Hi David, I understood what you plan, but analogue down conversion is another story. The example I sent you should demonstrate how this attached system calculation tool works. It shows a VHF frontend for digital down conversion, i.e. the ADC has to be sampled with 300 Msps. The BPF of 144 to 146 MHz exhibits an insertion loss of 2 dB in the stop band and consequently a noise figure of 2 dB. In your case NF or MDS is mainly not determined by the ADC in the base band, but on all analogue stages in front. It depends e.g. on what kind of mixer you intend to use: balanced passive mixers exhibit an insertion loss of 7-8 dB (equal NF 7- 8 dB), while modern active mixer provide even conversion gain. BTW negative NF doesnt exist. I guess you meant negative gain (loss) of the BPF. Nevertheless you can use that SYScal tool of course also for your approach to calculate the necessary gain of preamp and IF amp.
73, Helmut, DC6NY -----Ursprüngliche Nachricht----- Von: David Rowe [mailto:[email protected]] Gesendet: Mittwoch, 28. Oktober 2015 22:00 An: [email protected] Betreff: Re: [Freetel-codec2] gain before ADC 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 >> >> ---------------------------------------------------------------------------- -- >> _______________________________________________ >> Freetel-codec2 mailing list >> [email protected] >> https://lists.sourceforge.net/lists/listinfo/freetel-codec2 >> > ---------------------------------------------------------------------------- -- _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2 ------------------------------------------------------------------------------ _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2
