[hpsdr] new architecture - FFT front-end

Warren C. Pratt warren at wpratt.com
Thu Sep 4 10:10:30 PDT 2014


Hi David,

Let me preface this by saying that I have great interest in this 
project; however, I'm not personally doing any of the coding as I'm busy 
with other things.  So, those who are coding, and possibly learning as 
they go, may have alternative inputs.

First of all, one must process the entire sample stream.  It is not 
acceptable to skip any samples as the discontinuities create problems.

Secondly, my approach would be to build overlap-save filters where the 
forward FFT uses the original wideband sample stream and is common for 
all outputs being served.  However, each output has its own Inverse FFT 
and the time-domain decimation is done in the frequency domain before 
the inverse FFT which means that each inverse FFT is much smaller than 
the original forward FFT.  Using this approach, the Jetson (or 
alternative server) would be serving time domain samples and preserving 
all the flexibility that implies.

That's a brief explanation of ONE way to approach the solution. This is 
a very interesting project in that it should permit serving a very large 
number of "slices".

73,
Warren  NR0V

On 9/4/2014 9:23 AM, David Bridgham wrote:
> ***** High Performance Software Defined Radio Discussion List *****
>
>
>
> I've been mulling over this idea of using the Jetson board to do an 
> FFT of the entire (mostly) sample stream to then make the 
> frequency-domain signal available to applications.  Seems very 
> intriguing and I love hearing about entirely new ways of approaching 
> problems.
>
> I'm curious though.  When designing a codec, one chooses the 
> parameters for the FFT very carefully.  The sampling rate is given to 
> you by the sample stream coming in (though you may decimate to change 
> that) but you pick the size of the FFT, the overlap between blocks, 
> and the window function all to make the FFT bins show up just how you 
> want them.
>
> With this new architecture, all those choices are made up front and 
> all the codecs running behind it have to use them.  So my question is, 
> is there a way in the frequency domain to convert from one to 
> another?  Can you simply average a bunch of FFT bins together to get 
> larger bin sizes and interpolate between bins if you want smaller?  Or 
> do you have to do an IFFT back to the time domain and then FFT again 
> to change anything?
>
> I usually get about halfway through a DSP book before I get bogged 
> down so maybe this is all covered in later chapters.
>
>
>
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