Hi Matt, The Termination Insensitive Amplifier is a transistor circuit that uses feedback to present 50 ohms to the mixer IF port across a wide range of frequencies.
I've measured the return loss (i.e. match compared to a pure 50 ohm resistive load) of the input port of the TIA to be about 20dB between the IF frequency and twice the LO freq. All - a little about the philosophy and aims of this project: I'm not shooting for high end RF performance, e.g. contest grade IP3, low phase noise, high ACR, high tx power, multi-band operation. There are others who can do that much better than I. Rather, I want to use this radio to demonstrate advanced new VHF DV ideas, for example: i) completely open source (hardware, software, codec, modem, protocol stack) VHF DV. ii) Diplexor free TDMA repeaters on low cost hardware iii) variable bit rate/quality speech codecs iv) Diversity rx to address fading v) a 10dB gain over analog FM and current, 1st gen VHF DV systems (i) to (v) break a lot of new ground, and are "enough" for any one project. To demonstrate them I feel I need a custom radio: for example TDMA timing means we need control by a bare metal uC (no operating system) to get the timing right. I also need an up-conversion/SDR tx rather than direct FM to make sure the modem is done right. Cheers, David On 29/10/15 15:26, Matthew Cook wrote: > 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] > <mailto:[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 > >>>> > >>>> > > ------------------------------------------------------------------------------ > >>>> _______________________________________________ > >>>> Freetel-codec2 mailing list > >>>> [email protected] > <mailto:[email protected]> > >>>> https://lists.sourceforge.net/lists/listinfo/freetel-codec2 > >>>> > >> > > ------------------------------------------------------------------------------ > >> _______________________________________________ > >> Freetel-codec2 mailing list > >> [email protected] > <mailto:[email protected]> > >> https://lists.sourceforge.net/lists/listinfo/freetel-codec2 > >> > > > > > ------------------------------------------------------------------------------ > _______________________________________________ > Freetel-codec2 mailing list > [email protected] > <mailto:[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
