admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
25,884 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 3329be4f3d Roll chromium_revision b847e52039..d366835eb8 (631155:631269) 
Change log: b847e52039..d366835eb8
Full diff: b847e52039..d366835eb8

Changed dependencies
* src/base: cd67f57674..e3ee17f24b
* src/ios: b286b6856d..c90d925e9e
* src/testing: b76f7df43a..5be6e9c219
* src/third_party: bb0d6bd290..d1c4ec38bd
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/51e7b1437e..64102c0849
* src/third_party/depot_tools: 70ce8736cb..98a7e80352
* src/tools: 6c94f2caa5..8c2cc2a121
DEPS diff: b847e52039..d366835eb8/DEPS

No update to Clang.

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

Change-Id: I79a6f3e9867713c911e056b82ca447c6807a8dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/122412
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@{#26652}
2019-02-12 17:42:32 +00:00
api
AEC3: Fix delay hysteresis validation
2019-02-12 12:05:20 +00:00
audio
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
common_audio
Stop using Googletest legacy APIs.
2019-01-31 13:23:33 +00:00
common_video
Always add/rewrite VUI and set max_num_reorder_frames to 0.
2019-02-04 09:18:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Remove unused file.
2019-02-06 10:24:07 +00:00
logging
Stop hard-coding default IDs for RTP extensions
2019-02-09 01:04:35 +00:00
media
Delete video source proxying in WebRtcVideoSendStream
2019-02-11 12:43:31 +00:00
modules
Update H264EncoderImpl to use EncodedImage::Allocate
2019-02-12 13:33:23 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
rtc_tools
Do not build rtp_generator in Chromium builds.
2019-02-04 14:48:58 +00:00
sdk
Delete unused media constraints
2019-02-08 14:45:00 +00:00
stats
Expose video freeze metrics in GetStats.
2019-02-04 09:58:08 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Makes Clock interface fully mutable.
2019-01-31 08:12:05 +00:00
test
Introduce cross traffic for emulated network layer.
2019-02-12 13:18:33 +00:00
tools_webrtc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
video
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Allow use of functions in absl/algorithms
2019-01-28 18:42:16 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Set WEBRTC_USE_H264 in common_config
2019-02-08 10:17:04 +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
common_types.h
Delete constant RTP_PAYLOAD_NAME_SIZE
2019-01-25 07:59:52 +00:00
DEPS
Roll chromium_revision b847e52039..d366835eb8 (631155:631269)
2019-02-12 17:42:32 +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
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
…
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Add PRESUBMIT to enforce usage of new Googletest APIs.
2019-02-06 11:06:02 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
…
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Override default manifest from Chromium in WebRTC.
2019-01-24 13:00:49 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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%