[hpsdr] Mercury sampling rate

Alex, VE3NEA alshovk at dxatlas.com
Mon Apr 14 17:57:35 PDT 2008


Hi Alberto,

The data blocks in the ring buffer do not need to be the same size as the 
input blocks, you can make them as small as you want. I have found that 
10-ms blocks work best.

Your fractional resampler approach is good. As always, there is a 
compromise - between the bandwidth sacrifice due to resampling, extra CPU 
load, and sound quality.

73 Alex VE3NEA




----- Original Message ----- 
>  Alex,
>
>    I agree of course on the numerical part of your analysis, but not on 
> the conclusions.
> Suppose that the output rate is 11025 Hz, a commonly used sampling rate 
> for output. If you do the processing in blocks
> of 1k samples, then each block correspond to a duration of about 92.9 ms. 
> If you receive CW at a speed of 30 wpm, the
> dot length is about 40 ms. So, when you have to throw away one block, 
> because you are in a condition of buffer overflow,
> you throw away more than two elements of the incoming CW signal. Not nice.
>
> The same for buffer starvation/underflow. In such a circumstance you have 
> to send to the sound card either a replica of
> the last buffer sent, or a buffer of silence. In both cases the effects 
> are very negative for CW. I agree that for SSB
> reception, especially in conditions of high QRM, maybe the fact can go 
> unnoticed, but not for CW.
>
> So I don't see alternatives to a continuously adjustable fractional 
> resampler, if what is sought is the best audio
> quality. But I would like to be wrong, as this choice as other negative 
> implications, which, if possible, I would like
> to avoid altogether.
>
> 73  Alberto  I2PHD 


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