This is becoming a bit off-topic and too technical to be of general interest I fear. But I do agree with Sergei that the lower the frequency you want to process, then more taps you need on your filter. There is no way out. Just the same for analog filters you need a certain time for the response to settle. And distinguishing the edges of a 1/6 octave filter at 10 Hz and the passband is rather for the sake of giving it a name. You are talking about milliHertz range. Johan On Saturday 27 January 2007 22:58, Bill Unruh wrote: > On Sun, 28 Jan 2007, Sergei Steshenko wrote: > > > On Sat, 27 Jan 2007 14:23:19 -0800 (PST) > > Bill Unruh <unruh@xxxxxxxxxxxxxx> wrote: > > > >> On Sat, 27 Jan 2007, Sergei Steshenko wrote: > >> > >>> On Sat, 27 Jan 2007 12:45:06 -0800 (PST) > >>> Bill Unruh <unruh@xxxxxxxxxxxxxx> wrote: > >>> > >>>> 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. > >>>>> > >>>>> > >>>> > >>> > >>> Bill, > >>> > >>> think of: > >>> > >>> 1) relationship between Q factor of on oscillating loop and its > >>> ability to react to quickly changing envelope; > >> > >> It does not matter. The response of the loop is completely local in time. > >> The next instant's voltage output of the oscillator is simply determined by > >> the charge on the capacitor and the current through the > >> inductor/resistor. This purely local intereaction is what gives the system > >> its memory. Otherwise analog filters would not work either-- suffereing > >> from huge latencies. > >> > >>> > >>> 2) (non-equal for different frequencies in IIR/analog equalizer) > >>> group delay; > >> > >> Again, it does not matter. > >> These all arise purely from the local in time equations of motion, which > >> can be modeled by purely local equations for the digital stream (ie > >> depending only on a few buffer values which are updated at each time step , and the input value > >> now. > >> > >>> > >>> 3) possible pulse smudging in case of non-equal group delay. > >>> > >>> If Ronan opens the new thread, we'll discuss it all there. > >>> > >>> Regards, > >>> Sergei. > >>> > >>> > >> > > > > You may tell me is much as you can it doesn't matter. > > > > I did build an IIR bands per octave high Q (1/6 of octave can be > > suppressed by 30db) equalizer, and it works, just the difference > > in group delay is so big that drums + percussions don't sound > > naturally - the low frequency component comes much later than > > the high frequency one. > > > > Bill, again, have a look at Q <-> F <-> group delay <-> what > > we, humans, hear. > > > > Think of the whole thing differently - who what smallest time you > > have to observe a signal to understand that it has a 1Hz component ? > > > > The answer is 1 second. > > No. No human ear can detect 1 Hz-- 20 Hz is the minimum, but will accept > your point that the an analog equaliser at low frequencies can start > messing with the timing of the low frequency signals. This has nothing to > do with latency, or with locality, and it has nothing to do with digital > processing. > > > > Have you ever conducted laboratory work on how signals propagate > > through oscillating loops ? > > Sure. It is a standard undergraduate experiment. > > > > > Have you observed the delay in envelopes ? > > The "delay" occurs at the edges of the bandpass (for a bandpass filter)-- > ie places wheer the phase of the filter changes rapidly. So what you would > really like are low phase shift filter designs. > > > > > Do you agree at all agree that we can have good resolution either > > in frequency or in time domain, but not both ? > > ?? No idea what this has to do with the question at hand. > He wants a filter design to alter the frequency response at the ultra low > bass frequencies. Why he would want that I have no idea. But, assuming he > can do this with an analog equaliser board, he can also do it with a > digital one, with minimal extra latency. Whether he really wants to do that or > whether it is a good idea I have no comments on. > > > > > > > > > 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. 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