mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-19 08:37:54 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
No description
32550 commits 2 branches 92 tags 416 MiB
C++ 89.3%
Java 2.9%
C 2.8%
Objective-C++ 1.9%
Python 1.8%
Other 1%
Find a file
Artem Titov 28547e96cc Fix typos in network emulation default routing 
Bug: b/180750880
Change-Id: I8a927d5cb66af2292eff13382ed956def1585922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208481
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33318}
2021-02-22 14:25:27 +00:00
api Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
audio Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
build_overrides [build] Remove obsolete gn flag 2021-01-11 17:57:44 +00:00
call Use pixels from single active stream if set in CanDecreaseResolutionTo 2021-02-22 10:25:32 +00:00
common_audio Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
common_video Delete rtc::Callback0 and friends. 2021-02-16 12:41:35 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc: show how to build the fuzzers 2021-02-18 08:28:24 +00:00
examples Update the call to RuntimeEnvironment.application 2021-02-17 12:21:17 +00:00
logging Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
media Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
modules Extract sequencing from RtpSender 2021-02-22 14:00:06 +00:00
p2p Use CallbackList for DtlsState in dtls_transport. 2021-02-17 07:42:13 +00:00
pc Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
resources Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
rtc_base Delete unused sigslot SignalAddressReady and MSG_ID_ADDRESS_BOUND 2021-02-22 14:19:36 +00:00
rtc_tools Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
sdk Reland "Replace RecursiveCriticalSection with Mutex in RTCAudioSession." 2021-02-19 15:45:33 +00:00
stats Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
style-guide Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
system_wrappers Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
test Fix typos in network emulation default routing 2021-02-22 14:25:27 +00:00
tools_webrtc Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" 2021-02-17 12:28:07 +00:00
video Use pixels from single active stream if set in CanDecreaseResolutionTo 2021-02-22 10:25:32 +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 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h 2020-09-07 08:37:14 +00:00
.vpython Reland "Add protobuf-py2_py3 3.13.0 to .vpython." 2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS changed src\modules\audio_device\win\audio_device_core_win.cc , and it is working 2021-02-04 09:31:33 +00:00
BUILD.gn Move SequenceChecker header to API: step 1, move header only 2021-02-08 11:49:58 +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
DEPS Roll chromium_revision b5988d40c8..e1b9354ff4 (853388:854007) 2021-02-15 14:36:39 +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
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 the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Add deprecation section to webrtc style guide 2021-02-22 13:34:40 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +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: move bug reporting instructions to the repository 2020-10-21 14:47:49 +00:00
style-guide.md Add deprecation section to webrtc style guide 2021-02-22 13:34:40 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Add build argument rtc_exclude_system_time 2021-02-19 16:36:14 +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 Whitespace update 2021-02-14 19:14:44 +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