I've had a look at dspguru.com and am trying to implement a program that will 
resample an audio signal from 44.1kHz to 48kHz. I'm running into some problems 
- namely too many zeros in my converted file. My algorithm is basically:


Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159 zeros 
in order to oversample.

Apply a low pass filter with a stop band freq. of 24kHz

Downsample by keeping every 1 out of 147 samples

Do I need to LPF again?


Another thing that I'm not sure about is if there is any requirement on the 
number of filter taps. Obviously the more taps I have, the fewer zero-valued 
samples will be in my output file.

Does anyone have any suggestions to implementing this SRC correctly?

Newbie wrote:
> 

<snip>

> Does anyone have any suggestions to implementing this SRC 
> correctly?

You can do all the hard work yourself and probably not get
it right or you can use a library which is known to be 
correct:

    http://www.mega-nerd.com/SRC/

Erik
-- 
+-----------------------------------------------------------+
  Erik de Castro Lopo  nospam@mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
Java : A language from the C family of languages. It has all the bad
features of C++ but without the most powerful feature of C, pointers.

Newbie wrote:

> Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159 zeros 
> in order to oversample.
> 
> Apply a low pass filter with a stop band freq. of 24kHz

24KHz? Shouldn't be 22.05KHz?

bye,

-- 
   Piergiorgio Sartor

none@available.com (Newbie) writes:

> I've had a look at dspguru.com and am trying to implement a program that will 
> resample an audio signal from 44.1kHz to 48kHz. I'm running into some problems 
> - namely too many zeros in my converted file. My algorithm is basically:
>
>
> Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159 zeros 
> in order to oversample.

OK, so you're upsampling by a factor of 160 to a sample rate of 160 * 44100 = 7.056 MHz.

> Apply a low pass filter with a stop band freq. of 24kHz

No. The typical upconverter consists of upsampling by a factor of N
(inserting N-1 zeros) followed by a 1/N interpolating (lowpass)
filter. The typical downconverter consists of a 1/M lowpass filter
followed by a M:1 decimator (keeping 1 of M samples).

You can combine the two filters (1/N and 1/M) by simply choosing the lowest. 
In this case N = 160 and M = 147, so 1/N is the lowest. Thus you want to filter
by 1/N (relative to a sample rate of 7.056 MHz), which corresponds to a filter
of bandwidth (7.056 MHz/2) / 160 = 22.05 kHz. 

> Downsample by keeping every 1 out of 147 samples
>
> Do I need to LPF again?

Not if you did it right. 

> Another thing that I'm not sure about is if there is any requirement on the 
> number of filter taps. Obviously the more taps I have, the fewer zero-valued 
> samples will be in my output file.

Aha. Well, yes there is a requirement, but it's not black-and-white. Let's
just take the upsampling stage for a moment and think about what's happening
in the frequency domain. The original 44.1 kHz signal has a repetitive spectrum
out to infinity, repeating at multiples of 44.1 kHz. After you upsample by 160 
by inserting 159 zeros, the resulting spectrum hasn't changed. However, since
your sample rate is now 7.056 MHz, the stuff between 22.05 kHz and 7.056/2 MHz = 3.528
MHz has to be filtered out. If that filter isn't perfect, then, at the 7.056 MHz
rate, the stuff above 22.05 kHz that leaks through will sound like garbage (if 
you could hear it). So the object is to have you're 1/160 lowpass filter be
down as far as reasonable at 22.05 kHz. For high quality audio, it should
probably be at least 90 dB or so down. Note that this means that you will
necessarily have the cutoff frequency at something less than 22.05 kHz. The
closer you get the cutoff frequency to 22.05 kHz, and the more you want the
filter attenuation down at 22.05 kHz, the longer your filter will get. 

Another simple point: Obviously your filter was less than 161 taps long, 
hence you are getting the zeros you spoke of. That's a bad sign that
you probably haven't chosen the passband/stopband characteristics properly.
How did you design the filter?

> Does anyone have any suggestions to implementing this SRC correctly?

Erik has a good point - use his. Unless your goal is the learning 
experience.
-- 
%  Randy Yates                  % "Though you ride on the wheels of tomorrow,
%% Fuquay-Varina, NC            %  you still wander the fields of your
%%% 919-577-9882                %  sorrow."
%%%% <yates@ieee.org>           % '21st Century Man', *Time*, ELO
http://home.earthlink.net/~yatescr

You probably don't want to worry about this now since you are still trying
to get the basic operation correct, but you may notice some major
inefficiencies in your process.  In your filter, only 1 out of every 160
samples are non-zero.  So there's no reason to include them in the filtering
step since zero*filter coef = 0.  Then, you may notice that you are
calculating all these filtered values, and then throwing away 146 out of 147
of them.  So there is no need to calculate the 146 values that you are just
going to throw away anyway.  Once you take advantage of these 2 shortcuts,
you are left with the so-called polyphase implementation.

"Newbie" <none@available.com> wrote in message
news:tUHRb.10643$5G3.8881@newssvr22.news.prodigy.com...
> I've had a look at dspguru.com and am trying to implement a program that
will
> resample an audio signal from 44.1kHz to 48kHz. I'm running into some
problems
> - namely too many zeros in my converted file. My algorithm is basically:
>
> > Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159
zeros
> in order to oversample.
>
> Apply a low pass filter with a stop band freq. of 24kHz
>
> Downsample by keeping every 1 out of 147 samples
>
> Do I need to LPF again?
>
> Another thing that I'm not sure about is if there is any requirement on
the
> number of filter taps. Obviously the more taps I have, the fewer
zero-valued
> samples will be in my output file.
>
> Does anyone have any suggestions to implementing this SRC correctly?

