Re: [Discuss-gnuradio] problem with ofdm bandwidth and sampling rate

[BHN66]

hi marcus,

>>You will have to resample exactly inversely to your TX resampler, before trying to demodulate that.

 

done... and it works! thanks for your explanation. i hope that will help me to better understand the background and to avoid new mistakes.

 

>>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.

 

that part still works but was not implemented to this flowgraph. i have used a complex signal that i have found on the internet and the mixing up/down procedure works fine. on the rx side i get a real signal so i have to convert it to a complex signal by a hilbert filter for further processing (--> ofdm demod block). is that ok or are there some sticking points which i have not considered at the moment?

 

thanks a lot for your help!

 

alphonso

 

Gesendet: Freitag, 05. Juni 2015 um 02:02 Uhr
Von: "Marcus Müller" <address@hidden>
An: address@hidden
Betreff: Re: [Discuss-gnuradio] problem with ofdm bandwidth and sampling rate

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, address@hidden 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)" <address@hidden>
An: "address@hidden" <address@hidden>, "address@hidden" <address@hidden>
Betreff: Re: [Discuss-gnuradio] problem with ofdm bandwidth and sampling rate

 

On 03.06.2015 17:29, address@hidden 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

 

 

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Communications Engineering Lab (CEL)

Felix Wunsch, M. Sc.
Research Associate

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Phone: +49 721 608-46276
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