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chromium-webrtc-autoroll 83a5a20afe Roll chromium_revision 6afe1bbaba..d99eda17c0 (831717:831830) 
Change log: 6afe1bbaba..d99eda17c0
Full diff: 6afe1bbaba..d99eda17c0

Changed dependencies
* src/base: 84b4ee8c94..ba32a25741
* src/build: 425e151f06..3328d0bdee
* src/ios: eef3a69f73..b7dec422d2
* src/testing: 88c7098f26..08db542028
* src/third_party: 0ece487b4a..cf672b2448
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7598272651..146912bb62
* src/tools: bdd6677946..3692ffb6a3
DEPS diff: 6afe1bbaba..d99eda17c0/DEPS

No update to Clang.

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

Change-Id: Ia13808de69417efddb7a37f59200fa92586553c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195900
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@{#32720}
2020-11-30 14:33:21 +00:00
api
Remember the proxies
2020-11-27 12:51:54 +00:00
audio
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
common_audio
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
common_video
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Document purpose and wire format of extension http://www.webrtc.org/experiments/rtp-hdrext/video-layers-allocation00
2020-11-23 14:55:49 +00:00
examples
Defining API result types on VoIP API
2020-11-20 18:02:05 +00:00
logging
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
media
Adjust min bitrate for the first active stream
2020-11-26 17:36:45 +00:00
modules
Break RtpFrameReferenceFinder into descriptor specific parts.
2020-11-27 17:59:28 +00:00
p2p
Set TCP_NODELAY socket option before calling connect.
2020-11-26 01:00:22 +00:00
pc
Add create function for PeerConnection that can return an error.
2020-11-27 11:08:10 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
CallbackList: Add support for removing receivers
2020-11-26 08:41:36 +00:00
rtc_tools
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
sdk
Delete objc RTCRtpFragmentationHeader
2020-11-19 14:05:48 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
…
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
tools_webrtc
Build and run iOS tests as XCTests.
2020-11-27 15:39:58 +00:00
video
Enable initial frame drop for SVC 'singlecast'
2020-11-27 14:08:45 +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
Reland "Add protobuf-py2_py3 3.13.0 to .vpython."
2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Fix "control reaches end of non-void function" warnings
2020-10-27 10:22:23 +00:00
BUILD.gn
Rename RoboCaller to CallbackList.
2020-10-23 15:14:22 +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 6afe1bbaba..d99eda17c0 (831717:831830)
2020-11-30 14:33:21 +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 the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Add transition define RTC_OBJC_HAVE_LEGACY_RTC_RTP_FRAGMENTATION_HEADER
2020-11-17 13:51:29 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +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
Build and run iOS tests as XCTests.
2020-11-27 15:39:58 +00:00
whitespace.txt
Trigger bots for FYI waterfall.
2020-11-20 09:35:54 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%