Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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mszlazak

Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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SND-MAXSAMP gives the maximum absolute value of a wave.
Is there a signed (not absolute) version of this function?
I'm dealing with waves of only negative valued amplitues so the maximum amplitude is the one closest to the axis. A absolute minimum function would work here as well.

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David R. Sky

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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Hi Dave,

How about 'fudging' such a function:

(defun my-snd-maxsamp (s)
(* -1.0 (snd-minsamp (mult -1.0 s)))
) ; end defun

Assuming of course that snd-minsamp actually exists. And this would only
be for your special case where the signal is at or below the zero axis.

David

--
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http://www.shellworld.net/~davidsky/


On Sat, 27 Jun 2009, [hidden email] wrote:

> SND-MAXSAMP gives the maximum absolute value of a wave.
> Is there a signed (not absolute) version of this function?
> I'm dealing with waves of only negative valued amplitues so the maximum amplitude is the one closest to the axis. A absolute minimum function would work here as well.
>

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Roger Dannenberg

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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Some javascript/style in this post has been disabled (why?)
You can subtract your signal from a large number so that everything becomes positive and your minimum become the maximum.

For related problems, you can also use snd-maxv to take only positive samples (or negative samples if you scale the input by -1) and then use snd-maxsamp to get only the most positive or most negative samples if the signal has both positive and negative samples. There's nothing in Nyquist now to take the minimum or maximum if you can't guarantee whether the value is positive or negative (but you can process the signal sample-by-sample, which is slow, but for such a simple calculation might be ok. Is should be substantially faster than real time).

-Roger

[hidden email] wrote:
SND-MAXSAMP gives the maximum absolute value of a wave.
Is there a signed (not absolute) version of this function?
I'm dealing with waves of only negative valued amplitues so the maximum amplitude is the one closest to the axis. A absolute minimum function would work here as well.

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mszlazak

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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Hi Roger,

BTW, I think I figure out most of that paper we were exchanging on last week. It followed the work of one of the heavily cited references and as I implemented things others started to become clear.

This led to this post and has to do with normalization of energy.

Energy is estimated using g(t) = 10* lg (SUM of n frames in a 30 msec window centered on t) and here lg is the common logarithm.
All my original audio signals s have absolute amplitudes less than 1 so the energy waveform of s is always negative.
The least negative amplitude is the maximum value.

At first glance, scaling by -1 seem right but isn't that a flip around the horizontal axis not a shift/slide up into positive ranges?
To to the shift, I found the absolute maximum then added it to the energy signal, shifting it into positive ranges and then I found the maximum again.
So a built in function would be nice.

This second maximum value is part of a normalization procedure which then shifts the original energy signal g so it's minimum negative value is at zero.
That's a part of the energy normalization they were referring to and had me scratching my head way the formula gave only zero and negative values.

Anyway, thanks again for all that help last week. It was much appreciated.

Mark.


-----Original Message-----
From: Roger Dannenberg <[hidden email]>
To: Discussion of developing Nyquist plug-ins for Audacity <[hidden email]>
Sent: Sat, Jun 27, 2009 7:59 pm
Subject: Re: [Audacity-nyquist] Is there a signed version of snd-maxsamp or absolute minimum function for sound?

You can subtract your signal from a large number so that everything becomes positive and your minimum become the maximum.

For related problems, you can also use snd-maxv to take only positive samples (or negative samples if you scale the input by -1) and then use snd-maxsamp to get only the most positive or most negative samples if the signal has both positive and negative samples. There's nothing in Nyquist now to take the minimum or maximum if you can't guarantee whether the value is positive or negative (but you can process the signal sample-by-sample, which is slow, but for such a simple calculation might be ok. Is should be substantially faster than real time).

-Roger

[hidden email] wrote:
SND-MAXSAMP gives the maximum absolute value of a wave.
Is there a signed (not absolute) version of this function?
I'm dealing with waves of only negative valued amplitues so the maximum amplitude is the one closest to the axis. A absolute minimum function would work here as well.


