webrtc/src/modules
marpan@webrtc.org 2dad3fbe49 Media-opt: Added a filter type mode for the filtering of the received packet loss. This makes the filter selection explicit and easier to modify/test.
Removed the function UpdateLossPr(); the filter updates are done in the same function that returns the filtered loss.
Review URL: http://webrtc-codereview.appspot.com/333018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1361 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 18:18:36 +00:00
..
audio_coding Changed build settings for ARMv5 in Android. 2012-01-04 17:47:57 +00:00
audio_conference_mixer Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
audio_device Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
audio_processing Changed build settings for ARMv5 in Android. 2012-01-04 17:47:57 +00:00
interface Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
media_file Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
rtp_rtcp Made send pad data generic (audio and video) 2012-01-05 10:54:44 +00:00
udp_transport Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
utility Removed usage of the deprecated critical section constructor in modules/utility. 2011-12-16 19:53:46 +00:00
video_capture Updating capture module following latest libyuv api changes 2012-01-04 19:23:24 +00:00
video_coding Media-opt: Added a filter type mode for the filtering of the received packet loss. This makes the filter selection explicit and easier to modify/test. 2012-01-09 18:18:36 +00:00
video_processing/main Removed Version function from all modules. 2012-01-04 15:00:12 +00:00
video_render Fix for the build broken on Windows. 2012-01-04 22:38:05 +00:00
modules.gyp Restructuring and adding dummy unit test target. 2011-11-17 13:56:54 +00:00