Problem with ofdm bandwidth and sampling rate


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Hi Alphonso,

I implemented a transmitter for the DRM(30) standard in GNU Radio and
published it on github (https://github.com/kit-cel/gr-drm).

To (partly) answer your questions:

  1. DRM only accepts specific audio sample rates, make sure to read the
    standard documents carefully.
  2. Mixing is usually done by the USRP. You as the user only deal with
    the complex baseband centered around DC. 12 kHz would also be a VERY low
    carrier frequency, short-wave radio usually uses frequencies between 3
    and 30 MHz.

–Felix

On 03.06.2015 12:57, [email protected] wrote:

everything works fine but there is a problem with the spectrum of this
should i use a resampler block because there is a problem with
nyquist? how can i handle this problem?
2) the same problem goes with the output. if i mix up the ofdm signal
to a carrier frequency (for example 12khz) how can i handle this?
there is also a problem with nyquist.
thanks in advance!
alphonso


Karlsruhe Institute of Technology (KIT)
Communications Engineering Lab (CEL)

Felix W., M. Sc.
Research Associate

Kaiserstra??e 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-W??rttemberg and
National Laboratory of the Helmholtz Association

Hi,

please remember to always reply to the list.

If you mix your baseband signal in a way that no part of your spectrum
includes negative frequencies BEFORE doing the complex-to-real
conversion, this should work. At least from a signal processing
perspective. I never used my soundcard as a transmitter, so I can’t
really help you with that.

–Felix

On 03.06.2015 15:02, [email protected] wrote:

were problems with licensing and changed or removed functions in
2) i don’t have an USRP. i want to use a sound card to get my signal
Hi Alphonso,
low carrier frequency, short-wave radio usually uses frequencies
standard. so my first try was to implement the transmitter with
with the parameters above and the wish of a bandwidth of 10khz (to
this? there is also a problem with nyquist.
Kaiserstra??e 12
National Laboratory of the Helmholtz Association


Karlsruhe Institute of Technology (KIT)
Communications Engineering Lab (CEL)

Felix W., M. Sc.
Research Associate

Kaiserstra??e 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-W??rttemberg and
National Laboratory of the Helmholtz Association

I’d also like to point out that you should use an oscilloscope to verify
that a sine with a 17 kHz frequency has the same peak-to-peak voltage as
one with the same digital amplitude at 7 kHz – sound cards do have nice
DACs (bitdepth-wise), but the analog filtering that you must do after
DAC’ing tends to be implemented in a one-stage passive way (at least for
the cheap consumer-grade sound cards I’ve encountered), which will not
make your spectrum pretty.
However, you said there’s an existing Matlab implementation – was the
“Soundcard channel” measured with respect to frequency selectivity?

Best regards,
Marcus

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