mikhal@webrtc.org dbd6a6d653 Updating delay for first value ...

BUG=

Review URL: https://webrtc-codereview.appspot.com/1327005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3865 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-17 16:23:22 +00:00

..

audio_coding

Adding Opus unit test

2013-04-17 10:39:41 +00:00

audio_conference_mixer

Remove vim/emacs modelines from .gypi files

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

audio_device

Removing remaining WebRtc_Word32 not in typedefs.h

2013-04-10 17:59:17 +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

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

Remove vim/emacs modelines from .gypi files

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

rtp_rtcp

Remove vim/emacs modelines from .gypi files

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

utility

Remove vim/emacs modelines from .gypi files

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

video_capture

Change capture interface to use NTP capture time.

2013-04-12 11:56:23 +00:00

video_coding

Updating delay for first value

2013-04-17 16:23:22 +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

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

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