In article <llnsuxgv.fsf@ieee.org>, yates@ieee.org says...
>
>none@available.com (Newbie) writes:
>
>> I've had a look at dspguru.com and am trying to implement a program that 
will 
>> resample an audio signal from 44.1kHz to 48kHz. I'm running into some 
problems 
>> - namely too many zeros in my converted file. My algorithm is basically:
>>
>>
>> Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159 
zeros 
>> in order to oversample.
>
>OK, so you're upsampling by a factor of 160 to a sample rate of 160 * 44100 = 
7.056 MHz.
>
>> Apply a low pass filter with a stop band freq. of 24kHz
>
>No. The typical upconverter consists of upsampling by a factor of N
>(inserting N-1 zeros) followed by a 1/N interpolating (lowpass)
>filter. The typical downconverter consists of a 1/M lowpass filter
>followed by a M:1 decimator (keeping 1 of M samples).
>
>You can combine the two filters (1/N and 1/M) by simply choosing the lowest. 
>In this case N = 160 and M = 147, so 1/N is the lowest. Thus you want to 
filter
>by 1/N (relative to a sample rate of 7.056 MHz), which corresponds to a 
filter
>of bandwidth (7.056 MHz/2) / 160 = 22.05 kHz. 
>
>> Downsample by keeping every 1 out of 147 samples
>>
>> Do I need to LPF again?
>
>Not if you did it right. 
>
>> Another thing that I'm not sure about is if there is any requirement on the 
>> number of filter taps. Obviously the more taps I have, the fewer 
zero-valued 
>> samples will be in my output file.
>
>Aha. Well, yes there is a requirement, but it's not black-and-white. Let's
>just take the upsampling stage for a moment and think about what's happening
>in the frequency domain. The original 44.1 kHz signal has a repetitive 
spectrum
>out to infinity, repeating at multiples of 44.1 kHz. After you upsample by 
160 
>by inserting 159 zeros, the resulting spectrum hasn't changed. However, since
>your sample rate is now 7.056 MHz, the stuff between 22.05 kHz and 7.056/2 
MHz = 3.528
>MHz has to be filtered out. If that filter isn't perfect, then, at the 7.056 
MHz
>rate, the stuff above 22.05 kHz that leaks through will sound like garbage 
(if 
>you could hear it). So the object is to have you're 1/160 lowpass filter be
>down as far as reasonable at 22.05 kHz. For high quality audio, it should
>probably be at least 90 dB or so down. Note that this means that you will
>necessarily have the cutoff frequency at something less than 22.05 kHz. The
>closer you get the cutoff frequency to 22.05 kHz, and the more you want the
>filter attenuation down at 22.05 kHz, the longer your filter will get. 
>
>Another simple point: Obviously your filter was less than 161 taps long, 
>hence you are getting the zeros you spoke of. That's a bad sign that
>you probably haven't chosen the passband/stopband characteristics properly.
>How did you design the filter?




Thanks for all of the feedback. I think that you are correct about my filter 
not being designed correctly. 

I used ScopeFIR and chose Fs = 7056000, passband upper freq = 20kHz, 
stopband lower freq. = 22.05kHz, 1 dB passband ripple, and 100 dB stopband 
attenuation. I'm not sure what is going on here since the freq. response plot 
shows the signal to be attenuated ~-75dB at 0 Hz. Also, I've got several wide 
lobes above the cutoff freq. The lobes can be shorter if I increase the number 
of filter taps, but then I also get more lobes.

Please suggest a filter for me to use, including the number of taps. If the 
number can be < 32 that would be good as my program doesn't support any 
higher.

Lastly, I don't want to use someone elses program as I'm doing this for my own 
education. In separate projects, I've interpolated audio from 22.05kHz to 
44.1kHz and in another decimated from 48kHz down to 24kHz - these worked fine. 
The 44.1kHz to 48kHz resampling project so far has proven to be a little more 
difficult, namely, I believe because of the large interpolation and decimation 
factors and long filters required.


Thx.





>
>> Does anyone have any suggestions to implementing this SRC correctly?
>
>Erik has a good point - use his. Unless your goal is the learning 
>experience.
>-- 
>%  Randy Yates                  % "Though you ride on the wheels of tomorrow,
>%% Fuquay-Varina, NC            %  you still wander the fields of your
>%%% 919-577-9882                %  sorrow."
>%%%% <yates@ieee.org>           % '21st Century Man', *Time*, ELO
>http://home.earthlink.net/~yatescr

In article <bv8u8i$pj8dg$1@ID-210375.news.uni-berlin.de>, 
goldentully@hotmail.com says...
>
>You probably don't want to worry about this now since you are still trying
>to get the basic operation correct, but you may notice some major
>inefficiencies in your process.  In your filter, only 1 out of every 160
>samples are non-zero.  So there's no reason to include them in the filtering
>step since zero*filter coef = 0.  Then, you may notice that you are
>calculating all these filtered values, and then throwing away 146 out of 147
>of them.  So there is no need to calculate the 146 values that you are just
>going to throw away anyway.  Once you take advantage of these 2 shortcuts,
>you are left with the so-called polyphase implementation.
>


Yes, I've made provisions not to multiply zero-valued samples by the filter 
coeffs. I took advantage of this when I did my separate interpolation project.
It surely will be beneficial in this case as well since my buffers can be much 
smaller. In this resampling project, I'm now troubleshooting my program to 
make sure that I'm aligning my filter coeffs with the correct sample value. 
Things were a lot easier when I did my interpolation project (x2) with a 
single zero pad between samples. Now I've got an L=160, and M=147. Thanks.




>"Newbie" <none@available.com> wrote in message
>news:tUHRb.10643$5G3.8881@newssvr22.news.prodigy.com...
>> I've had a look at dspguru.com and am trying to implement a program that
>will
>> resample an audio signal from 44.1kHz to 48kHz. I'm running into some
>problems
>> - namely too many zeros in my converted file. My algorithm is basically:
>>
>> > Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159
>zeros
>> in order to oversample.
>>
>> Apply a low pass filter with a stop band freq. of 24kHz
>>
>> Downsample by keeping every 1 out of 147 samples
>>
>> Do I need to LPF again?
>>
>> Another thing that I'm not sure about is if there is any requirement on
>the
>> number of filter taps. Obviously the more taps I have, the fewer
>zero-valued
>> samples will be in my output file.
>>
>> Does anyone have any suggestions to implementing this SRC correctly?
>
>

"Newbie" <none@available.com> wrote in message
news:usTRb.11943$7s7.4291@newssvr23.news.prodigy.com...
> In article <llnsuxgv.fsf@ieee.org>, yates@ieee.org says...
> >
> >none@available.com (Newbie) writes:
> >
>
> Thanks for all of the feedback. I think that you are correct about my
filter
> not being designed correctly.
>
> I used ScopeFIR and chose Fs = 7056000, passband upper freq = 20kHz,
> stopband lower freq. = 22.05kHz, 1 dB passband ripple, and 100 dB stopband
> attenuation. I'm not sure what is going on here since the freq. response
plot
> shows the signal to be attenuated ~-75dB at 0 Hz. Also, I've got several
wide
> lobes above the cutoff freq. The lobes can be shorter if I increase the
number
> of filter taps, but then I also get more lobes.
>
> Please suggest a filter for me to use, including the number of taps. If
the
> number can be < 32 that would be good as my program doesn't support any
> higher.

