Re: [Freetel-codec2] gain before ADC

Wed, 28 Oct 2015 17:02:01 -0700

Not direct conversion Matthew- that is a superhet- David will undersample the IF.
On 29/10/2015 10:42 AM, Matthew Cook wrote:
David,

Ok that's just a straight direct conversion receiver.  As Helmut has said this is a different beast.  The majority of your design decisions will be made in the analogue world.

Here's a few good worked examples for you to look at, it doesn't matter these are for HF the same rule applies you just need too substitute for the components you can source at your frequency of interest.


Somewhere I've got a copy of Doug DeMaw (W1FB) receiver design book, I must try to find it and let you have a read.  Dougs book discusses the trade off's moving amps and filters around the mixer etc.

Oh and with your ring diode mixer, look at putting a diplexer in there to terminate all the unwanted energy in a resistor and pick off the band of interest before feeding to your post mixer amp.  Lots of references on the net for doing this.

73

Matthew
VK5ZM

On 29 October 2015 at 07:30, David Rowe <[email protected]> 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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-- 
- 
Glen English
RF Communications and Electronics Engineer

CORTEX RF
&
Pacific Media Technologies Pty Ltd

ABN 40 075 532 008

PO Box 5231 Lyneham ACT 2602, Australia.
au mobile : +61 (0)418 975077 


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