bug: None
Change-Id: Iab9cc66cf569458af9fd9ba5b938d83186c78c73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369700
Reviewed-by: Erik SprĂĄng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43488}
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Comment 110•18 days ago
|
||
Essentially a no-op since we're going to see this change
reverted when we vendor in e046787a5a.
Upstream commit: https://webrtc.googlesource.com/src/+/e5048949b0fcc275264e24f3b2a4c658fcc84aa3
Use PayloadTypePicker for video PT assignment
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
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Comment 111•18 days ago
|
||
We already cherry-picked this when we vendored e5048949b0.
Upstream commit: https://webrtc.googlesource.com/src/+/e046787a5a80a9d292b3aec7e946644e025a2b95
Revert "Use PayloadTypePicker for video PT assignment"
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
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Comment 112•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/ab4f8e1fde2b021fa6200d3e2f7d2d27a239b713
Update WebRTC code version (2024-12-04T04:04:17).
Bug: None
Change-Id: I5d175f8f850225bd95b50b3e598c605f68425587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370300
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43491}
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Comment 113•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/6c7ac74e8b17ef8c6dd7048fc3706afdf0f84c42
Add dependancy on the java side of the generated code
The next step of the migration is to use the generated java wrappers
which requires depending on the generated java targets.
Bug: webrtc:353174456
Change-Id: I834da78f9ab6050f3be148f6557252897aa68711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369781
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mohamed Heikal <mheikal@google.com>
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43492}
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Comment 114•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/fac1bafd449b3116b304a1c73757814589428219
Make PC capability APIs pure virtual
Bug: None
Change-Id: I22fdc44d5e164cab025c9d7884881eebd5160816
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370123
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43493}
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Comment 115•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1d2f30b8b9bc5e8e942f26246bdac55ba7350683
Add utility WaitUntil for testing for an eventual condition
This replaces the WaitUntilCondition function that was used in the
peer_connection_encodings_integrationtest previously. Along with that it
adds tests and improved error message printing.
As a drive-by, matchers were added for RTCError as these are the return
type of this utility function.
Bug: webrtc:381524905
Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43494}
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Comment 116•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b0acde349cee0b05b34d5db96ad31880385d6746
dcsctp: Add handover test for interleaved streams
This test was missing, which made me believe that it wasn't supported as
the handover state only included SSN and not MID. But when adding tests,
I saw that the current implementation used the SSN field to handover the
MID information for ordered streams which is sufficient given the 32 bit
type used for that (SSNs are only 16 bits).
For unordered streams, there is no need to handover any state there are
no expected next MID for unordered streams (they can be received in any
order).
So, adding tests and removing the handover state I just added.
Bug: webrtc:41481008
Change-Id: If1799cb1def5bd9f585a87cff6d835f4a9053b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370121
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43495}
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Comment 117•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/67de21e407de59b53fa4d1c5179d35173a329cb9
Update WebRTC code version (2024-12-05T04:09:18).
Bug: None
Change-Id: Icc4ee1a17ecaa131dc509bc07f62f7e79395da45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370401
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Cr-Commit-Position: refs/heads/main@{#43496}
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Comment 118•18 days ago
|
||
We cherry-picked this in bug 1920530
Upstream commit: https://webrtc.googlesource.com/src/+/0c282c471b794933dfa7a38bc5a4ee55f70769a1
Reset encoder on simulcast layer maxFramerate changes
Without this, Firefox wasn't passing WPT
webrtc/simulcast/setParameters-maxFramerate.https.html.
The main issue is the SetRates API's RateControlParameters doesn't have
a way to model maxFramerate for simulcast layers.
A long term fix would probably be to represent maxFramerate for all
simulcast layers in RateControlParameters. This change is a short term
fix, and resets the encoder iff a simulcast layer's maxFramerate has
changed, and also differs from the maxFramerate of the codec (passed to
SetRates), which matches the layer with the highest maxFramerate.
