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chromium-webrtc-autoroll 0bf7082260 Roll chromium_revision 0d8179a1dc..663e2bfeb3 (833508:833628) 
Change log: 0d8179a1dc..663e2bfeb3
Full diff: 0d8179a1dc..663e2bfeb3

Changed dependencies
* src/base: 19f1957572..5eca0806a0
* src/build: 0741699cec..b85da6c931
* src/ios: bbda1cbd02..0e62f4538f
* src/testing: f7e915587b..cd2895768d
* src/third_party: 6df6bdaebb..3fa3afaa6c
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5537c03942..d1cf5db415
* src/third_party/depot_tools: 1ec57124a6..968b1fe7d7
* src/third_party/perfetto: 8ddf5ae4b8..0fdb2247e8
* src/tools: 2ffec9146d..abb5818782
DEPS diff: 0d8179a1dc..663e2bfeb3/DEPS

No update to Clang.

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

Change-Id: I8d222ca12244fa14045fa3d32786ec674dc66a44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196482
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@{#32769}
2020-12-04 06:39:31 +00:00
api
Revert "Ignore frames that are comming to DVQA after Stop is called"
2020-12-02 18:42:58 +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
Use frame rate in video overhead calculation.
2020-12-01 09:42:40 +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
doc: chrome://media-internals is useful for camera issues
2020-12-03 07:15:46 +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
test: do not consider flexfec-03 a normal codec
2020-12-01 10:23:26 +00:00
modules
Add support for PipeWire 0.3
2020-12-04 05:36:18 +00:00
p2p
Change TestStunServer::Create to take a SocketServer rather than a thread as argument.
2020-12-03 08:31:41 +00:00
pc
Change TestStunServer::Create to take a SocketServer rather than a thread as argument.
2020-12-03 08:31:41 +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
Add support for toggling builtin voice processing on iOS
2020-12-02 16:07:01 +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
Set default max_threads_count to DVQA injection helper
2020-12-02 18:49:22 +00:00
tools_webrtc
Fix 'iOS API Framework Builder'.
2020-12-01 15:21:23 +00:00
video
Delete SetRtcpXrRrtrStatus, make it a construction-time setting
2020-12-03 10:01:01 +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 0d8179a1dc..663e2bfeb3 (833508:833628)
2020-12-04 06:39:31 +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
Add support for PipeWire 0.3
2020-12-04 05:36:18 +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%