webrtc/test
andrew@webrtc.org daacee81b8 Use better reference files with audioproc_unittest.
The files are shorter (7 s) with one set provided for each sample rate.

Will be accompanied by the following set of files in the resource bundle:
far8_stereo.pcm
far16_stereo.pcm
far32_stereo.pcm
near8_stereo.pcm
near16_stereo.pcm
near32_stereo.pcm

BUG=114
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/380003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 00:01:04 +00:00
..
data Use better reference files with audioproc_unittest. 2012-02-07 00:01:04 +00:00
functional_test Updated test web page info for PeerConnection v2. 2012-02-01 13:10:48 +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 Removed default cases causing clang errors, -Wcovered-switch-default. 2012-02-06 10:11:25 +00:00
metrics.gyp Changing all PSNR/SSIM calculations to use libyuv. 2012-01-04 08:09:32 +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 Changing all PSNR/SSIM calculations to use libyuv. 2012-01-04 08:09:32 +00:00