mikhal@webrtc.org 1248d4effc VCM/JB: Using last decoded state for waiting for key ...

Review URL: https://webrtc-codereview.appspot.com/1323006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3892 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-23 20:57:06 +00:00

..

audio_coding

Buf fix for r3883.

2013-04-23 16:08:29 +00:00

audio_conference_mixer

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

audio_device

Add lock to prevent possible rare race condition in Win coreAudio capture implementation.

2013-04-18 09:32:07 +00:00

audio_processing

WebRtc_Word32 -> int32_t in audio_processing/

2013-04-10 07:50:54 +00:00

bitrate_controller

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

desktop_capture

Add desktop_capture directory for screen and window capturers.

2013-04-17 18:21:42 +00:00

interface

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

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

media_file

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

pacing

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

remote_bitrate_estimator

revert r3871

2013-04-18 20:26:28 +00:00

rtp_rtcp

Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."

2013-04-23 16:48:32 +00:00

utility

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

video_capture

BUG=1351

2013-04-23 11:57:56 +00:00

video_coding

VCM/JB: Using last decoded state for waiting for key

2013-04-23 20:57:06 +00:00

video_processing/main

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

video_render

WebRTCDemo: handle stride!=width from first frame.

2013-04-10 23:21:10 +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.gyp

emove desktop_capture.gypi from modules.gyp

2013-04-22 13:13:49 +00:00