admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
38,895 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 ec8b84b740 Roll chromium_revision d5af113b9f..8024fc261c (1121345:1121474) 
Change log: d5af113b9f..8024fc261c
Full diff: d5af113b9f..8024fc261c

Changed dependencies
* fuchsia_vesion: version:12.20230323.1.1..version:12.20230323.3.1
* src/base: 312d1816ca..bb32b46a85
* src/build: 3de6b706e7..97631fbec5
* src/buildtools: a23318c669..63e1771c68
* src/buildtools/third_party/libc++/trunk: e44019bfac..5622befaf8
* src/ios: cbf3a5585a..c7f98ff25b
* src/testing: 3bbd269488..a7c5a5c493
* src/third_party: b74b114656..d00a252294
* src/third_party/androidx: Bd5GuYdu-KY7DAsP-QU0XofyaFYU7ySZX8QQpca7j3sC..NiTVtksLm4vhVv9RrW0oOiw5xdUpMuEuyGmD1_YR5kkC
* src/third_party/depot_tools: 479e1e9055..c799f76f30
* src/third_party/perfetto: 134b110423..bfdb3fc5cc
* src/tools: e1a96c1a92..ff6fca47e6
DEPS diff: d5af113b9f..8024fc261c/DEPS

No update to Clang.

BUG=None

Change-Id: I712a07cc46bb18e35a12432aea1b709c77bd778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298965
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@{#39663}
2023-03-24 01:10:11 +00:00
api
Make VP9 simulcast behave like singlecast for single active layer cases.
2023-03-23 14:49:22 +00:00
audio
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
2023-03-16 06:46:19 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Update WebRTC code version (2023-03-22T04:02:52).
2023-03-22 05:36:24 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
Changed sps parser and sps parser unit test case for h264, and it is working
2023-03-14 12:15:54 +00:00
data
…
docs
doc: rename index.md to README.md
2023-03-13 13:16:22 +00:00
examples
New PeerConnectionFactory::CreateVideoTrack with refcounted source
2023-03-22 09:10:27 +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
Add webrtc_nonparallel_tests to Fuchsia bots.
2023-03-15 08:44:59 +00:00
logging
Ensure correct destruction order in RtcEventLogImplTest
2023-03-23 11:50:41 +00:00
media
Ship ability to opt-in to VP9/AV1 simulcast (re-land).
2023-03-23 17:53:05 +00:00
modules
Combine concealment decision logic in NetEq.
2023-03-23 15:02:53 +00:00
net/dcsctp
dcsctp: Make use of log_prefix consistent
2023-03-15 22:15:05 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Disable flaky PeerConnectionSimulcastWithMediaFlowTests on ASAN.
2023-03-23 18:21:42 +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
Per default endable reading incoming packet timestamp from socket
2023-03-21 14:41:37 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Allow const pointer in NativeToJavaPointer
2023-03-23 11:11:05 +00:00
stats
Remove deprecated RTCStatsReport(int64) and timestamp_us
2023-03-22 08:00:53 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
Use an audio file when AudioConfig::input_file_name is specified.
2023-03-22 17:08:26 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Make VP9 simulcast behave like singlecast for single active layer cases.
2023-03-23 14:49:22 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.gn
Increase android32_ndk_api_level to 21.
2023-03-13 12:37:57 +00:00
.mailmap
…
.style.yapf
…
.vpython
…
.vpython3
Add python grpc to .vpython3 for ios test runner
2022-09-16 12:26:48 +00:00
AUTHORS
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Always check out google_benchmark, part 5.
2023-03-15 07:52:04 +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 d5af113b9f..8024fc261c (1121345:1121474)
2023-03-24 01:10:11 +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
…
webrtc.gni
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +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%