webrtc/test
bjornv@webrtc.org 7270a6bcc2 Merged apm-buffer branch [r1293] back to trunk.
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.

TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 08:44:17 +00:00
..
data Merged apm-buffer branch [r1293] back to trunk. 2011-12-28 08:44:17 +00:00
functional_test Review URL: http://webrtc-codereview.appspot.com/269019 2011-11-22 14:34:44 +00:00
sanity_check Make the sanity check test a little more robust, and add a README file. 2011-10-14 13:56:26 +00:00
testsupport Integration test that tracks dropped frames and compares video output. 2011-12-21 16:11:25 +00:00
OWNERS Fix Amy's email address. 2011-11-16 02:08:52 +00:00
run_all_unittests.cc Changing the namespace of TestSuite to webrtc::test. 2011-11-04 01:19:16 +00:00
test_suite.cc Changing the namespace of TestSuite to webrtc::test. 2011-11-04 01:19:16 +00:00
test_suite.h Changing the namespace of TestSuite to webrtc::test. 2011-11-04 01:19:16 +00:00
test.gyp Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp. 2011-12-08 07:42:18 +00:00