Bug: None
Change-Id: I088dda0fe88092fe5a5cc61114e10847f072a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370124
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43497}
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Comment 119•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1cf342a321fcbd2bf293470f1b190d473393b64d
Add IceConfig getter to IceTransportInternal(Interface)
and misc cleanup
BUG=webrtc:367395350
No-Iwyu: remaining IWYU failure is deep inside gtest which is unrelated to the changes and needs to be investigated separately
Change-Id: I5c2b7a6cc6b15fc5474c55eb98635cb9145b7373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370180
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43498}
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Comment 120•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b7bc1aa180d6277f4adbd55b56a222ad1aef47b6
Make MapCodecs return error rather than empty list when failing.
The video engine MapCodecs function returned an empty list of
codecs when errors occured, which caused crashes downstream.
This created issues with diagnosing errors caused by PT redesign.
Bug: webrtc:360058654
Change-Id: I0b5bdc9f95814ac4cfb99f749075990c3077e7a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370420
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43499}
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Comment 121•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/9d0799f9432c295842a89eb20107d711970eed92
Add android_build_tools/nullaway to DEPS
Bug: None
Change-Id: Ic8a0ac65343c482ab56ad0485385aa1201cbd83b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370441
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43500}
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Comment 122•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/89b0f2ef7169bf8c401ceb6a448acf01590384e5
Introduce p2p DTLS utils
for (partially) parsing DTLS packets and extracting the msg_seqs
BUG=webrtc:367395350
Change-Id: Ieb0fc121c6dc82118ced5939c1a9ebe2d72e3cb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370181
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43501}
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Comment 123•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/cab60a0842e180b4aafe4cb280c473d58594f6dd
Provide default implementation of IceTransportInternal::config()
to be deleted when downstream consumers are upgraded
BUG=webrtc:367395350
Change-Id: I35f1fefdc6535ad443b86176ea600455c2361834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370284
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43502}
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Comment 124•18 days ago
|
||
Essentially a no-op since we're going to see this change
reverted when we vendor in 414706543d.
Upstream commit: https://webrtc.googlesource.com/src/+/03f56d75d5a4bbbc6b6fe93e119f73c69ff98267
Remove stun_prober
The STUN prober shows the old RFC 3489 way of determining the NAT type
by pinging two different servers. This is known to be faulty as pointed
out by
https://datatracker.ietf.org/doc/html/rfc5389#section-2
Chromium dependency removed in
https://chromium-review.googlesource.com/c/chromium/src/+/6036622
BUG=None
Change-Id: I2b61dfe2ff899ce71ec9d2253dc836c5908cf8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368182
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43503}
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Comment 125•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/002d00486ba26a4d3c3466fcdd9b6347eceb9185
Update WebRTC code version (2024-12-06T04:03:34).
Bug: None
Change-Id: Ia92f9b26b9ff1bf11b5b312be8a7502f41addac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370285
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43504}
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Comment 126•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/711e1a8bebf10f5560b3b374a0260f3d09ad97ea
Create a custom test launcher for android
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
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Comment 127•18 days ago
|
||
We already cherry-picked this when we vendored 03f56d75d5.
Upstream commit: https://webrtc.googlesource.com/src/+/414706543d88922b21bc4619079ff5522212ae9e
Revert "Remove stun_prober"
This reverts commit 03f56d75d5a4bbbc6b6fe93e119f73c69ff98267.
Reason for revert: Breaks downstream project.
Original change's description:
> Remove stun_prober
>
> The STUN prober shows the old RFC 3489 way of determining the NAT type
> by pinging two different servers. This is known to be faulty as pointed
> out by
> https://datatracker.ietf.org/doc/html/rfc5389#section-2
>
> Chromium dependency removed in
> https://chromium-review.googlesource.com/c/chromium/src/+/6036622
>
> BUG=None
>
> Change-Id: I2b61dfe2ff899ce71ec9d2253dc836c5908cf8c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368182
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43503}
Bug: None
Change-Id: I08d01d4c9d882aca883e1c889aed8bddbca65b91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43506}
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Comment 128•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c8b66944d427b3ae817d84710fdf455ed38a042d
Update infra/tools/luci DEPS
Bug: None
Change-Id: I66a299e07e9949d0be088c79163a6393eab237fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370560
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43507}
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||
Comment 129•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/c23fe2fbf9ebb8f9a8760be0f79e95d632361dbb
Update WebRTC code version (2024-12-07T04:03:07).
