signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Mirko Bonadei c3cb9b7247 Add .rustfmt.toml file 
Bug: webrtc:403297821
Change-Id: Ia395519524948da6c4b83ac6f510214e75d90534
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/381240
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44127}
2025-03-14 05:46:20 -07:00
api Move rtc_certificate and rtc_certificate_generator to webrtc namespace 2025-03-13 04:50:39 -07:00
audio Move function_view.h to webrtc namespace 2025-03-12 04:33:54 -07:00
build_overrides build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
call Update WebRTC code version (2025-03-14T04:09:46). 2025-03-13 22:41:18 -07:00
common_audio Move sanitizer.h to webrtc namespace 2025-03-13 02:59:39 -07:00
common_video Move bit_buffer.h to webrtc namespace 2025-03-09 22:25:10 -07:00
data
docs Format fuzzer documentation 2025-03-06 10:53:27 -08:00
examples Move all files in p2p/test to webrtc namespace 2025-03-13 07:39:35 -07:00
experiments Add field trial that change g2g metric to use abs. capture time. 2025-03-14 05:24:05 -07:00
g3doc Update freshness for the h-cc-pairs style document 2025-03-14 03:19:54 -07:00
infra Disable 'rtc_unittests' on iOS simulator. 2025-03-10 07:03:04 -07:00
logging Add RtcEventProcessor TieBreaker for LoggedRtcpPacketSenderReport. 2025-03-13 03:17:51 -07:00
media Improve test outcomes for WebRTC-PayloadTypesInTransport 2025-03-12 07:20:15 -07:00
modules Avoids log spam in AudioDeviceGeneric 2025-03-13 06:33:27 -07:00
net/dcsctp Move socket_address.h to webrtc namespace 2025-03-13 02:19:57 -07:00
p2p dtls-in-stun: Fix late SDP answer 2025-03-14 05:11:02 -07:00
pc Run DataChannelIntegrationTests w/o media enginge (for most tests) 2025-03-14 04:35:25 -07:00
resources Increase precision of high pass filter 2025-03-12 05:10:36 -07:00
rtc_base Refresh abseil-in-webrtc rules and documentation 2025-03-13 07:46:35 -07:00
rtc_tools Move socket_address.h to webrtc namespace 2025-03-13 02:19:57 -07:00
sdk Move rtc_certificate and rtc_certificate_generator to webrtc namespace 2025-03-13 04:50:39 -07:00
stats Reland "IWYU stats related files" 2025-02-27 11:49:52 -08:00
system_wrappers Move safe_conversions.h to webrtc namespace 2025-02-25 00:45:04 -08:00
test Enable 0Hz capture for test::FrameGeneratorCapturer. 2025-03-13 07:57:14 -07:00
tools_webrtc Roll chromium_revision 36f59af467..767b207241 (1424618:1426198) 2025-03-04 05:31:53 -08:00
video Add field trial that change g2g metric to use abs. capture time. 2025-03-14 05:24:05 -07:00
.clang-format Make .clang-format ObjC respect Chromium column limit length 2025-01-07 02:05:31 -08:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add the Emacs #*# pattern to .gitignore 2025-02-24 00:51:28 -08:00
.gn Enable rust toolchain for bots that depend on chromium base/. 2024-11-11 08:06:35 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.rustfmt.toml Add .rustfmt.toml file 2025-03-14 05:46:20 -07:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Fix issue with protobuf that is blocking perf tests. 2025-03-13 01:39:25 -07:00
AUTHORS Revert "fix: h26x packet buffer video artifacts" 2025-03-13 00:43:06 -07:00
BUILD.gn Improve testing in stun_port_unittest 2025-02-25 06:54:55 -08:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
codereview.settings
DEPS Refresh abseil-in-webrtc rules and documentation 2025-03-13 07:46:35 -07:00
DIR_METADATA
ENG_REVIEW_OWNERS
LICENSE
license_template.txt
native-api.md AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add .rustfmt.toml file 2025-03-14 05:46:20 -07:00
PATENTS
presubmit_test_mocks.py
presubmit_test.py
PRESUBMIT.py Advise to use [[deprecated]], not ABSL_DEPRECATED 2024-10-08 20:59:53 +00:00
pylintrc Configure Pylint to follow PEP-8 2023-09-25 15:56:09 +00:00
pylintrc_old_style Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
README.chromium Add Security Critical field to README.chromium. 2024-08-27 07:38:26 +00:00
README.md doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c 2023-05-26 09:20:16 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc In tests replace AudioProcessingBuilder with BuiltinAudioProcessingBuilder 2024-11-01 12:38:34 +00:00
webrtc.gni OpenH264 library enabled in Android Chromium build 2025-02-20 03:51:15 -08:00
whitespace.txt Test CQ 2024-05-27 12:46:57 +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