PSN-L Email List Message

Subject: Re: Modified sound card and datalogging and geophones
From: Gordon Couger gcouger@..........
Date: Wed, 20 Jul 2005 15:02:44 -0500


If you are recording Audio and use the mixer a high LO to invert 
  the signal then at the end of the day play back the recording 
into a computer and use digital filters to pull out what you 
want and store it on CD ROM or DVD with software that has been 
developed in public domain, GNU and share ware. TAPR is a good starting point.

Recoding everything in real time then doing the signal 
processing one pass at a time is a great deal easier to do than 
doing it all in real time. Of course it must be automated and 
take less time to process than record.


ChrisAtUpw@....... wrote:
> In a message dated 20/07/2005, gcouger@.......... writes:
>     Hi Chris,
>     Have you considered using a mixer with local oscillator
>     frequency that puts the unwanted image above the range of the
>     sound card.
>     It inverts the spectrum but that's no problem with computers as
>     fast as we have.
>     Gordon Couger
>     Stillwater, OK
> Hi Gordon,
>     No I haven't. You get into sharp cut filters for the end of the 
> band, which tends to covert the problem into a headache. I did not 
> mention the simple removal of the input capacitor and fitting an input 
> opamp, for the same reason. You are still left with the 8,000 sps 
> minimum, which generates ~2.7 G bytes / day / channel. Until we are able 
> to control the sample rate as desired, to say 20 sps, the method is 
> unsatisfactory for long term recording. It must be possible in 
> principle, but I suspect that it is hidden in the commercial software.
>     The way that I have used small disk stereo audio recorders for data 
> recording is to convert the analogue signal into a variable frequency 
> sine wave and record that. The only problem is that while you can get 
> very fast V->F changes, you can't easily recover fast F->V changes, so 
> you need low bandwidth. You can add maybe 3 different frequency 
> ranges per audio channel and sort the output with bandpass filters, but 
> it is complicated.
>     Regards,
>     Chris Chapman


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