admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
28,830 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 2040dcf9fc Roll chromium_revision f656c810e4..d7338c33b2 (708845:708965) 
Change log: f656c810e4..d7338c33b2
Full diff: f656c810e4..d7338c33b2

Changed dependencies
* src/base: 7367c832c4..d3316195f5
* src/build: 493ce92c35..08c0f52820
* src/ios: 973179bedd..2343142a49
* src/testing: 8debb1f16e..74115ea5c2
* src/third_party: 5e39a9d8df..ee908d6807
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9988f9da40..bf69ed0d31
* src/third_party/depot_tools: ea98ebbe65..ad557c41da
* src/tools: 2651d12003..ee184ccf04
DEPS diff: f656c810e4..d7338c33b2/DEPS

No update to Clang.

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

Change-Id: I550c1ab2e287db1ec87a224c3124ab9c118e348f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158185
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@{#29598}
2019-10-24 10:38:19 +00:00
api
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
audio
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
common_audio
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
common_video
Adding missing RTC_EXPORT for component build.
2019-10-18 09:17:56 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
Fix gtk color-space conversion in peerconnection_client
2019-10-23 11:25:06 +00:00
logging
Move rtc_base/ignore_wundef.h to its own target.
2019-10-19 10:50:36 +00:00
media
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
modules
Remove trial WebRTC-Bwe-ProbeRateFallback
2019-10-24 08:30:42 +00:00
p2p
Stop pairing local relay candidates with remote mDNS candidates.
2019-10-18 23:36:17 +00:00
pc
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
resources
Delete voice_detection() pointer to submodule
2019-10-07 13:06:05 +00:00
rtc_base
Fix RTC_LOCKABLE RTC_EXPORT order for rtc::Thread.
2019-10-23 18:50:18 +00:00
rtc_tools
Move rtc_base/ignore_wundef.h to its own target.
2019-10-19 10:50:36 +00:00
sdk
Add multipleTouchEnabled for subview of RTCMTLVideoView and RTCEAGLVideoView
2019-10-23 09:53:36 +00:00
stats
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
test
change PacketBuffer to return it's result rather that use callback
2019-10-23 16:50:57 +00:00
tools_webrtc
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
video
change PacketBuffer to return it's result rather that use callback
2019-10-23 16:50:57 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +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
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Add RTCStatisticsReport.h to WebRTC.framework.
2019-10-21 11:16:10 +00:00
BUILD.gn
Revert "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
2019-10-23 16:56:30 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision f656c810e4..d7338c33b2 (708845:708965)
2019-10-24 10:38:19 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
Update template to follow chromium copyright style
2013-04-24 01:01:28 +00:00
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add #COMPONENT to WebRTC.
2019-10-08 12:20:39 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +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
Add possibility to skip check_includes presubmit check.
2019-10-22 19:35:31 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +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
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +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%