There is no way a filter with less than 32 taps will be close to adequate
for this problem.  For 160x oversampling, you would need something more like
321 taps just to do a relatively lousy job!  Either you need to get a
different program that supports more taps, or switch to a filter design
method that allows more easily calculating the coefficients yourself, such
as a windowed sinc.

Alternatively, just to get up and running for now you could use linear
interpolation as a crude low pass filter.  This could even be implemented as
a standard FIR filter to verify your code--just set up an array that varies
from 0 to 1 linearly over 160 samples, then back to zero over .

> Lastly, I don't want to use someone elses program as I'm doing this for my
own
> education. In separate projects, I've interpolated audio from 22.05kHz to
> 44.1kHz and in another decimated from 48kHz down to 24kHz - these worked
fine.
> The 44.1kHz to 48kHz resampling project so far has proven to be a little
more
> difficult, namely, I believe because of the large interpolation and
decimation
> factors and long filters required.

That's certainly a noble goal and you'll certainly learn more than if you
use a canned routine.  Go for it!

In article <bv929v$phlpa$1@ID-210375.news.uni-berlin.de>, 
goldentully@hotmail.com says...
>
>"Newbie" <none@available.com> wrote in message
>news:usTRb.11943$7s7.4291@newssvr23.news.prodigy.com...
>> In article <llnsuxgv.fsf@ieee.org>, yates@ieee.org says...
>> >
>> >none@available.com (Newbie) writes:
>> >
>>
>> Thanks for all of the feedback. I think that you are correct about my
>filter
>> not being designed correctly.
>>
>> I used ScopeFIR and chose Fs = 7056000, passband upper freq = 20kHz,
>> stopband lower freq. = 22.05kHz, 1 dB passband ripple, and 100 dB stopband
>> attenuation. I'm not sure what is going on here since the freq. response
>plot
>> shows the signal to be attenuated ~-75dB at 0 Hz. Also, I've got several
>wide
>> lobes above the cutoff freq. The lobes can be shorter if I increase the
>number
>> of filter taps, but then I also get more lobes.
>>
>> Please suggest a filter for me to use, including the number of taps. If
>the
>> number can be < 32 that would be good as my program doesn't support any
>> higher.
>
>There is no way a filter with less than 32 taps will be close to adequate
>for this problem.  For 160x oversampling, you would need something more like
>321 taps just to do a relatively lousy job!  Either you need to get a
>different program that supports more taps, or switch to a filter design
>method that allows more easily calculating the coefficients yourself, such
>as a windowed sinc.
>
>Alternatively, just to get up and running for now you could use linear
>interpolation as a crude low pass filter.  This could even be implemented as
>a standard FIR filter to verify your code--just set up an array that varies
>from 0 to 1 linearly over 160 samples, then back to zero over .
>
>> Lastly, I don't want to use someone elses program as I'm doing this for my
>own
>> education. In separate projects, I've interpolated audio from 22.05kHz to
>> 44.1kHz and in another decimated from 48kHz down to 24kHz - these worked
>fine.
>> The 44.1kHz to 48kHz resampling project so far has proven to be a little
>more
>> difficult, namely, I believe because of the large interpolation and
>decimation
>> factors and long filters required.
>
>That's certainly a noble goal and you'll certainly learn more than if you
>use a canned routine.  Go for it!
>
>


Thanks. Do you have any recommendations on software I can use to design the 
filter - windowed sinc?

Jon Harris wrote:
> There is no way a filter with less than 32 taps will be close to adequate
> for this problem.  For 160x oversampling, you would need something more like
> 321 taps just to do a relatively lousy job!  Either you need to get a
> different program that supports more taps, or switch to a filter design
> method that allows more easily calculating the coefficients yourself, such
> as a windowed sinc.

Another approach would be for you to interpolate to 160 in several 
smaller steps.  160=(2^5)*5, so you could interpolate by 2 five times, 
and then by 5 once.  In each stage, set the cutoff frequency to 20KHz. 
The first stage will require a good (that is, long) filter.  I think 32 
taps would prolly be OK.  After interpolating by 2 (and filtering) in 
the first stage, you'll have a spectrum something like this:

______               ________
       \______|______/
       fs/4  fs/2

fs/4 is roughly 20KHz.

The next stage can get by with a much smaller filter, because the 
spectrum between 20K and 60K has already been supressed by a decent 
amount.  The later stages should still roll off starting at 20KHz, but 
you don't have to worry about stopband attenuation until you get into 
the 60KHz neighborhood.  That's a pretty relaxed filter.  You could 
prolly get away with interpolating in fewer stages - say in four stages 
(2,4,4,5).


Or as Jon suggested, you could use a windowed sinc and do it in one 
massive stage.  The rule of thumb I use is that I use 16*N taps for 
interpolation by N (at least in the first stage).  To do it in one stage 
would require 2560 taps.


-- 
Jim Thomas            Principal Applications Engineer  Bittware, Inc
jthomas@bittware.com  http://www.bittware.com          (703) 779-7770
The sooner you get behind, the more time you'll have to catch up

> Thanks. Do you have any recommendations on software I can use to design
the
> filter - windowed sinc?

Matlab, but I presume you don't have that.  You can actually just do it
yourself with Excel or a C program from the basic equations for sin(x)/x and
say a Hamming window.  I'm not all that familiar with other filter packages
out there but maybe someone else can help?

Or Jim's recommendation on multiple stages is a good idea too.

"Jon Harris" <goldentully@hotmail.com> wrote in message
news:bv97om$pe1oi$1@ID-210375.news.uni-berlin.de...
> > Thanks. Do you have any recommendations on software I can use to design
> the
> > filter - windowed sinc?
>
> Matlab, but I presume you don't have that.  You can actually just do it
> yourself with Excel or a C program from the basic equations for sin(x)/x
and
> say a Hamming window.  I'm not all that familiar with other filter
packages
> out there but maybe someone else can help?

Just found this link: http://www.dspguru.com/sw/tools/filtdsn.htm
Or you could cough up the $100 for the full version of ScopeFIR and help our
buddy Grant Griffin put food on his table!  :-)

Hello Newbie,
Instead of directly doing a 147:160 conversion, just do a series of simpler
conversions. Since 147 factors into 3*7*7, I'd look at doing a cascade of 3
interpolator/decimator functions in series. Just write your code so it can
work recursively.