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edgar-rft

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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In reply to this post by mszlazak


[hidden email] schrieb:
> SND-MAXSAMP gives the maximum absolute value of a wave.
> Is there a signed (not absolute) version of this function?
> I'm dealing with waves of only negative valued amplitues so the maximum
> amplitude is the one closest to the axis. A absolute minimum function
> would work here as well.
>

You can "invert" a sound (make the negative samples become positive and
vice versa) by using Nyquist "scale" or "mult" with a factor of "-1".

- edgar




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mszlazak

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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Thank you but this doesn't work. See my response to Roger.


-----Original Message-----
From: [hidden email]
To: Discussion of developing Nyquist plug-ins for Audacity <[hidden email]>
Sent: Sat, Jun 27, 2009 9:02 pm
Subject: Re: [Audacity-nyquist] Is there a signed version of snd-maxsamp or absolute minimum function for sound?



[hidden email] schrieb:
> SND-MAXSAMP gives the maximum absolute value of a wave.
> Is there a signed (not absolute) version of this function?
> I'm dealing with waves of only negative valued amplitues so the maximum
> amplitude is the one closest to the axis. A absolute minimum function
> would work here as well.
>

You can "invert" a sound (make the negative samples become positive and
vice versa) by using Nyquist "scale" or "mult" with a factor of "-1".

- edgar




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edgar-rft

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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 > Edgar wrote in a former mail:
 > You can "invert" a sound (make the negative samples become positive
 > and vice versa) by using Nyquist "scale" or "mult" with a factor
 > of "-1".

 > Thank you but this doesn't work. See my response to Roger.

Sorry, I was cleaning up my mailbox and hadn't realized that Roger
already had answered before.

 > All my original audio signals s have absolute amplitudes
 > less than 1 so the energy waveform of s is always negative.

I assume you're talking about "normalized float" samples (like used
in Nyquist and Audacity) with maxPeak=1.0, minPeak=-1.0, what leads
to absolute peak values between min=0.0 and max=1.0, so lg(absPeak)
is always negative.

One of the first things I would try whould look like this:

(scale -1 (s-log (s-abs sound)))

 > That's a part of the energy normalization they were referring to
 > and had me scratching my head way the formula gave only zero and
 > negative values.

Who is "they"? Can you tell what document? Is it one of the
CCRMA/JOSIII papers? I wasn't able to find out by reading the
mails from last week.

In Nyquist you often have to "compute around the corner", so maybe
we need to re-arrange the formula, but therefore I would need the
complete context of the original paper.

Other ideas: In Nyquist there is e.g. "snd-avg", which can compute
the RMS-power of blocks of samples.

- edgar

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mszlazak

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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Hi Edgar.

The paper I mentioned is here:

http://figment.csee.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf

and it cites this paper a lot which gave me clues to what's going on:

http://lilabs.com/info/pdf/journal/asp-ept02.pdf

Thanks. Mark.


-----Original Message-----
From: [hidden email]
To: Discussion of developing Nyquist plug-ins for Audacity <[hidden email]>
Sent: Sun, Jun 28, 2009 1:05 am
Subject: Re: [Audacity-nyquist] Is there a signed version of snd-maxsamp or absolute minimum function for sound?

 > Edgar wrote in a former mail:
> You can "invert" a sound (make the negative samples become positive
> and vice versa) by using Nyquist "scale" or "mult" with a factor
> of "-1".

> Thank you but this doesn't work. See my response to Roger.

Sorry, I was cleaning up my mailbox and hadn't realized that Roger
already had answered before.

> All my original audio signals s have absolute amplitudes
> less than 1 so the energy waveform of s is always negative.

I assume you're talking about "normalized float" samples (like used
in Nyquist and Audacity) with maxPeak=1.0, minPeak=-1.0, what leads
to absolute peak values between min=0.0 and max=1.0, so lg(absPeak)
is always negative.

