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chromium-webrtc-autoroll 90e958b4cd Roll chromium_revision cc31f06a47..6e17f9390b (827051:827170) 
Change log: cc31f06a47..6e17f9390b
Full diff: cc31f06a47..6e17f9390b

Changed dependencies
* src/base: f6c14e7c53..57c8fdb432
* src/build: fc3fd8947b..cc35ecde81
* src/ios: 0410044229..1b6f5e66d0
* src/testing: 6739932951..e299061351
* src/third_party: d5aaf207bc..6f1583e3cb
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ac95abdb4b..2a9b38f2da
* src/third_party/depot_tools: f0517095b8..e61ccc59a2
* src/tools: e13b1915fa..6709467b3e
DEPS diff: cc31f06a47..6e17f9390b/DEPS

No update to Clang.

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

Change-Id: I91640d62ac5761f0e4c96e959094f3b00ca65567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193020
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@{#32601}
2020-11-13 07:10:08 +00:00
api
Delay VoipCore initialization.
2020-11-12 18:05:19 +00:00
audio
Delay VoipCore initialization.
2020-11-12 18:05:19 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +00:00
common_audio
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
common_video
Reland "Add scaling interface to VideoFrameBuffer"
2020-10-09 08:30:50 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: port release notes
2020-11-12 19:18:59 +00:00
examples
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
logging
Fix incorrect ToUnsigned in RTC event log.
2020-11-02 16:50:36 +00:00
media
Make SEA to be codec agnostic.
2020-11-11 18:34:35 +00:00
modules
Add delay manager config options.
2020-11-11 17:30:36 +00:00
p2p
Check for oversized TURN usernames
2020-11-02 13:46:16 +00:00
pc
Eliminate sigslot from RtpTransmissionManager
2020-11-10 14:41:45 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
CallbackList: Don't allow reentrancy
2020-11-11 12:26:14 +00:00
rtc_tools
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
sdk
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +00:00
stats
Fix -Wrange-loop-analysis.
2020-11-05 15:55:33 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Add support for loopback route on emulated endpoints
2020-11-12 17:01:59 +00:00
tools_webrtc
Add possibility of upload check for pc perf tests
2020-11-11 17:51:15 +00:00
video
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +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
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 cc31f06a47..6e17f9390b (827051:827170)
2020-11-13 07:10:08 +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
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
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +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
Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
2020-10-22 11:29:48 +00:00
whitespace.txt
Trigger try and CI bots.
2020-11-12 07:31:34 +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%