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Niels Möller 5b747233a3 Add method Mutex::AssertHeld
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.

Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
2021-07-27 07:46:32 +00:00
api Use backticks not vertical bars to denote variables in comments for /api 2021-07-26 18:27:34 +00:00
audio Adding packetsDiscarded to RTCReceivedRtpStreamStats. 2021-07-13 20:34:45 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-07-24T04:02:31). 2021-07-24 05:22:12 +00:00
common_audio Fix -Wunreachable-code-aggressive. 2021-06-30 11:14:37 +00:00
common_video Ignore prefix NAL unit for slice parsing. 2021-07-22 06:25:53 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update links to point at main branch 2021-07-22 16:41:26 +00:00
examples peer_connection_server: fix static analysis issues 2021-07-24 16:42:03 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
logging Update links to point at main branch 2021-07-22 16:41:26 +00:00
media Move helpers to parse base rtp packet fields to rtp_rtcp module 2021-07-19 14:27:27 +00:00
modules Add method Mutex::AssertHeld 2021-07-27 07:46:32 +00:00
net/dcsctp Update links to point at main branch 2021-07-22 16:41:26 +00:00
p2p Rename rtc_base/ssl_stream_adapter.h constants. 2021-07-26 16:33:54 +00:00
pc Rename rtc_base/ssl_stream_adapter.h constants. 2021-07-26 16:33:54 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Add method Mutex::AssertHeld 2021-07-27 07:46:32 +00:00
rtc_tools Replace assert() with RTC_DCHECK(). 2021-07-09 07:49:43 +00:00
sdk Rename rtc_base/ssl_stream_adapter.h constants. 2021-07-26 16:33:54 +00:00
stats Adding packetsDiscarded to RTCReceivedRtpStreamStats. 2021-07-13 20:34:45 +00:00
system_wrappers Replace assert() with RTC_DCHECK(). 2021-07-09 07:49:43 +00:00
test Migrate away from legacy rtp parser in test/ 2021-07-26 13:35:08 +00:00
tools_webrtc build_ios_libs.py: add support for building for Catalyst 2021-07-24 19:42:39 +00:00
video NackModule2: Rename to NackRequester. 2021-07-23 08:30:33 +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 Increase iOS deployment target from 10 to 12. 2021-07-02 17:02:27 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
AUTHORS Added PeerConnectionObserverJni::OnRemoveTrack() 2021-06-03 19:24:55 +00:00
BUILD.gn Remove duplicated or non effective build flags. 2021-07-25 11:32:09 +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 89e5d40511..03a29cf406 (904223:904346) 2021-07-22 17:25:16 +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 kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Fix PRESUBMIT.py to not run pylint on deleted files. 2021-07-23 06:45:57 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +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 Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Reland "Add WebRTC specific dcheck_always_on." 2021-07-21 13:26:14 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Revert "Reland "Trigger postsubmit tests."" 2021-06-28 19:44:42 +00:00

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