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 >>>> >>>> ------------------------------------------------------------------------------ >>>> _______________________________________________ >>>> 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