-- 
Clay S. Turner, V.P.
Wireless Systems Engineering, Inc.
Satellite Beach, Florida 32937
(321) 777-7889
www.wse.biz
csturner@wse.biz



"Newbie" <none@available.com> wrote in message
news:usTRb.11943$7s7.4291@newssvr23.news.prodigy.com...
> In article <llnsuxgv.fsf@ieee.org>, yates@ieee.org says...
> >
> >none@available.com (Newbie) writes:
> >
> >> I've had a look at dspguru.com and am trying to implement a program
that
> will
> >> resample an audio signal from 44.1kHz to 48kHz. I'm running into some
> problems
> >> - namely too many zeros in my converted file. My algorithm is
basically:
> >>
> >>
> >> Take 44.1kHz signal and interpolate by padding the signal with L-1 =
159
> zeros
> >> in order to oversample.
> >
> >OK, so you're upsampling by a factor of 160 to a sample rate of 160 *
44100 =
> 7.056 MHz.
> >
> >> Apply a low pass filter with a stop band freq. of 24kHz
> >
> >No. The typical upconverter consists of upsampling by a factor of N
> >(inserting N-1 zeros) followed by a 1/N interpolating (lowpass)
> >filter. The typical downconverter consists of a 1/M lowpass filter
> >followed by a M:1 decimator (keeping 1 of M samples).
> >
> >You can combine the two filters (1/N and 1/M) by simply choosing the
lowest.
> >In this case N = 160 and M = 147, so 1/N is the lowest. Thus you want to
> filter
> >by 1/N (relative to a sample rate of 7.056 MHz), which corresponds to a
> filter
> >of bandwidth (7.056 MHz/2) / 160 = 22.05 kHz.
> >
> >> Downsample by keeping every 1 out of 147 samples
> >>
> >> Do I need to LPF again?
> >
> >Not if you did it right.
> >
> >> Another thing that I'm not sure about is if there is any requirement on
the
> >> number of filter taps. Obviously the more taps I have, the fewer
> zero-valued
> >> samples will be in my output file.
> >
> >Aha. Well, yes there is a requirement, but it's not black-and-white.
Let's
> >just take the upsampling stage for a moment and think about what's
happening
> >in the frequency domain. The original 44.1 kHz signal has a repetitive
> spectrum
> >out to infinity, repeating at multiples of 44.1 kHz. After you upsample
by
> 160
> >by inserting 159 zeros, the resulting spectrum hasn't changed. However,
since
> >your sample rate is now 7.056 MHz, the stuff between 22.05 kHz and
7.056/2
> MHz = 3.528
> >MHz has to be filtered out. If that filter isn't perfect, then, at the
7.056
> MHz
> >rate, the stuff above 22.05 kHz that leaks through will sound like
garbage
> (if
> >you could hear it). So the object is to have you're 1/160 lowpass filter
be
> >down as far as reasonable at 22.05 kHz. For high quality audio, it should
> >probably be at least 90 dB or so down. Note that this means that you will
> >necessarily have the cutoff frequency at something less than 22.05 kHz.
The
> >closer you get the cutoff frequency to 22.05 kHz, and the more you want
the
> >filter attenuation down at 22.05 kHz, the longer your filter will get.
> >
> >Another simple point: Obviously your filter was less than 161 taps long,
> >hence you are getting the zeros you spoke of. That's a bad sign that
> >you probably haven't chosen the passband/stopband characteristics
properly.
> >How did you design the filter?
>
>
>
>
> Thanks for all of the feedback. I think that you are correct about my
filter
> not being designed correctly.
>
> I used ScopeFIR and chose Fs = 7056000, passband upper freq = 20kHz,
> stopband lower freq. = 22.05kHz, 1 dB passband ripple, and 100 dB stopband
> attenuation. I'm not sure what is going on here since the freq. response
plot
> shows the signal to be attenuated ~-75dB at 0 Hz. Also, I've got several
wide
> lobes above the cutoff freq. The lobes can be shorter if I increase the
number
> of filter taps, but then I also get more lobes.
>
> Please suggest a filter for me to use, including the number of taps. If
the
> number can be < 32 that would be good as my program doesn't support any
> higher.
>
> Lastly, I don't want to use someone elses program as I'm doing this for my
own
> education. In separate projects, I've interpolated audio from 22.05kHz to
> 44.1kHz and in another decimated from 48kHz down to 24kHz - these worked
fine.
> The 44.1kHz to 48kHz resampling project so far has proven to be a little
more
> difficult, namely, I believe because of the large interpolation and
decimation
> factors and long filters required.
>
>
> Thx.
>
>
>
>
>
> >
> >> Does anyone have any suggestions to implementing this SRC correctly?
> >
> >Erik has a good point - use his. Unless your goal is the learning
> >experience.
> >-- 
> >%  Randy Yates                  % "Though you ride on the wheels of
tomorrow,
> >%% Fuquay-Varina, NC            %  you still wander the fields of your
> >%%% 919-577-9882                %  sorrow."
> >%%%% <yates@ieee.org>           % '21st Century Man', *Time*, ELO
> >http://home.earthlink.net/~yatescr
>

Newbie wrote:
> 
> 
> Thanks. Do you have any recommendations on software I can use to design the
> filter - windowed sinc?

Would you please trim the inclusion of posts you respond to
down what is necessasary to establish context for what you
add to it.


Thanks,

Bob
-- 

"Things should be described as simply as possible, but no
simpler."

                                             A. Einstein

"Newbie" <none@available.com> wrote in message
news:OWURb.11694$2y5.9162@newssvr24.news.prodigy.com...
> In article <bv929v$phlpa$1@ID-210375.news.uni-berlin.de>,
> goldentully@hotmail.com says...
> >

> Thanks. Do you have any recommendations on software I can use to design
the
> filter - windowed sinc?
>

Well, there's a couple of ways to look at this objective.  One has to do
with the overall filtering objective and the other has to do with the design
approach for the filter.

So, it's easy enough to window a sinc in a spreadsheet as was suggested.

The other way is to recognize that it's just a lowpass filter anyway and use
the Parks-McClellan program or filter design program of your choice.  The
zero-crossing spacing of the sinc is directly related to the bandwidth of
the lowpass filter.  So, you have both pieces of information anyway.

Here's a way to look at it:

A sinc is of infinite extent and has a perfect lowpass / brick wall
frequency response.  Not realizable.
A windowed sinc is a sinc that's been truncated to a finite length.
A rectangular window does the simplest truncation.
As a result, the frequency response is no longer perfect and there are
noticeable peaks in the stopband.
Changing the shape of the window so that it's more tapered has an effect on
the frequency response.
The more gradual the window, the sharper the frequency response and the
lower the stopband ripples - as a very general statement.
Note that the zero crossings of the sinc in time don't change when you apply
a window - unless the window has zeros itself so that those zeros are
potentially adde - which is most unlikely or unusual at least.

If you turn the process around and design a lowpass filter:
The impulse response will resemble a windowed sinc and the "window" is
generally unknown.
The zero crossings in time need not remain the same as for a sinc.
You can optimize the frequency response more directly this way.

