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8dfe8ff590
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pbos@webrtc.org 8dfe8ff590 Disable capture test for FrameRate on Windows. 
Flaky on Windows, has been for a while.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=3270

Review URL: https://webrtc-codereview.appspot.com/19389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5994 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:27:36 +00:00
..
build Set include_internal_video_capture=1 for video_capture_tests 2014-04-16 12:59:49 +00:00
common_audio Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
common_video Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
examples Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
modules Disable capture test for FrameRate on Windows. 2014-04-28 11:27:36 +00:00
system_wrappers Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
test Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
tools Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
video Disabling flaky CanReceiveFec. 2014-04-28 09:00:50 +00:00
video_engine Disable flaky RunsRtpRtcpTestWIthoutErrors. 2014-04-28 08:49:07 +00:00
voice_engine Replace scoped_array<T> with scoped_ptr<T[]>. 2014-04-25 23:10:28 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
call.h Remove TraceCallback use from Call. 2014-04-24 11:35:33 +00:00
common_types.h Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-04-17 10:45:01 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.h Wire up statistics in video receive stream of new API 2014-02-07 12:06:29 +00:00
engine_configurations.h Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-04-14 14:12:50 +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 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 2014-04-14 20:08:03 +00:00
PATENTS Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
PRESUBMIT.py Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. 2013-07-22 22:32:50 +00:00
README.chromium Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
supplement.gypi Removes script for generating supplement.gypi also adds git ignore for tools/gn. 2014-01-21 15:54:56 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Fix ARM64 detection. 2014-04-01 01:19:08 +00:00
video_engine_tests.isolate Merge metrics_unittests into video_engine_tests. 2013-12-13 14:31:47 +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 Rename Start/Stop in Video{Send,Receive}Streams. 2014-04-24 11:13:21 +00:00
webrtc_examples.gyp Make WebRTC Android examples build without sourcing envsetup.sh 2014-04-15 08:35:00 +00:00
webrtc_perf_tests.isolate Move realtime tests to webrtc_perf_tests. 2013-12-13 12:48:05 +00:00
webrtc_tests.gypi Fix the Android compilation (better structure for NetEq test libs) 2014-04-24 13:19:04 +00:00
webrtc.gyp Integrate fake_network_pipe into direct_transport. 2013-12-18 20:28:25 +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.