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chromium-webrtc-autoroll 5e482c1613 Roll chromium_revision d76cada124..6aac6345f8 (1107948:1108803)

Change log: d76cada124..6aac6345f8
Full diff: d76cada124..6aac6345f8

Changed dependencies
* fuchsia_vesion: version:11.20230221.2.1..version:11.20230222.4.1
* src/base: 0fc91d5d90..106feb9958
* src/build: 07283ef796..153b10edb9
* src/buildtools: d110f6238f..c9ed0b613f
* src/buildtools/third_party/libc++/trunk: abad4bb576..09f68a400f
* src/buildtools/third_party/libc++abi/trunk: 5559ffe8ca..87d8fe050b
* src/buildtools/third_party/libunwind/trunk: aecf8d01b9..56a849f05a
* src/ios: 143d6295ff..134bf6f441
* src/testing: b93e19cb40..fcbe55967d
* src/third_party: 82835c7ebd..e4c483947e
* src/third_party/androidx: waVlDuvM1_o0siiUN39LBaTpj0pfqoBRglXQgdicJicC..OVxvEebhJr_xMeJW1cAq-rwMYIL1KI0r_mJ7Bbu1awUC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a22c2597b5..c824231b87
* src/third_party/depot_tools: 30136f0cb7..d85c5bc6ff
* src/third_party/fontconfig/src: 2ef790a0db..06929a556f
* src/third_party/freetype/src: 7f9499044e..3f01161ff2
* src/third_party/r8: PwglNZFRNPkBBXdnY9NfrZFk2ULWDTRxhV9rl2kvkpUC..If-y2Dxou76tPxzbjAxhESv2MK1f7UTlgmP5Zg1JfUQC
* src/tools: 6e72739081..0ddfcb9964
* src/tools/luci-go: git_revision:c543f47ae455dbfe7e8fed5baa61a14d9068e98c..git_revision:8a8b4f2ea65c7ff5fde8a0c522008aed78d42d9d
* src/tools/luci-go: git_revision:c543f47ae455dbfe7e8fed5baa61a14d9068e98c..git_revision:8a8b4f2ea65c7ff5fde8a0c522008aed78d42d9d
DEPS diff: d76cada124..6aac6345f8/DEPS

No update to Clang.

BUG=None

Change-Id: Iddf3057c359daf2c953dfed78c9d53c5d78671d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294661
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@{#39377}

2023-02-23 09:12:19 +00:00

api

Introduce capture_time_identifier in webrtc::EncodedImage

2023-02-22 17:08:53 +00:00

audio

Create unit test for the population of capture_start_ntp_time

2023-02-06 14:00:39 +00:00

build_overrides

Use default values provided by PartitionAlloc instead of hard-coded ones

2022-12-07 09:11:35 +00:00

call

Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."

2023-02-21 18:30:35 +00:00

common_audio

Check FMA3 support before use it in SincResampler

2023-01-31 17:28:55 +00:00

common_video

Add 444 10 bits support for H264 and VP9

2023-01-17 12:32:26 +00:00

data

…

docs

Update documentation links in docs/native-code/development/index.md

2023-02-20 16:35:23 +00:00

examples

Remove all usage of //rtc_base target

2023-01-16 14:36:06 +00:00

experiments

Add tool for generating field trial registry header

2022-10-18 07:25:43 +00:00

g3doc

Fix doc path

2023-01-31 10:14:47 +00:00

infra

Remove use of refs/heads/master mirror in WebRTC infra.

2023-02-21 09:24:41 +00:00

logging

RtcEventLogImpl: Add test cases

2023-02-03 09:55:33 +00:00

media

Add plumbing for video NACK to be coupled between channels.

2023-02-22 14:54:38 +00:00

modules

Add unittest for RtpSenderVideoFrameTransformerDelegate

2023-02-22 20:17:35 +00:00

net/dcsctp

Remove all usage of //rtc_base target

2023-01-16 14:36:06 +00:00

p2p

Add an ICE switch reason for a switch requested by an application.

2023-02-06 16:19:49 +00:00

Add plumbing for video NACK to be coupled between channels.

