stefan@webrtc.org 791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late. ... BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/200001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-10-11 07:53:43 +00:00
..
audio_coding	Adding neteq_tests to modules tests	2011-10-05 20:38:19 +00:00
audio_conference_mixer	refactor the gyp file to gypi file.	2011-09-12 12:24:39 +00:00
audio_device	Remove trailing whitespace in AudioDevice.	2011-10-06 16:34:36 +00:00
audio_processing	Some code style changes in audio_processing/ns/main/source/ by Astyle,	2011-10-05 23:36:01 +00:00
interface	Adding missing parts related to VP8 partitions	2011-09-08 10:11:24 +00:00
media_file	refactor the gyp file to gypi file.	2011-09-12 12:24:39 +00:00
rtp_rtcp	Removed code under a non-existing define.	2011-10-06 18:14:25 +00:00
udp_transport	Removed code under a non-existing define.	2011-10-06 18:14:25 +00:00
utility	refactor the gyp file to gypi file.	2011-09-12 12:24:39 +00:00
video_capture	Return the number of /dev/video* without trying to open it.	2011-09-21 16:57:15 +00:00
video_coding	Add API to get the number of packets discarded by the video jitter buffer due to being too late.	2011-10-11 07:53:43 +00:00
video_processing/main	Fix MSVC issues in AEC to enable SSE2 optimization on Windows.	2011-09-16 22:51:57 +00:00
video_render	refactor the gyp file to gypi file.	2011-09-12 12:24:39 +00:00
modules.gyp	First version of video quality measurement program and test framework.	2011-10-06 06:44:54 +00:00