webrtc

History

henrike@webrtc.org d72a7599d4 Create a copy of talk/xmllite under webrtc/xmllite. BUG=3379 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7027 4adac7df-926f-26a2-2b94-8c16560cd09d		2014-09-02 15:41:12 +00:00
..
base	GN: Update webrtc/base to recent GYP changes.	2014-09-02 11:22:06 +00:00
build	Create a copy of talk/xmllite under webrtc/xmllite.	2014-09-02 15:41:12 +00:00
common_audio	Unpacking aecdumps generates wav files	2014-09-02 07:51:51 +00:00
common_video	Android APK tests built from a normal WebRTC checkout.	2014-09-01 11:06:37 +00:00
examples	Remove former team members from OWNERS and WATCHLISTS	2014-08-26 06:12:08 +00:00
libjingle/xmllite	Create a copy of talk/xmllite under webrtc/xmllite.	2014-09-02 15:41:12 +00:00
modules	Disable video_engine_tests and webrtc_perf_tests on Android.	2014-09-02 15:13:55 +00:00
overrides/webrtc/base	Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.	2014-05-21 16:52:14 +00:00
sound	Add talk owners to migrated talk folders	2014-08-28 16:03:58 +00:00
system_wrappers	Android APK tests built from a normal WebRTC checkout.	2014-09-01 11:06:37 +00:00
test	Disable video_engine_tests and webrtc_perf_tests on Android.	2014-09-02 15:13:55 +00:00
tools	RTCBot is a framework that allows to write tests where logic runs on a single	2014-09-02 10:52:54 +00:00
video	Disable video_engine_tests and webrtc_perf_tests on Android.	2014-09-02 15:13:55 +00:00
video_engine	Android APK tests built from a normal WebRTC checkout.	2014-09-01 11:06:37 +00:00
voice_engine	Android APK tests built from a normal WebRTC checkout.	2014-09-01 11:06:37 +00:00
.gitignore	.gitignore: Add *.mk, created as part of ChromiumOS build	2013-01-04 21:25:42 +00:00
BUILD.gn	GN: Disable Chromium clang plugins for standalone build.	2014-08-25 14:15:35 +00:00
call.h	Remove GetDefaultConfigs() from Call.	2014-07-07 04:45:15 +00:00
common_types.h	Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.	2014-07-11 13:44:02 +00:00
common.gyp	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +00:00
common.h	Add a Config class interface to AudioProcessing for passing options.	2013-07-25 18:28:29 +00:00
config.cc	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +00:00
config.h	Check before send/receive rtp header extensions.	2014-07-20 15:27:35 +00:00
engine_configurations.h	Add boilerplate code for H.264.	2014-07-04 12:42:07 +00:00
experiments.h	Remove no longer used SkipEncodingUnusedStreams.	2014-07-22 07:17:17 +00:00
frame_callback.h	Wire up statistics in video receive stream of new API	2014-02-07 12:06:29 +00:00
LICENSE	Move src/ -> webrtc/	2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY	Consolidate all third party licenses in LICENSE_THIRD_PARTY.	2013-05-05 18:54:10 +00:00
OWNERS	GN: Add BUILD.gn files + kjellander to OWNERS	2014-06-23 19:21:07 +00:00
PATENTS	Move src/ -> webrtc/	2012-10-22 18:19:23 +00:00
PRESUBMIT.py	PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.	2014-05-23 17:27:18 +00:00
README.chromium	Move src/ -> webrtc/	2012-10-22 18:19:23 +00:00
supplement.gypi	Roll chromium_revision 289723:291647	2014-08-25 14:16:32 +00:00
transport.h	Rename newapi::Transport::SendRTP()->SendRtp().	2013-11-20 12:17:04 +00:00
typedefs.h	Add CHECK and friends from Chromium.	2014-08-28 16:28:26 +00:00
video_engine_tests.isolate	Pass GYP DEPTH variable to isolate.	2014-06-13 09:02:15 +00:00
video_receive_stream.h	Remove GetDefaultConfigs() from Call.	2014-07-07 04:45:15 +00:00
video_renderer.h	Separate Call API/build files from video_engine/.	2013-10-28 16:32:01 +00:00
video_send_stream.h	Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.	2014-07-11 13:44:02 +00:00
webrtc_examples.gyp	WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.	2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate	Pass GYP DEPTH variable to isolate.	2014-06-13 09:02:15 +00:00
webrtc_tests.gypi	Disable video_engine_tests and webrtc_perf_tests on Android.	2014-09-02 15:13:55 +00:00
webrtc.gyp	Create a copy of talk/xmllite under webrtc/xmllite.	2014-09-02 15:41:12 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.