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Alessio Bazzica 29accefbb2 Export script bug fixed. 
Bug: webrtc:7218
Change-Id: Ie8b512290578111b8eae5f9ee2535bb015da7cb2
Reviewed-on: https://webrtc-review.googlesource.com/3781
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19990}
2017-09-27 09:47:16 +00:00
api Remove unnecessary audio references in PeerConnectionFactory 2017-09-23 14:36:14 +00:00
audio Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
build_overrides Use hermetic toolchain on Mac, except for local iOS builds 2017-09-11 18:38:48 +00:00
call Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. 2017-09-26 19:11:38 +00:00
common_audio Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_video Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Reland "Remove precompiled header for AppRTCMobile." 2017-09-27 09:02:15 +00:00
infra Update webrtc CQ to commit to new location 2017-09-13 18:05:33 +00:00
logging Remove #include of rtc_stream_config.h from rtc_event_log.h 2017-09-21 09:05:54 +00:00
media Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
modules Export script bug fixed. 2017-09-27 09:47:16 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Reland: Completed the functionalities of SrtpTransport. 2017-09-26 18:12:45 +00:00
pc Reland: Completed the functionalities of SrtpTransport. 2017-09-26 18:12:45 +00:00
resources Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtc_base Remove backwards compatibilty header for Optional 2017-09-20 19:17:42 +00:00
rtc_tools Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
sdk Android: Suppress lint warnings in JNI generator header 2017-09-27 09:22:15 +00:00
stats Added RTCMediaStreamTrackStats.concealmentEvents 2017-09-18 08:58:06 +00:00
system_wrappers Delete unused Atomic32 overloads of binary +/- operators. 2017-09-22 08:54:23 +00:00
test Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
tools_webrtc Fix isac_fix_test on swarming perf bot. 2017-09-27 06:32:15 +00:00
video Continuously request keyframes if decoding does not recover. 2017-09-26 09:54:58 +00:00
voice_engine Remove unnecessary send codec initialization from voe::Channel. 2017-09-27 00:13:19 +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 gitignore: Remove webrtc/ 2017-09-15 10:51:29 +00:00
.gn Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) 2017-09-25 13:37:12 +00:00
BUILD.gn Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) 2017-09-22 11:30:08 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
common_types.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_types.h Added RTCMediaStreamTrackStats.concealmentEvents 2017-09-18 08:58:06 +00:00
DEPS Roll chromium_revision 69fe0e1a5f..ff8cef57fe (504538:504574) 2017-09-27 04:22:45 +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
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Presubmit: Add check to support b/xxx entry in bug reference. 2017-09-26 13:50:25 +00:00
presubmit_test.py Presubmit: Add check to support b/xxx entry in bug reference. 2017-09-26 13:50:25 +00:00
presubmit_test_mocks.py Reland "Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++."" 2017-09-19 14:23:00 +00:00
pylintrc Fix Python shebang and license for presubmit_test.py 2017-09-15 09:22:50 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
style-guide.md Style guide: Attempt to make the L2 and L3 headings more visually distinct 2017-09-09 03:52:23 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Add myself to the watchlist for webrtc/api/ and webrtc/base/ 2017-05-04 13:22:46 +00:00
webrtc.gni Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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.

More info