webrtc

History

bjornv@webrtc.org 6a9835d59c Delay estimator structural changes.

Improved the way we handle different data types (float vs fixed) and reduced the complexity by nearly 50%.
We now have a generic struct for both float and fixed delay estimators and a core struct for the binary spectrum based delay estimator. All wrapper codes (for both fixed and float) are gathered in delay_estimator_wrappers.*.
Moved out the far end history buffer to AEC(M).
Added a union to handle difference types when create.
Review URL: http://webrtc-codereview.appspot.com/277004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@973 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-18 08:30:34 +00:00

audio_coding

Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix.

2011-11-18 02:50:55 +00:00

audio_conference_mixer

Add RMS computation for the RTP level indicator.

2011-11-15 16:57:56 +00:00

audio_device

This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend().

2011-11-18 02:17:28 +00:00

audio_processing

Delay estimator structural changes.