Hi Alphonso,
keep in mind that if you don't resample on RX side your signal
processing has to deal with the oversampled signal. If you use the same
chain to demodulate your signal I can see why it only works when you
resample on RX side.
-- Felix
On 05.06.2015 10:15, [email protected] 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)" <[email protected]>
*An:* "[email protected]" <[email protected]>, "[email protected]"
<[email protected]>
*Betreff:* Re: [Discuss-gnuradio] problem with ofdm bandwidth and
sampling rate
On 03.06.2015 17:29, [email protected] 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:[email protected]
www.cel.kit.edu
KIT -- University of the State of Baden-Württemberg and
National Laboratory of the Helmholtz Association
--
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: [email protected]
www.cel.kit.edu
KIT -- University of the State of Baden-Württemberg and
National Laboratory of the Helmholtz Association
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