xians@webrtc.org 0dffc6449a To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. ... This patch disbale some codecs which are not considered necessary. Review URL: http://webrtc-codereview.appspot.com/299001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-11-30 15:35:44 +00:00
..
audio_coding	To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory.	2011-11-30 15:35:44 +00:00
audio_conference_mixer	Added empty unit tests to get code coverage measured.	2011-11-24 07:20:00 +00:00
audio_device	Adding a check to ensure that the memcpy does not exceed bounds of the arrays.	2011-11-29 18:49:54 +00:00
audio_processing	Move stream_delay check to ProcessStream().	2011-11-29 18:28:57 +00:00
interface	Updating to VP8 RTP spec rev -02	2011-11-24 12:52:40 +00:00
media_file	Added empty unit tests to get code coverage measured.	2011-11-24 07:20:00 +00:00
rtp_rtcp	FEC: Fix to valgrind warning.	2011-11-28 22:10:05 +00:00
udp_transport	Added empty unit tests to get code coverage measured.	2011-11-24 07:20:00 +00:00
utility	Added empty unit tests to get code coverage measured.	2011-11-24 07:20:00 +00:00
video_capture	Included modules in webrtc.gyp and fixed build errors.	2011-11-16 15:36:44 +00:00
video_coding	Fix bug introduced when enabling VP8 frame dropping.	2011-11-30 14:41:58 +00:00
video_processing/main	Fixing Release compilation errors	2011-11-23 12:20:35 +00:00
video_render	Fix one more Objective-C clang error.	2011-11-30 00:54:04 +00:00
modules.gyp	Restructuring and adding dummy unit test target.	2011-11-17 13:56:54 +00:00