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39372 commits 2 branches 91 tags 415 MiB
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Java 2.9%
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Philipp Hancke 1cc41ea84f Remove unused Win32Window class 
BUG=None

Change-Id: I1d6b4e64a01076166d841c7c72eb0e2b968dd812
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306441
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40140}
2023-05-25 10:48:40 +00:00
api Delete RTC[NonStandard/Restricted]StatsMember. 2023-05-25 08:39:48 +00:00
audio Pass rtcp message to RtpTransportController through newer interface 2023-05-17 17:19:23 +00:00
build_overrides Define enable_safe_libcxx in build_overrides/build.gni. 2023-05-03 08:18:25 +00:00
call Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. 2023-05-25 07:33:39 +00:00
common_audio Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
common_video webrtc_libyuv: Add support for more video types for consistency 2023-04-24 19:06:25 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs docs: explain release note process 2023-05-24 14:09:54 +00:00
examples Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Add docs about adding a new test binary. 2023-03-07 11:12:33 +00:00
infra Add video_codec_perf_tests to desktop and android perf test suites 2023-05-23 12:13:29 +00:00
logging Use DD encoder/decoder in RTC event log encoder/parser. 2023-04-24 10:35:22 +00:00
media Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. 2023-05-25 07:33:39 +00:00
modules In RtcpTransciever refactor outgoing transport interface 2023-05-24 14:14:53 +00:00
net/dcsctp Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
p2p Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
pc Delete RTC[NonStandard/Restricted]StatsMember. 2023-05-25 08:39:48 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base Remove unused Win32Window class 2023-05-25 10:48:40 +00:00
rtc_tools Format the rest 2023-05-03 12:56:39 +00:00
sdk Add EglThread class wrapping EglConnection and handler. 2023-05-23 14:02:21 +00:00
stats Delete RTC[NonStandard/Restricted]StatsMember. 2023-05-25 08:39:48 +00:00
system_wrappers Format the rest 2023-05-03 12:56:39 +00:00
test Format the rest 2023-05-03 12:56:39 +00:00
tools_webrtc Rewrite 'generate_sslroots' w/o OpenSSL. 2023-05-10 12:57:37 +00:00
video Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. 2023-05-25 07:33:39 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Set use_cxx to true. 2023-05-17 06:30:04 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Rewrite 'generate_sslroots' w/o OpenSSL. 2023-05-10 12:57:37 +00:00
AUTHORS Expose setCodecPreferences/getCapabilities for android 2023-05-15 19:24:15 +00:00
BUILD.gn Add video_codec_perf_tests to desktop and android perf test suites 2023-05-23 12:13:29 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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 8f46ad499d..be3e47cd99 (1148441:1148555) 2023-05-24 17:27:34 +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
LICENSE
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +00:00
PATENTS
PRESUBMIT.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +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 Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
webrtc_lib_link_test.cc Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
whitespace.txt Trigger bots 2023-05-16 08:24:54 +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