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chromium-webrtc-autoroll cf4e534997 Roll chromium_revision 0026868b2d..e35e978ee1 (842481:842656) 
Change log: 0026868b2d..e35e978ee1
Full diff: 0026868b2d..e35e978ee1

Changed dependencies
* src/base: ad94a75854..e981b96f03
* src/ios: 384df7a771..9bb27183e8
* src/testing: 6e80dc5c14..3ea0ecba00
* src/third_party: a055fcd127..7a85aaaa2d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d1a3011cd9..3889691dd6
* src/third_party/perfetto: 53a231c0ae..002c03c4b6
* src/tools: 44a1ead5a4..c60a44e69e
DEPS diff: 0026868b2d..e35e978ee1/DEPS

No update to Clang.

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

Change-Id: I56496ed66b903d840b759307c813fb9e408d27d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201440
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@{#32955}
2021-01-12 21:15:13 +00:00
api
ChannelStatistics used for RTP stats in VoipStatistics.
2021-01-12 18:55:41 +00:00
audio
ChannelStatistics used for RTP stats in VoipStatistics.
2021-01-12 18:55:41 +00:00
build_overrides
[build] Remove obsolete gn flag
2021-01-11 17:57:44 +00:00
call
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
common_audio
Organize iLBC headers as per style guide
2020-12-31 20:57:18 +00:00
common_video
Delete KeepBufferRefs helpers, and use of rtc::Bind.
2021-01-12 12:35:38 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: update working-with-native-branches information
2020-12-16 09:02:46 +00:00
examples
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
logging
Move logged types for RTC event log into event headers.
2021-01-12 16:45:40 +00:00
media
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
modules
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
p2p
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
pc
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
resources
Erle metric: avoid the decreasing of the metric when there is no render activity.
2020-12-09 11:48:11 +00:00
rtc_base
Delete KeepBufferRefs helpers, and use of rtc::Bind.
2021-01-12 12:35:38 +00:00
rtc_tools
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
sdk
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Use rtc::CopyOnWriteBuffer::MutableData instead of data in fuzzers
2021-01-12 12:36:43 +00:00
tools_webrtc
Make RTC_OBJC_EXPORT respect is_component_build.
2020-12-17 15:52:46 +00:00
video
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +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
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +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
DEPS
Roll chromium_revision 0026868b2d..e35e978ee1 (842481:842656)
2021-01-12 21:15:13 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Fix Authors Check presubmit.
2020-12-15 11:59:03 +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
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +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
Get rid of recently added rtc_openssl_is_boringssl build argument.
2020-12-15 08:34:42 +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%