David,

If you're going to use a diode mixer then terminating *is* important, for
your amusement/reading;

http://www.robkalmeijer.nl/techniek/electronica/radiotechniek/hambladen/qex/2001/05_06/page45/index.html

There are termination insensitive mixers to go with your termination
insensitive amplifier;

https://www.miteq.com/docs/MITEQ_Mixers_TermInsensitve.pdf

But they cost a lot and don't really get you that much further than say
terminating your mixer in a -6dB pad and living with the additional
conversion loss.

The mixer that Glen suggests is a nice unit with great specs, that sounds
like a serious leg up to me.

Also watch those IP3's from end to end.

73

Matthew
VK5ZM

On 29 October 2015 at 12:01, David Rowe <[email protected]> wrote:

> Hi Guys,
>
> Thanks again for all your tips and insight.  I'm following up yr
> references and doing lots of reading.
>
> Glen - I haven't decided if I'll undersample yet or use a 2nd mixer from
> 10.7MHz to low IF.
>
> Steve - I have built up a spreadsheet to work out cascaded NF, MDS, and
> ADC NF. Am working through each calculation step nice and slowly!  Also
> prototyping some of this and getting results a few dB from my
> calculations which is encouraging.
>
> Matt I'm using a termination insensitive amp at the mixer IF port, have
> measured it's return loss as 20dB between 10.7MHz and 300MHz (2LO is
> 272MHz).
>
> Glen - what problems should I look out for with a limiting amp? Any
> tests I can do to spot issues early?  I guess two-tone wouldn't work.
>
> Cheers,
>
> David
>
> On 29/10/15 10:28, glen english wrote:
> > Hi David
> > Glad my description of NF made sense.
> >
> > If your limiter is perfectly doing its job,  then yes 1 bit is all you
> need.
> >
> > I like seeing the xtal filter in there ! This reduces the  umming and
> > ahhhing by two to three orders of magnitude.
> >
> > Of course the filter stop-bands are finite, so you must consider this,
> also.
> >
> > I'm guessing you are going to undersample. Goes without saying that ADC
> > performance degrades quickly on super nyquist but this is unlikely to be
> > any issue in this design.
> >
> > Final sample rate needs to be at least two times the IF filter BW, 4x if
> > you want to not make life hell for your digital filters.
> >
> > My guess is, you could use a single  pin of the STM32 inconjunction with
> > some timer  and input compare block to  get a very nice and periodic 1
> > bit sampler.
> >
> > Dont expect miracles though. Analog limiters are dreadful things. if you
> > want some REALLY good limiter lessons  tuition , go look at the FM
> > limiter in a analog VIDEO CASSETTE RECORDER. They are the very best,
> > high bandwidth (10 MHz) limiters around....
> >
> > For a IF width of 15kHz, I'd suggest a SR of about 60 k. ( an enormous
> > undersample) ..That will help also preventing strong adjacent channels
> > aliasing.
> >
> > regards
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On 29/10/2015 8:00 AM, David Rowe wrote:
> >> Thank you Helmut, Glen, and Steve,
> >>
> >> 1/ Ok it's starting to make sense, lets see if I can work through a
> >> contrived example:
> >>
> >> We have a 16 bit sound blaster ADC, with a SNR of 6*15 = 90dB.  Its FSD
> >> is 1Vrms (10dBm). It samples at 48 kHz:
> >>
> >> The ADC noise floor is at 10-96=-86dBm.  This is in a Nyquist BW of
> >> 48kHz, so the noise power normalised to a 1Hz BW is -86-10*log10(24E3) =
> >> -129.8dBm/Hz.  The thermal noise floor is -174dBm/Hz so our ADC NF is
> >> 174-129.8 = 44.2dB.
> >>
> >> Help me understand what NF is too, and why a filter has a negative NF -
> >> it moves the signal closer to the thermal noise floor.
> >>
> >> 2/ I'm actually working on narrow band constant envelope radio, So:
> >>
> >>      BPF- LNA - Mixer - Xtal filter - limiting amp - ADC
> >>
> >> If I am sampling a constant envelope signal (FM, FSK, GMSK), what are
> >> the SNR/SFDR requirements for the ADC?  The limiting amp has removed all
> >> amplitude information.  So do we just need the sign bit of the ADC?
