Re: Help/advice on RME cards and Linux ALSA

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On Sat, 27 Jan 2007 14:58:32 -0800 (PST)
Bill Unruh <unruh@xxxxxxxxxxxxxx> 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.
> >

I said "observe", not "hear". I am saying that to roughly estimate that
a signal contains a 1Hz spectral component one has to observe it (with
his/her eyes, or scope, supersensitive to infra low frequencies ears) for
at least one second.

> > 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.

The delay also occurs in band pass filters (oscillating loops) which
by construction are HPF + LPF, and the bigger the Q factor is, the bigger
the delay is.

> 
> >
> > 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.

To be able to arbitrarily form response at low frequencies Ronan needs
narrow band bandpass filters, i.e. for, say, 10Hz with 1Hz -3db level
Ronan needs a BPF with Q factor 10, which will have group delay of
about 1 second (roughly), which is as bad (at these frequencies) as
1 second latency.

> 
> >
> >
> >
> > Regards,
> >  Sergei.
> >
> >
> 

Regards,
  Sergei.

-- 
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