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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audio_coding	Switch to new sqrt in NetEQ	2011-09-12 16:44:37 +00:00
audio_device	Uploaded test files for ADM functional tests.	2011-07-06 08:34:04 +00:00
audio_processing	Add RMS computation for the RTP level indicator.	2011-11-15 16:57:56 +00:00
rtp_rtcp	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.	2011-07-08 17:16:47 +00:00
voice_engine	Creates new test folder for VoiceEngine test files and adds the required files.	2011-07-04 15:39:40 +00:00