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Gustaf Ullberg ed51a6e665 AEC3: Avoid static initializers
Bug: webrtc:9288,chromium:846615
Change-Id: I9df7f07454bdba45181972b7ed3dff77c370abb3
Reviewed-on: https://webrtc-review.googlesource.com/81750
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23538}
2018-06-07 18:13:01 +00:00
api Cleanup webrtc:: namespace from leaked TimingFrameFlags 2018-06-05 13:52:04 +00:00
audio Delete RtpFeedback. The ssrc for a receive stream should be known at 2018-05-28 11:05:19 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Delete unused stats for preferred_bitrate. 2018-06-07 08:11:07 +00:00
common_audio Remove typedefs.h from webrtc/ root (part 1) 2018-05-23 12:07:10 +00:00
common_video Delete unused stats for preferred_bitrate. 2018-06-07 08:11:07 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Broadcast extension for AppRTCMobile on iOS 2018-06-04 08:49:21 +00:00
infra Flip luci.webrtc.try to production 2018-05-30 08:30:00 +00:00
logging Split IceCandidatePairEventType enum. 2018-05-31 08:42:10 +00:00
media Delete unused stats for preferred_bitrate. 2018-06-07 08:11:07 +00:00
modules AEC3: Avoid static initializers 2018-06-07 18:13:01 +00:00
ortc New class FakePeriodicVideoTrackSource, simplifying shutdown logic. 2018-05-21 10:27:55 +00:00
p2p Fixing bug with PseudoTcp that corrupts data if initial packet is lost. 2018-05-31 18:54:58 +00:00
pc Delete deprecated CreateAudioSource method, with constraints. 2018-06-04 08:19:30 +00:00
resources AGC2 RNN VAD: Polishing. 2018-05-15 16:41:02 +00:00
rtc_base Re-enabling SanitizerTest.MsanUninitialized. 2018-06-07 11:39:15 +00:00
rtc_tools Split LoggedBweProbeResult into -Success and -Failure. 2018-05-29 13:41:04 +00:00
sdk Android: Throw exception in CallSessionFileRotatingLogSink if dir is null 2018-06-07 13:16:40 +00:00
stats PeerConnectionInterface::GetStats() with selector argument added. 2018-03-26 12:08:20 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Clean up posix-only files. 2018-04-18 00:18:04 +00:00
test Removing warning suppression flags from test/. 2018-06-07 11:54:56 +00:00
third_party Roll chromium_revision f2d1e453de..cf1645bec7 (563863:563963) 2018-06-03 09:05:38 +00:00
tools_webrtc Remove MIPS MB config since we don't need it anymore. 2018-06-05 12:35:20 +00:00
video Delete unused stats for preferred_bitrate. 2018-06-07 08:11:07 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Remove third_party from DEPS file to prepare to check it into webrtc. 2018-05-11 09:30:12 +00:00
.gn Opt out of "Migrate the Android Support Lib to android_deps". 2018-04-05 13:40:53 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
AUTHORS ObjC: Notify local video track 2018-05-30 22:36:14 +00:00
BUILD.gn Rely on use_fuzzing_engine && optimize_for_fuzzing to define WEBRTC_UNSAFE_FUZZER_MODE. 2018-05-30 04:28:28 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Delete RTP-specific values from the VideoCodecType enum. 2018-06-07 07:49:27 +00:00
common_types.h Delete RTP-specific values from the VideoCodecType enum. 2018-06-07 07:49:27 +00:00
DEPS Roll chromium_revision cf1645bec7..42930fc83a (563963:564069) 2018-06-04 13:06:07 +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
LICENSE_THIRD_PARTY Remove custom MD5 / SHA-1 implementations. 2018-02-19 15:03:35 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add phoglund as root owner. 2018-05-18 15:57:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Add presubmit error if third_party/.git exists. 2018-05-23 09:17:30 +00:00
presubmit_test.py Roll chromium_revision 95336cb92b..191d55580e (557816:557824) 2018-05-11 11:17:05 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Add style guidance about forward declarations. 2018-03-28 20:58:27 +00:00
THIRD_PARTY_CHROMIUM_DEPS.json Change structure of deps file and tool for adding chromium dep. 2018-05-17 08:52:31 +00:00
THIRD_PARTY_WEBRTC_DEPS.json Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
typedefs.h Remove typedefs.h from webrtc/ root (part 1) 2018-05-23 12:07:10 +00:00
WATCHLISTS Fixing root_files WATCHLIST regex. 2018-04-19 06:52:18 +00:00
webrtc.gni Broadcast extension for AppRTCMobile on iOS 2018-06-04 08:49:21 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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