mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
No description
31141 commits 2 branches 91 tags 415 MiB
C++ 89.3%
Java 2.9%
C 2.8%
Objective-C++ 1.9%
Python 1.8%
Other 1%
Find a file
Artem Titov bcb42f1e4b Move initialization of GoogleMock and flags to main from test_main_lib 
Bug: None
Change-Id: Ie3aed382d4e468c4adbfdbcc1bdb3f069d3eaae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181364
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31909}
2020-08-11 11:46:50 +00:00
api Reland "Implement transceiver.stop()" 2020-08-11 10:46:23 +00:00
audio Added Error Checking in Ingress/Egress and Extra Unit Tests 2020-08-06 20:48:13 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call [Adaptation] Remove processing_in_progress_ from ResourceAdaptationProcessor 2020-08-11 08:57:10 +00:00
common_audio Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
common_video Move function PrintVideoFrame to the test file where it is used. 2020-08-06 15:20:19 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Search and replace gendered terms according to style guide: 2020-06-12 14:12:54 +00:00
examples Implemented Android Demo Application for VoIP API 2020-07-21 16:34:22 +00:00
logging Add missing tests for DTLS state logging in RTC event log. 2020-08-07 11:49:43 +00:00
media Add new fmtp parameter for H.264 2020-08-07 10:32:41 +00:00
modules Delete deprecated variant of VideoCodingModule::RegisterReceiveCodec 2020-08-11 08:44:50 +00:00
p2p Delete left-over TODO comment 2020-08-07 11:36:51 +00:00
pc Reland "Implement transceiver.stop()" 2020-08-11 10:46:23 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Update settings for balanced degradation. 2020-08-10 10:59:17 +00:00
rtc_tools video_replay: wrap main thread 2020-08-11 11:28:09 +00:00
sdk Reland "Implement transceiver.stop()" 2020-08-11 10:46:23 +00:00
stats Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
test Move initialization of GoogleMock and flags to main from test_main_lib 2020-08-11 11:46:50 +00:00
tools_webrtc Fix clang revision regexp in chromium autoroller to match new format 2020-08-07 09:22:03 +00:00
video Delete deprecated variant of RtpVideoStreamReceiver::AddReceiveCodec 2020-08-11 09:53:35 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
BUILD.gn Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +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
common_types.h Replaces OverheadObserver with simple getter. 2020-05-07 17:33:45 +00:00
DEPS Roll chromium_revision 48eed58536..5246cdb214 (796566:796691) 2020-08-11 04:37:47 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +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 Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Revert "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-07-30 17:35:30 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Trigger CI bots. 2020-07-15 17:50:55 +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