Bug: None
Change-Id: I150cfc4e8a795b454e6bc0561b4db3981d5d0d26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370663
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43508}
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Comment 130•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/1537a8d34bd677589d2e613553a049f1b7dd0515
Update WebRTC code version (2024-12-08T04:05:53).
Bug: None
Change-Id: I8bcf06b32200b080e70ecd3ebd3f4a76ff9fb037
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370670
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43509}
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Comment 131•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/2ca8f15991ba593509aae828d8ab28cb763ee1a1
Add target_environment on iOS bots.
* It is now mandatory since https://chromium-review.googlesource.com/c/chromium/src/+/6054045.
* Also apply yapf formatting on mb_config.pyl.
Change-Id: Ida7df21ca077f11751bbf04a79518cefc849db2c
Bug: webrtc:383026404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370720
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43510}
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Comment 132•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/67f9d7b4edc20235b3f77a2452864a97cad7e33d
Add first L4S test using PeerScenario framework
The purpose is to be able to add more tests that verify that BWE still work and verify ECN behaviour e2e.
Bug: webrtc:42225697
Change-Id: Ie178d29d7870bfa3211d10925d00c621617ddf48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370561
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43511}
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Comment 133•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/768f78f097126841c4fe74105b3d0aff11e6943e
Add missing include in native_test_launcher.cc
Bug: webrtc:42223878
Change-Id: Ice9f4f92e32b6f824b2ded6e84f99a414a7c80ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370760
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43512}
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Comment 134•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/4c56c9ff9a63fa756d55205ce40cb6569fb5a383
Add helper class for determining ACD filter settings.
Further, add use of it in libvpx_vp8_encoder and with tuning for keyframes and lower bound of std_dev = 1.25 to work around some edge cases. Plus some minor cleanup.
Bug: webrtc:358039777
Change-Id: I6f624a6a8c7ccfe2fe656e4c089c225296f0264f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik SprĂĄng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43513}
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Comment 136•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/a01f34cdf16fe0ac5039614d9f4a37ca9e952008
Suppress "UnusedMethod" warning on methods only used on native code.
Change-Id: Ide048fd06d20b6a7a7ef0f74db9d6d267ab61f01
Bug: webrtc:383026404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43514}
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Comment 137•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/15543544b9dbfd0f67e3ad64fc8fa68de334e61a
Test that caller adapts to link capacity using CCFB
Fix todo to ensure TransportSequence numbers are generated if CCFB according to RFC 8888 is used. Transport sequence numbers are used in BWE algorithms regardless of feedback format.
Bug: webrtc:42225697
Change-Id: I6eab95c0241d590f6e7a90d19c82d13ab8692f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43515}
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Comment 138•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/93f5f9e8677d61eff3e8669ade6224195a421ebf
Restrict max number of threads in dav1d settings
Dav1d expects Dav1dSettings::n_threads to be in range [0,256]. http://google3/third_party/dav1d/src/lib.c;l=130;rcl=674317898
Bug: none
Change-Id: I4a7dfd15be733b84809aa1fc496b261d097ca5e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370800
Reviewed-by: Ă…sa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43516}
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Comment 139•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/65fbc51b61da11173da6495c8f080e20e4dad9b9
Remove srtpfilter_unittest rule from DEPS
srtpfilter was a SDES thing which is gone.
BUG=None
Change-Id: I060582b5ba9e72d1fdad3662e2b478042f0c780c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370640
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43517}
![]() |
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Comment 140•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/b00c469cad3f8c926fcf81ded90b90b6e1e62b9c
Add build_java dependency explicitely.
This is required after https://chromium-review.googlesource.com/c/chromium/src/+/6069515.
