admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
39,293 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 5998ddfad8 Roll chromium_revision 5cdde4d6df..bd7ba3acbb (1143390:1143522) 
Change log: 5cdde4d6df..bd7ba3acbb
Full diff: 5cdde4d6df..bd7ba3acbb

Changed dependencies
* fuchsia_vesion: version:12.20230511.4.1..version:12.20230512.2.1
* src/base: 7a31914328..6c44bd2dbc
* src/build: 9c82c2f2b3..d7b02720d9
* src/ios: e9c71bd6a5..0592df026b
* src/testing: e32ce90894..49245997df
* src/third_party: affcba5c6b..4990b5ef1a
* src/third_party/android_build_tools/bundletool: xnKkaX2u7XVfsUzExic0KW3jYMINpI16Ll9QYESBoI8C..Xt6znihJ60Ym06wn163_2B0HOt2tNsV5fVy-BpYJZcMC
* src/third_party/depot_tools: f949f4c167..a1e578320b
* src/third_party/perfetto: edd765afa0..a6abd99444
* src/tools: 3e0c860bca..944d65f545
* src/tools/luci-go: git_revision:c53344de69401cd5d608484b37cd6545f328406c..git_revision:6ac770095bc0e289138f6f29aa91ff2f4886b81c
* src/tools/luci-go: git_revision:c53344de69401cd5d608484b37cd6545f328406c..git_revision:6ac770095bc0e289138f6f29aa91ff2f4886b81c
DEPS diff: 5cdde4d6df..bd7ba3acbb/DEPS

No update to Clang.

BUG=None

Change-Id: If6c312c703a05c11adb48e0c6dc9c688ad8d157d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305141
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@{#40061}
2023-05-12 22:57:11 +00:00
api
[Stats] Align RTCStatsMember<T> closer to absl::optional<T>.
2023-05-11 07:33:54 +00:00
audio
Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT
2023-05-09 14:54:50 +00:00
build_overrides
Define enable_safe_libcxx in build_overrides/build.gni.
2023-05-03 08:18:25 +00:00
call
Update WebRTC code version (2023-05-12T04:01:51).
2023-05-12 09:16:11 +00:00
common_audio
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
common_video
webrtc_libyuv: Add support for more video types for consistency
2023-04-24 19:06:25 +00:00
data
…
docs
fix some more minor typos
2023-05-11 12:26:25 +00:00
examples
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
Increase timeout to 3h for bots that compile Chromium.
2023-05-09 09:18:40 +00:00
logging
Use DD encoder/decoder in RTC event log encoder/parser.
2023-04-24 10:35:22 +00:00
media
Initialize RTP modes from callback
2023-05-12 12:30:15 +00:00
modules
PipeWire capturer: fix fcntl call when duplicating a file descriptor
2023-05-12 20:39:07 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
pc
sdp: accept bundle-only media section without rtcp-mux
2023-05-12 13:38:23 +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
Remove WEBRTC_EXTERNAL_JSON.
2023-05-10 21:34:52 +00:00
rtc_tools
Format the rest
2023-05-03 12:56:39 +00:00
sdk
Removed unused members of UIDevice extension.
2023-05-04 14:48:05 +00:00
stats
stats: remove media_type which was an alias for kind
2023-05-09 11:46:52 +00:00
system_wrappers
Format the rest
2023-05-03 12:56:39 +00:00
test
Format the rest
2023-05-03 12:56:39 +00:00
tools_webrtc
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
video
Delete unused member in VideoSendStream
2023-05-12 11:42:00 +00:00
.clang-format
…
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
…
.gn
Increase android32_ndk_api_level to 21.
2023-03-13 12:37:57 +00:00
.mailmap
…
.style.yapf
…
.vpython
…
.vpython3
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
AUTHORS
Video: add new metric for VP9/AV1 hw encoding with softwareBRC
2023-04-20 12:54:06 +00:00
BUILD.gn
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
…
DEPS
Roll chromium_revision 5cdde4d6df..bd7ba3acbb (1143390:1143522)
2023-05-12 22:57:11 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +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
…
presubmit_test.py
…
PRESUBMIT.py
…
pylintrc
…
README.chromium
…
README.md
…
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
whitespace.txt
Trigger builds
2023-04-12 07:09:41 +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%