webrtc_m130

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chromium-webrtc-autoroll 89f1aa59b6 Roll chromium_revision 0755db8caa..866506397b (944150:944390)

Change log: 0755db8caa..866506397b
Full diff: 0755db8caa..866506397b

Changed dependencies
* src/base: e704213601..379395e10f
* src/build: 5867e704e3..368880db83
* src/ios: ee53859d89..f5aa54daec
* src/testing: 7597ad2c88..83d4511de4
* src/third_party: 55fa5116e5..9a1aa8fd33
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib: version:2@1.5.31.cr0..version:2@1.6.0.cr0
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib_common: version:2@1.5.31.cr0..version:2@1.6.0.cr0
* src/third_party/androidx: 5_6hGqfW9hh3pWzd73P3qsKKKFcswNcKOHp0z_Xen94C..jXiiROMn0WffhrxtV5WAwhgkPccRtz7S0sTKt5pboZsC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/69030a0cea..3a667d10e9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8a112e2bdf..7ae3bc4766
* src/third_party/depot_tools: b6c1ed40d5..a29f589a15
* src/third_party/freetype/src: 81912a1385..3cabd142ce
* src/third_party/perfetto: 715528ea10..d5cb19a57d
* src/tools: 020ad36eaf..91e4d58d2c
DEPS diff: 0755db8caa..866506397b/DEPS

No update to Clang.

BUG=None

Change-Id: I680205721eae6b0fd3de0e34d253d6bfabea76e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238700
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@{#35402}

2021-11-23 06:40:06 +00:00

api

Explicitly set encoder and decoder format in codec tests.

2021-11-22 08:18:25 +00:00

audio

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

build_overrides

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

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

call

Fix WebRTC version updater bot.

2021-11-22 08:44:15 +00:00

common_audio

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

common_video

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

data

…

docs

Add contributing.md

2021-11-03 14:59:46 +00:00

examples

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

g3doc

Add static AsString functions for PeerConnectionInterface enums

2021-11-02 12:29:50 +00:00

logging

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

media

Revert "Add dav1d decoder to WebRTC."

2021-11-19 18:47:42 +00:00

modules

Explicitly set encoder and decoder format in codec tests.

2021-11-22 08:18:25 +00:00

net/dcsctp

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

p2p

Use a FQDN hostname as an invalid hostname

2021-11-16 12:59:20 +00:00

Allow setting volume on unstarted AudioRtpReceivers

2021-11-22 18:58:56 +00:00

resources

Disable high-pass filtering of the AEC reference

2021-02-23 07:06:11 +00:00

rtc_base

Modify global variables to not call the destructor on exit.

2021-11-19 13:32:21 +00:00

rtc_tools

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

sdk

Modify global variables to not call the destructor on exit.

2021-11-19 13:32:21 +00:00

stats

Implement missing candidate pair packets/bytes sent/received stats.

2021-09-28 23:27:05 +00:00

system_wrappers

Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED

2021-11-15 21:44:59 +00:00

test

[DVQA] Remove sender from DVQA StatsKey

2021-11-22 14:42:13 +00:00

tools_webrtc

Fixed CFBundleVersion version in Python 3.

2021-11-19 13:01:25 +00:00

video

VideoStreamEncoder: move PostTasks to WebRtcVideoChannel.

2021-11-19 11:25:11 +00:00

.clang-format

Add IncludeBlocks to clang-format.

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

.git-blame-ignore-revs

Let git-hyper-blame ignore new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Add .cache to .gitignore.

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

.gn

Roll chromium_revision 867b87eb29..b56d8868f3 (933713:941610)

2021-11-15 17:34:08 +00:00

.vpython

Update links to point at main branch

2021-07-22 16:41:26 +00:00

.vpython3

Add .vpython3 file to webrtc

2021-09-15 16:56:30 +00:00

AUTHORS

Added main profile to supported H264 codecs

2021-11-10 10:56:13 +00:00

BUILD.gn

Modify global variables to not call the destructor on exit.

2021-11-19 13:32:21 +00:00

CODE_OF_CONDUCT.md

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

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

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

DEPS

Roll chromium_revision 0755db8caa..866506397b (944150:944390)

2021-11-23 06:40:06 +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

Add .vpython3 file to webrtc

2021-09-15 16:56:30 +00:00

PATENTS

…

presubmit_test_mocks.py

Add presubmit check to guard against assert() usage.

2021-07-22 17:08:26 +00:00

presubmit_test.py

Add presubmit check to guard against assert() usage.

2021-07-22 17:08:26 +00:00

PRESUBMIT.py

fix some typos

2021-08-12 18:37:10 +00:00

pylintrc

Undo enforcing of PEP-8 pylint changes for method and function names.

2020-11-10 18:26:25 +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

Fix typo in comment.

2021-10-12 15:10:50 +00:00

whitespace.txt

Test LKGR finder.

2021-09-24 20:09:34 +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%