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chromium-webrtc-autoroll 7184016a6a Roll chromium_revision 422b5c30cb..8113154ab7 (1076590:1076706) 
Change log: 422b5c30cb..8113154ab7
Full diff: 422b5c30cb..8113154ab7

Changed dependencies
* src/base: 4546310f77..b4ae821cf6
* src/buildtools/linux64: git_revision:1c4151ff5c1d6fbf7fa800b8d4bb34d3abc03a41..git_revision:70d6c60823c0233a0f35eccc25b2b640d2980bdc
* src/buildtools/mac: git_revision:1c4151ff5c1d6fbf7fa800b8d4bb34d3abc03a41..git_revision:70d6c60823c0233a0f35eccc25b2b640d2980bdc
* src/buildtools/third_party/libc++abi/trunk: 7362046cc8..9231726c61
* src/buildtools/third_party/libunwind/trunk: 5870472fdd..fd49ee4ee9
* src/buildtools/win: git_revision:1c4151ff5c1d6fbf7fa800b8d4bb34d3abc03a41..git_revision:70d6c60823c0233a0f35eccc25b2b640d2980bdc
* src/ios: df6e08138d..31d059e03a
* src/testing: 9fbafd799e..760f5a33d6
* src/third_party: 1dd824ccd1..2f339b3151
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4f0d759433..41982a4a32
* src/third_party/depot_tools: 84edf22d0e..fc04ff1628
* src/tools: e6109b2345..30a9474044
DEPS diff: 422b5c30cb..8113154ab7/DEPS

No update to Clang.

BUG=None

Change-Id: I8142a4b6cd8ddabd8eabc8bb601fc683d60bf94b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285401
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@{#38751}
2022-11-29 08:29:45 +00:00
api
Add RTPVideoHeader::GetAsMetadata().
2022-11-25 14:40:30 +00:00
audio
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +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 (2022-11-28T04:02:58).
2022-11-28 05:11:28 +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
Update webrtc build command instruction to autoninja.
2022-11-25 09:19:37 +00:00
examples
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Explicitly forbid dynamic_cast
2022-11-04 12:37:57 +00:00
infra
Enable sharding for fuchsia bots.
2022-11-28 19:39:08 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
generatekeyframe: allow simulcast_encoder_adapter to request per-layer
2022-11-25 12:12:05 +00:00
modules
WebRTC APM: Add missing channel format check
2022-11-28 18:19:06 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
Convert StreamInterfaceChannel to new API
2022-11-16 10:01:28 +00:00
pc
Unvirtualize more wasteful functions
2022-11-22 22:44:21 +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
Fix SetCurrentThreadName() for Fuchsia.
2022-11-24 09:56:39 +00:00
rtc_tools
Actually build frame_analyzer when not on Chromium.
2022-11-18 17:32:33 +00:00
sdk
Remove unused field trial string kRTCFieldTrialAudioForceNoTWCCKey
2022-11-24 15:11:55 +00:00
stats
Add power efficient stats to RTC stats
2022-11-08 08:35:47 +00:00
system_wrappers
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
test
Enable sharding for fuchsia bots.
2022-11-28 19:39:08 +00:00
tools_webrtc
Replace dash by underscore in the command line argument before absl flag parsing.
2022-11-22 11:03:33 +00:00
video
Add RTPVideoHeader::GetAsMetadata().
2022-11-25 14:40:30 +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
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
Allow maxptime equal to minptime in the SDP.
2022-11-16 10:54:03 +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 422b5c30cb..8113154ab7 (1076590:1076706)
2022-11-29 08:29:45 +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
Trigger bots
2022-11-17 21:29:53 +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
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