> >> Could we just sample the signal with a flip flop?
> >>
> >> Thanks,
> >>
> >> David
> >>
> >>
> >> On 28/10/15 17:03, glen english wrote:
> >>> Howdy
> >>> A good response Helmut
> >>>
> >>> The oversampling ratio  (OSR) , clock purity will dominate.
> >>>
> >>> What is your planned OSR , and sampler rate ?
> >>> quadruple the sample rate , gain a bit of ADC of course.
> >>>
> >>> With an OSR = 1, for -120dBm , and full scale of say +12dBm (roughly
> >>> what you have) , and 60dB of SNR will put your noise floor at 12-60 =
> >>> -48dBm, so you'll need 72dB of gain.  And so it will overload on the
> >>> slightest thing out there !
> >>>
> >>> With an OSR of 4x, you are 6dB or 1 bit better off, and so on. I know
> >>> you understand this stuff so I wont elaborate.
> >>>
> >>> The SFDR of the converter will dominate what it can USEFULLY hear,
> >>> because below the SFDR , REGARDLESS of the OSR, there will be all sorts
> >>> of funny converter artifacts, and the intermods will be there also.
> >>> So the SFDR , not the OSR ultimately determines the performance
> capability.
> >>>
> >>> IE the SNR might be 60dB, say a 10 bit converter,
> >>> if the OSR = 4 then you'll get 66dB SNR, BUTthe SFDR does not change,
> >>> the SFDR is still 60dB.  So you can improve dynamic range, but not the
> >>> SFDR. The SFDR, or more likely, where the third order two tone
> intermods
> >>> are, won't change. Some LT converters have incredibly good SFDRs via
> >>> internal digital dithering (later subtracted out in your receiver) .
> >>>
> >>> For my commercial SDR, I use a 12 bit converter at 200 Msps.
> >>> The SFDR is 96dB, approx.
> >>>
> >>> The converter input FSD is abotu +12dBm, so the IMD will be always
> 12-96
> >>> = -84dBm
> >>>
> >>> so if I want my IMD down at -120dBm, then I need 36dB gain in front of
> >>> the receiver.
> >>> With such a high OSR (200M/ 10k)=43dB , the SNR is off in the
> >>> stratosphere, but the IMD dominates....
> >>>
> >>> In my experience the SFDR is what will limit the sensitivity.
> >>>
> >>> Watch out for ALIASED noise also. don't forget your converter is also
> >>> equally (almost) bringing in noise form 2fs, 3fs 4fs etc. SO important
> >>> that the convertor is seeing a low pass (nyquist) or band pass filter
> >>> (super nyquist sampling)
> >>>
> >>> NOW what you can do is vary the voltage that the converter sees by
> >>> fiddlign with the termination and the nosie figure can be usefully
> >>> manipulated +/- 12dB (improved at the expense of full scale level)
> >>>
> >>> The noise figure of the converter is approx (the input level - the SNR)
> >>> - 174
> >>>
> >>> IE +12dBm FSD, SNR  = 70, noise floor = -58dBm.
> >>> Now, that is for a 200 Msps, or 100 Msps nyquist bandwidth, that is
> >>> 10log10(1e8) or 80dB
> >>> so -58 - 80 = -138dBm dBcHzSNR or 174 - 138 = 36dB.
> >>>
> >>>
> >>> cheers
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> On 28/10/2015 6:47 AM, David Rowe wrote:
> >>>> Hello Glen/Matt,
> >>>>
> >>>> I'm working on a VHF radio prototype for testing some of my open
> source
> >>>> DV ideas.  Could you pls explain how to work out the gain required in
> >>>> front of my ADC?
> >>>>
> >>>> For example if I have a MDS of -120dBm (0.224uV), and an ADC with a
> 3Vpp
> >>>> (1.06Vrms) clipping point, and SFDR of say 60dB.
> >>>>
> >>>> Is the gain rqd simply Av=1.06/0.224E-16?  That would mean the minimum
> >>>> signal would hit full scale on the ADC.  Perhaps we could scale that
> >>>> back by 60dB plus some margin such that the MDS is still a few dB
> above
> >>>> the floor of the ADC.
> >>>>
> >>>> I'm a bit mixed up by the idea of NF and ADCs.  A worked example
> would help.
> >>>>
> >>>> Anyone else on the list with receiver design skills, pls feel free to
> >>>> comment. If a good reference exists I'm happy to dig that up.
> >>>>
> >>>> Thanks,
> >>>>
> >>>> David
> >>>>
> >>>>
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> >>>>
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