One of the first things I would try whould look like this:

(scale -1 (s-log (s-abs sound)))

> That's a part of the energy normalization they were referring to
> and had me scratching my head way the formula gave only zero and
> negative values.

Who is "they"? Can you tell what document? Is it one of the
CCRMA/JOSIII papers? I wasn't able to find out by reading the
mails from last week.

In Nyquist you often have to "compute around the corner", so maybe
we need to re-arrange the formula, but therefore I would need the
complete context of the original paper.

Other ideas: In Nyquist there is e.g. "snd-avg", which can compute
the RMS-power of blocks of samples.

- edgar

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mszlazak

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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In reply to this post by edgar-rft
BTW Edgar, below is what I have so far.
If I've got things right to now then I still need to add together normalized energy and normalized edge information.
After that would be the automaton part. Learning as I go.

Roger D. was very helpful and gave me a couple items that may make this all work with different energy calculations:
----------

"All these measures are related -- peak absolute value, average of squared signal, square root of average, etc. For a given type of sound (like speech) peak amplitude is generally a constant factor different from the RMS amplitude, and once you take logs, that constant factor becomes an offset, and whether you work with intensity (amplitude) or power (amplitude squared) is just a scale factor. The "normal" thing is use RMS because RMS squared is average power (energy / time), but it would not be too surprising to use some other related measure.
...
log(x) is monotonic: x > y > 0 => log(x) > log(y), so if you want the max(log(x)) you can take log(max(x)), provided x > 0."
----------

(defun cubic (frame window-size)
    (let    
        (
            (k (/ 27.0 16.0))       
            (i (round (* frame (/ 2581.0 (1- window-size)))))
        )
                (cond ((> i 1290) (setf i 1290)) ((< i -1290) (setf i -1290)))
        (* k (+ (* i i i) (* 2.0 i i) i))
    )
)
(defun energy (s window-size frame-size)
    (scale 4.34294481 (snd-log (snd-avg (mult s s) window-size frame-size OP-AVERAGE)))
)
(defun norm-g (g)
        (setf neg (scale -1 g))
    (setf shift (snd-maxsamp (snd-maxv neg (const 0.0 (snd-length neg 10000000)))))
    (if (> shift 0.0) (setf g (sum g shift)))
    (setf gmax (snd-maxsamp g))
    (scale (/ 20.0 gmax) (diff g gmax))
)
(defun make-filter (rate)
    (setf ar-f (make-array 15))
    (dotimes (i 15)
            (setf frame (- i 7))
            (setf (aref ar-f i) (cubic frame 15))
    )
    (snd-from-array 0 rate ar-f)
)
(setf g (energy s 240 80))
(setf gnorm (norm-g g))
(setf f (make-filter (snd-srate g)))
(setf edge (convolve g f))
(setf edgenorm (scale (/ 1.0 (snd-maxsamp edge)) edge))








-----Original Message-----
From: [hidden email]
To: Discussion of developing Nyquist plug-ins for Audacity <[hidden email]>
Sent: Sun, Jun 28, 2009 1:05 am
Subject: Re: [Audacity-nyquist] Is there a signed version of snd-maxsamp or absolute minimum function for sound?

 > Edgar wrote in a former mail:
> You can "invert" a sound (make the negative samples become positive
> and vice versa) by using Nyquist "scale" or "mult" with a factor
> of "-1".

> Thank you but this doesn't work. See my response to Roger.

Sorry, I was cleaning up my mailbox and hadn't realized that Roger
already had answered before.

> All my original audio signals s have absolute amplitudes
> less than 1 so the energy waveform of s is always negative.

I assume you're talking about "normalized float" samples (like used
in Nyquist and Audacity) with maxPeak=1.0, minPeak=-1.0, what leads
to absolute peak values between min=0.0 and max=1.0, so lg(absPeak)
is always negative.

One of the first things I would try whould look like this:

(scale -1 (s-log (s-abs sound)))

> That's a part of the energy normalization they were referring to
> and had me scratching my head way the formula gave only zero and
> negative values.