Change-Id: I0d8439197add23cd646c094225dca07fd2c3bf73
Bug: webrtc:383026404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370803
Auto-Submit: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43518}
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Comment 141•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/8d5b344b05e2415be08d31a21c09f89c1d220741
Roll chromium_revision 030af8fbf0..edc5771bf9 (1368717:1393593)
Change log: https://chromium.googlesource.com/chromium/src/+log/030af8fbf0..edc5771bf9
Full diff: https://chromium.googlesource.com/chromium/src/+/030af8fbf0..edc5771bf9
Changed dependencies
* fuchsia_version: version:24.20241014.3.1..version:26.20241203.6.1
* reclient_version: re_client_version:0.168.0.c46e68bc-gomaip..re_client_version:0.172.0.3cf60ba5-gomaip
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/69f3676cdb..189a044d69
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/05874e6c94..351a8c7f44
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/db0eae9640..ba1ffdb859
* src/buildtools/linux64: git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a..git_revision:468c6128db7fabe32a29d4753460ef53594406fc
* src/buildtools/mac: git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a..git_revision:468c6128db7fabe32a29d4753460ef53594406fc
* src/buildtools/reclient: re_client_version:0.168.0.c46e68bc-gomaip..re_client_version:0.172.0.3cf60ba5-gomaip
* src/buildtools/win: git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a..git_revision:468c6128db7fabe32a29d4753460ef53594406fc
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/452c8ab216..7bcd9c24de
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/eac4c18f74..aab53ebd79
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9d80a193fb..832ec8a32c
* src/third_party/android_build_tools/aapt2/cipd: cfGQ9GV4juNnGZIPzTmaL3ehiZM1hs6UsB5HUA8fT6oC..JPmxYHMZl4VH2Umf2-flGSdTVHtNDdFBmXd7v1Gs7EoC
* src/third_party/android_build_tools/bundletool/cipd: sZ24OFOduSafn8fvR3ajsGS6KP_oS_Tq0Cw3SA8XiD0C..Q-7KJYcc15Cn9iVy-7bk0tPH9-oUrhqHis6HmUzck80C
* src/third_party/android_build_tools/error_prone/cipd: hUxlP8GvC1xhmZ6r9xjYau2laPlzHbs_P2emx4ZL4jgC..ClNqEKjHuNFTgplEe3HujivpADFDjRs0H0ijhyVUg2gC
* src/third_party/android_build_tools/manifest_merger/cipd: rnIeJMlGw7adxOKZofLsm7tdYaOy1nHivJn9ck7ocVkC..ltqylfWRbafoft_Pjhfb4NciQwCb3J4PAkpcTZA7lW0C
* src/third_party/android_deps/cipd/libs/com_google_android_gms_play_services_auth: version:2@20.1.0.cr1..version:2@21.1.1.cr1
* src/third_party/android_deps/cipd/libs/com_google_android_gms_play_services_auth_api_phone: version:2@18.0.1.cr1..version:2@18.0.2.cr1
* src/third_party/android_deps/cipd/libs/com_google_android_gms_play_services_auth_base: version:2@18.0.2.cr1..version:2@18.0.10.cr1
* src/third_party/android_deps/cipd/libs/com_squareup_okio_okio_jvm: version:2@3.9.0.cr1..version:2@3.9.1.cr1
* src/third_party/androidx/cipd: k1wif7sS51pJGSFGN7FAeGWDorxgPart9E1f383TQL4C..w6ABIi35Hbkb4RHJCx4sSWDpt7W9rBvJFkcPVufgZMIC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0fa9014d86..9559c4566a
* src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/6b0c5b7ee1..02dd5c3ffb
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4479191661..4899cc831c
* src/third_party/clang-format/script: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git/+log/3c0acd2d4e..37f6e68a10
* src/third_party/dav1d/libdav1d: https://chromium.googlesource.com/external/github.com/videolan/dav1d.git/+log/389450f61e..d242c47b43
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/20b9bdcace..8d20c1e0b5
* src/third_party/ffmpeg: https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/686d694450..591ae4b02e
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/f02bffad0f..0ae7e60737
