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chromium-webrtc-autoroll d9b9c29143 Roll chromium_revision a89ca86efe..63f3ebd5cc (1054956:1055093) 
Change log: a89ca86efe..63f3ebd5cc
Full diff: a89ca86efe..63f3ebd5cc

Changed dependencies
* src/base: 99cbc26ce2..011568bb79
* src/build: 242b1e9a8d..59d8cf7cf1
* src/ios: 97d769d08d..8fcb1bfbc0
* src/testing: 3f2c107017..bc014bd96a
* src/third_party: e9743e2c2c..fcda1c9916
* src/third_party/androidx: GH4chA5uw-SC8v-7t3AgSUWvQIztAZ5UIkiwTaMPRSEC..X1dMN9cFCDTpYog4ymN04-yRr_kxzwcv3OH20tUGfHMC
* src/third_party/depot_tools: fe24047cdc..1795358f90
* src/tools: 99f77edb6a..f6bb4365a7
* src/tools/luci-go: git_revision:45c4b0ebf2c594bbd64adf3f042f3f491c16bbdf..git_revision:4fd960efe8027d39520e9aea3bbc8d669b53218f
* src/tools/luci-go: git_revision:45c4b0ebf2c594bbd64adf3f042f3f491c16bbdf..git_revision:4fd960efe8027d39520e9aea3bbc8d669b53218f
DEPS diff: a89ca86efe..63f3ebd5cc/DEPS

No update to Clang.

BUG=None

Change-Id: I1f098904b48a45e48ca1df6db0d10e892441818f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278024
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@{#38295}
2022-10-05 02:49:47 +00:00
api
stats: add missing ice candidate stats
2022-10-04 18:02:28 +00:00
audio
More audio stack traces
2022-10-04 14:31:52 +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 (2022-10-04T04:11:45).
2022-10-04 06:36:27 +00:00
common_audio
Fix math involving enums in C++20
2022-09-27 06:55:31 +00:00
common_video
Move IncomingVideoStream closer to its only user.
2022-09-30 10:37:37 +00:00
data
…
docs
doc: backfill M90-M105 release notes
2022-08-29 21:04:32 +00:00
examples
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
g3doc
stats: add missing ice candidate stats
2022-10-04 18:02:28 +00:00
infra
Add reclient properties to perf builders
2022-09-09 09:04:35 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
Populate ScalabilityMode of subcodecs in SimulcastEncoderAdapter.
2022-10-01 17:59:54 +00:00
modules
Add utility class MaybeWorkerThread
2022-10-04 11:39:38 +00:00
net/dcsctp
Fix several UBsan issues with memcpy
2022-09-14 09:35:39 +00:00
p2p
Add test coverage for P2PTransportChannel refactor (#8/8)
2022-09-21 16:49:02 +00:00
pc
stats: add missing ice candidate stats
2022-10-04 18:02:28 +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
Implement RTC_LOG_IF and RTC_DLOG_IF
2022-09-30 12:01:28 +00:00
rtc_tools
Add --disable_decoding flag to video_replay
2022-10-04 06:52:17 +00:00
sdk
Remove the libaom av1 decoder.
2022-09-30 08:42:25 +00:00
stats
stats: add missing ice candidate stats
2022-10-04 18:02:28 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
Add flag to export in new MetricSet proto format
2022-09-30 13:14:09 +00:00
tools_webrtc
Add flags_compatibility script for fuchsia bots.
2022-10-03 14:05:29 +00:00
video
QualityScalingTest: Use EncoderStreamFactory instead of test::DefaultVideoStreamFactory
2022-10-03 08:54:53 +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
loongarch: Add loongarch architecture support.
2022-09-13 10:44:59 +00:00
BUILD.gn
Introduce MetricsExporter API with stdout implementation
2022-09-17 00:56:29 +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 a89ca86efe..63f3ebd5cc (1054956:1055093)
2022-10-05 02:49:47 +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
Add Fuchsia platform deps
2022-09-23 07:49:47 +00:00
whitespace.txt
Trigger CI.
2022-08-12 11:03:03 +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
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C 2.2%
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