cduvivier@google.com d0159d8eb0 aec_rdft_128: one entry point for each sign. ... Review URL: http://webrtc-codereview.appspot.com/61007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@153 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-07-06 23:35:37 +00:00
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audio_coding	Moving two testfiles, audio coding module.	2011-07-05 09:17:37 +00:00
audio_conference_mixer	add android makefile, some modification in vpx makefile to build encoder from c source for now	2011-06-07 17:24:39 +00:00
audio_device	Adds sanity checks related to IAudioCaptureClient::GetBuffer.	2011-06-23 09:44:59 +00:00
audio_processing	aec_rdft_128: one entry point for each sign.	2011-07-06 23:35:37 +00:00
interface	git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d	2011-05-30 11:22:19 +00:00
media_file	add android makefile, some modification in vpx makefile to build encoder from c source for now	2011-06-07 17:24:39 +00:00
rtp_rtcp	Disabling DEBUG_FILE in the overuse detector by default.	2011-07-05 14:47:23 +00:00
udp_transport	Fixed valgrind warning in the udp_module.	2011-06-20 23:06:04 +00:00
utility	Route CodecSpecificInfo from encoder to packetizer	2011-07-01 08:32:57 +00:00
video_capture	Remove the full header file path to:	2011-06-22 21:17:43 +00:00
video_coding	VCM: 1. Updating handling of empty packets. 2. Updating JB test. 3. Removing un-used code.	2011-07-01 18:15:11 +00:00
video_processing/main	VPLIB/Interpolation - Delete decode buffer only if too small, this required an API change. In addition, done some clean up and updated test and related code in VCM.	2011-06-29 17:00:03 +00:00
video_render	Disable ChangeWindow function for chromium build.	2011-06-17 22:18:43 +00:00