* src/third_party/google_benchmark/src: https://chromium.googlesource.com/external/github.com/google/benchmark.git/+log/344117638c..761305ec3b
* src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/62df7bdbc1..35d0c36560
* src/third_party/harfbuzz-ng/src: https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git/+log/1da053e87f..1c249be96e
* src/third_party/icu: https://chromium.googlesource.com/chromium/deps/icu.git/+log/4239b1559d..ba7ed88cc5
* src/third_party/jdk/current: BXZwbslDFpYhPRuG8hBh2z7ApP36ZG-ZfkBWrkpnPl4C..U-h0kbYoIUjL__n_sfiudzFLH7-C9vKVsLSOA2XHkBEC
* src/third_party/kotlin_stdlib/cipd: XJ7_doI-Qt7GFaSQ9BNo-3qF7Gv2--9Sa8GEUdjxMTUC..O9XlYa2GU1tlWmQ40OrK2Bekxjt372NtG5pMD-QyNaYC
* src/third_party/kotlinc/current: FNZSCjJ6yKsi6oRcgQrt-lX0MDlaWoxT7gPTz0CjLhMC..HckNA1Q54JBiHrpY1gP5qblVCdeuVmGsiGCdv1XjqywC
* src/third_party/libFuzzer/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git/+log/487e793763..e31b999178
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/840f879787..ff9badefdf
* src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/6a68fd412b..619ccf0515
* src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/9a1d90c3b4..1bf8357272
* src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/000edd7914..a52756acb1
* src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/efc3baa2d1..6066b14957
* src/third_party/libunwindstack: https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git/+log/a3bb4cd02e..215bddfd8e
* src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/906334ac1d..6f0c446c7b
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/a8e59d2074..307b951229
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/98ed093183..ab4e0ad953
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/e57316a6ae..d3990a1c9d
* src/third_party/r8/cipd: -i5fwP_NzM6Ylg5AsSGEotYN7hQgV852gXCslvXIrRwC..hmRsAQA2HG0DR196Odv3OKZYk-2Ul43tCtzrCyfWpEQC
* src/third_party/r8/d8/cipd: 3KCj5eRYCvGGYs5i90pRaeihkzsqgUGc4OkICT8AOlIC..o7pbwAFk68RQBPYLlCUm0mQ6B3Bj9pykXOPvv8akR5kC
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/third_party/turbine/cipd: vSia3h9tzpwpP_goLj4HMdl7_FEB5iVCv9nU5ZXOfIMC..c4onccFJcd8rKHM_6js-X_ucjXdFqhDWK8aUX4ARSkkC
* src/third_party/zstd/src: https://chromium.googlesource.com/external/github.com/facebook/zstd.git/+log/7fb5347e88..b0a179d469
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c809c41331..ae13111c68
Added dependencies
* src/third_party/android_deps/cipd/libs/com_google_android_libraries_identity_googleid_googleid
* src/third_party/android_deps/cipd/libs/com_google_android_gms_play_services_auth_blockstore
* src/third_party/android_deps/cipd/libs/com_google_android_gms_play_services_fido
DEPS diff: https://chromium.googlesource.com/chromium/src/+/030af8fbf0..edc5771bf9/DEPS
Clang version changed llvmorg-20-init-6794-g3dbd929e:llvmorg-20-init-13894-g8cb44859
Details: https://chromium.googlesource.com/chromium/src/+/030af8fbf0..edc5771bf9/tools/clang/scripts/update.py
BUG=webrtc:383026404
Change-Id: I9570d31fc173189c3728f90e33d4ea5947615c0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370710
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43519}
![]() |
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Comment 142•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/da2a3d6c4d34c9f58082eaddad10550c1855c5a7
Roll chromium_revision edc5771bf9..91cb2eade9 (1393593:1393832)
Change log: https://chromium.googlesource.com/chromium/src/+log/edc5771bf9..91cb2eade9
Full diff: https://chromium.googlesource.com/chromium/src/+/edc5771bf9..91cb2eade9
Changed dependencies
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/189a044d69..9144618bb1
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/351a8c7f44..1140f85dd1
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/ba1ffdb859..853eb76b5f
