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37776 commits 2 branches 91 tags 415 MiB
C++ 89.3%
Java 2.9%
C 2.8%
Objective-C++ 1.9%
Python 1.8%
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Philipp Hancke 42e5ed38a7 stats: more consistent use of has_value() for optionals 
replacing
  if (optional) { ...}
with the more explicit
  if (optional.has_value()) { ... }

No functional changes.

BUG=None

Change-Id: I005fd3df307880b07cfda0cbe435efb0e0717a88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281362
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38544}
2022-11-03 14:09:26 +00:00
api Revert "Change default NetEq sample rate to 48k." 2022-11-02 16:00:16 +00:00
audio audio: make packets lost a signed integer 2022-11-01 11:46:49 +00:00
build_overrides Add stub for build_overrides/partition_alloc.gni 2022-08-29 12:17:02 +00:00
call Update WebRTC code version (2022-11-02T04:07:27). 2022-11-02 05:20:18 +00:00
common_audio Make header files self contained. 2022-10-08 08:38:36 +00:00
common_video Make header files self contained. 2022-10-08 08:38:36 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc,ios/index.md: fix Xcode project reference 2022-11-02 12:54:11 +00:00
examples Replace Thread::Invoke with Thread::BlockingCall 2022-09-09 10:44:17 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Stop revalidating STUN packets with the wrong password 2022-10-19 09:38:09 +00:00
infra Reenable mac_rel on CQ 2022-11-02 15:00:39 +00:00
logging Add missing dependencies. 2022-10-10 15:51:33 +00:00
media audio: make packets lost a signed integer 2022-11-01 11:46:49 +00:00
modules Revert "Change default NetEq sample rate to 48k." 2022-11-02 16:00:16 +00:00
net/dcsctp Add missing dependencies. 2022-10-10 10:18:37 +00:00
p2p Reland "move relay server priority assignment to port_allocator" 2022-11-03 12:16:08 +00:00
pc stats: more consistent use of has_value() for optionals 2022-11-03 14:09:26 +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 Add IWYU pragmas for some api headers 2022-10-31 15:43:16 +00:00
rtc_tools Add --extend_run_time_duration flag to video_replay. 2022-10-19 06:16:12 +00:00
sdk Remove the possibility to disable IPv6 in Java and ObjC. 2022-10-27 19:45:58 +00:00
stats stats: implement candidate-pair lastPacket(Sent|Received)Timestamp 2022-11-02 12:16:21 +00:00
system_wrappers Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +00:00
test Remove unused MetricsLoggerAndExporter 2022-11-02 07:35:47 +00:00
tools_webrtc Update IWYU mappings with a few more lines. 2022-10-26 22:58:55 +00:00
video Remove WebRTC-PreStreamDecoders fieldtrial 2022-11-01 16:10:08 +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 Set Fuchsia Api level + update SDK version 2022-09-14 08:49:56 +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 Add python grpc to .vpython3 for ios test runner 2022-09-16 12:26:48 +00:00
AUTHORS Fix bug where RTCTransportStats.dtlsCipher was missing when using OpenSSL 2022-10-24 20:51:33 +00:00
BUILD.gn Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +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 7e68c5600d..938bf905b9 (1066800:1066903) 2022-11-03 08:22:18 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Fix add some eng prod owners to PRESUBMIT.py. 2022-03-18 13:19:07 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Delete modules/video_processing 2022-09-30 13:50:49 +00:00
webrtc.gni Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Triggering CI and LKGR. 2022-10-14 08:53:38 +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