This may be a FAQ, but I don't see one, and you guys seem pretty smart. Is there a standard algorithm for doing audio compression in a DSP context? I mean the type of compression that recording studios and radio stations used to use a Universal LA2A or Universal Audio 1176LN. While these are typically thought of as compression (making the loud ampliture software) they usually work by making the loud sounds software, and then using make-up gain to restore the overall level. This means they really make soft sounds louder and loud sounds stay the same. Perferably simple and in the time domain. Seems like a natural for a look ahead algorithm. It also seems that the naive algorithm, roughtly: look at the signal over some small look ahead window, if the absolute value of the signal exceeds some threshold, multiply all values from the current zero crossing to the max by .9 (or 0.8 or whatever). would do bad things to signals other than the one you are trying to compress (say a soft higher frequency signal over a loud low Hz signal) Thanks -- Pat
Standard audio compression (LA-2A, not MP3) algorithm
Started by ●October 14, 2005
Reply by ●October 15, 20052005-10-15
all single band compressors apply the same gain change to all signal frequencies... so called multi-band compressors can apply individual gain changes to individual frequency bands.. i.e. there may be one compressor for the bass, one for the mid and one for the treble and they work together... the classic LA2A is a single band compressor The art in the design of a compressor involves the tuning of the time constants for the speed of the gain reduction and the speed of the gain increases.... and the transfer function of the gain changes vs input level.. Mark