admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
31,298 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 76a627bf26 Roll chromium_revision aa7a6a963a..6e37a7f772 (805300:805451) 
Change log: aa7a6a963a..6e37a7f772
Full diff: aa7a6a963a..6e37a7f772

Changed dependencies
* src/base: df45521e2e..1fc0fc9476
* src/build: b557c091de..108b5fbeab
* src/ios: e4c12e8daa..0cbe7b0775
* src/testing: bd629bc759..2728ab1bd4
* src/third_party: 3cc3fa56cd..a8f61c16d5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/dc9083e483..6acdaf066c
* src/third_party/perfetto: ca9406bac4..21205c3720
* src/tools: 48f6d0967f..9ab1339b3f
DEPS diff: aa7a6a963a..6e37a7f772/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I7b8852ea3421d6f2016d497a25c9ff344a96308c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183783
Reviewed-by: 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/master@{#32066}
2020-09-09 21:07:46 +00:00
api
Add NV12 video buffer type
2020-09-09 15:26:02 +00:00
audio
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
common_audio
Refactor cpu_features_wrapper.h functions from C to C++.
2020-09-07 08:44:24 +00:00
common_video
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Adding documentation to become a WebRTC committer.
2020-09-02 07:49:04 +00:00
examples
Switch from "rtc_ios_xctest_test" to "test".
2020-09-09 20:44:35 +00:00
logging
Ensure RtcEventLogEncoderNewFormat::EncodeRemoteEstimate handles infite
2020-08-25 09:22:49 +00:00
media
Adding field trial to control send buffer size
2020-09-09 08:24:14 +00:00
modules
Fixing a crash when capturing non-DPI aware windows at high DPIs
2020-09-09 20:55:05 +00:00
p2p
Make MessageHandler cleanup optional.
2020-09-07 12:57:15 +00:00
pc
Make MessageHandler cleanup optional.
2020-09-07 12:57:15 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Use field trial list in CpuSpeedExperiment.
2020-09-08 06:56:49 +00:00
rtc_tools
Migrate android_resources targets to not create R.java files
2020-09-01 20:29:36 +00:00
sdk
Switch from "rtc_ios_xctest_test" to "test".
2020-09-09 20:44:35 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Refactor cpu_features_wrapper.h functions from C to C++.
2020-09-07 08:44:24 +00:00
test
Use time controller based task queue factory in PC level tests.
2020-09-08 11:08:52 +00:00
tools_webrtc
Switch from "rtc_ios_xctest_test" to "test".
2020-09-09 20:44:35 +00:00
video
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add 'requests' to .vpython.
2020-09-09 14:36:03 +00:00
abseil-in-webrtc.md
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
authors: add comments about individual / org sections
2020-09-07 18:33:21 +00:00
BUILD.gn
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
CODE_OF_CONDUCT.md
…
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 aa7a6a963a..6e37a7f772 (805300:805451)
2020-09-09 21:07:46 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Fix PRESUBMIT.py warnings.
2020-09-07 08:34:44 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Switch from "rtc_ios_xctest_test" to "test".
2020-09-09 20:44:35 +00:00
whitespace.txt
Trigger CI bots.
2020-09-06 10:13:23 +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%