[Lf] The best LF Receivers]

[Andre' Kesteloot]

Talbot Andrew wrote:

> Alternatively, if you want to do a bit more analogue signal processing
> to get some good filtering of adjacent strong signals, and have fun
> building a custom top grade receiver :-
>
> Use a DDS derived LO locked to a master reference to mix up to an IF of
> 455 kHz.  Get hold of some mechanical filters with 300 Hz bandwidth or
> lower and and bandpass filter / amplify.  Mix back down with a 456k BFO,
> again derived from the same reference to give a signal centred on 1kHz.
> A sampling mixer could be used here driven with 8kHz at this is easy to
> get from a 5 or 10MHz reference by division without resorting to a
> second DDS or synth.   The output can be fed to an audio amp for
> inefficient ear / brain based reception and is the ideal centre
> frequency for off-the-shelf Soundcard software such as PSK31,  SPECTRAN
> etc.  The -60dB  bandwidth has to be 400 Hz or less..
>
> Undersample the 1kHz in a 12 - 16 bit A/D converter, at a clock rate of
> 800Hz (also locked to the reference) which will have the effect of
> apparantly mixing the signal down by the clock to the band 0 - 400 Hz.
> For looking at signals in the noise, even 8 bit sampling will work
> adequately, the greater dynamic range is only relevant when the S/N is
> high enough to require this level of digitisation.   Feed the digitised
> samples to a PC via the serial port and write (or get written) custom
> software that works on this data instead of all the complications of
> using the Soundcard.   Alternatively, sample in I/Q at 1kHz, mixing down
> to zero IF in the process.
>
> Believe me, driving the Soundcard is by far the most complex part of
> doing any DSP work on a PC and puts off many software authors who want
> to concentrate on writing decent software than interfacing to Windows !
> It is so messy being forced to use Windows that I haven't even tried
> going this route.   When time and other projects permit, I will be
> making an LF Rx to this design and writing my own software for display
> and filtering that will run on any old PC using DOS rather than having
> to use the latest Win 95 technology just to drive the A/D converter.
>
> In the past, using an 8 bit A/D based on a 16C71 PIC plus PC I have seen
> signals that are -20dB S/N in a 300Hz CW filter, using custom
> Spectrogram type software written in the Basic programming language.
>
> Andy  G4JNT
>
> > ----------
> > From:         Klaus von der Heide[SMTP:v.d.heide at on-line.de]
> > Reply To:     rsgb_lf_group at blacksheep.org
> > Sent:         2000-07-06 10:01
> > To:   rsgb_lf_group at blacksheep.org
> > Subject:      Re: LF: What is the best RX for 136 kHz?
> >
> > Date sent:            Wed, 05 Jul 2000 11:29:11 +0200
> > From:                 Alberto di Bene <dibene at usa.net>
> > Organization:         Undisclosed
> > To:                   rsgb_lf_group at blacksheep.org
> > Subject:              Re: LF: What is the best RX for 136 kHz?
> > Send reply to:        rsgb_lf_group at blacksheep.org
> >
> >
> > Klaus, DJ5HG wrote:
> >
> > > > After an input filter a low-noise operational amplifier
> > > > is followed by an analog to digital converter. The receiver
> > > > structure, the demodulation, noise blanking etc... is entirely
> > > > programmed on a DSP or on one of the new Million-Gates FPGAs.
> >
> > Alberto, I2PHD wrote:
> >
> > > I substantially agree, with a precisation :
> > > depending on the type of demodulation you wish to do on the signal,
> > > having the I and Q components is of great help. So, why not mix down
> > > the 137 kHz signal to baseband, using a couple of mixers fed with
> > > an oscillator that produce two signals in quadrature ? Some DDS have
> > > this capability, so the +- 90 degrees shifting could also be done
> > > digitally, reducing the analog part of the receiver to the bare
> > minimum.
> > > Then the two I and Q components could be fed to the left and right
> > > inputs of a stereo sound card, and all the processing done on a
> > Pentium,
> > > which nowadays rivals in processing speed with DSPs that only a few
> > years
> > > ago were considered state of the art.
> > > Just food for thought...
> >
> >
> > Yes, that is a good design if the hardware effort of the whole system
> > is to be minimized. Also, if you want to use one of the well-known
> > DSP-cards as the DSP56002EVM that is the best solution.
> > The DDS-module of your external front end should be clocked
> > synchronously with the sampling rate of the analog to digital
> > converter. That is a little technical problem when you use the
> > sound card. The A/D-converter of the EVM-board can be clocked from
> > an external clock.
> > The second point in favour of the direct way of sampling the 136 kHz
> > is the better linearity of the digital mixer.
> > That are the reasons why I prefer to do the DDS and quadrature mix
> > down to baseband by the DSP. Your design in principle is the same
> > as mine, only the selection what is done in hardware or software
> > is different with the result of minimum extra hardware in the case
> > of the sound card, nice!
> >
> > 73 de Klaus, DJ5HG
> >
> >
>
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