andrew@webrtc.org 755b04a06e Add RMS computation for the RTP level indicator. ... - Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer. - We now use the entire packet rather than the last 10 ms frame. - Restore functionality to LevelEstimator. - Fix a bug in the splitting filter. - Fix a number of bugs in process_test related to a poorly named AudioFrame member. - Update the unittest protobuf and float reference output. - Add audioproc unittests. - Reenable voe_extended_tests, and add a real function test. - Use correct minimum level of 127. TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test Review URL: http://webrtc-codereview.appspot.com/279003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-11-15 16:57:56 +00:00
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android	Add unit test output. webrtc r319, ran on Xoom, synced source code on 8/8.	2011-08-09 18:13:15 +00:00
aec_far.pcm	Creating a new directory for test data files, and moving audio_processing files there.	2011-06-23 11:45:12 +00:00
aec_near.pcm	Creating a new directory for test data files, and moving audio_processing files there.	2011-06-23 11:45:12 +00:00
output_data_fixed.pb	Update fixed point audio processing output.	2011-10-25 03:29:08 +00:00
output_data_float.pb	Add RMS computation for the RTP level indicator.	2011-11-15 16:57:56 +00:00