glaznev@webrtc.org a4da771914 Fix deadlock in Android stopCapture() call. ...

BUG=3467
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6673 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-14 17:01:53 +00:00

..

audio_coding

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

audio_conference_mixer

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

audio_device

Remove unnecessary build message.

2014-07-11 11:15:35 +00:00

audio_processing

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

bitrate_controller

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

desktop_capture

GN: Fix include paths for WebRTC in Chromium build.

2014-07-13 09:02:54 +00:00

interface

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

media_file

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

pacing

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

remote_bitrate_estimator

GN: Fix include paths for WebRTC in Chromium build.

2014-07-13 09:02:54 +00:00

rtp_rtcp

Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .

2014-07-11 20:33:39 +00:00

utility

Some refactoring inside rtp_rtcp/.

2014-07-08 12:10:51 +00:00

video_capture

Fix deadlock in Android stopCapture() call.

2014-07-14 17:01:53 +00:00

video_coding

Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .

2014-07-11 20:33:39 +00:00

video_processing

Add initial gn build files for video_coding and video_processing.

2014-07-07 11:46:43 +00:00

video_render

Add tkchin@ to OWNERS.

2014-07-01 18:38:28 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Receiver bit-exactness test for AudioCoding Module

2014-06-26 19:07:04 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00