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webrtc-version-updater ac615b6766 Update WebRTC code version (2021-02-27T04:03:14). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If05da5731bedf3310c0748de02ade5c8597e94d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208920
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33352}
2021-02-27 06:02:56 +00:00
api
Add ability to configure sampling rate for input/output video dumps in PC level framework
2021-02-26 11:24:52 +00:00
audio
Feed the clock skew to AbsoluteCaptureTimeReceiver in audio receiver.
2021-02-24 16:02:52 +00:00
build_overrides
[build] Remove obsolete gn flag
2021-01-11 17:57:44 +00:00
call
Update WebRTC code version (2021-02-27T04:03:14).
2021-02-27 06:02:56 +00:00
common_audio
Remove from chromium build targets that are not compatible with it.
2021-02-01 13:46:19 +00:00
common_video
Delete rtc::Callback0 and friends.
2021-02-16 12:41:35 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: show how to build the fuzzers
2021-02-18 08:28:24 +00:00
examples
Update the call to RuntimeEnvironment.application
2021-02-17 12:21:17 +00:00
logging
Replace RTC_DEPRECATED with ABSL_DEPRECATED
2021-02-22 12:53:23 +00:00
media
Change the safe SCTP MTU size to 1191
2021-02-25 00:05:35 +00:00
modules
modules/desktop_capture: replace memcpy with libyuv::CopyPlane
2021-02-25 21:50:30 +00:00
p2p
Delete unused sigslot variables.
2021-02-24 06:47:39 +00:00
pc
Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
2021-02-26 10:35:22 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Use default ResolutionBitrateLimits for simulcast with one active stream if not configured
2021-02-25 12:14:45 +00:00
rtc_tools
Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
2021-02-25 10:48:55 +00:00
sdk
AndroidVideoDecoder: Ignore format updates with zero dimensions
2021-02-24 11:37:32 +00:00
stats
Remove from chromium build targets that are not compatible with it.
2021-02-01 13:46:19 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Don't use SystemTimeNanos() for current wallclock time on WINUWP
2021-02-23 14:53:22 +00:00
test
Add ability to configure sampling rate for input/output video dumps in PC level framework
2021-02-26 11:24:52 +00:00
tools_webrtc
Reland "Reland "Split peer_connection_integrationtest.cc into pieces""
2021-02-17 12:28:07 +00:00
video
Use default ResolutionBitrateLimits for simulcast with one active stream if not configured
2021-02-25 12:14:45 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +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
Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits)
2021-02-24 08:40:13 +00:00
BUILD.gn
Move SequenceChecker header to API: step 1, move header only
2021-02-08 11:49:58 +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 src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits)
2021-02-24 08:40:13 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +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
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +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 deprecation section to webrtc style guide
2021-02-22 13:34:40 +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
Add deprecation section to webrtc style guide
2021-02-22 13:34:40 +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
Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
2021-02-25 10:48:55 +00:00
whitespace.txt
Whitespace update
2021-02-14 19:14:44 +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
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Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%