mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-15 06:40:43 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
No description
21405 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
Gustaf Ullberg f35c6667d6 Separate build targets for aec3 and aec3_unittests 
Bug: webrtc:8844
Change-Id: Id6a98eae19aaedc87c3f402a004f58f0290d5c28
Reviewed-on: https://webrtc-review.googlesource.com/56580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22173}
2018-02-23 13:16:16 +00:00
api Separate build targets for aec3 and aec3_unittests 2018-02-23 13:16:16 +00:00
audio Separate build targets for aec3 and aec3_unittests 2018-02-23 13:16:16 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Added some margin to ramp down target in perf test. 2018-02-23 12:15:15 +00:00
common_audio Conditionally include real_fourier_openmax.h. 2018-02-23 09:02:36 +00:00
common_video Drop dependency of common_video on api:libjingle_peerconnection_api. 2018-02-19 13:20:24 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Add support for saving local audio input to file in AppRTCMobile 2018-02-21 14:09:56 +00:00
infra Shorten Chromium compile trybot names 2018-02-22 10:06:20 +00:00
logging Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. 2018-02-07 10:07:28 +00:00
media Move default thresholds from QualityScaler to encoders. 2018-02-23 13:12:36 +00:00
modules Separate build targets for aec3 and aec3_unittests 2018-02-23 13:16:16 +00:00
ortc Removed new calls on RtpTransportControllerSend. 2018-02-22 12:54:43 +00:00
p2p Create the JsepTransportController and JsepTransport2. 2018-02-23 00:13:45 +00:00
pc Create the JsepTransportController and JsepTransport2. 2018-02-23 00:13:45 +00:00
resources Adding FourPeople_1280x720_30.yuv. 2018-02-12 15:55:00 +00:00
rtc_base ClosePlatformFile() on non-Windows: Return true on success, false on failure 2018-02-22 14:18:49 +00:00
rtc_tools Preparing for task queue in congenstion controller 2018-02-20 12:35:15 +00:00
sdk Move default thresholds from QualityScaler to encoders. 2018-02-23 13:12:36 +00:00
stats Removing useless dependencies on //testing/gmock. 2018-01-26 13:34:12 +00:00
system_wrappers Reland Use runtime enabled features API to enable dual stream mode 2018-01-18 12:22:49 +00:00
test Separate build targets for aec3 and aec3_unittests 2018-02-23 13:16:16 +00:00
tools_webrtc Delete unused sample project code. 2018-02-20 09:51:52 +00:00
video Move default thresholds from QualityScaler to encoders. 2018-02-23 13:12:36 +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 Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Re-enabling 'gn check' on //examples/*. 2018-02-19 15:07:45 +00:00
.vpython vpython: Specify dependency on pywin32 2018-02-14 13:56:39 +00:00
AUTHORS Expose a link-local network interfaces enumeration option 2018-02-06 19:12:04 +00:00
BUILD.gn Removing definition of FEATURE_ENABLE_VOICEMAIL. 2018-02-19 15:51:24 +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 Add flexfec payload name to string-type conversions 2018-01-31 08:58:39 +00:00
common_types.h Remove unused fields from VideoCodecVP8. 2018-02-09 15:55:59 +00:00
DEPS Roll chromium_revision eb957d794e..1cf758f803 (538454:538618) 2018-02-23 07:43:57 +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@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Delete unused MediaFile module. 2018-01-29 11:18:18 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +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 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
typedefs.h Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix 2018-02-05 11:24:59 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Removing obsolete defines. 2018-02-19 14:35:45 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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 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