com> wrote:
>
>
> Original Message ----
> Subject: Re: [music-dsp] Sampling theory "best" explanation
> From: "Ethan Duni" <ethan.d...@gmail.com>
> Date: Wed, September 6, 2017 4:49 pm
> To
So when is a "system", or better put, it's mathematical model, a linear system ?
In a way the scalar high school definition suffices, when the inputs are added, the result
must be the same as when the inputs are processed separately, and a mulitplied input gives
a multiplied output. Simple
It is a most fascinating thread. The more one looks into it, the more
one has to marvel that the process works at all.
Richard Dobson
On 07/09/2017 07:16, Nigel Redmon wrote:
Somehow, combining the term "rat's ass" with a clear and concise
explanation of your viewpoint makes it especially
- Original Message ----------------
> Subject: Re: [music-dsp] Sampling theory "best" explanation
> From: "Ethan Duni" <ethan.d...@gmail.com>
> Date: Wed, September 6, 2017 4:49 pm
> To: "robert bristow-johnson" <r...@audioimagi
Original Message
Subject: Re: [music-dsp] Sampling theory "best" explanation
From: "Ethan Duni" <ethan.d...@gmail.com>
Date: Wed, September 6, 2017 4:49 pm
To: "robert bristow-johnson" <r.
Okay, no big deal. It's easy to come off the wrong way in complicated, fast
moving email threads.
Ethan D
On Wed, Sep 6, 2017 at 6:37 PM, Nigel Redmon wrote:
> Ethan, I wasn't taking a swipe at you, by any stretch. In fact, I wasn't
> even addressing your ADC comment. It
Of course I mean that we store a representation of an impulse. I've said many
times that the sample values "represent" impulses.
> On Sep 7, 2017, at 5:34 AM, Ethan Duni wrote:
>
> >For ADC, we effectively measure an instantaneous voltage and store it as an
> >impulse.
>
Nigel Redmon wrote:
>As an electrical engineer, we find great humor when people say we can't do
impulses.
I'm the electrical engineer who pointed out that impulses don't exist and
are not found in actual ADCs. If you have some issue with anything I've
posted, I'll thank you to address it to me
sion-making purposes it clearly
is, it seem you've defeated the purpose.
> On Sep 5, 2017, at 11:57 PM, robert bristow-johnson
> <r...@audioimagination.com> wrote:
>
>
>
> Original Message ------------
> Subject: Re: [music
Original Message
Subject: Re: [music-dsp] Sampling theory "best" explanation
From: "Nigel Redmon" <earle...@earlevel.com>
Date: Tue, September 5, 2017 4:05 am
To: mu
Original Message
Subject: Re: [music-dsp] Sampling theory "best" explanation
From: "Ethan Duni" <ethan.d...@gmail.com>
Date: Tue, September 5, 2017 1:07 am
To: "A discussion list fo
>
> If not, then the phrase "resampling is LTI" - without some kind of "ideal"
> qualifier - seems misleading. If it's LTI then what are all these aliases
> doing in my outputs?
>
Yeah, I think you had it right when you pointed out that the existence of
aliasing shows that resampling is not LTI if
As an electrical engineer, we find great humor when people say we can't do
impulses. What constitutes an impulse depends on the context—nano seconds,
milliseconds...
For ADC, we effectively measure an instantaneous voltage and store it as an
impulse. Arguing that we don't really do
rbj wrote:
>1. resampling is LTI **if**, for the TI portion, one appropriately scales
time.
Have we established that this holds for non-ideal resampling? It doesn't
seem like it does, in general.
If not, then the phrase "resampling is LTI" - without some kind of "ideal"
qualifier - seems
> The fact that 5,17,-12,2 at sample rate 1X and
> 5,0,0,0,17,0,0,0,-12,0,0,0,2,0,0,0 at sample rate 4X are identical is obvious
> only for samples representing impulses.
>
> I agree that the zero-stuff-then-lowpass technique is much more obvious when
> we you consider the impulse train
OTOH, just about everything we do with digital audio doesn’t exactly work.
Start with sampling. Do we give up if we can’t ensure absolutely no signal at
and above half the sample rate? Fortunately, our ears have limitations (whew!).
;-) Anyway, the aliasing occurred to me as I wrote that, but
ASRC chip uses), that means
(taking advantage of symmetry) 8K coefficients needed in a table, 64 MAC
instructions, and one linear interpolation per output sample. �doesn't matter
what the sample-rate conversion ratio is (as long as we don't worry about
aliasing in downsampling).
�
bestest,
r
b-j
�
-
>
> First, I want to be clear that I don’t think people are crippled by
> certain viewpoint—I’ve said this elsewhere before, maybe not it this thread
> or the article so much.