Who is "they"? Can you tell what document? Is it one of the
CCRMA/JOSIII papers? I wasn't able to find out by reading the
mails from last week.

In Nyquist you often have to "compute around the corner", so maybe
we need to re-arrange the formula, but therefore I would need the
complete context of the original paper.

Other ideas: In Nyquist there is e.g. "snd-avg", which can compute
the RMS-power of blocks of samples.

- edgar

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Roger Dannenberg

Re: Is there a signed version of snd-maxsamp or absolute minimum function for sound?

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 Is there a signed version of snd-maxsamp or absolute minimum function for
sound?

No. You can use snd-maxv and snd-minv to remove the negative or positive
part of a signal, but it only helps you find the minimum if you know the
minimum is negative. -Roger


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paul beach-2

sine of sine don't work

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Sine of a sine does not seem to work,
(hzosc (hzosc 11))

If something is moving in a frequency modulated field, it creates quite
a commotion. It seems difficult to set up a car engine in Nyquist.

http://www.climatehoax.ca/music/engine.mp3

Accelerating frequency,
sin((t*55)^1.25+ 2*sin(3*(t*55)^1.25))
Not in rads; starting radian frequency is 55/(2*pi)
--
  paul beach
  [hidden email]


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Roger Dannenberg

Re: sine of sine don't work

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What did you expect to get from this and what did you actually get?
hzosc(hzosc(f)) is not the same as sin(sin(t))

paul beach wrote:
> Sine of a sine does not seem to work,
> (hzosc (hzosc 11))
>
> If something is moving in a frequency modulated field, it creates quite
> a commotion. It seems difficult to set up a car engine in Nyquist.
>
>  

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paul beach-2

Re: sine of sine don't work

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I expect sin(sin t) look like another sine wave.
This:
(hzosc (hzosc f))
attenuates the amplitude in proportion to the frequency, at f = 11, the
amplitude is about 0.1 of full scale.
This: (osc (osc 11)) does not return audio.

Paul

On Sun, 28 Jun 2009 23:11:16 -0400, "Roger Dannenberg" <[hidden email]>
said:

> What did you expect to get from this and what did you actually get?
> hzosc(hzosc(f)) is not the same as sin(sin(t))
>
> paul beach wrote:
> > Sine of a sine does not seem to work,
> > (hzosc (hzosc 11))
> >
> > If something is moving in a frequency modulated field, it creates quite
> > a commotion. It seems difficult to set up a car engine in Nyquist.
> >
> >  
>
> ------------------------------------------------------------------------------
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  [hidden email]


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Roger Dannenberg

Re: sine of sine don't work

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paul beach wrote:
> I expect sin(sin t) look like another sine wave.
>  
But we know from FM theory that the result has sidebands, so it cannot
be a sine wave.
> This:
> (hzosc (hzosc f))
> attenuates the amplitude in proportion to the frequency, at f = 11, the
> amplitude is about 0.1 of full scale.
>  
That all sounds right to me -- the instantaneous frequency of the
"carrier" never exceeds one (the maximum value of the input), so the
phase of the outer hzosc is going to wobble around zero.
> This: (osc (osc 11)) does not return audio.
>  
The first argument to OSC is a scalar step number, so this doesn't
return anything, it generates an error.

-Roger



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paul beach-2

Equalizer

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I was asked about equalizer and tried a 20khz signal. The spectrum drops
about 30dB. Is it just my old Windows computer + sound card, that does
that? To be expected since capacitive reactance is inversely propotional
to frequency?
--
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  [hidden email]


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David R. Sky

Re: Equalizer

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Hi Paul,

You didn't specify any of several parameters that can be 'fed' to a
Nyquist equalizer function.

There's at least one on my plug-ins page, a zip file with .ny plug and
.txt help file, go to this section:

http://www.shellworld.net/~davidsky/plug-ins.htm#misc

There's at least one plug which uses eq-band if I remember correctly, you
need to specify center frequency, width of band in octaves, amount to
boost or attenuate, and perhaps one or two other argument fields.

