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Mirko Bonadei cc2294d9b7 Remove references to xstream from WebRTC codebase. 
No-Presubmit: True
Bug: chromium:1245605
Change-Id: I02e9459c0a41c95e8ae08551350d1a5f4ca6cb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231121
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34900}
2021-09-02 08:51:14 +00:00
api Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
audio Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +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 Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Replace legacy getStats with standard getStats in the Android example 2021-08-30 10:45:15 +00:00
g3doc Use absl instead of self-made function for low-level bit counting 2021-08-26 08:56:37 +00:00
logging Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
media Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
modules Change ParseUncompressedVp9Header implementation to use BitstreamReader 2021-09-01 22:52:15 +00:00
net/dcsctp dcsctp: Support unlimited max_retransmissions 2021-08-31 10:57:48 +00:00
p2p Rename VirtualSocketServer::SetDefaultRoute --> SetDefaultSourceAddress 2021-09-01 14:27:29 +00:00
pc Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Rename VirtualSocketServer::SetDefaultRoute --> SetDefaultSourceAddress 2021-09-01 14:27:29 +00:00
rtc_tools Delete legacy rtp header parser as no longer used 2021-08-09 12:14:52 +00:00
sdk Fix NPE when setting the camera2 stabilization mode 2021-08-30 12:25:15 +00:00
stats Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test Reland "Enable WebRTC-Vp9DependencyDescriptor by default" 2021-09-01 09:27:39 +00:00
tools_webrtc Remove references to xstream from WebRTC codebase. 2021-09-02 08:51:14 +00:00
video frame transformer: expose payload type 2021-08-25 08:33:20 +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 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 _hRecThread,_hPlayThread RTC_DCHECK reverse bug. 2021-08-18 11:36:46 +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 Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
DEPS Use absl instead of self-made function for low-level bit counting 2021-08-26 08:56:37 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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.gni Revert "PipeWire capturer: implement proper DMA-BUFs support" 2021-09-01 11:32:42 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots. 2021-08-23 15:29:25 +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