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Steve Anton 8acdd1a1dc Parameterize PeerConnection signaling tests for Unified Plan 
Bug: webrtc:8765
Change-Id: I50fbcfab66edb70c069b2dcb803c78b516e428c9
Reviewed-on: https://webrtc-review.googlesource.com/47582
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21977}
2018-02-09 19:48:09 +00:00
api Remove unused fields from VideoCodecVP8. 2018-02-09 15:55:59 +00:00
audio Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. 2018-02-07 10:07:28 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Reland "Moved congestion controller to task queue." 2018-02-06 08:38:49 +00:00
common_audio Stop using public_deps in common_audio. 2018-02-06 09:44:20 +00:00
common_video Prevent potential integer overflow in sps parser 2018-02-09 13:52:48 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Update AppRTCMobile AAR-project to SDK version 27. 2018-02-07 16:22:31 +00:00
infra Remove win_chromium_webrtc_compile_rel_ng from CQ 2018-02-01 19:26:24 +00:00
logging Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. 2018-02-07 10:07:28 +00:00
media Remove dead code GetSimulcastSsrcs from simulcast.{cc,h}. 2018-02-09 15:57:39 +00:00
modules Add callback when new audio data is ready 2018-02-09 19:31:49 +00:00
ortc Change FakePeriodicVideoCapturer to use a TaskQueue instead of Thread. 2018-02-07 20:51:51 +00:00
p2p Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." 2018-02-08 16:25:31 +00:00
pc Parameterize PeerConnection signaling tests for Unified Plan 2018-02-09 19:48:09 +00:00
resources Add a new NetEq decoding unit test for Opus with DTX 2017-11-28 10:45:38 +00:00
rtc_base Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." 2018-02-08 16:25:31 +00:00
rtc_tools Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. 2018-02-07 10:07:28 +00:00
sdk Implements JavaToNativeStringMap and adds tests for native API. 2018-02-09 10:34:44 +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 Simplify CodecSettings helper function. 2018-02-08 14:38:59 +00:00
tools_webrtc MB: Add a way to run tests on swarming without using gtest-parallel. 2018-02-08 19:37:19 +00:00
video Includes the time(ms) that spent in network to test results. 2018-02-09 17:12:59 +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 libyuv 'gn check'. 2018-02-09 10:31:04 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Expose a link-local network interfaces enumeration option 2018-02-06 19:12:04 +00:00
BUILD.gn Implements JavaToNativeStringMap and adds tests for native API. 2018-02-09 10:34:44 +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 Revert "Roll chromium_revision 6fcad09d4a..004f3b4b40 (535596:535645) + sysroot fix" 2018-02-09 12:56:59 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +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 Enable building WebRTC without built-in software codecs 2018-01-31 08:33:59 +00:00
whitespace.txt Moving src/webrtc into src/. 2017-09-15 04:25:06 +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