In that case I'd suggest some more editing is in order, since the article
stated this pretty overtly at least a couple
>
> Time variance is a bit subtle in the multi-rate context. For integer
> downsampling, as you point out, it might make more sense to replace the
> classic n-shift-in/n-shift-out definition of time invariance with one that
> works in terms of the common real time represented by the different
>
Hmm this is quite a few discussions of LTI with respect to resampling that
have gone badly on the list over the years...
Time variance is a bit subtle in the multi-rate context. For integer
downsampling, as you point out, it might make more sense to replace the
classic n-shift-in/n-shift-out
Interesting comments, Ethan.
Somewhat related to your points, I also had a situation on this board years ago
where I said that sample rate conversion was LTI. It was a specific context,
regarding downsampling, so a number of people, one by one, basically quoted
back the reason I was wrong.
Ethan F wrote:
>I see your nitpick and raise you. :o) Surely there are uncountably many
such functions,
>as the power at any apparent frequency can be distributed arbitrarily
among the bands.
Ah, good point. Uncountable it is!
Nigel R wrote:
>But I think there are good reasons to understand the
Hi Ethan,
Good comments and questions…I’m going to have to skip the questions for now
(I’m in a race against time the next few days, then will been off the grid,
relatively speaking, for a couple of weeks—but I didn’t want to seem like I was
ignoring your reply; I think any quick answers to
>
> This needs an additional qualifier, something about the bandlimited
> function with the lowest possible bandwidth, or containing DC, or
> "baseband," or such.
Yes, by bandlimited here I mean bandlimited to [-Nyquist, Nyquist].
Otherwise, there are a countably infinite number of bandlimited
>I'm one of those people who prefer to think of a discrete-time signal as
>representing the unique bandlimited function interpolating its samples.
This needs an additional qualifier, something about the bandlimited
function with the lowest possible bandwidth, or containing DC, or
"baseband," or
Thanks for posting this! It's always interesting to get such a good glimpse
at someone else's mental model.
I'm one of those people who prefer to think of a discrete-time signal as
representing the unique bandlimited function interpolating its samples. And
I don't think this point of view has
Hi Remy,
> On Aug 28, 2017, at 2:16 AM, Remy Muller wrote:
>
> I second Sampo about giving some more hints about Hilbert spaces,
> shift-invariance, Riesz representation theorem… etc
I think you’ve hit upon precisely what my blog isn’t, and why it exists at all.
;-)
>
Nigel Redmon wrote:
Well, it’s quiet here, why not…
Right, a good reiteration never hurts! I quickly read through and find your explanation
fine, it's not right to expect everybody to be theoretically sound and solid up to the
level of mathematical proof, but I'm a strong proponent of
I second Sampo about giving some more hints about Hilbert spaces,
shift-invariance, Riesz representation theorem... etc
Correct me if you said it somewhere and I didn't saw it, but an
important /implicit/ assumption in your explanation is that you are
talking about "uniform bandlimited
>
> r b-j
>
>
>
> Original Message
> Subject: Re: [music-dsp] Sampling theory "best" explanation
> From: "Sampo Syreeni" <de...@iki.fi>
> Date: Sun, August 27, 2017 2:20 am
> To: &quo
Sampo, the purpose was to convince people that samples are impulses, and why
that means the spectrum represented by a series of samples is the intended
spectrum plus aliased images, forever. in the simplest, most intuitive way I
could think of.
That’s why I put those points up front, before
Well, it’s a DSP blog. The intended audience is whoever reads it, I’m not
judgmental. So, the question is probably more like “who can benefit from it”.
At the novice end, I’d say they can probably benefit at least from the
revelation that it comes from solving issues in analog communication,
Le 2017-08-26 03:21, Nigel Redmon a écrit :
http://www.earlevel.com/main/tag/sampling-theory-series/?order=asc
Hi Nigel,
For me the best sampling theory explanation I ever saw is probably also
one of the oldest (1980!)
This explanation can be found in the second chapter of the 2920
On 2017-08-25, Nigel Redmon wrote:
http://www.earlevel.com/main/tag/sampling-theory-series/?order=asc
Personally I'd make it much simpler at the top. Just tell them sampling
is what it is: taking an instantaneous value of a signal at regular
intervals. Then tell them that is all it takes to
Please check out my new series on sampling theory, and feel free to comment
here or there. The goal was to be brief, but thorough, and avoid abstract
mathematical explanations. In other words, accurate enough that you can
deduce correct calculations from it, but intuitive enough for the math-shy.
thank you very much !
2017-08-26 4:21 GMT+03:00 Nigel Redmon :
> Well, it’s quiet here, why not…
>
> Please check out my new series on sampling theory, and feel free to
> comment here or there. The goal was to be brief, but thorough, and
> avoid abstract mathematical
This is neat, thanks for sharing Nigel
On Aug 25, 2017 6:22 PM, "Nigel Redmon" wrote:
> Well, it’s quiet here, why not…
>
> Please check out my new series on sampling theory, and feel free to
> comment here or there. The goal was to be brief, but thorough, and
> avoid
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