leozwang@google.com
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325bca7ccf
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Add unit test output. webrtc r319, ran on Xoom, synced source code on 8/8.
Review URL: http://webrtc-codereview.appspot.com/100005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@338 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-08-09 18:13:15 +00:00 |
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andrew@webrtc.org
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14acdbc14d
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Update fixed-point profile output due to r313.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@333 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-08-09 01:54:03 +00:00 |
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ajm@google.com
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59e41405d1
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Add a fixed-point profile to the APM unit test.
It uses fixed-point NS, AECM and adaptive digital AGC. It's selected by enabling "prefer_fixed_point" in common.gypi.
Review URL: http://webrtc-codereview.appspot.com/88009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@266 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-28 17:34:04 +00:00 |
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ajm@google.com
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a769fa51c0
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Adding more output data checks to APM unittest. Blowing out the protobuf definition (changing the tags) since we're still in the formative stages. Later, this would be very bad. Leaving a Frame message in case we want frame-by-frame data, but we prefer to keep the output storage small in general so avoiding it thus far.
Review URL: http://webrtc-codereview.appspot.com/68004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@203 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-13 21:57:58 +00:00 |
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hellner@google.com
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1b627c72b5
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Tests using the rtp_rtcp test data should now be run from inside trunk/test/data/rtp_rtcp. I.e. all test files were moved to the test folder.
Review URL: http://webrtc-codereview.appspot.com/60006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@185 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-08 17:16:47 +00:00 |
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tlegrand@google.com
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3675f9b121
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Review URL: http://webrtc-codereview.appspot.com/56003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@181 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-08 06:43:34 +00:00 |
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ajm@google.com
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7c4469bf61
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Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up.
Review URL: http://webrtc-codereview.appspot.com/56002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@151 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-06 17:45:37 +00:00 |
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henrika@google.com
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c5758f8c51
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Uploaded test files for ADM functional tests.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@150 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-06 08:34:04 +00:00 |
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tlegrand@google.com
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0adca82c35
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Move iLBC test and reference files to new location.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@147 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-05 09:10:23 +00:00 |
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henrika@google.com
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2e8a1a2092
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Creates new test folder for VoiceEngine test files and adds the required files.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@144 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-07-04 15:39:40 +00:00 |
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ajm@google.com
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95fa29ec96
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Creating a new directory for test data files, and moving audio_processing files there.
Review URL: http://webrtc-codereview.appspot.com/48004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@121 4adac7df-926f-26a2-2b94-8c16560cd09d
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2011-06-23 11:45:12 +00:00 |
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