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

..

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