David

--
David R. Sky
http://www.shellworld.net/~davidsky/


On Fri, 3 Jul 2009, paul beach wrote:

> I was asked about equalizer and tried a 20khz signal. The spectrum drops
> about 30dB. Is it just my old Windows computer + sound card, that does
> that? To be expected since capacitive reactance is inversely propotional
> to frequency?
>

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paul beach-2

Re: requency response

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I was checking the frequency response of the sound card. It should fail
somwhere at greater than 20khz because of Shanon's theorem. David's Buzz
generator was used, with frequencies set from 1000 to 21000 in steps of
a thousand.

http://www.shellworld.net/~davidsky/buzz.zip

Amplitude is set at .5
My respone is flat up to 19 khz and then takes off after that. It may be
that some sound cards trap or equalize so that it does not alias and
create a hash around 10 khz.


On Fri, 03 Jul 2009 21:28:58 -0700, "paul beach"
<[hidden email]> said:

> I was asked about equalizer and tried a 20khz signal. The spectrum drops
> about 30dB. Is it just my old Windows computer + sound card, that does
> that? To be expected since capacitive reactance is inversely propotional
> to frequency?
> --
>   paul beach
>   [hidden email]
>
>
> ------------------------------------------------------------------------------
> _______________________________________________
> Audacity-nyquist mailing list
> [hidden email]
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--
  paul beach
  [hidden email]


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David R. Sky

Re: requency response

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Hi Paul,

That buzz.ny file is a simple 'wrap' around the Nyquist buzz function and
should only produce the number of overtones you specify in the plug-in.
For example, frequency 440 hz with 4 overtones should result with the
following frequencies returned: 440 880 1320 1760 and possibly also 2200 .

Also, I wrote that plug when there was an error in the buzz function
which Dominic Mazzoni had pointed out to me, so I'd put his corrected code
in the plug, some time back in Audacity 1.2.x. Something might have
changed since then.

What version of Audacity are you using?

David

--
David R. Sky
http://www.shellworld.net/~davidsky/


On Sat, 4 Jul 2009, paul beach wrote:

> I was checking the frequency response of the sound card. It should fail
> somwhere at greater than 20khz because of Shanon's theorem. David's Buzz
> generator was used, with frequencies set from 1000 to 21000 in steps of
> a thousand.
>
> http://www.shellworld.net/~davidsky/buzz.zip
>
> Amplitude is set at .5
> My respone is flat up to 19 khz and then takes off after that. It may be
> that some sound cards trap or equalize so that it does not alias and
> create a hash around 10 khz.
>
>
> On Fri, 03 Jul 2009 21:28:58 -0700, "paul beach"
> <[hidden email]> said:
>> I was asked about equalizer and tried a 20khz signal. The spectrum drops
>> about 30dB. Is it just my old Windows computer + sound card, that does
>> that? To be expected since capacitive reactance is inversely propotional
>> to frequency?
>> --
>>   paul beach
>>   [hidden email]
>>
>>
>> ------------------------------------------------------------------------------
>> _______________________________________________
>> Audacity-nyquist mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/audacity-nyquist
>

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paul beach-2

Re: requency response

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Hi,

Buzz worked without a problem, I used 11 overtones. This seems quite
useful to check the bandpass of a complex filter, where there is
generally no way to do a numerical evaluation.

Paul

On Sat, 4 Jul 2009 10:34:44 -0700 (PDT), "David R. Sky"
<[hidden email]> said:

