[PD] band-limited square wave

[matthew jones]

cheers for replying,

yes, that's all straightforward.  BUT - if you _construct_ the signal in the
digital domain, it will automatically get those ripples when it goes through
the reconstruction filters in the DAC's (they are designed to take out the
steps from the sample-and-hold circuitry anyway).  you only get aliasing
when going from analogue TO digital without applying antialiasing filters
before the ADC's.

you get what I'm driving at?  ;)

Matt

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http://www.loopit.org/
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----- Original Message -----
From: "Johannes Taelman" <j0 at advalvas.be>
To: "matthew jones" <m.jones at signal.dra.hmg.gb>
Sent: Friday, March 05, 2004 4:22 PM
Subject: Re: [PD] band-limited square wave


> Hi Matt,
>
> A non-bandlimited square wave will only look like straight squares when
> viewed in the continue (non-sampled) time-domain. If you'd pass it to an
> ADC, the ADC would need to brick-wall filter the signal (in the
> frequency domain) to remove frequency components (harmonics) above half
> the sampling rate (nyquist frequency). After this filter the square wave
> will have overshoot and ringing at the edges, which you need to
> reproduce when synthesizing a digital domain alias-free signal. If you
> don't, all the harmonics will be wrapped to the spectrum between DC and
> nyquist frequency.
>
> Filtering after sampling a non-band-limited square wave does not help,
> since the filter cannot distinguish aliased from non-aliased harmonics.
>
> Perfect bandlimited waveforms can be generated with additive synthesis
> but this is very expensive. It can be approximated in several ways. I
> think Tom has written a nice paper on it Check also the music-dsp
> mailinglist archives where this was/is a frequenct topic.
>
> DSP is can of tricky worms. It's full of approximations involving more
> complex math processing than immediatly obvious, and trade-offs between
> quality and cpu usage.
>
> regards,
>
> j#|@
>
> matthew jones wrote:
>
> > Hi Johannes,
> >
> > this is strange.  I'm certainly being a dunce, but I can't see how you
can
> > get aliasing when you're already in the digital domain...??
> > the DAC's have reconstruction filters so there won't be any aliasing
when
> > converted to analogue either.....????!!?!?!
> >
> > any hints/prods appreciated.
> > regards,
> >
> > Matt
> >
> > -=-=-=-=-=-=-=-=-=-=-=-=-
> > http://www.loopit.org/
> > -=-=-=-=-=-=-=-=-=-=-=-=-
> > ----- Original Message -----
> > From: "Johannes Taelman" <j0 at advalvas.be>
> > To: "matthew jones" <m.jones at signal.dra.hmg.gb>
> > Cc: "PD-List" <pd-list at iem.kug.ac.at>; "Martin Dupras"
> > <martin.dupras at uwe.ac.uk>
> > Sent: Friday, March 05, 2004 3:42 PM
> > Subject: Re: [PD] band-limited square wave
> >
> >
> >
> >>Matt,
> >>
> >>None of these methods are band-limited in any sense, and will sound
> >>aliased.
> >>
> >>Check out Tom Schouten's creb package. I started porting it to win32,
> >>some objects work, but continued working other projects before finishing
> >>it...
> >>
> >>  j#|@
> >>
> >>matthew jones wrote:
> >>
> >>
> >>>why not an [osc~] or [phasor~] connected to a [>~]? then bandpass
> >
> > filter?
> >
> >>>phasor is probably nicest cos you can change the 'larger than' value
> >>>linearly to get pulse width modulation.
> >>>
> >>>ideally you want to subtract 0.5 ([-~ 0.5]) from the result and
multiply
> >
> > by
> >
> >>>two ([*~ 2]) to make it a true square wave.
> >>>
> >>>Matt
> >>
> >> >
> >>
> >>>----- Original Message -----
> >>>From: "Martin Dupras" <martin.dupras at uwe.ac.uk>
> >>>
> >>>>Does anyone know how to generate band-limited square waves (or other
> >>>>classi waeforms) in PD?
> >>>>
> >>>>Thanks.
> >>>>
> >>>>- martin
> >>
> >>
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> >>
> >
> >
>
>