On 12/5/2014 10:08 PM, rickman wrote:> On 12/5/2014 9:37 PM, Phil Hobbs wrote: >> On 12/5/2014 7:12 PM, George Herold wrote: >>> On Thursday, December 4, 2014 3:04:06 PM UTC-5, Phil Hobbs wrote: >>>> Hi, all, >>>> >>>> I have a gig coming in that will have me revisiting my thesis research >>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) >>>> >>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>> flipflop). With quite a lot of calibration, that got me a 13-bit, >>>> 2-pi, >>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit >>>> came from deciding which null to head for, which is why I needed the >>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>> card, and (eventually) worked great. I published one of my only two >>>> instruments papers on it (this was before I realized the total futility >>>> of almost all instruments papers). >>> >>> Hi Phil, I've been sorta half following this thread, >>> and I wonder if you could tell me what a nulling type phase digitizer >>> is? >>> (I "turn" the phase knob of a lockin type mixer/detector till the >>> signal goes to zero?) >>> Maybe just a reference to your instrument paper...? >>> >>> George H. >> >> >> Hi, George, >> >> The idea is to use a phase detector wrapped in a successive >> approximation loop. Like other SAR ADCs, you run the register to null >> out the error signal to N bits' accuracy, and read off the value from >> the DAC control word corresponding to the null. In this case, the 'DAC' >> is a phase shifter. > > How do you get the amplitude? Don't you need to work the two together? > Or I guess you can get the phase from the DAC setting at null and then > calculate the amplitude from the depth of the null? >The amplitude comes from the DLVA output, suitably calibrated. One of my favourite ways to calibrate log detectors is to hit a crystal with a long tone burst at its series resonance, then turn that off and use the ring-down transient. It's very exponential, so the desired log characteristic turns into a straight line. A 10 MHz crystal with a Q of 10**6 decays with a time constant of about 0.12 dB/ms, which is pretty convenient. Most DLVAs work by cascading a bunch of differential pairs, each with a gain of about 10 dB, detecting the output of each stage, and summing the results. Every time one stage saturates, the gain of the whole drops by 10 dB. That means that the output is ideally a piecewise-linear approximation to the logarithm of the input level. The detection is often done by summing the tail currents of all the diff pairs. When a stage is running linear, the junction of the two emitters sits pretty nearly still, but once the stage saturates, it bounces up and down every half cycle, and the average current goes up, making it a reasonably decent amplitude detector. Have a look at the SA604A datasheet. Cheers Phil Hobbs -- Dr Philip C D Hobbs Principal Consultant ElectroOptical Innovations LLC Optics, Electro-optics, Photonics, Analog Electronics 160 North State Road #203 Briarcliff Manor NY 10510 hobbs at electrooptical dot net http://electrooptical.net
DDS wisdom
Started by ●December 4, 2014
Reply by ●December 5, 20142014-12-05
Reply by ●December 6, 20142014-12-06
On 12/5/2014 10:19 PM, John Larkin wrote:> On Fri, 05 Dec 2014 21:29:46 -0500, Phil Hobbs > <hobbs@electrooptical.net> wrote: > >> On 12/5/2014 7:44 PM, Joe Gwinn wrote: >>> In article <5481273A.6010107@electrooptical.net>, Phil Hobbs >>> <hobbs@electrooptical.net> wrote: >>> >>>> On 12/4/2014 7:44 PM, Joe Gwinn wrote: >>>>> In article <cYydnTNwFPGvIx3JnZ2dnUU7-W-dnZ2d@supernews.com>, Phil Hobbs >>>>> <pcdhSpamMeSenseless@electrooptical.net> wrote: >>>>> >>>>>> Hi, all, >>>>>> >>>>>> I have a gig coming in that will have me revisiting my thesis research >>>>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) >>>>>> >>>>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>>>> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, >>>>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit >>>>>> came from deciding which null to head for, which is why I needed the >>>>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>>>> card, and (eventually) worked great. I published one of my only two >>>>>> instruments papers on it (this was before I realized the total futility >>>>>> of almost all instruments papers). >>>>>> >>>>>> The advantage of nulling detection is that you only need 1-D calibration >>>>>> tables for phase shift and amplitude, whereas getting that sort of >>>>>> accuracy with I/Q techniques requires a 2-D calibration table, which is >>>>>> a gigantic pain. >>>>>> >>>>>> I need to do this again, 2015 style. The speed requirements are set by >>>>>> the acoustic delay in the AO scanner, so 50-100 