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/7bcd9c24de..53d89b481b
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/aab53ebd79..cc75aabda5
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/832ec8a32c..7195bd6ed2
* src/third_party/android_build_tools/error_prone/cipd: ClNqEKjHuNFTgplEe3HujivpADFDjRs0H0ijhyVUg2gC..AYoMbR1EWdl0sGTYnDZkWcRZOQnKpArFh-Mq-qpoq3gC
* src/third_party/android_build_tools/manifest_merger/cipd: ltqylfWRbafoft_Pjhfb4NciQwCb3J4PAkpcTZA7lW0C..hyODeowQwjjxMHe11I3-FOtBGbTg6MQdr3Uh9ZUUHA4C
* src/third_party/androidx/cipd: w6ABIi35Hbkb4RHJCx4sSWDpt7W9rBvJFkcPVufgZMIC..htV4MvGlfVbpqGSk_BC6R-OR2p5JKa7M1TmDGgIgSioC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/9559c4566a..b6eec48a57
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4899cc831c..350ba7c712
* src/third_party/jdk/current: U-h0kbYoIUjL__n_sfiudzFLH7-C9vKVsLSOA2XHkBEC..G-WIGWdFIBTPPhv2xPK_lTfI5N7WmfCBh0oTKqX-c5sC
* src/third_party/kotlin_stdlib/cipd: O9XlYa2GU1tlWmQ40OrK2Bekxjt372NtG5pMD-QyNaYC..VqqnQS2oBxM8b3Q7CDJaGfa-QQoOPx9weNRX1avHGNoC
* src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/619ccf0515..60a3e18387
* src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/1bf8357272..61ad6ef5b6
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/ab4e0ad953..4d75039545
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/d3990a1c9d..1a0d053d2f
* src/third_party/r8/cipd: hmRsAQA2HG0DR196Odv3OKZYk-2Ul43tCtzrCyfWpEQC..BkjdEJg2icSYHSYeOMbEI4Kf_1Zn4-fkpg6vqgvFkAEC
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/ae13111c68..91ade50a97
DEPS diff: https://chromium.googlesource.com/chromium/src/+/edc5771bf9..91cb2eade9/DEPS
No update to Clang.
BUG=None
Change-Id: I2055143f98d63c49c55b7aef6ae2aed266a90d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370676
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43520}
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Comment 143•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/a2c84a713ff9bda1add0c1c1f7a03983fe110b15
Roll chromium_revision 91cb2eade9..872845b531 (1393832:1393983)
Change log: https://chromium.googlesource.com/chromium/src/+log/91cb2eade9..872845b531
Full diff: https://chromium.googlesource.com/chromium/src/+/91cb2eade9..872845b531
Changed dependencies
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/9144618bb1..27bc0e3425
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/1140f85dd1..5fe89da99b
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/53d89b481b..1396ffe390
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/cc75aabda5..837d6b96ef
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7195bd6ed2..ea01f26b2f
* src/third_party/androidx/cipd: htV4MvGlfVbpqGSk_BC6R-OR2p5JKa7M1TmDGgIgSioC..vSj92wjvvjLnN0i_PCgtEw7ddiRrnqtkDeAoAmxGjGEC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/b6eec48a57..470d9b07dd
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/8d20c1e0b5..c30219e4a7
* src/third_party/instrumented_libs: https://chromium.googlesource.com/chromium/third_party/instrumented_libraries.git/+log/bb6dbcf2df..3cc43119a2
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/4d75039545..ab2f32b660
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/1a0d053d2f..b4c1dfa247
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/91ade50a97..9cfb48ac51
DEPS diff: https://chromium.googlesource.com/chromium/src/+/91cb2eade9..872845b531/DEPS
No update to Clang.
BUG=None
Change-Id: Ib9257b97f7f4dd9bcd82c20cbe9c440da63742e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370678
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43521}
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Comment 144•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/77485c7954b6750385ae14105f5e76406c9a183d
Update WebRTC code version (2024-12-10T04:06:01).
