pwestin@webrtc.org 52aa019e98 Avoid adding duplicates in pacer lists. ...

Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-25 17:35:56 +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

Avoid adding duplicates in pacer lists.

2013-04-25 17:35:56 +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

Make sure timestamps are monotonically increasing.

2013-04-24 23:54:31 +00:00

video_coding

Revert 3892 "VCM/JB: Using last decoded state for waiting for key"

2013-04-24 02:13:18 +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