ks/s is about all I can >>>>>> use. Rather than all that squishy analogue stuff, I'm planning to do >>>>>> the SAR in software and use a pair of AD9951 DDS chips, one to generate >>>>>> the desired signal and one to be the phase shifted comparison signal. >>>>>> >>>>>> So far so straightforward. >>>>>> >>>>>> What I'm less sure about is being able to keep the two channels >>>>>> sufficiently isolated to be able to maintain 12 or ideally 14 bits of >>>>>> phase accuracy. Even with a full-scale input, I'll need 85 dB of >>>>>> isolation to get 14 bits, and it gets harder with weaker signals. >>>>>> (There'll be a DLVA/limiter ahead of the phase detector, which will help.) >>>>>> >>>>>> I've never used DDSes before, and I'd appreciate some wisdom from folks >>>>>> who have. How hard is that likely to be, and what should I particularly >>>>>> watch out for? >>>>> >>>>> DDSs have a forest of rational-multiple (but not necessarily harmonic) >>>>> spurs, and it can be difficult to get them below -60 dBc unless you can >>>>> place some restrictions on the frequency resolution. >>>> >>>> I can pick my IF to be anything I like, which I expect will help. >>>>> >>>>> Also beware phase jumps when the DDS phase wheel rolls over. >>>> >>>> Could you elaborate a bit? I thought the whole idea was to keep phase >>>> continuity. >>> >>> Lots of people have elaborated on the point, so I won't recite it. >>> >>> It's true that choosing tuning words with the lower k (one chooses a >>> suitable value such that nothing is truncated in lookup tables) bits >>> zero will greatly reduce the number of spurs, and get rid of the phase >>> bump when the phase wheel rolls over, but there will still be lots of >>> spurs from the limited width of the lookup tables and DACs. >> >> If the 'hidden' bits in the phase register are always zero, then the >> output of the DAC should be strictly periodic at f_out. That means that >> all, and I mean *all*, of the artifacts will be harmonics of f_out. >> Isn't that so? > > Sure. Absolutely everything repeats at Fout.No, that's not correct. All of the phase words will repeat every cycle of Fout only if the modulus is a multiple of the phase step which in the case of a 2^N modulus means the step size is power of 2 as well. Or in other words, the clock rate is a power of 2 harmonic of Fout or octaves. Having the lower order bits of the phase step be zero is not even necessary for the phase pattern to repeat each cycle of Fout. If the phase step word were all zeros except the largest bit that is truncated is a 1, the phase pattern will repeat each cycle of Fout, but there will be spurs. The same spurs you would have if the clock rate were halved for the same Fout. This idea of arbitrarily truncating the phase value to suit a lookup table is a bit simplistic. The only real limitation is the DAC resolution. The generation of a sine value to input to the DAC can incorporate a lot of bits from the phase accumulator and does not need to be limited by the size of an actual lookup table. BTW, has anyone considered the CORDIC algorithm for generating the sine function? I know it is used for DDS of sine waves, but I have not seen an analysis. -- Rick
Reply by ●December 6, 20142014-12-06
On 12/5/2014 10:25 PM, Phil Hobbs wrote:> On 12/5/2014 10:08 PM, rickman wrote: >> On 12/5/2014 9:37 PM, Phil Hobbs wrote: >>> On 12/5/2014 7:12 PM, George Herold wrote: >>>> On Thursday, December 4, 2014 3:04:06 PM UTC-5, Phil Hobbs wrote: >>>>> Hi, all, >>>>> >>>>> I have a gig coming in that will have me revisiting my thesis research >>>>> from nearly 30 years ago, on interferometric laser microscopes. >>>>> (Fun.) >>>>> >>>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>>> flipflop). With quite a lot of calibration, that got me a 13-bit, >>>>> 2-pi, >>>>> 50 ks/s phase measurement that I was pretty happy with. (The extra >>>>> bit >>>>> came from deciding which null to head for, which is why I needed the >>>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>>> card, and (eventually) worked great. I published one of my only two >>>>> instruments papers on it (this was before I realized the total >>>>> futility >>>>> of almost all instruments papers). >>>> >>>> Hi Phil, I've been sorta half following this thread, >>>> and I wonder if you could tell me what a nulling type phase digitizer >>>> is? >>>> (I "turn" the phase knob of a lockin type mixer/detector till the >>>> signal goes to zero?) >>>> Maybe just a reference to your instrument paper...? >>>> >>>> George H. >>> >>> >>> Hi, George, >>> >>> The idea is to use a phase detector wrapped in a successive >>> approximation loop. Like other SAR ADCs, you run the register to null >>> out the error signal to N