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chromium-webrtc-autoroll 983cc2d992 Roll chromium_revision 1b992fc712..c792190b02 (1026226:1026401)

Change log: 1b992fc712..c792190b02
Full diff: 1b992fc712..c792190b02

Changed dependencies
* src/base: 311317bc94..5a0537725a
* src/build: ac02acec7e..c3ca167fb9
* src/ios: daa442d05b..8fa48df2d3
* src/testing: 84daac4837..266df76833
* src/third_party: c337bbeacc..cf8e76aecf
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ab2ff5cdd4..45c6b725b5
* src/third_party/depot_tools: f3edc50445..7298625c85
* src/tools: 3d196512c1..3520d10448
DEPS diff: 1b992fc712..c792190b02/DEPS

No update to Clang.

BUG=None

Change-Id: Id560257148573ab86bdb6abd0fdd4a14ea68e2b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269022
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@{#37578}

2022-07-20 20:33:03 +00:00

api

Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

2022-07-20 09:14:03 +00:00

audio

Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

2022-07-20 09:14:03 +00:00

build_overrides

Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions

2021-05-27 16:16:01 +00:00

call

Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

2022-07-20 09:14:03 +00:00

common_audio

Adopt absl::string_view in common_audio/

2022-05-13 15:00:14 +00:00

common_video

Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable

2022-07-20 08:15:08 +00:00

data

…

docs

[ios] Remove the support for bitcode

2022-07-04 09:01:52 +00:00

examples

Delete ProcessThread and related Module interface

2022-07-04 10:20:35 +00:00

g3doc

Clarify how to reference WebRTC bugs in TODOs

2022-07-01 08:03:34 +00:00

infra

Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing."

2022-07-06 20:28:26 +00:00

logging

Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable

2022-07-20 08:15:08 +00:00

media

Adopt absl::string_view in modules/audio_coding/

2022-07-20 13:34:23 +00:00

modules

AgcManagerDirect parametrized unit tests

2022-07-20 15:44:13 +00:00

net/dcsctp

Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable

2022-07-20 08:15:08 +00:00

p2p

[TCPConnection] Check for valid port_ in OnClose and OnConnect.

2022-07-19 15:56:43 +00:00

Fix degradation_preference setting being ignored using RtpSender.SetParameters.

2022-07-20 13:48:27 +00:00

resources

AEC3: Changing the default for the use_conservative_tail_frequency_response flag.

2021-12-21 17:35:26 +00:00

rtc_base

Update TaskQueueWin implementation to absl::AnyInvocable

2022-07-20 12:49:44 +00:00

rtc_tools

Add lower/upper link capacity to the outgoing bitrate graph.

2022-07-19 13:22:32 +00:00

sdk

Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable

2022-07-20 08:15:08 +00:00

stats

Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

2022-07-20 09:14:03 +00:00

system_wrappers

Delete rtc_base/atomic_ops.h

2022-06-28 08:32:13 +00:00

test

Adopt absl::string_view in modules/audio_coding/

2022-07-20 13:34:23 +00:00

tools_webrtc

Clobber win bots

2022-07-19 11:34:02 +00:00

video

Delete tests with 4 temporal layers

2022-07-20 14:05:13 +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

Prevent jsoncpp from hiding deprecated declarations in WebRTC

2022-04-11 12:33:47 +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

Update protobuf-py2_py3 wheel.

2022-07-01 15:17:36 +00:00

AUTHORS

Add missing header to fix build error when using linux system libraries

2022-07-19 12:25:42 +00:00

BUILD.gn

SVC: Add end to end tests for VP8 and VP9

2022-06-22 11:07:01 +00:00

CODE_OF_CONDUCT.md

Reference "main" branches instead of "master" branches.

2021-07-15 11:07:44 +00:00

codereview.settings

…

DEPS

Roll chromium_revision 1b992fc712..c792190b02 (1026226:1026401)

2022-07-20 20:33:03 +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

g3doc.lua

Improve webrtc documentation infra. Preview at:

2021-03-30 10:29:30 +00:00

LICENSE

…

license_template.txt

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native-api.md

Reference "main" branches instead of "master" branches.

2021-07-15 11:07:44 +00:00

OWNERS

Fix add some eng prod owners to PRESUBMIT.py.

2022-03-18 13:19:07 +00:00

PATENTS

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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

Add CPEPrefix.

2020-07-13 11:42:07 +00:00

README.md

doc: add g3doc sitemap to toplevel readme

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

WATCHLISTS

Update WATCHLISTS

2021-08-23 13:37:55 +00:00

webrtc_lib_link_test.cc

Deprecate PeerConnectionFactory::CreatePeerConnection

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

webrtc.gni

[Cast Convergence] Replace is_chromecast with new args

2022-06-16 00:50:08 +00:00

whitespace.txt

Trigger CI bots

2021-12-16 17:45:31 +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%