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C++ 89.3%
Java 2.9%
C 2.8%
Objective-C++ 1.9%
Python 1.8%
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chromium-webrtc-autoroll 01859d41bf Roll chromium_revision 07c11578f5..74acb31550 (1228659:1228767) 
Change log: 07c11578f5..74acb31550
Full diff: 07c11578f5..74acb31550

Changed dependencies
* src/base: db662aea20..341add321e
* src/build: 6355ffc8d8..8b922e8b99
* src/buildtools: 2a8d2c5eab..06a9897f24
* src/ios: e08da60964..f6e22b9c75
* src/testing: 5417139dbe..317c96423d
* src/third_party: b2e6642fb9..a676e4c7f6
* src/third_party/freetype/src: df39b017d9..63d3a37eb7
* src/third_party/kotlinc/current: y9zd2-JF5FESwNZEVJnnejmk6J97g7fjlmwxNaMuJAoC..WKNG-_aQcnsBG-F7SS-yUGLlN9roxcWYt1K_8uw27zQC
* src/third_party/libc++/src: 4b348d339b..c03569abfc
* src/third_party/perfetto: d8864cd183..eb9117dab7
* src/tools: 12cdf74665..b7510e8017
DEPS diff: 07c11578f5..74acb31550/DEPS

No update to Clang.

BUG=None

Change-Id: Ib6d252b022978daf846ef0e8e4715ac823821b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328486
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41233}
2023-11-24 14:42:29 +00:00
api VideoStreamEncoder: Clean up drop handling and update rects. 2023-11-23 17:19:33 +00:00
audio Support shortcircuiting encoded transforms 2023-11-17 13:03:27 +00:00
build_overrides Roll chromium_revision 01dc2965ca..917876224a (1209117:1211391) 2023-10-18 15:15:07 +00:00
call Update WebRTC code version (2023-11-24T04:13:54). 2023-11-24 05:43:23 +00:00
common_audio Fix pointer overflow in neon implemenation of audio filters 2023-10-13 06:41:08 +00:00
common_video Add rtp packetizer for H265 2023-11-08 15:49:37 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add note about two-byte extension to VLA docs 2023-10-11 11:20:19 +00:00
examples Use rtc::ReceivedPacket in Stun and TurnServer 2023-11-23 10:40:56 +00:00
experiments Remove SSRCs from libSRTP when removing them from the rtp_demuxer 2023-11-08 10:24:10 +00:00
g3doc Update TODO example in the style guide. 2023-11-21 23:11:09 +00:00
infra Add xcode caches to dimensions. 2023-11-17 14:36:35 +00:00
logging Try removing RTC_PUSH_IGNORING_WUNDEF() around proto includes 2023-11-01 08:21:05 +00:00
media Make Codec::Matches also consider packetization 2023-11-14 08:14:14 +00:00
modules Per default enable WebRTC-PaddingMode-RecentLargePacket 2023-11-22 17:43:43 +00:00
net/dcsctp Move StrJoin to rtc_base/strings 2023-11-15 12:10:28 +00:00
p2p Use AsyncPacketSocket::RegisterReceivedPacketCallback in StunProber 2023-11-24 09:37:27 +00:00
pc Refactor MediaSession to unify audio/video codec handling 2023-11-23 14:27:54 +00:00
resources Ignore .binarypb files. 2023-10-30 14:56:36 +00:00
rtc_base Use rtc::ReceivedPacket in Stun and TurnServer 2023-11-23 10:40:56 +00:00
rtc_tools Move some users to use webrtc::RefCountInterface 2023-11-02 14:45:57 +00:00
sdk Trace render window state using a counter. 2023-11-23 14:56:31 +00:00
stats stats: implement fecSsrc on inbound-rtp 2023-08-04 12:54:48 +00:00
system_wrappers Use //third_party/cpu_features directly 2023-06-02 07:17:36 +00:00
test fuzzer: add testonly to fuzzer tests 2023-11-24 11:33:33 +00:00
tools_webrtc Remove not-needed webrtc:: prefixes in pc/ 2023-11-13 13:23:04 +00:00
video Fix parameter types inconsistency in FrameCadanceAdapter 2023-11-24 13:40:11 +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 [Fuchsia] Remove fuchsia_target_api_level from .gn 2023-09-04 07:26:36 +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
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Update vpython3 requests 2023-06-02 07:49:24 +00:00
AUTHORS fix: Handle out-of-range device index after GetDevicesInfo 2023-09-19 12:13:39 +00:00
BUILD.gn AsyncPacketSocket::RegisterReceivedPacketCallback 2023-11-15 11:17:53 +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 07c11578f5..74acb31550 (1228659:1228767) 2023-11-24 14:42:29 +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 Add google-truth to WEBRTC_ONLY_DEPS to unblock Chromium roll. 2023-10-17 16:40:08 +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 Make AEC3 json parsing code testonly 2023-10-26 12:03:02 +00:00
webrtc_lib_link_test.cc Migrate webrtc_link_test to include EnableMedia api 2023-11-02 22:45:40 +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