2023-02-22 14:54:38 +00:00

resources

Clarify and extend test support for certain sample rates in audio processing

2022-08-03 14:26:36 +00:00

rtc_base

Changed OutputToDebug to create a CFString at compile-time, rather than runtime

2023-02-19 22:41:59 +00:00

rtc_tools

Handling NetEqSetMinimumDelay events in neteq_rtpplay.

2023-02-09 09:39:29 +00:00

sdk

Handle frame_types=null in VideoEncoderWrapper::Encode()

2023-02-16 11:09:51 +00:00

stats

stats: Deprecate RTCStatsReport(int64 timestamp_us)

2023-02-22 12:32:02 +00:00

system_wrappers

Check FMA3 support before use it in SincResampler

2023-01-31 17:28:55 +00:00

test

Allow setting the network labels in NetworkQualityMetricsReporter

2023-02-21 20:01:54 +00:00

tools_webrtc

Noop change to trigger bots

2023-02-13 10:30:38 +00:00

video

Introduce capture_time_identifier in webrtc::EncodedImage

2023-02-22 17:08:53 +00:00

.clang-format

Add IncludeBlocks to clang-format.

2021-02-03 16:29:07 +00:00

.git-blame-ignore-revs

…

.gitignore

Add .cache to .gitignore.

2021-01-20 15:01:07 +00:00

.gn

Set Fuchsia Api level + update SDK version

2022-09-14 08:49:56 +00:00

.mailmap

Add .mailmap for git.

2022-02-20 14:22:13 +00:00

.style.yapf

Fix mb.py presubmit issues.

2021-12-08 08:53:00 +00:00

.vpython

Remove unused script webrtc_dashboard_upload.py

2022-03-21 12:54:42 +00:00

.vpython3

Add python grpc to .vpython3 for ios test runner

2022-09-16 12:26:48 +00:00

AUTHORS

Changed OutputToDebug to create a CFString at compile-time, rather than runtime

2023-02-19 22:41:59 +00:00

BUILD.gn

Delete unused Audio Bwe integration test.

2023-01-26 09:31:44 +00:00

CODE_OF_CONDUCT.md

Reland "Migrate WebRTC documentation to new renderer"

2023-01-31 09:30:04 +00:00

codereview.settings

…

DEPS

Roll chromium_revision d76cada124..6aac6345f8 (1107948:1108803)

2023-02-23 09:12:19 +00:00

DIR_METADATA

Move metadata in OWNERS files to DIR_METADATA files.

2021-02-08 19:09:33 +00:00

ENG_REVIEW_OWNERS

Remove phoglund from ENG_REVIEW_OWNERS

2021-10-08 08:29:42 +00:00

LICENSE

…

license_template.txt

…

native-api.md

Reland "Migrate WebRTC documentation to new renderer"

2023-01-31 09:30:04 +00:00

OWNERS

Add infra owners file

2022-12-02 09:21:47 +00:00

OWNERS_INFRA

Add infra owners file

2022-12-02 09:21:47 +00:00

PATENTS

…

presubmit_test_mocks.py

tools_webrtc dir converted to py3 + top level PRESUBMIT script

2022-02-08 14:42:26 +00:00

presubmit_test.py

tools_webrtc dir converted to py3 + top level PRESUBMIT script

2022-02-08 14:42:26 +00:00

PRESUBMIT.py

Update portaudio to the latest

2022-05-13 09:01:34 +00:00

pylintrc

tools_webrtc dir converted to py3 + top level PRESUBMIT script

2022-02-08 14:42:26 +00:00

README.chromium

…

README.md

doc: add g3doc sitemap to toplevel readme

2021-07-23 07:55:17 +00:00

WATCHLISTS

Remove xooglers from WATCHLISTS and OWNERS

2022-11-30 15:33:25 +00:00

webrtc_lib_link_test.cc

Deprecate PeerConnectionFactory::CreatePeerConnection

2021-05-10 08:47:48 +00:00

webrtc.gni

Verify field trials looked up through field_trial::FindFullName

2022-10-20 10:46:01 +00:00

whitespace.txt

Trigger bots

2022-11-17 21:29:53 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
Coding style guide
Code of conduct
Reporting bugs
Documentation

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%