Bug: None
Change-Id: I8f6eea2d364ee715971c1dc480d48d788dc0f176
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370716
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@{#43522}
![]() |
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Comment 145•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/33501e4286b3b771a02761acc332b8bcc09acc1e
Roll chromium_revision 872845b531..8c8eda9309 (1393983:1394101)
Change log: https://chromium.googlesource.com/chromium/src/+log/872845b531..8c8eda9309
Full diff: https://chromium.googlesource.com/chromium/src/+/872845b531..8c8eda9309
Changed dependencies
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/27bc0e3425..46020f2812
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/5fe89da99b..0ef8e99239
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/853eb76b5f..88979f2947
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/1396ffe390..4aa2709912
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/837d6b96ef..255cfb7e0e
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ea01f26b2f..0095b9ed98
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/470d9b07dd..57f525e425
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/350ba7c712..40c51c5d3e
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/c30219e4a7..25fede45b3
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/ff9badefdf..0063380307
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/ab2f32b660..de6fc6ccb3
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/b4c1dfa247..b5c4a751ca
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/9cfb48ac51..b3b8ae063f
DEPS diff: https://chromium.googlesource.com/chromium/src/+/872845b531..8c8eda9309/DEPS
No update to Clang.
BUG=None
Change-Id: I4b28806dec75df0997964df706366900c8ba95d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370717
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43523}
![]() |
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Comment 146•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/50387647ec02ff7b10263043930f1321c0e4bb1b
Update fft/README.chromium to conform to guidelines.
No-Try: true
Bug: webrtc:42234774, chromium:362397270
Change-Id: Id8d7368ded541b2bc31b53fc86158c366bf1d57d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370842
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43524}
![]() |
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Comment 147•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/dbd4bcaeb21e686d77c696ccc06db72f42ab739b
Update info for spl_sqrt_floor
There were a few typos in the README file.
Bug: chromium:362397579
Change-Id: Ib0aa84f57f3d83851f085e595ffa72a53ec8311d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43525}
![]() |
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Comment 148•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/919e3acd139aed6e88931a91d03d4ae28158fa2f
Set version to N/A in spl_sqrt_floor README file
Bug: chromium:362397579
Change-Id: Ibf15f0790995aea755c00ff4e6d0c136f2973dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370843
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43526}
![]() |
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Comment 149•18 days ago
|
||
Upstream commit: https://webrtc.googlesource.com/src/+/6d0ecac3dd29ff7937c161a92c8dee543609f213
Roll chromium_revision 8c8eda9309..0148002d14 (1394101:1394244)
Change log: https://chromium.googlesource.com/chromium/src/+log/8c8eda9309..0148002d14
Full diff: https://chromium.googlesource.com/chromium/src/+/8c8eda9309..0148002d14
Changed dependencies
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/4aa2709912..6e79596f12
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/255cfb7e0e..8acc8daf2b
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0095b9ed98..ee24687ac9
* src/third_party/androidx/cipd: vSj92wjvvjLnN0i_PCgtEw7ddiRrnqtkDeAoAmxGjGEC..SQ6WZpWJXzqBAsM3nJMdxbVaDCaTTMrUUHr0PmWLCdoC
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/25fede45b3..719139c30a
* src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/de6fc6ccb3..b4a4af5376
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/b5c4a751ca..0d9f42bd7d
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b3b8ae063f..606a6fd0be
DEPS diff: https://chromium.googlesource.com/chromium/src/+/8c8eda9309..0148002d14/DEPS
No update to Clang.
BUG=None
Change-Id: I20f6442732d97fcc06dcddabac1d79b77a903a8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370719
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43527}
![]() |
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Comment 150•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/740d72673928ecc1009dc7e14761feb68ba8e326
Move DTLS related code from p2p/base to p2p/dtls
BUG=webrtc:367395350
Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43528}
![]() |
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Comment 151•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/d004aee4a41f3fba3b29ef33be2c4f07f2ceb1e7
Allow absl map and set containers in WebRTC
Bug: None
No-Try: True
Change-Id: I84218daf71de115af74e0484bc26e798f125297a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370762
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43529}
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Comment 152•18 days ago
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Upstream commit: https://webrtc.googlesource.com/src/+/dcf0ffa639bff681fad31edb1463c38b15c50049
dtls-1.3
Add SSLProtocolVersion for TLS13 and DTLS13
Allow setting max version to 13 (for BoringSSL)
Don't change any defaults.
