andresp@webrtc.org 8d375c95b7 Fix deadlock on register/unregister observer while there is a an going callback. ...

BUG=2835
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-01-23 23:09:25 +00:00

..

audio_coding

Add full path to headers

2014-01-16 09:01:39 +00:00

audio_conference_mixer

Removes the remaining uses of the list wrapper class and the list wrapper class.

2014-01-13 22:41:34 +00:00

audio_device

Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback

2014-01-17 23:12:51 +00:00

audio_processing

Fix array declarations in aec_rdft.h.

2014-01-23 20:55:14 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

2013-10-24 09:24:06 +00:00

desktop_capture

Implement screen enumeration and individual screen capturing for Windows.

2014-01-17 17:19:16 +00:00

interface

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

media_file

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +00:00

pacing

Fix for libtalkmobile build error

2014-01-21 16:16:58 +00:00

remote_bitrate_estimator

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

rtp_rtcp

Add callbacks for receive channel RTP statistics

2014-01-23 10:00:39 +00:00

utility

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +00:00

video_capture

Fix deadlock on register/unregister observer while there is a an going callback.

2014-01-23 23:09:25 +00:00

video_coding

Remove empty VideoCodecGeneric struct.

2014-01-14 12:55:59 +00:00

video_processing/main

Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.

2013-11-25 09:06:33 +00:00

video_render

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +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

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules_unittests.isolate

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules.gyp

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00