On Nov 18, 2012, at 2:33 PM, Shashank Kumar (shanxS) wrote: > @ Bjorn: > > Yes, you are right. What I thought was scaling is actually clipping. I > removed it and it worked. > Here is the o/p: > http://trystwithdsp.wordpress.com/2012/11/19/basic-lpf-part-2/
Great. I guess that means LADSPA does not use the usual [-1,1] range. There's nothing really wrong with that, I used to not use the 16-bit integer range in some of my software even for floating point, but that's an unusual choice. > And I have mentioned you and linked that to your blog. If you want me > to link it to some other page let me know. I'll be more than happy to > do it. Thanks :) Glad to help. > I have one more question: > Why so many people use analog prototypes to get a digital filter ? Why > not just put a few constraints on location of poles/zeros on Z plane > and get done with it ? This is a really great question. One answer is that analog filter design is a highly developed art, and therefore serves as an excellent starting point. Anecdote: I know one guy who's a DSP genius. He sent me some designs and told me he did them "directly in the digital domain". He must've forgotten that his starting point was a set of prototypes that came, ultimately, from analog filters. Analog prototypes make great digital filters if you take care with a few things. All that said, some folks have started to think along the same lines as you. After all 1. there may be unique digital solutions (and I'm not just talking about FIR filters), and 2. you should be able to learn digital filter design without also having to learn analog filter design. To that end, here is one interesting paper: http://www.elec.qmul.ac.uk/people/josh/documents/Reiss-2011-TASLP-ParametricEqualisers.pdf As a side note, there are, indeed, problems associated with filters designed with an analog prototype. For example, let's say you design a bell filter in the analog domain, and map it into the digital domain with a sample rate of, say 44100 Hz. Let's further assume that in the analog domain, the gain at the niquist frequency of this filter is 1 dB. That means the filer will boost 20 kHz by 1 dB. When you use the bilinear transform, though, the resulting filter will have a gain a the niquist frequency of zero. This is usually not a serious problem, and often not a problem at all. However, if you are designing a parametric EQ, the difference will be noticeable at extreme settings. You could say this is one reason to do audio production at sample rates > 44100 Hz but there are arguments both ways there as well. bjorn ----------------------------- Bjorn Roche http://www.xonami.com Audio Collaboration http://blog.bjornroche.com -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
