Hi Alphonso,

well, your OFDM mod block uses the whole nyquist bandwidth of your
sampled signal, ie. at the output of the OFDM mod block, all frequencies
in [-pi;pi] (relative frequencies, as sampling rate doesn't really exist
in this world) are used. After the resampler, only frequencies in
[-(11/48)pi;(11/48)pi] are occupied. You feed that signal directly into
the OFDM demod block. How is that block supposed to know to only look
into that sub band? You will have to resample exactly inversely to your
TX resampler, before trying to demodulate that.

You still have to move the upsampled signal from 0 Hz to your desired
center frequency, and then you'll have to convert to a real signal,
before being able to feed things to a sound card. On the receive side,
you'd need to do the inverse.

Best regards,
Marcus
On 06/05/2015 10:15 AM, bh...@web.de wrote:
> hi felix,
> i have tried it with the resampler block but it didn't work well.
>  
> on the tx side i have used it to resample from 11khz, which i need for
> my 10khz bandwidth, to 48khz. on the rx side i can see the spectrum
> with 48khz and the bandwidth of my signal still remains at 10khz (as
> requested). the problem is that i can't detect any signal at rx side.
> when i adjust both parts (tx and rx) with a sampling rate at 11khz it
> works.
>  
> for better understanding i have attached my flowgraphs and sink outputs.
>  
> alphonso
> *Gesendet:* Donnerstag, 04. Juni 2015 um 23:40 Uhr
> *Von:* "Wunsch, Felix (CEL)" <felix.wun...@kit.edu>
> *An:* "bh...@web.de" <bh...@web.de>, "discuss-gnuradio@gnu.org"
> <discuss-gnuradio@gnu.org>
> *Betreff:* Re: [Discuss-gnuradio] problem with ofdm bandwidth and
> sampling rate
>  
> On 03.06.2015 17:29, bh...@web.de wrote:
>
>     hello felix,
>     >>please remember to always reply to the list.
>     i'm sorry, i'm new at this. i have checked my configuration and i
>     hope it works the next time.
>      
>     to mix up the baseband signal to 12khz or higher to short wave
>     (3-30mhz) should be the same procedure. i have tested it by using
>     a multiply block and a signal source block as the carrier frequency.
>      
>     but there is an other question: if i configured my system as
>     described, can i be sure that the bandwidth of each sub carrier
>     has the bandwidth and symbol duration as shown by the equations?
>      
>     used_bw = occupied_carriers * df
>     df = used_bw / occupied_carriers = 10khz / 228
>     df = 43.86hz   -->   tsym = 22.8ms
>      
>     it's very confusing why i have to adjust my used bw by changing
>     the sampling rate. when i follow the equations and the standard i
>     choose one mode (RM A, SO 3). that means i have to use 228 carriers
>     and have a symbol duration with 24ms which leads to 41.66hz
>     carrier spacing. so the used bw is 228 * 41.66hz = 9.5khz (approx.
>     10khz). this bw should be fix and should not change by the
>     sampling rate.
>      
>     sorry for the maybe silly questions but i want to keep on that stuff.
>
> Well, the measurable on-air bandwidth is always defined by the
> speed/rate at which your soundcard processes your digital samples. If
> you remember that B ~ 1/T and that the rate of your soundcard controls
> T it becomes obvious that changing the sample rate changes the
> bandwidth. If you don't want that, you could use a resampler that
> adapts its resampling ratio to the sampling rate of the soundcard.
>
> Felix
>
>      
>     @Marcus
>     you mean if there were pilots for a channel estimation in the
>     transmitted signal?
>      
>     alphonso
>      
>
>  
> -- 
> Karlsruhe Institute of Technology (KIT)
> Communications Engineering Lab (CEL)
>
> Felix Wunsch, M. Sc.
> Research Associate
>
> Kaiserstraße 12
> Building 05.01
> 76131 Karlsruhe
>
> Phone: +49 721 608-46276
> Fax: +49 721 608-46071
> E-Mail: felix.wun...@kit.edu
>
> www.cel.kit.edu
>
> KIT -- University of the State of Baden-Württemberg and
> National Laboratory of the Helmholtz Association
>
>
>
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