Fred

Fred Marshall wrote:

   ...

> A windowed sinc is a sinc that's been truncated to a finite length.
> A rectangular window does the simplest truncation.
> As a result, the frequency response is no longer perfect and there are
> noticeable peaks in the stopband.
> Changing the shape of the window so that it's more tapered has an effect on
> the frequency response.
> The more gradual the window, the sharper the frequency response and the
> lower the stopband ripples - as a very general statement.

Sharper frequency response? I think you mean lower stopband ripple
(smaller side lobes) and more gradual transition to cutoff. No?

   ...

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

In article <llnsuxgv.fsf@ieee.org>, Randy Yates  <yates@ieee.org> wrote:
>none@available.com (Newbie) writes:
>No. The typical upconverter consists of upsampling by a factor of N
>(inserting N-1 zeros) followed by a 1/N interpolating (lowpass)
>filter. The typical downconverter consists of a 1/M lowpass filter
>followed by a M:1 decimator (keeping 1 of M samples).

This upsample and downsample stuff sounds like a bunch of confusing
extra work.  How about just applying your lowpass FIR filter directly
to the input data, calculating the correct phase taps for any fractional
point between input samples, rational or irrational, and getting your
output data in a single pass?  Then it becomes a matter of approximating
or interpolating your windowed sync, or other choosen filter function,
instead of all this artificial upsample, downsample work.  Does anyone
really cares about the intermediate 7 MHz upsampled "signal" with all
those stuffed zeros?  If not, why bother generating it?

So, does anyone have some good polynomial approximations for each
lobe of a windowed sync?  Or is phase table interpolation the only
method worth looking at?


IMHO. YMMV.
-- 
Ron Nicholson   rhn AT nicholson DOT com   http://www.nicholson.com/rhn/ 
#include <canonical.disclaimer>        // only my own opinions, etc.

"Fred Marshall" <fmarshallx@remove_the_x.acm.org> wrote in message
news:aLydnYdO7ub10oXd4p2dnA@centurytel.net...
>
> "Newbie" <none@available.com> wrote in message
> news:OWURb.11694$2y5.9162@newssvr24.news.prodigy.com...
> > In article <bv929v$phlpa$1@ID-210375.news.uni-berlin.de>,
> > goldentully@hotmail.com says...
> > >
>
> > Thanks. Do you have any recommendations on software I can use to design
> the
> > filter - windowed sinc?
> >
>
> Well, there's a couple of ways to look at this objective.  One has to do
> with the overall filtering objective and the other has to do with the
design
> approach for the filter.
>
> So, it's easy enough to window a sinc in a spreadsheet as was suggested.
>
> The other way is to recognize that it's just a lowpass filter anyway and
use
> the Parks-McClellan program or filter design program of your choice.  The
> zero-crossing spacing of the sinc is directly related to the bandwidth of
> the lowpass filter.  So, you have both pieces of information anyway.

The OP said that his filter design program only supported filters up to 32
taps, hence the suggestion for using a spreadsheet.  The simplest approach
would be of course to get a better filter design program!

Jerry Avins wrote:
> 
> Fred Marshall wrote:
>
> > The more gradual the window, the sharper the frequency response and the
> > lower the stopband ripples - as a very general statement.
> 
> Sharper frequency response? I think you mean lower stopband ripple
> (smaller side lobes) and more gradual transition to cutoff. No?

I just tried it going between a straight truncated and a
Hahn window, the cutoff is _much_ sharper.  The test was a
65536 point sinc with 100 samples between zero crossings.


Bob
-- 

"Things should be described as simply as possible, but no
simpler."

                                             A. Einstein

"Jerry Avins" <jya@ieee.org> wrote in message
news:40185603$0$11471$61fed72c@news.rcn.com...
> Fred Marshall wrote:
>
>    ...
>
> > A windowed sinc is a sinc that's been truncated to a finite length.
> > A rectangular window does the simplest truncation.
> > As a result, the frequency response is no longer perfect and there are
> > noticeable peaks in the stopband.
> > Changing the shape of the window so that it's more tapered has an effect
on
> > the frequency response.
> > The more gradual the window, the sharper the frequency response and the
> > lower the stopband ripples - as a very general statement.
>
> Sharper frequency response? I think you mean lower stopband ripple
> (smaller side lobes) and more gradual transition to cutoff. No?


Jerry,

Yes. As a general statement that's what I should have said.  So now I'm
curious about what Bob Cain did....

For general filter design there's a tradeoff between sidelobe level and
transition width.  The higher the sidelobes, the narrower the transition can
be - largely because the definition of width in that case is dependent on
the sidelob levels!!  But, a given length filter should be limited to a
minimum transition width as you say - as long as we don't get too picky
about how to define the width to -80dB.....

Fred

"Bob Cain" <arcane@arcanemethods.com> wrote in message
news:4018A3B2.F354DA8C@arcanemethods.com...
>
>
> Jerry Avins wrote:
> >
> > Fred Marshall wrote:
> >
> > > The more gradual the window, the sharper the frequency response and
the
> > > lower the stopband ripples - as a very general statement.
> >
> > Sharper frequency response? I think you mean lower stopband ripple
> > (smaller side lobes) and more gradual transition to cutoff. No?
>
> I just tried it going between a straight truncated and a
> Hahn window, the cutoff is _much_ sharper.  The test was a
> 65536 point sinc with 100 samples between zero crossings.
>

Bob,

I decided I was wrong.  So now I'm curious about this.  How did you define
cutoff?
Did you mean von Hann / hanning?  or really mean Hahn?

Here's a nice little program that demonstrates various windows and
transition width:

http://web.mit.edu/6.555/www/fir.html

Fred

Fred Marshall wrote:
> >
> > I just tried it going between a straight truncated and a
> > Hahn window, the cutoff is _much_ sharper.  The test was a
> > 65536 point sinc with 100 samples between zero crossings.
> >
> 
> Bob,
> 
> I decided I was wrong.  So now I'm curious about this.  How did you define
> cutoff?

Fs/100

> Did you mean von Hann / hanning?  or really mean Hahn?

No, I actually meant von Hann (didn't know about the von
though) but I just realized that my little test was
meaningless because the FFT I used (that in the Adobe
Audition audio editor) applies a window on top of the one I
generated over the data and that there is no option to
specify no window (or rectangular window as some would have
it) so that I was comparing a Hann window with a Hann window
applied twice.  Sorry for the noise.


Bob
-- 

"Things should be described as simply as possible, but no
simpler."