> Hi Paul,
>
> That buzz.ny file is a simple 'wrap' around the Nyquist buzz function and
> should only produce the number of overtones you specify in the plug-in.
> For example, frequency 440 hz with 4 overtones should result with the
> following frequencies returned: 440 880 1320 1760 and possibly also 2200
> .
>
> Also, I wrote that plug when there was an error in the buzz function
> which Dominic Mazzoni had pointed out to me, so I'd put his corrected
> code
> in the plug, some time back in Audacity 1.2.x. Something might have
> changed since then.
>
> What version of Audacity are you using?
>
> David
>
> --
> David R. Sky
> http://www.shellworld.net/~davidsky/
>
>
> On Sat, 4 Jul 2009, paul beach wrote:
>
> > I was checking the frequency response of the sound card. It should fail
> > somwhere at greater than 20khz because of Shanon's theorem. David's Buzz
> > generator was used, with frequencies set from 1000 to 21000 in steps of
> > a thousand.
> >
> > http://www.shellworld.net/~davidsky/buzz.zip
> >
> > Amplitude is set at .5
> > My respone is flat up to 19 khz and then takes off after that. It may be
> > that some sound cards trap or equalize so that it does not alias and
> > create a hash around 10 khz.
> >
> >
> > On Fri, 03 Jul 2009 21:28:58 -0700, "paul beach"
> > <[hidden email]> said:
> >> I was asked about equalizer and tried a 20khz signal. The spectrum drops
> >> about 30dB. Is it just my old Windows computer + sound card, that does
> >> that? To be expected since capacitive reactance is inversely propotional
> >> to frequency?
> >> --
> >>   paul beach
> >>   [hidden email]
> >>
> >>
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> >
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David R. Sky

Re: requency response

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Then what did you mean in your first post by "going off the scale"? it
sounded like you were complaining about something that was wrong with
buzz.ny.

Please be specific!

Thanks

David

--
David R. Sky
http://www.shellworld.net/~davidsky/


On Sat, 4 Jul 2009, paul beach wrote:

> Hi,
>
> Buzz worked without a problem, I used 11 overtones. This seems quite
> useful to check the bandpass of a complex filter, where there is
> generally no way to do a numerical evaluation.
>
> Paul
>
> On Sat, 4 Jul 2009 10:34:44 -0700 (PDT), "David R. Sky"
> <[hidden email]> said:
>> Hi Paul,
>>
>> That buzz.ny file is a simple 'wrap' around the Nyquist buzz function and
>> should only produce the number of overtones you specify in the plug-in.
>> For example, frequency 440 hz with 4 overtones should result with the
>> following frequencies returned: 440 880 1320 1760 and possibly also 2200
>> .
>>
>> Also, I wrote that plug when there was an error in the buzz function
>> which Dominic Mazzoni had pointed out to me, so I'd put his corrected
>> code
>> in the plug, some time back in Audacity 1.2.x. Something might have
>> changed since then.
>>
>> What version of Audacity are you using?
>>
>> David
>>
>> --
>> David R. Sky
>> http://www.shellworld.net/~davidsky/
>>
>>
>> On Sat, 4 Jul 2009, paul beach wrote:
>>
>>> I was checking the frequency response of the sound card. It should fail
>>> somwhere at greater than 20khz because of Shanon's theorem. David's Buzz
>>> generator was used, with frequencies set from 1000 to 21000 in steps of
>>> a thousand.
>>>
>>> http://www.shellworld.net/~davidsky/buzz.zip
>>>
>>> Amplitude is set at .5
>>> My respone is flat up to 19 khz and then takes off after that. It may be
>>> that some sound cards trap or equalize so that it does not alias and
>>> create a hash around 10 khz.
>>>
>>>
>>> On Fri, 03 Jul 2009 21:28:58 -0700, "paul beach"
>>> <[hidden email]> said:
>>>> I was asked about equalizer and tried a 20khz signal. The spectrum drops
>>>> about 30dB. Is it just my old Windows computer + sound card, that does
>>>> that? To be expected since capacitive reactance is inversely propotional
>>>> to frequency?
>>>> --
>>>>   paul beach
>>>>   [hidden email]
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>> _______________________________________________
>>>> Audacity-nyquist mailing list
>>>> [hidden email]
>>>> https://lists.sourceforge.net/lists/listinfo/audacity-nyquist
>>>
>>
>> ------------------------------------------------------------------------------
>> _______________________________________________
>> Audacity-nyquist mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/audacity-nyquist
>

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