Band limited signal into soundcard with sample & hold?

On 2006-07-09, Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
> On Fri, 07 Jul 2006 14:07:40 -0500, Ben Jackson <ben@ben.com> wrote:
>
>>On 2006-07-07, Jerry Avins <jya@ieee.org> wrote:
>>> My reasoning was, that if the S&H's sample window drifts with respect to 
>>> the DAC's, there's no way to assure that the DAC always samples a 
>>> settled signal.
>>
>>But the ADC (I think you mean) on the soundcard is behind the card's
>>anti-aliasing filter.  In effect it can't "see" those high frequency
>>transitions at all.  If there were no filter in the system, I'd agree
>>with what you're saying.
>
> Do you mean that the signal is fairly narrow and contained within the
> passband of the 20kHz anti-alias filter, or do you mean that the
> signal is less than 20kHz bandwidth and outside of the the filter
> passband?

I meant that initially it would be narrow (<20kHz bw) and outside
the 20kHz.  Then by chopping it would appear as an alias within
the 20kHz passband of the soundcard.

-- 
Ben Jackson AD7GD
<ben@ben.com>
http://www.ben.com/

On Sun, 09 Jul 2006 19:24:13 -0500, Ben Jackson <ben@ben.com> wrote:

>On 2006-07-09, Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
>> On Fri, 07 Jul 2006 14:07:40 -0500, Ben Jackson <ben@ben.com> wrote:
>>
>>>On 2006-07-07, Jerry Avins <jya@ieee.org> wrote:
>>>> My reasoning was, that if the S&H's sample window drifts with respect to 
>>>> the DAC's, there's no way to assure that the DAC always samples a 
>>>> settled signal.
>>>
>>>But the ADC (I think you mean) on the soundcard is behind the card's
>>>anti-aliasing filter.  In effect it can't "see" those high frequency
>>>transitions at all.  If there were no filter in the system, I'd agree
>>>with what you're saying.
>>
>> Do you mean that the signal is fairly narrow and contained within the
>> passband of the 20kHz anti-alias filter, or do you mean that the
>> signal is less than 20kHz bandwidth and outside of the the filter
>> passband?
>
>I meant that initially it would be narrow (<20kHz bw) and outside
>the 20kHz.  Then by chopping it would appear as an alias within
>the 20kHz passband of the soundcard.

In that case it's a futile exercise as the anit-alias filter will
already have attenuated the signal into oblivion.   I think that's
what Jerry was trying to say earlier.

What you're describing is often called "sampled IF" when used in
communications systems.   For those type of systems the "anit-alias"
filter is actually a bandpass filter that passes the signal of
interest and rejects anything that may already be at baseband within
the normal Nyquist region of the converter.   This leaves only the
aliased energy within the bandpass filter input frequency, with the
low-frequency alias left for subsequent digital processing.

If the anti-alias filter removes the desired signal before
digitization and you don't want to change the anti-alias filter then
you don't have much hope of recovering the signal.
Eric Jacobsen
Minister of Algorithms, Intel Corp.
My opinions may not be Intel's opinions.
http://www.ericjacobsen.org

On 2006-07-10, Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
> On Sun, 09 Jul 2006 19:24:13 -0500, Ben Jackson <ben@ben.com> wrote:
>>
>>I meant that initially it would be narrow (<20kHz bw) and outside
>>the 20kHz.

Here I was talking about still outside the soundcard, already
band-limited.

>> Then by chopping

...outside the soundcard...

>> it would appear as an alias within
>>the 20kHz passband of the soundcard.

...after going through the soundcard's anti-aliasing filter.

> What you're describing is often called "sampled IF" when used in
> communications systems.   For those type of systems the "anit-alias"
> filter is actually a bandpass filter that passes the signal of
> interest and rejects anything that may already be at baseband within
> the normal Nyquist region of the converter.   This leaves only the
> aliased energy within the bandpass filter input frequency, with the
> low-frequency alias left for subsequent digital processing.

Well, the "low frequency alias" doesn't exist until someone comes along
and samples at Fs < 2 * Fmax.  I was proposing to do that undersampling
external to the soundcard (since the soundcard's input anti-alising
filter and bandwidth are unsuitable, as you describe), essentially
introducing a zero-order hold, and then use the soundcard to sample
the aliased signal.

What we were discussing was what (if any) phase/frequency alignment
requirements would be for such an external chopper.

I think the question was basically equivalent to asking if you could
mix an IF down to baseband and then feed it to a soundcard without
subsequent filtering, relying on the soundcard's anti-aliasing filter
to remove unwanted mixing products.  And that's probably a false economy!

-- 
Ben Jackson AD7GD
<ben@ben.com>
http://www.ben.com/

On Mon, 10 Jul 2006 18:22:39 -0500, Ben Jackson <ben@ben.com> wrote:

>On 2006-07-10, Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
>> On Sun, 09 Jul 2006 19:24:13 -0500, Ben Jackson <ben@ben.com> wrote:
>>>
>>>I meant that initially it would be narrow (<20kHz bw) and outside
>>>the 20kHz.
>
>Here I was talking about still outside the soundcard, already
>band-limited.
>
>>> Then by chopping
>
>...outside the soundcard...
>
>>> it would appear as an alias within
>>>the 20kHz passband of the soundcard.
>
>...after going through the soundcard's anti-aliasing filter.
>
>> What you're describing is often called "sampled IF" when used in
>> communications systems.   For those type of systems the "anit-alias"
>> filter is actually a bandpass filter that passes the signal of
>> interest and rejects anything that may already be at baseband within
>> the normal Nyquist region of the converter.   This leaves only the
>> aliased energy within the bandpass filter input frequency, with the
>> low-frequency alias left for subsequent digital processing.
>
>Well, the "low frequency alias" doesn't exist until someone comes along
>and samples at Fs < 2 * Fmax.  I was proposing to do that undersampling
>external to the soundcard (since the soundcard's input anti-alising
>filter and bandwidth are unsuitable, as you describe), essentially
>introducing a zero-order hold, and then use the soundcard to sample
>the aliased signal.
>
>What we were discussing was what (if any) phase/frequency alignment
>requirements would be for such an external chopper.
>
>I think the question was basically equivalent to asking if you could
>mix an IF down to baseband and then feed it to a soundcard without
>subsequent filtering, relying on the soundcard's anti-aliasing filter
>to remove unwanted mixing products.  And that's probably a false economy!

Duh, okay, now that I've caught up to everyone else...   ;)

I think you got pretty good dialogue on this previously.  Personally,
if the sample clock to the ADC on the card is reasonably stable
(enough to drive the SHA without degrading SNR), it might be as simple
as driving an external SHA with the existing ADC clock.

Getting that clock out cleanly might be tricky, but I think that'd be
my first approach.

Eric Jacobsen
Minister of Algorithms, Intel Corp.
My opinions may not be Intel's opinions.
http://www.ericjacobsen.org