                                             A. Einstein

Bob Cain wrote:

   ...

> No, I actually meant von Hann (didn't know about the von
> though) but I just realized that my little test was
> meaningless because the FFT I used (that in the Adobe
> Audition audio editor) applies a window on top of the one I
> generated over the data and that there is no option to
> specify no window (or rectangular window as some would have
> it) so that I was comparing a Hann window with a Hann window
> applied twice.  Sorry for the noise.

Aw chucks! And here I thought a Huhn window was for watching the sky
fall.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

Fred Marshall wrote:

   ...

> For general filter design there's a tradeoff between sidelobe level and
> transition width.  The higher the sidelobes, the narrower the transition can
> be - largely because the definition of width in that case is dependent on
> the sidelob levels!!  But, a given length filter should be limited to a
> minimum transition width as you say - as long as we don't get too picky
> about how to define the width to -80dB.....

With analog bandpass filters, skirt selectivity is a good measure, and I
tend to look at all filters in terms of edge slope after a hard-to-pin-
down knee is rounded. On a log plot, the angle tells all.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

Bob Cain <arcane@arcanemethods.com> wrote in
news:40193D36.77A68501@arcanemethods.com: 

> 
> 
> Fred Marshall wrote:
>> >
>> > I just tried it going between a straight truncated and a
>> > Hahn window, the cutoff is _much_ sharper.  The test was a
>> > 65536 point sinc with 100 samples between zero crossings.
>> >
>> 
>> Bob,
>> 
>> I decided I was wrong.  So now I'm curious about this.  How did you
>> define cutoff?
> 
> Fs/100
> 
>> Did you mean von Hann / hanning?  or really mean Hahn?
> 
> No, I actually meant von Hann (didn't know about the von
> though) but I just realized that my little test was
> meaningless because the FFT I used (that in the Adobe
> Audition audio editor) applies a window on top of the one I
> generated over the data and that there is no option to
> specify no window (or rectangular window as some would have
> it) so that I was comparing a Hann window with a Hann window
> applied twice.  Sorry for the noise.
> 
> 
> Bob

Poor von Hann! His window is commonly called a Hanning Window only 
because there is also a Hamming Window.

Of course, very few of us come up with something important that is also 
tied to our name. I guess that I would be happy if there was a "Clarking 
Window" ;-)



-- 
Al Clark
Danville Signal Processing, Inc.
--------------------------------------------------------------------
Purveyors of Fine DSP Hardware and other Cool Stuff
Available at http://www.danvillesignal.com

After looking over some windowed sinc implementations, I am a little concerned 
about the passband performance. Isn't this one of the tradeoffs in going 
with this approach? Since my signals are audio, I'd like to keep the passband 
as flat as possible. 

Does anyone know how the pro audio guys design their 44.1kHz<->48kHz SRCs? I'm 
referring to using a series of cascaded filters (for the interpolation 
w/decimation) or is a windowed sinc design more common? 

Right now I'm looking towards a cascaded set of filters.

Thanks for all of the responses to my post.

Jerry Avins wrote:
....
> Aw chucks! And here I thought a Huhn window was for watching the sky
> fall.

Now Jerry, I would guess a Huhn window would be something you would
find on a chicken shack :).

Andor

Andor wrote:
> Jerry Avins wrote:
> ...
> 
>>Aw chucks! And here I thought a Huhn window was for watching the sky
>>fall.
> 
> 
> Now Jerry, I would guess a Huhn window would be something you would
> find on a chicken shack :).
> 
> Andor

Yes. And it was Chicken Little who cried, "The sky is falling! The sky
is falling!"

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

>Another approach would be for you to interpolate to 160 in several 
>smaller steps.  160=(2^5)*5, so you could interpolate by 2 five times, 
>and then by 5 once.  In each stage, set the cutoff frequency to 20KHz. 
>The first stage will require a good (that is, long) filter.  I think 32 
>taps would prolly be OK.  After interpolating by 2 (and filtering) in 
>the first stage, you'll have a spectrum something like this:
>
>______               ________
>       \______|______/
>       fs/4  fs/2
>
>fs/4 is roughly 20KHz.
>
>The next stage can get by with a much smaller filter, because the 
>spectrum between 20K and 60K has already been supressed by a decent 
>amount.  The later stages should still roll off starting at 20KHz, but 
>you don't have to worry about stopband attenuation until you get into 
>the 60KHz neighborhood.  That's a pretty relaxed filter.  You could 
>prolly get away with interpolating in fewer stages - say in four stages 
>(2,4,4,5).
>


Jim,

I'm working on this as issue as well. Given your example of (2^5)*5 with a 
20kHz signal, after interpolating your signal in the first stage by 2 and 
filtering, how do you approach the second stage of the filter in terms of the 
sampling freq.?

In other words, when designing the filter for this 2nd stage to preserve the 
20kHz content, would I be basing the sampling freq. for this stage off of: 

fs = 80 x 2 = 160kHz with fs/2 = 80kHz 

or would it be fs = 40 x 2 = 80kHz with fs/2 = 40kHz (again, like stage 1?)

I assume that the fs is increasing with each stage. Is this correct? 

This concept is what isn't sinking in to my thick skull.

-Rick S.

On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
wrote:

   (snipped)
>
>Poor von Hann! His window is commonly called a Hanning Window only 
>because there is also a Hamming Window.
>
>Of course, very few of us come up with something important that is also 
>tied to our name. I guess that I would be happy if there was a "Clarking 
>Window" ;-)

Hi Al,
   don't worry, at least you have a 
candy bar named after you.

That's better than the rest of us here.

[-Rick-]

"Rick Lyons" <r.lyons@_BOGUS_ieee.org> wrote in message
news:401af766.185204703@news.west.earthlink.net...
> On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
> wrote:
>
>    (snipped)
> >
> >Poor von Hann! His window is commonly called a Hanning Window only
> >because there is also a Hamming Window.
> >
> >Of course, very few of us come up with something important that is also
> >tied to our name. I guess that I would be happy if there was a "Clarking
> >Window" ;-)
>
> Hi Al,
>    don't worry, at least you have a
> candy bar named after you.
>
> That's better than the rest of us here.
>
> [-Rick-]

Rick, you've got the "Lyons share".  ;-)
All I have is the semiconductor company that is now called Intersil.

