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webrtc/webrtc
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henrike@webrtc.org 7ea71de396 clock.h: Removed GUARDED_BY annotation as it breaks som builds. 
BUG=N/A
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-26 16:13:58 +00:00
..
base pkg-config-wrapper should not be run when build_nss is disabled (=0). 2014-06-25 20:14:13 +00:00
build GN: BUILD.gn for system_wrappers 2014-06-21 14:25:16 +00:00
common_audio GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
common_video GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
examples AppRTCDemo(android): support app (UI) & capture rotation. 2014-06-06 18:40:44 +00:00
modules Fixes a bug causing NACKs to be dropped excessively at the send-side. 2014-06-26 11:44:49 +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
system_wrappers clock.h: Removed GUARDED_BY annotation as it breaks som builds. 2014-06-26 16:13:58 +00:00
test Add RTCP packet types to packet builder: 2014-06-25 12:22:17 +00:00
tools Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
video_engine Bump version number to 3.55 2014-06-25 22:04:27 +00:00
voice_engine GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
call.h Implements start bitrate for new video API. 2014-06-16 08:57:39 +00:00
common_types.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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 Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-05-12 12:19:19 +00:00
experiments.h Adding API for setting bandwidth estimation configurations. 2014-03-25 10:37:31 +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
sanitizer_options.gyp Roll chromium_revision 272489:277350 + fix sanitizer options 2014-06-21 19:30:29 +00:00
supplement.gypi Roll chromium_revision 272489:277350 + fix sanitizer options 2014-06-21 19:30:29 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_receive_stream.h Rename Start/Stop in Video{Send,Receive}Streams. 2014-04-24 11:13:21 +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 Enable pacing by default and remove the option to disable it from the new API. 2014-06-12 15:12:25 +00:00
webrtc_examples.gyp Add webrtc field trials API. 2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Rename neteq4 folder to neteq 2014-06-09 08:10:28 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.