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Basic Sampling Theory Question

Started by old_ee August 17, 2015
On 8/25/15 8:23 AM, Bob Masta wrote:
> On Mon, 24 Aug 2015 15:53:49 -0400, robert bristow-johnson > <rbj@audioimagination.com> wrote: > >> i wonder how many people use a soundcard to do a controller for a >> real-time control app? >> >> just wondering. sometimes people use technology for a different >> application field to do stuff. like audio people using a graphic >> accelerator to do massively computational real-time audio stuff. (i can >> sorta see it for reverberation or maybe de-reverberation.) > > One big limitation with sound cards as controllers is that > they are all AC coupled. (The ones that claim "DC coupled" > really mean they are using servo-controlled stages that take > the input capacitor out of the signal path, but it's still > there in the feedback integrator giving a high-pass > response.)
well, the old Turtle Beach Tahiti and Fiji cards were DC coupled. i dunno what they used for an ADC. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
glen herrmannsfeldt  <gah@ugcs.caltech.edu> wrote:

>Tim Wescott <seemywebsite@myfooter.really> wrote:
>> On Fri, 21 Aug 2015 22:43:54 +0000, Steve Pope wrote: >>> This is why you need (at least slightly) more than two samples per >>> period.
>> And how much that "at least slightly" is depends on a lot of other >> factors.
>For real Nyquist sampling, you need an infinite number of points >(or periodic boundary conditions), in which case slightly is >very slight.
>But in reality, we don't have an infinite number of points.
Intuitively it seems that what's (informally) called the "coherence interval" -- the interval over which time the signal is short-term stationary -- is going to figure into this, more than the actual lenght of the input sample (which could be very long, and could suggest that one could sample very close to Nyquist). Steve
On 8/21/2015 3:14 AM, Evgeny Filatov wrote:
> On 19.08.2015 23:51, Eric Jacobsen wrote: >> On Wed, 19 Aug 2015 22:52:52 +0300, Evgeny Filatov >> <e.v.filatov@ieee.org> wrote: >> >>> On 18.08.2015 21:29, gyansorova@gmail.com wrote: >>> (snip) >>> >>>> Just a small correction, sampling theory was not due to Nyquist and >>>> Shannon but Whittaker and a Russian gentleman. Shannon just put it >>>> in an engineering framework. >>>> >>> >>> That gentleman's name is Kotelnikov. >>> >>> http://en.wikipedia.org/wiki/Vladimir_Kotelnikov >>> >>> Evgeny. >>> >> >> It's sad that the cold war kept a lot of science and engineering >> developments of the two sides separated for so long. I hope that >> doesn't ever happen again. >> >> >> Eric Jacobsen >> Anchor Hill Communications >> http://www.anchorhill.com >> > > Actually one of the reasons I like this place is that it's apolitical. > > A decade ago, when I was a college student, once we've got the attention > of a "telephone terrorist". There was a phone call to the police > (falsely) claiming there's a bomb somewhere in the college. All classes > were cancelled. Except for, guess what? A lecture in calculus, which > went on like if nothing extraordinary had happened. > > Imho, that tells something about the relative importance of math and > politics.
I think it says something about the lack of perception of danger by mathematicians. I know any number of students who would have bombed the math building if they could have gotten the materials... and knew enough chemistry. -- Rick
On Thu, 27 Aug 2015 04:54:51 +0000, Steve Pope wrote:

> glen herrmannsfeldt <gah@ugcs.caltech.edu> wrote: > >>Tim Wescott <seemywebsite@myfooter.really> wrote: > >>> On Fri, 21 Aug 2015 22:43:54 +0000, Steve Pope wrote: >>>> This is why you need (at least slightly) more than two samples per >>>> period. > >>> And how much that "at least slightly" is depends on a lot of other >>> factors. > >>For real Nyquist sampling, you need an infinite number of points (or >>periodic boundary conditions), in which case slightly is very slight. > >>But in reality, we don't have an infinite number of points. > > Intuitively it seems that what's (informally) called the "coherence > interval" -- the interval over which time the signal is short-term > stationary -- is going to figure into this, more than the actual lenght > of the input sample (which could be very long, and could suggest that > one could sample very close to Nyquist).
That may apply. For setting sampling rates in practice, though, you can make decisions based on the amount of aliased signal that gets through. You know what filters you're going to use, so if you have a good idea of what the input signal is then you can calculate what's going to get sampled, and decide if the amount of aliasing is acceptable. Because in the real world, you're _always_ going to have some amount of aliasing! -- Tim Wescott Wescott Design Services http://www.wescottdesign.com
robert bristow-johnson  <rbj@audioimagination.com> wrote:

>On 8/24/15 1:33 AM, Tim Wescott wrote:
>> On Sat, 22 Aug 2015 05:00:37 +0000, glen herrmannsfeldt wrote: >> For signal processing I'm pretty sure that the two biggest issues are how >> much fewer than an infinite amount of points you have,
>i thought that modeling zero-padding a brick-wall LPF FIR definitively >models the fewer. (then in our analysis we can pretend we have an >infinite number of points when we Fourier transform stuff.)
>> and how much phase distortion you can stand (they're probably related).
>dunno for sure about what is "phase distortion", but for me the tradeoff >is the number of coefs in the polyphase FIR vs how much aliasing >distortion you can stand.
Note that even a linear-phase FIR has a non-zero delay spread so this can be considered a form of phase distortion. More coefficients, so as to make the filter more brick-wall, generally implies a longer normalized delay spread. Steve
> >Note that even a linear-phase FIR has a non-zero delay spread >so this can be considered a form of phase distortion. > >More coefficients, so as to make the filter more brick-wall, generally >implies a longer normalized delay spread. > >Steve
fixed delay (whether piece of wire,space,fir) can't be distortion. That is the whole point about linear phase. Kaz --------------------------------------- Posted through http://www.DSPRelated.com
On Tuesday, August 25, 2015 at 1:24:11 PM UTC+1, Bob Masta wrote:

> One big limitation with sound cards as controllers is that > they are all AC coupled. (The ones that claim "DC coupled" > really mean they are using servo-controlled stages that take > the input capacitor out of the signal path, but it's still > there in the feedback integrator giving a high-pass > response.) > > That fact means custom modifications or extra custom-built > hardware are needed to get DC into and out of the card, > which limits a lot of the interesting controller > applications. (I have a couple of DC input circuits at > <http://www.daqarta.com/dw_ggmm.htm>, and DC pulse output > circuits at <http://www.daqarta.com/dw_gg0o.htm>.) >
There's a handy list of DC coupled sound cards here: http://www.expert-sleepers.co.uk/siwacompatibility.html You can use them for controlling your sprawling eurocrack modular synth...
>On Tuesday, August 18, 2015 at 5:26:05 AM UTC+12, Tim Wescott wrote: >> On Mon, 17 Aug 2015 05:29:31 -0500, old_ee wrote: >>
> >Just a small correction, sampling theory was not due to Nyquist and
Shannon
>but Whittaker and a Russian gentleman. Shannon just put it in an
engineering
>framework.
Hi, The Russian's name was Vladimir Kotelnikov. There was also a Japanese mathematician/scientist (darn, I forgot his name) that was trying to make sense out of the notion of "sampling" around the same time as those other pioneers. [-Rick-] --------------------------------------- Posted through http://www.DSPRelated.com