signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Java 2.8%
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Tomas Gunnarsson dfd69c2210 Move VideoTrack's content_hint property to the signaling thread. 
Bug: webrtc:13673, webrtc:13681
Change-Id: I06810338bf5e44665e4d005d35636e9a98b1bd0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251684
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36010}
2022-02-15 21:55:59 +00:00
api Calculate max/avg encode/decode latency in codec tests 2022-02-15 18:14:41 +00:00
audio Revert "Reland "Remove unused APM voice activity detection sub-module"" 2022-02-14 12:25:51 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Integrate sync decoding in video_receive_stream 2022-02-14 16:59:20 +00:00
common_audio Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely" 2022-01-27 12:55:44 +00:00
common_video Reland: Added support for H264 YUV444 (I444) decoding. 2022-02-09 11:57:55 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add contributing.md 2021-11-03 14:59:46 +00:00
examples WebRTC examples/ should not depend on base/ 2022-02-07 12:22:46 +00:00
g3doc Allow designated initializers in WebRTC 2022-02-15 13:23:12 +00:00
logging Revert "Refactor AnalyzerConfig to use Timestamps instead of microseconds." 2022-02-14 15:01:51 +00:00
media Revert "Reland "Remove unused APM voice activity detection sub-module"" 2022-02-14 12:25:51 +00:00
modules Ensure PipeWire doesn't use a Null SourceId 2022-02-15 20:03:33 +00:00
net/dcsctp Use kHigh timer precision for 'delayed-ack' timers. 2022-01-27 11:09:24 +00:00
p2p stun/turn: use hostname when reconstructing the url 2022-02-03 12:18:50 +00:00
pc Move VideoTrack's content_hint property to the signaling thread. 2022-02-15 21:55:59 +00:00
resources AEC3: Changing the default for the use_conservative_tail_frequency_response flag. 2021-12-21 17:35:26 +00:00
rtc_base Fix LOGGING_INSIDE_WEBRTC propagation in Chromium builds. 2022-02-10 15:20:16 +00:00
rtc_tools Revert "Refactor AnalyzerConfig to use Timestamps instead of microseconds." 2022-02-14 15:01:51 +00:00
sdk Remove unused audio options and corresponding media constraints 2022-02-14 10:50:20 +00:00
stats Implement missing candidate pair packets/bytes sent/received stats. 2021-09-28 23:27:05 +00:00
system_wrappers Remove RTC_DISALLOW_COPY_AND_ASSIGN more. 2022-01-20 11:00:18 +00:00
test [PCLF] Move FEC and bitrate mulitplier into per peer configs 2022-02-15 18:10:31 +00:00
tools_webrtc Encode data for compression + add initial tests 2022-02-15 14:24:11 +00:00
video [PCLF] Move FEC and bitrate mulitplier into per peer configs 2022-02-15 18:10:31 +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 Roll chromium_revision 867b87eb29..b56d8868f3 (933713:941610) 2021-11-15 17:34:08 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
.vpython3 tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
AUTHORS Fix for payload type id collision 2022-02-08 09:13:33 +00:00
BUILD.gn Add restrictive visibility to all targets in //pc 2022-02-15 14:40:52 +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 Roll chromium_revision f85538e9c5..1d65e161b8 (967938:968255) 2022-02-08 10:46:34 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +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 Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
PATENTS
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
PRESUBMIT.py Encode data for compression + add initial tests 2022-02-15 14:24:11 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +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 APM: Make echo detector an optionally compilable and injectable component 2021-12-16 17:39:11 +00:00
whitespace.txt Trigger CI bots 2021-12-16 17:45:31 +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