On Sat, 27 Jan 2007, Sergei Steshenko wrote: > On Sat, 27 Jan 2007 10:32:30 -0800 > "ronan mcallister" <bass.woofer@xxxxxxxxx> wrote: > >> Sergei, >> >> For the moment forgetting about the Xover's, how would I use ecasound or >> another tool to implement an arbitrary EQ function with sliders / user >> controls? I've got JACK now running better (mainly a problem related to >> configuration) and I'd like to have maybe a dozen or more bands of very LF >> EQ (eg, fc: 5hz, 8hz, 12hz,.... 100hz) for subsonic equalization. >> >> So far it appears brutefir can do this but sans a GUI? What plugin would I >> need and is it extensible? >> >> should I start a new topic to discuss the IIR based EQ you hinted about? >> >> Thank you very much, >> Ronan >> >> > > Yes, please start a new topic about IIR vs FIR, but, anyway, if you > want low latencies AND equalization at 5hz, 8hz, 12hz, forget about > it - it's impossible physically/mathematically - regardless of OS > and sound system, and regardless of digital/analog. > > I.e, you can either have > > low (latency/group delay) AND equalization only at high frequencies What? What are you trying to say here? Most equalizers are just realisations of second order differential equations ( or fourth order) that is why analog systems can create them. The behaviour at the next instant of time depends only on the values of certain variables at this instant of time. That is local and is locally simulatable digitally. There is no need to wait for many periods of the signal. Thus if o_i is the ith output and f_i is the ith input o_i+1= ((1-a)o_i -2afi)/(1+a) is a low pass single pole filter with the low passband frequency determined by a. Even a 12 pole filter can be done using only 13 immediate frequencies. and you do not need to wait, you just save the last 12 in a buffer. Ie, this filter has as latency only the time required to actually carry out the calculation. You certainly would not impliment this by doing a Fourier transform. Just as the analog filter does not do it by instituting a fourier transform-- it impliments the filter by storing information in the charge on capacitors, or currents in inductors and the next value of the output depends only on the immediate values of those few variables. Now if your purpose is to do frequency shifting or resampling that is far more difficult, because there things really are non-local in time. > > OR > > big (latency/group delay) AND equalization also at low frequencies. Or low latency and equalisation at low frequencies. > > Regards, > Sergei. > > -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | unruh@xxxxxxxxxxxxxx Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/ ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
