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chromium-webrtc-autoroll 12ce334ce3 Roll chromium_revision 9e85c2dcfd..62c473bc47 (1065838:1065959) 
Change log: 9e85c2dcfd..62c473bc47
Full diff: 9e85c2dcfd..62c473bc47

Changed dependencies
* src/build: 8907941989..5d9f3dcc6a
* src/buildtools: 7f385a95b7..83176ba1ee
* src/buildtools/third_party/libc++/trunk: b093b39e7b..f8571eaba6
* src/ios: 2827fcb271..53a6955f72
* src/testing: 2d3e655a42..4b0f3d071c
* src/third_party: 4c84f6c537..a9c0a4786c
* src/tools: ef3f0048c5..7cfe2a872a
DEPS diff: 9e85c2dcfd..62c473bc47/DEPS

No update to Clang.

BUG=None

Change-Id: I81474b7d6d72c2559a4b8ffa1320d91dbbab9c01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281248
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@{#38526}
2022-11-01 16:48:32 +00:00
api
Add IWYU pragmas for some api headers
2022-10-31 15:43:16 +00:00
audio
audio: make packets lost a signed integer
2022-11-01 11:46:49 +00:00
build_overrides
Add stub for build_overrides/partition_alloc.gni
2022-08-29 12:17:02 +00:00
call
audio: make packets lost a signed integer
2022-11-01 11:46:49 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Make header files self contained.
2022-10-08 08:38:36 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: align VLA documentation with code
2022-11-01 11:51:32 +00:00
examples
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Stop revalidating STUN packets with the wrong password
2022-10-19 09:38:09 +00:00
infra
Temporarily remove mac_rel from CQ
2022-10-31 14:51:25 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
audio: make packets lost a signed integer
2022-11-01 11:46:49 +00:00
modules
doc: align VLA documentation with code
2022-11-01 11:51:32 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
Revert "move relay server priority assignment to port_allocator"
2022-11-01 13:29:28 +00:00
pc
Revert "move relay server priority assignment to port_allocator"
2022-11-01 13:29:28 +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
Add IWYU pragmas for some api headers
2022-10-31 15:43:16 +00:00
rtc_tools
Add --extend_run_time_duration flag to video_replay.
2022-10-19 06:16:12 +00:00
sdk
Remove the possibility to disable IPv6 in Java and ObjC.
2022-10-27 19:45:58 +00:00
stats
Unship track.totalFramesDuration/sumSquaredFrameDurations.
2022-10-31 15:09:10 +00:00
system_wrappers
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
test
[PCLF] Improve error handling and test coverage for AnalyzingVideoSink
2022-10-31 12:54:17 +00:00
tools_webrtc
Update IWYU mappings with a few more lines.
2022-10-26 22:58:55 +00:00
video
Remove WebRTC-PreStreamDecoders fieldtrial
2022-11-01 16:10:08 +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
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
Fix bug where RTCTransportStats.dtlsCipher was missing when using OpenSSL
2022-10-24 20:51:33 +00:00
BUILD.gn
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +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 9e85c2dcfd..62c473bc47 (1065838:1065959)
2022-11-01 16:48:32 +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
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
Delete modules/video_processing
2022-09-30 13:50:49 +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
Triggering CI and LKGR.
2022-10-14 08:53:38 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%