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chromium-webrtc-autoroll 1bed90b80e Roll chromium_revision 921d7ed344..803466977a (828216:828632) 
Change log: 921d7ed344..803466977a
Full diff: 921d7ed344..803466977a

Changed dependencies
* src/base: feb49e2ae9..9508820097
* src/build: ed0f5ec48a..ec6d3bb29a
* src/ios: b42ed99c65..5606b590c7
* src/testing: d40df8f462..47e2c83db6
* src/third_party: 27d756f4b9..fddcd50323
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f92a7636da..309eb9b672
* src/third_party/depot_tools: 9abde8c3f3..19da6ff2e8
* src/third_party/libvpx/source/libvpx: 94384b5c68..b5d77a48d7
* src/third_party/r8: jIJhJ4I22Dx5V-URd7bp-x1RILv5vrLEJdjjt77MRPcC..GXNupZ4wd3_EsDohjcNOss3iuVb5gcv0rAeHUiH05JkC
* src/tools: 4f0e6cc798..a0517a4af0
DEPS diff: 921d7ed344..803466977a/DEPS

No update to Clang.

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

Change-Id: Ib63628beb18c266a766d531449ebc23f09624a1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193791
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@{#32629}
2020-11-18 09:08:02 +00:00
api
Add TURN server to Emulated Network infrastructure
2020-11-17 21:07:56 +00:00
audio
Revert "Rename FATAL() into RTC_FATAL()."
2020-11-18 07:03:54 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
common_audio
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
common_video
Revert "Rename FATAL() into RTC_FATAL()."
2020-11-18 07:03:54 +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
Remove resolution limited max bitrate for simulcast screenshare.
2020-11-17 18:04:44 +00:00
modules
Revert "Rename FATAL() into RTC_FATAL()."
2020-11-18 07:03:54 +00:00
p2p
Check for oversized TURN usernames
2020-11-02 13:46:16 +00:00
pc
guard flexfec usage with fieldtrial
2020-11-18 09:04:52 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
Revert "Rename FATAL() into RTC_FATAL()."
2020-11-18 07:03:54 +00:00
rtc_tools
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
sdk
Add support for turn logging id in ios sdk.
2020-11-17 22:34:07 +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
Revert "Rename FATAL() into RTC_FATAL()."
2020-11-18 07:03:54 +00:00
tools_webrtc
Introduce length checking of all STUN byte string attributes
2020-11-13 12:31:37 +00:00
video
Ensure EncoderSink::OnBitrateAllocationUpdated not called after Stop
2020-11-13 16:26:41 +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 921d7ed344..803466977a (828216:828632)
2020-11-18 09:08:02 +00:00
ENG_REVIEW_OWNERS
…
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
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
Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
2020-10-22 11:29:48 +00:00
whitespace.txt
Trigger tests.
2020-11-16 19:06:26 +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%