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chromium-webrtc-autoroll 1287eadaea Roll chromium_revision 70262e3c97..31e4c0f7f9 (1066346:1066645) 
Change log: 70262e3c97..31e4c0f7f9
Full diff: 70262e3c97..31e4c0f7f9

Changed dependencies
* src/base: a0ed3aa522..25e7e9b990
* src/build: fd0a10d3ba..14b08d3de0
* src/buildtools: 83176ba1ee..33b52eafd5
* src/buildtools/third_party/libc++/trunk: f8571eaba6..fc6bbc5eb0
* src/ios: 8f59e3e013..b46b363967
* src/testing: f053a63179..4e758ee81a
* src/third_party: d068e8427d..5e2006a7ce
* src/third_party/perfetto: 0cbb37ad93..06c0150ef0
* src/third_party/r8/d8: zoicZU9zhNbJmgq7fa0FZxcPTPlmRi907Rn2MZ21rAsC..qGtBu6TtxyR5XNy4cmsslb7c946YtkZF5_QCjVP-wc8C
* src/tools: f22e2e69c9..45722ff738
DEPS diff: 70262e3c97..31e4c0f7f9/DEPS

Clang version changed llvmorg-16-init-8697-g60809cd2:llvmorg-16-init-9369-g87a20868
Details: 70262e3c97..31e4c0f7f9/tools/clang/scripts/update.py

BUG=None

Change-Id: I0fa38f432830fa2487ba6d0509dda72e4310bb46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281560
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@{#38539}
2022-11-02 21:56:55 +00:00
api
Revert "Change default NetEq sample rate to 48k."
2022-11-02 16:00:16 +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
Update WebRTC code version (2022-11-02T04:07:27).
2022-11-02 05:20:18 +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
…
docs
doc,ios/index.md: fix Xcode project reference
2022-11-02 12:54:11 +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
Reenable mac_rel on CQ
2022-11-02 15:00:39 +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
Revert "Change default NetEq sample rate to 48k."
2022-11-02 16:00:16 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
stats: implement candidate-pair lastPacket(Sent|Received)Timestamp
2022-11-02 12:16:21 +00:00
pc
stats: implement candidate-pair lastPacket(Sent|Received)Timestamp
2022-11-02 12:16:21 +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
stats: implement candidate-pair lastPacket(Sent|Received)Timestamp
2022-11-02 12:16:21 +00:00
system_wrappers
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
test
Remove unused MetricsLoggerAndExporter
2022-11-02 07:35:47 +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
…
.git-blame-ignore-revs
…
.gitignore
…
.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
…
DEPS
Roll chromium_revision 70262e3c97..31e4c0f7f9 (1066346:1066645)
2022-11-02 21:56:55 +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
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%