Re: [Freetel-codec2] gain before ADC

Wed, 28 Oct 2015 23:33:51 -0700

adl5350- and it is cheap.

and the performance is sensational .

but you must watch the harmonic products.

For a small radio I'd use the CML994
It's not bad actually.   a few caveat sand problems  but its $7 so who cares.





On 29/10/2015 3:56 PM, Matthew Cook wrote:
David,

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


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


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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-- 
- 
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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