[op25-dev] Re: "Could not find a decoder to use"

Matt D md123 at nycap.rr.com
Thu Apr 18 15:29:46 UTC 2013


On 04/17/2013 09:15 PM, Souf Hadoken wrote:
> Hi Matt,
> I would like first to thank you for your help the last weeks (I spend my spring break locked in my room trying to make the OP25 block and gnu radio work).
> You gave me really good indications and instruction. After another atemp staring from a clean disk I manage to make the file work and could listen to the sound.
> 
> No I'm in the phase where I'm using an USRP. I manage to make it work and detected but Im stuck at a point where you have been (according to the archive of the groupe discussion):
> 
> -I can see the 4 ligne of the 4FSK BUT I HAVE NO AUDIO.
> 
> I think it's a probleme of sample rate, channel rate and things like that.
> Could you help me by giving me some explanations like do the channel rate from the OP25 block should be perfectly maching the audio sink?
> what is the difference between channel rate and sample rate?
> What is the bandwith in the USRP and how should it be?
> 
> thanks for your help
> 
I don't know much about the USRP (I use a USB tuner) because I am kind
of low on cash.

For the audio: you've got the 4 lines in the Dibits tab so its decoding.
 First I would say to apply the r307 patch.  I leave the gain set at 1,
maybe you can set it lower using USRP, but don't leave at 0.

For the sample rate:  To get a handle on sampling rates read chapter3 at
"http://www.dspguide.com/copyrite.htm".  For me an important, perhaps
the most important, aspect of sampling rates was determining what rate
to capture the signal at.  I found that oversampling (I settled on
1.024Msps) and then increasing the decimation and sample_per_symbol to
get the channel rate right (between 19.2k and 38.4k) was the way to go
with my little USB tuner.  You probably wont have to deal with this
weirdness on your USRP but your still going to oversample somewhat.

The channel rate:  the channel rate is rate the bits are transmitted
(usually at 9.6k) on the carrier frequency AND in our case the rate we
are telling the decoder to decode the information at.  I have found that
receiving works best set between 19.2k and 38.4k (2 to 4 times the
transmitted rate).  This rate does not need to match the audio sink
although it may work if it did.

Audio sink:  The audio sink sample rate is set at the rate that your
sound-card uses.  the channel rate is resampled to the sample rate of
your sound-card.  set interpolation in the resampler block to the
sound-card's sample rate.

Bandwidth:  I leave mine at around 25khz although this is more than wide
enough.  12.5khz I think is standard and I think FCC is trying to shrink
it down to 6.25khz (maybe already have?).  What is very important is to
get the active carrier frequency exactly at zero on the FFT Plot (you'll
see how much bandwidth is being used) using the fine offset.

Hope this helps.
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