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Björn Terelius 08b649b6b7 Include-what-you-use api/rtc_event_log_output* 
Bug: webrtc:42226242
Change-Id: Ibf28c25900776f1223dfe9685d2fc299d4da7269
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354680
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42491}
2024-06-16 15:13:29 +00:00
api Include-what-you-use api/rtc_event_log_output* 2024-06-16 15:13:29 +00:00
audio Include-what-you-use api/rtc_event_log/ 2024-06-16 13:53:56 +00:00
build_overrides Import config.gni from partition_alloc.gni's build_override. 2024-06-04 14:24:36 +00:00
call Update WebRTC code version (2024-06-15T04:02:13). 2024-06-15 06:05:03 +00:00
common_audio Skip tests failing with the new version of UBSan. 2024-06-07 11:14:55 +00:00
common_video Deprecate rtc::RefCountInterface 2024-06-07 09:47:26 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add Monorail -> Google Issue Tracker map. 2024-04-29 19:08:57 +00:00
examples Split "helpers" from SSL target to "crypto_random" and rename 2024-06-07 06:41:51 +00:00
experiments Remove WebRTC-AutomaticAnimationDetectionScreenshare experiment 2024-05-22 10:36:49 +00:00
g3doc Deprecate absl_deps in templates and update documentation. 2024-05-27 12:49:03 +00:00
infra Use iOS 17.5.1 for perf 2024-06-14 08:12:55 +00:00
logging Include-what-you-use logging/rtc_event_log/ 2024-05-28 14:33:25 +00:00
media Support WebRTC-DataChannelMessageInterleaving 2024-06-12 13:21:00 +00:00
modules Remove kMaxNalusPerPacket hard limit for H264 frames 2024-06-14 16:29:42 +00:00
net/dcsctp dcsctp: Pack state cookie 2024-05-31 13:20:31 +00:00
p2p Fix UBsan error in TurnPortTest.TestChannelBindGetErrorResponse. 2024-06-07 09:42:13 +00:00
pc Deprecate rtc::RefCountInterface 2024-06-07 09:47:26 +00:00
resources Ignore .binarypb files. 2023-10-30 14:56:36 +00:00
rtc_base Remove more (D)TLS1.0 legacy code 2024-06-12 19:57:31 +00:00
rtc_tools Update PushResampler to use a single buffer for source, destination. 2024-05-30 14:37:50 +00:00
sdk [Android] Add RtcError class and use it in RtpTransceiver.setCodecPreferences 2024-06-13 13:57:21 +00:00
stats Mass removal of absl_deps in all BUILD.gn files 2024-05-23 15:09:46 +00:00
system_wrappers Mass removal of absl_deps in all BUILD.gn files 2024-05-23 15:09:46 +00:00
test Remove kMaxNalusPerPacket hard limit for H264 frames 2024-06-14 16:29:42 +00:00
tools_webrtc Remove instrumented_libraries_release=focal for msan. 2024-06-10 06:23:46 +00:00
video Remove kMaxNalusPerPacket hard limit for H264 frames 2024-06-14 16:29:42 +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 Use vpython3 as the default interpreter for gn. 2024-04-09 14:14:16 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Update to vpython 3.11 and remove .vpython (v2.x) 2024-01-25 11:12:20 +00:00
AUTHORS Fix 'Screen flickering on ScreenCapturerWinDirectx' 2024-04-25 21:18:27 +00:00
BUILD.gn Reland "Add SchedulableNetworkBehavior and tests." 2024-05-29 13:17:07 +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 837c81d9f7..e80ae6ea68 (1315145:1315265) 2024-06-14 16:42:13 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py gn: Remove the need for absl_deps 2024-05-22 13:32:41 +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 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 [ssci] Added Shipped field to READMEs 2023-07-12 07:31:06 +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.gni Remove BWE logging functionality 2024-05-29 12:18:44 +00:00
webrtc_lib_link_test.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
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