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Sergey Sukhanov 225cd47445 dcsctp: implement handover in DataTracker 
Bug: webrtc:13154
Change-Id: Ia8c41dcffd95dafd904ee630f2131b575fe833dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231955
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34996}
2021-09-15 07:30:18 +00:00
api
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +00:00
audio
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2021-09-15T04:03:10).
2021-09-15 05:54:38 +00:00
common_audio
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
common_video
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
data
…
docs
Update Mac prerequisites
2021-08-23 19:52:17 +00:00
examples
Replace AV1X with AV1
2021-09-14 08:29:02 +00:00
g3doc
Use absl instead of self-made function for low-level bit counting
2021-08-26 08:56:37 +00:00
logging
Migrate rtc event log from rtc::BitBuffer to BitstreamReader
2021-09-07 14:03:27 +00:00
media
Fix potential crash during SimulcastEncoderAdapter tear down.
2021-09-14 09:15:22 +00:00
modules
Remove support for nack threshold.
2021-09-14 16:51:17 +00:00
net/dcsctp
dcsctp: implement handover in DataTracker
2021-09-15 07:30:18 +00:00
p2p
Delete BasicPacketSocketFactory default constructor
2021-09-03 10:46:29 +00:00
pc
Replace AV1X with AV1
2021-09-14 08:29:02 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Don't invoke custom certificate verifier unless there is an error.
2021-09-15 00:04:27 +00:00
rtc_tools
Improve points visualization in metrics_plotter.
2021-09-10 10:59:58 +00:00
sdk
Replace AV1X with AV1
2021-09-14 08:29:02 +00:00
stats
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
system_wrappers
Use GTEST_SKIP() instead of early return.
2021-08-12 15:24:13 +00:00
test
Add danilchap@webrtc.org as owner of test/fuzzers/
2021-09-14 08:48:03 +00:00
tools_webrtc
Fix iOS sim bot to for the new Chromium roll
2021-09-14 16:16:33 +00:00
video
VideoSendStreamTest: Add tests for encoder reconfiguration.
2021-09-13 13:14:22 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Increase iOS deployment target from 10 to 12.
2021-07-02 17:02:27 +00:00
.vpython
Update links to point at main branch
2021-07-22 16:41:26 +00:00
AUTHORS
Fix potential crash during SimulcastEncoderAdapter tear down.
2021-09-14 09:15:22 +00:00
BUILD.gn
Allow export of Obj-C symbols without C++ ones.
2021-07-30 22:54:59 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Roll chromium_revision 58d05cfd21..48501b3f18 (917624:917742)
2021-09-02 18:38:08 +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
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Remove inactive owners.
2021-08-31 14:27:49 +00:00
PATENTS
…
presubmit_test_mocks.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
presubmit_test.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
PRESUBMIT.py
fix some typos
2021-08-12 18:37:10 +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: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Revert "Reland "PipeWire capturer: implement proper DMA-BUFs support"""
2021-09-03 11:28:26 +00:00
whitespace.txt
Trigger bots
2021-09-08 18:30:03 +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%