This is a NOP.
BUG=webrtc:383141571
Change-Id: I11303c14e8d79c09d9437d44e44003c67d2fc31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370900
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43530}
![]() |
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Comment 153•18 days ago
|
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Upstream commit: https://webrtc.googlesource.com/src/+/e528a2c5f28aba97a65212f2d891f449b97e4d05
Roll chromium_revision 0148002d14..88a4a3f183 (1394244:1394390)
Change log: https://chromium.googlesource.com/chromium/src/+log/0148002d14..88a4a3f183
Full diff: https://chromium.googlesource.com/chromium/src/+/0148002d14..88a4a3f183
Changed dependencies
* src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/88979f2947..863e50ee3c
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6e79596f12..155d98f0fa
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8acc8daf2b..0730f23eb1
* src/third_party/androidx/cipd: SQ6WZpWJXzqBAsM3nJMdxbVaDCaTTMrUUHr0PmWLCdoC..jiSw6hD706772Ci0lovUctvq0azknqPQ5bXgSdeLRLoC
* src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/0ae7e60737..59320b2d3c
* src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/60a3e18387..92b37caa42
* src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/0d9f42bd7d..fcccdd3940
* src/third_party/r8/cipd: BkjdEJg2icSYHSYeOMbEI4Kf_1Zn4-fkpg6vqgvFkAEC..jnP8YjPJaWAXthtBv-bT_5HD3c6ywpAmGL31ZURB3fgC
* src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Mac_arm64/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz,Win/rust-toolchain-a522d78598415cdd614ccc6d961160f192f64b5c-1-llvmorg-20-init-13894-g8cb44859.tar.xz
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/606a6fd0be..e2ad157859
DEPS diff: https://chromium.googlesource.com/chromium/src/+/0148002d14..88a4a3f183/DEPS
No update to Clang.
BUG=None
Change-Id: I471445c04c6e9a2dbe88535f75624793d161cacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370921
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43531}
Comment 154•18 days ago
|
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Comment 155•17 days ago
|
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bugherder |
https://hg.mozilla.org/mozilla-central/rev/3d69d4aec0ed
https://hg.mozilla.org/mozilla-central/rev/dfb0b4e59fc6
https://hg.mozilla.org/mozilla-central/rev/483e5b1c9478
https://hg.mozilla.org/mozilla-central/rev/0495f011a58e
https://hg.mozilla.org/mozilla-central/rev/9cfae4a13aca
https://hg.mozilla.org/mozilla-central/rev/fdd139fea8b1
https://hg.mozilla.org/mozilla-central/rev/b331d02393b5
https://hg.mozilla.org/mozilla-central/rev/8793540a49f4
https://hg.mozilla.org/mozilla-central/rev/ea67c8af3fc9
https://hg.mozilla.org/mozilla-central/rev/f568027f84a0
https://hg.mozilla.org/mozilla-central/rev/eb702e3d7e79
https://hg.mozilla.org/mozilla-central/rev/c5278af736c0
https://hg.mozilla.org/mozilla-central/rev/dc8744c7090d
https://hg.mozilla.org/mozilla-central/rev/0d3e07c1d177
https://hg.mozilla.org/mozilla-central/rev/9902d7f0222d
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https://hg.mozilla.org/mozilla-central/rev/407a1160901b
https://hg.mozilla.org/mozilla-central/rev/7f73ca5231ee
https://hg.mozilla.org/mozilla-central/rev/a8a84d19a316
https://hg.mozilla.org/mozilla-central/rev/197017ea1f36
https://hg.mozilla.org/mozilla-central/rev/293e9b263c95
https://hg.mozilla.org/mozilla-central/rev/9954146e3802
https://hg.mozilla.org/mozilla-central/rev/d94956840f4a
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https://hg.mozilla.org/mozilla-central/rev/d8df5e219705
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https://hg.mozilla.org/mozilla-central/rev/9e7f49ef5984
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