-Jon Harris

On Sat, 31 Jan 2004 00:09:34 GMT, disregard@disregard.com (Rick S.)
wrote:

>
>>Another approach would be for you to interpolate to 160 in several 
>>smaller steps.  160=(2^5)*5, so you could interpolate by 2 five times, 
>>and then by 5 once.  In each stage, set the cutoff frequency to 20KHz. 
>>The first stage will require a good (that is, long) filter.  I think 32 
>>taps would prolly be OK.  After interpolating by 2 (and filtering) in 
>>the first stage, you'll have a spectrum something like this:
>>
>>______               ________
>>       \______|______/
>>       fs/4  fs/2
>>
>>fs/4 is roughly 20KHz.
>>
>>The next stage can get by with a much smaller filter, because the 
>>spectrum between 20K and 60K has already been supressed by a decent 
>>amount.  The later stages should still roll off starting at 20KHz, but 
>>you don't have to worry about stopband attenuation until you get into 
>>the 60KHz neighborhood.  That's a pretty relaxed filter.  You could 
>>prolly get away with interpolating in fewer stages - say in four stages 
>>(2,4,4,5).
>>
>
>
>Jim,
>
>I'm working on this as issue as well. Given your example of (2^5)*5 with a 
>20kHz signal, after interpolating your signal in the first stage by 2 and 
>filtering, how do you approach the second stage of the filter in terms of the 
>sampling freq.?
>
>In other words, when designing the filter for this 2nd stage to preserve the 
>20kHz content, would I be basing the sampling freq. for this stage off of: 
>
>fs = 80 x 2 = 160kHz with fs/2 = 80kHz 
>
>or would it be fs = 40 x 2 = 80kHz with fs/2 = 40kHz (again, like stage 1?)
>
>I assume that the fs is increasing with each stage. Is this correct? 
>
>This concept is what isn't sinking in to my thick skull.
>
>-Rick S.

Hi,

   well, ... after the first interpolation-by-2, 
the spectrum of the first filter's output will be


     ____    44.1   ________
         \____|____/    |   \____
     0    |             |
        22.05          88.2   

fs = 88.2 kHz.

----------------------------------------------------
After the second interpolation-by-2, the 
spectrum of the second filter's output will be


       __    44.1                        ____
         \____|_______________...______/  |  \________
     0   |              |               176.4
        22.05          88.2   

fs = 176.4 kHz.

Yes, the fs is increasing with each stage.


I like Jim Thomas' idea because the filters for the 
first five stages could be "half-band" filters which 
are computationally efficient.

Good Luck,
[-Rick-]

r.lyons@_BOGUS_ieee.org (Rick Lyons) wrote in news:401af766.185204703
@news.west.earthlink.net:

> On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
> wrote:
> 
>    (snipped)
>>
>>Poor von Hann! His window is commonly called a Hanning Window only 
>>because there is also a Hamming Window.
>>
>>Of course, very few of us come up with something important that is also 
>>tied to our name. I guess that I would be happy if there was a 
"Clarking 
>>Window" ;-)
> 
> Hi Al,
>    don't worry, at least you have a 
> candy bar named after you.
> 
> That's better than the rest of us here.
> 
> [-Rick-]
> 

That reminds me of an April Fools Day prank:

I once told by boss that a Mr Lyon needed to talk to him right away. When 
he called the zoo, they told him that Mr Lion wasn't taking calls, but 
that Mr Monkey could help him.

There is actually a Clark's Law (mine and certainly discovered by every 
other engineer and programmer)

Clark's Rule of Ten:

1. All bugs take 10 secs, 10 minutes, 10 hours, 10 days, etc to solve.
2. They're all dumb shits after you find them.
3. If you find yourself in the 10 hour variety, its time to reboot or you 
may be headed towards 10 days. 

Clark's Corollary to the Rule of Ten:

1. Most bugs can be solved on the 494 bridge. (Two blocks from my old 
office - on the way home).


Now I know this Law is not as profound as Murphy or as edible as a Clark 
bar, but I claim it anyway.....


-- 
Al Clark
Danville Signal Processing, Inc.
--------------------------------------------------------------------
Purveyors of Fine DSP Hardware and other Cool Stuff
Available at http://www.danvillesignal.com

>   well, ... after the first interpolation-by-2, 
>the spectrum of the first filter's output will be
>
>
>     ____    44.1   ________
>         \____|____/    |   \____
>     0    |             |
>        22.05          88.2   
>
>fs = 88.2 kHz.
>
>----------------------------------------------------
>After the second interpolation-by-2, the 
>spectrum of the second filter's output will be
>
>
>       __    44.1                        ____
>         \____|_______________...______/  |  \________
>     0   |              |               176.4
>        22.05          88.2   
>
>fs = 176.4 kHz.
>
>Yes, the fs is increasing with each stage.
>
>
>I like Jim Thomas' idea because the filters for the 
>first five stages could be "half-band" filters which 
>are computationally efficient.
>
>Good Luck,
>[-Rick-]
>


Got it. Thanks.

Jon Harris wrote:
> 
> "Rick Lyons" <r.lyons@_BOGUS_ieee.org> wrote in message
> news:401af766.185204703@news.west.earthlink.net...
> > On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
> > wrote:
> >
> >    (snipped)
> > >
> > >Poor von Hann! His window is commonly called a Hanning Window only
> > >because there is also a Hamming Window.
> > >
> > >Of course, very few of us come up with something important that is also
> > >tied to our name. I guess that I would be happy if there was a "Clarking
> > >Window" ;-)
> >
> > Hi Al,
> >    don't worry, at least you have a
> > candy bar named after you.
> >
> > That's better than the rest of us here.
> >
> > [-Rick-]
> 
> Rick, you've got the "Lyons share".  ;-)
> All I have is the semiconductor company that is now called Intersil.
> 
> -Jon Harris

Hell, all I got was a C&W duo.

Rob Judd

Jerry Avins wrote:

> >>Aw chucks! And here I thought a Huhn window was for watching the sky
> >>fall.
> > 
> > 
> > Now Jerry, I would guess a Huhn window would be something you would
> > find on a chicken shack :).
> > 
> > Andor
> 
> Yes. And it was Chicken Little who cried, "The sky is falling! The sky
> is falling!"

Ah, I thought I missed some literary pointer. Who is Chicken Little?

Andor wrote:
> Jerry Avins wrote:
> 
> 
>>>>Aw chucks! And here I thought a Huhn window was for watching the sky
>>>>fall.
>>>
>>>
>>>Now Jerry, I would guess a Huhn window would be something you would
>>>find on a chicken shack :).
>>>
>>>Andor
>>
>>Yes. And it was Chicken Little who cried, "The sky is falling! The sky
>>is falling!"
> 
> 
> Ah, I thought I missed some literary pointer. Who is Chicken Little?

A version of the story: http://www.edsanders.com/chickenlittle/ (Ignore
the moral it draws. Even those with paranoia can have enemies.)

