admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
38,360 Commits 1 Branch 0 Tags
Go to file
Clone
Open with VS Code Open with VSCodium Open with Intellij IDEA
Download ZIP Download TAR.GZ Download BUNDLE
chromium-webrtc-autoroll d05daeb16d Roll chromium_revision 2762031951..8496a82e1d (1093486:1093621) 
Change log: 2762031951..8496a82e1d
Full diff: 2762031951..8496a82e1d

Changed dependencies
* src/base: 3c1a98de5d..1f7bc02188
* src/build: da3104dcca..14b0a21a0a
* src/ios: b6b3020f9f..0971740636
* src/testing: 10453cac03..baa990ec41
* src/third_party: 222fdc35b1..b085d968b0
* src/third_party/depot_tools: 7879da9e9d..4f50adb332
* src/third_party/freetype/src: 6a179ff7d5..f80be4e959
* src/tools: b5ce3027c4..9eb8161069
DEPS diff: 2762031951..8496a82e1d/DEPS

No update to Clang.

BUG=None

Change-Id: I88ed9a0545195712433d2b876cfcfb377d92f6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291066
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@{#39128}
2023-01-18 01:14:14 +00:00
api
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
audio
Avoid waking up encoder thread when audio send stream is stopped.
2023-01-17 15:52:45 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Update WebRTC code version (2023-01-17T04:11:09).
2023-01-17 05:57:42 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
data
…
docs
Update WebRTC doc related to webrtc.org accounts.
2023-01-16 09:34:28 +00:00
examples
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Update WebRTC style guide about GN.
2022-12-16 10:19:13 +00:00
infra
Remove 'iOS Debug (simulator)' from LKGR.
2023-01-13 11:02:01 +00:00
logging
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
media
Only fill send/recv stats if there are send/receive streams
2023-01-17 11:44:32 +00:00
modules
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
pc
Ensure RTCRtpSenders are always created with one encoding
2023-01-17 18:05:49 +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 time functions for Fuchsia.
2023-01-16 23:06:21 +00:00
rtc_tools
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
sdk
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
stats
Add RTCAudioPlayoutStats to GetStats().
2023-01-16 13:19:45 +00:00
system_wrappers
[Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper
2023-01-11 17:14:41 +00:00
test
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
tools_webrtc
Add "iOS Debug (simulator)" CI bot.
2023-01-13 08:16:10 +00:00
video
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.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
Remove dimension check in SimulcastUtility::ValidSimulcastParameters
2023-01-11 13:41:55 +00:00
BUILD.gn
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +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 2762031951..8496a82e1d (1093486:1093621)
2023-01-18 01:14:14 +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 infra owners file
2022-12-02 09:21:47 +00:00
OWNERS_INFRA
Add infra owners file
2022-12-02 09:21:47 +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
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%