webrtc_m130

Go to file

chromium-webrtc-autoroll 9927cb752f Roll chromium_revision efe4047c2d..89179a0330 (1020566:1020743)

Change log: efe4047c2d..89179a0330
Full diff: efe4047c2d..89179a0330

Changed dependencies
* src/base: 6875905167..dcc9e0f0ad
* src/build: 919e8a4290..9ea9d4931d
* src/buildtools/linux64: git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264..git_revision:03ce92df5f9875bd9929b564be4b612713569aa9
* src/buildtools/mac: git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264..git_revision:03ce92df5f9875bd9929b564be4b612713569aa9
* src/buildtools/win: git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264..git_revision:03ce92df5f9875bd9929b564be4b612713569aa9
* src/ios: cd8a70b5e0..656c312427
* src/testing: cfb7d3bf6a..61e1b984d1
* src/third_party: 9545dcb1bd..536cd52d78
* src/third_party/androidx: 4gPri5A_WLHmRIG0GHdvmd3LeWiNvBj1i5IP7kEXAgsC..LQ9vdgkVWAv1399Rmm15OrPsglaFUcAF8fkDjzH06o4C
* src/third_party/perfetto: dd682e48aa..80dd4d929a
* src/third_party/r8: iMLEt10uXASDfG2AlATR1fO8xYhBoF24nQvDDXLY6Q8C..YYmB-DSqgEMUFtrSQw6plpnZygVruQmxrc3Qqeac8ZEC
* src/tools: 53e227512d..ddde2729cd
DEPS diff: efe4047c2d..89179a0330/DEPS

No update to Clang.

BUG=None

Change-Id: I05ebcddf2acc6b8ff522638d252525d041003221
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267744
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@{#37428}

2022-07-05 08:14:34 +00:00

api

Add support for scalability modes L2T3 and S2T3

2022-07-01 08:17:04 +00:00

audio

Delete ProcessThread and related Module interface

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

build_overrides

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

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

call

Update WebRTC code version (2022-07-05T04:05:09).

2022-07-05 05:30:02 +00:00

common_audio

Adopt absl::string_view in common_audio/

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

common_video

Delete useless test fixture H264SpsParserTest

2022-07-05 08:10:43 +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

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

2022-07-02 07:41:21 +00:00

logging

Change RTCEventLogFactory to have a const Create function

2022-06-28 23:48:37 +00:00

media

When VP9 SVC is used, use SvcConfig to set max bitrate for the stream.

2022-06-29 14:23:08 +00:00

modules

Remove WebRTC-Bwe-AlrLimitedBackoff field trial.

2022-07-04 16:29:42 +00:00

net/dcsctp

dcsctp: Support lifecycle events in send queue

2022-07-04 14:06:32 +00:00

p2p

Adopt absl::string_view in p2p/

2022-06-30 13:19:18 +00:00

Rename StatsCollector to LegacyStatsCollector.

2022-07-05 07:49:43 +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

Use generate_stubs without //base dependency

2022-07-01 10:50:54 +00:00

rtc_tools

network_tester: Remove usage of rtc::Thread::socketserver() and cleanup

2022-07-03 08:44:52 +00:00

sdk

Always include the actual decoder implementation when RTCVideoDecoderAV1 is used.

2022-07-04 12:27:02 +00:00

stats

Implement Outbound/InboundRtpStreamStats.mid.

2022-06-17 08:44:09 +00:00

system_wrappers

Delete rtc_base/atomic_ops.h

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

test

Adopt absl::string_view in p2p/

2022-06-30 13:19:18 +00:00

tools_webrtc

Always include the actual decoder implementation when RTCVideoDecoderAV1 is used.

2022-07-04 12:27:02 +00:00

video

Delete rtc_base/atomic_ops.h

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

.clang-format

Add IncludeBlocks to clang-format.

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

.git-blame-ignore-revs

…

.gitignore

…

.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

GCC: Avoid symbol clash in RenderBuffer

2022-05-24 10:47:56 +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 efe4047c2d..89179a0330 (1020566:1020743)

2022-07-05 08:14:34 +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

…

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

…

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

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%