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

an2or@mailcircuit.com (Andor) wrote in message news:<ce45f9ed.0402020001.1ca738d8@posting.google.com>...
> Jerry Avins wrote:
> 
> > >>Aw chucks! And here I thought a Huhn window was for watching the sky
> > >>fall.
> > > 
> > > 
> > > Now Jerry, I would guess a Huhn window would be something you would
> > > find on a chicken shack :).
> > > 
> > > Andor
> > 
> > Yes. And it was Chicken Little who cried, "The sky is falling! The sky
> > is falling!"
> 
> Ah, I thought I missed some literary pointer. Who is Chicken Little?

At the riskof starting another 'why the Americans don't speak English'
thread...

Chicken Little is the American version of Chicken Licken who, together
with Hhenny Penny, Goosey Loosey, Turkey Lurkey and Ducky Lucky, ends
up as dinner for Foxy Loxy.

Ian

Ian Okey wrote:

> 
> At the riskof starting another 'why the Americans don't speak English'
> thread...
>

First you would have to define "English" .]

Is it Cockney, Valley Girl, BBC, etc etc etc ....

Richard Owlett <rowlett@atlascomm.net> writes:

> Ian Okey wrote:
>
>> At the riskof starting another 'why the Americans don't speak
>> English'
>> thread...
>>
>
> First you would have to define "English" .]
>
> Is it Cockney, Valley Girl, BBC, etc etc etc ....

Yes.
-- 
%  Randy Yates                  % "Watching all the days go by...    
%% Fuquay-Varina, NC            %  Who are you and who am I?"
%%% 919-577-9882                % 'Mission (A World Record)', 
%%%% <yates@ieee.org>           % *A New World Record*, ELO
http://home.earthlink.net/~yatescr

On Fri, 30 Jan 2004 16:57:37 -0800, "Jon Harris" 

(snipped)

>> >
>> >Of course, very few of us come up with something important that is also
>> >tied to our name. I guess that I would be happy if there was a "Clarking
>> >Window" ;-)
>>
>> Hi Al,
>>    don't worry, at least you have a
>> candy bar named after you.
>>
>> That's better than the rest of us here.
>>
>> [-Rick-]
>
>Rick, you've got the "Lyons share".  ;-)
>All I have is the semiconductor company that is now called Intersil.
>
>-Jon Harris

Hi,
  Whew!!!  You scared me for a moment there.
For an instant I thought, "No way.  This can't 
really be the the "Harris" of Harris Semiconductors?"
What's he doin' on comp.dsp? (!!!)
He should be drinking champagne on his yacht off 
the coast of Greece.

[-Rick-]
 

On Mon, 02 Feb 2004 18:18:43 +1100, Rob Judd <judd@ob-wan.com> wrote:

>Jon Harris wrote:
>> 
>> "Rick Lyons" <r.lyons@_BOGUS_ieee.org> wrote in message
>> news:401af766.185204703@news.west.earthlink.net...
>> > On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
>> > wrote:
>> >
>> >    (snipped)
>> > >
>> > >Poor von Hann! His window is commonly called a Hanning Window only
>> > >because there is also a Hamming Window.
>> > >
>> > >Of course, very few of us come up with something important that is also
>> > >tied to our name. I guess that I would be happy if there was a "Clarking
>> > >Window" ;-)
>> >
>> > Hi Al,
>> >    don't worry, at least you have a
>> > candy bar named after you.
>> >
>> > That's better than the rest of us here.
>> >
>> > [-Rick-]
>> 
>> Rick, you've got the "Lyons share".  ;-)
>> All I have is the semiconductor company that is now called Intersil.
>> 
>> -Jon Harris
>
>Hell, all I got was a C&W duo.
>
>Rob Judd

Hi Rob,

  there are four things like the Judd's  ;-)

Hey, didn't the daughter Judd just get into 
some kind of trouble.  (I have a vague recollection.)

[-Rick-]

"Rick Lyons" <r.lyons@_BOGUS_ieee.org> wrote in message
news:4020385f.6737234@news.west.earthlink.net...
> On Fri, 30 Jan 2004 16:57:37 -0800, "Jon Harris"
>
> (snipped)
>
> >> >
> >> >Of course, very few of us come up with something important that is
also
> >> >tied to our name. I guess that I would be happy if there was a
"Clarking
> >> >Window" ;-)
> >>
> >> Hi Al,
> >>    don't worry, at least you have a
> >> candy bar named after you.
> >>
> >> That's better than the rest of us here.
> >>
> >> [-Rick-]
> >
> >Rick, you've got the "Lyons share".  ;-)
> >All I have is the semiconductor company that is now called Intersil.
> >
> >-Jon Harris
>
> Hi,
>   Whew!!!  You scared me for a moment there.
> For an instant I thought, "No way.  This can't
> really be the the "Harris" of Harris Semiconductors?"
> What's he doin' on comp.dsp? (!!!)
> He should be drinking champagne on his yacht off
> the coast of Greece.

I wish!

Rick Lyons wrote:
> 
> On Mon, 02 Feb 2004 18:18:43 +1100, Rob Judd <judd@ob-wan.com> wrote:
> 
> >Jon Harris wrote:
> >>
> >> "Rick Lyons" <r.lyons@_BOGUS_ieee.org> wrote in message
> >> news:401af766.185204703@news.west.earthlink.net...
> >> > On Thu, 29 Jan 2004 23:13:54 GMT, Al Clark <dsp@danvillesignal.com>
> >> > wrote:
> >> >
> >> >    (snipped)
> >> > >
> >> > >Poor von Hann! His window is commonly called a Hanning Window only
> >> > >because there is also a Hamming Window.
> >> > >
> >> > >Of course, very few of us come up with something important that is also
> >> > >tied to our name. I guess that I would be happy if there was a "Clarking
> >> > >Window" ;-)
> >> >
> >> > Hi Al,
> >> >    don't worry, at least you have a
> >> > candy bar named after you.
> >> >
> >> > That's better than the rest of us here.
> >> >
> >> > [-Rick-]
> >>
> >> Rick, you've got the "Lyons share".  ;-)
> >> All I have is the semiconductor company that is now called Intersil.
> >>
> >> -Jon Harris
> >
> >Hell, all I got was a C&W duo.
> >
> >Rob Judd
> 
> Hi Rob,
> 
>   there are four things like the Judd's  ;-)
> 
> Hey, didn't the daughter Judd just get into
> some kind of trouble.  (I have a vague recollection.)
> 
> [-Rick-]

Naah, Rene Russo was in Big Trouble. Ashley was in High Crimes. :)

Rob