bits' accuracy, and read off the value from >>> the DAC control word corresponding to the null. In this case, the 'DAC' >>> is a phase shifter. >> >> How do you get the amplitude? Don't you need to work the two together? >> Or I guess you can get the phase from the DAC setting at null and then >> calculate the amplitude from the depth of the null? >> > The amplitude comes from the DLVA output, suitably calibrated. One of > my favourite ways to calibrate log detectors is to hit a crystal with a > long tone burst at its series resonance, then turn that off and use the > ring-down transient. It's very exponential, so the desired log > characteristic turns into a straight line. A 10 MHz crystal with a Q of > 10**6 decays with a time constant of about 0.12 dB/ms, which is pretty > convenient. > > Most DLVAs work by cascading a bunch of differential pairs, each with a > gain of about 10 dB, detecting the output of each stage, and summing the > results. Every time one stage saturates, the gain of the whole drops by > 10 dB. That means that the output is ideally a piecewise-linear > approximation to the logarithm of the input level. > > The detection is often done by summing the tail currents of all the diff > pairs. When a stage is running linear, the junction of the two emitters > sits pretty nearly still, but once the stage saturates, it bounces up > and down every half cycle, and the average current goes up, making it a > reasonably decent amplitude detector.So the amplitude is measured with a totally separate circuit? I had the impression you were getting both phase and amplitude from the same circuit. My bad. -- Rick
Reply by ●December 6, 20142014-12-06
Bill Sloman <bill.sloman@gmail.com> wrote:> On Saturday, 6 December 2014 00:08:38 UTC+11, Lasse Langwadt Christensen wrote: >> Den fredag den 5. december 2014 13.54.58 UTC+1 skrev Bill Sloman: >>> On Friday, 5 December 2014 14:27:37 UTC+11, Phil Hobbs wrote: >>>> On 12/4/2014 7:10 PM, rickman wrote: >>>>> On 12/4/2014 3:04 PM, Phil Hobbs wrote: >>>>>> Hi, all, >>>>>> >>>>>> I have a gig coming in that will have me revisiting my thesis research >>>>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) >>>>>> >>>>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>>>> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, >>>>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit >>>>>> came from deciding which null to head for, which is why I needed the >>>>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>>>> card, and (eventually) worked great. I published one of my only two >>>>>> instruments papers on it (this was before I realized the total futility >>>>>> of almost all instruments papers). >>>>>> >>>>>> The advantage of nulling detection is that you only need 1-D calibration >>>>>> tables for phase shift and amplitude, whereas getting that sort of >>>>>> accuracy with I/Q techniques requires a 2-D calibration table, which is >>>>>> a gigantic pain. >>>>>> >>>>>> I need to do this again, 2015 style. The speed requirements are set by >>>>>> the acoustic delay in the AO scanner, so 50-100 ks/s is about all I can >>>>>> use. Rather than all that squishy analogue stuff, I'm planning to do >>>>>> the SAR in software and use a pair of AD9951 DDS chips, one to generate >>>>>> the desired signal and one to be the phase shifted comparison signal. >>>>>> >>>>>> So far so straightforward. >>>>>> >>>>>> What I'm less sure about is being able to keep the two channels >>>>>> sufficiently isolated to be able to maintain 12 or ideally 14 bits of >>>>>> phase accuracy. Even with a full-scale input, I'll need 85 dB of >>>>>> isolation to get 14 bits, and it gets harder with weaker signals. >>>>>> (There'll be a DLVA/limiter ahead of the phase detector, which will >>>>>> help.) >>>>>> >>>>>> I've never used DDSes before, and I'd appreciate some wisdom from folks >>>>>> who have. How hard is that likely to be, and what should I particularly >>>>>> watch out for? >>>>> >>>>> I've read all the posts so far and it seems you are generating a VHF >>>>> sine wave to compare to a VHF signal you wish to measure the phase and >>>>> amplitude of. I think I get that. But it seems the modulation of the >>>>> VHF signal is pretty low rate so that 50 kSPS is good enough. >>>>> >>>>> Then you ask about how to maintain enough isolation to preserve 14 bits >>>>> of phase measurement. I think the isolation you are worried about it in >>>>> the VHF range, no? That is the domain of RF design and not at all >>>>> trivial. I think you will need to provide more info on design specifics. >>>>> >>>>> I'm not clear on how you plan to do the phase detector. Is this just >>>>> subtracting the reference signal from the signal being measured? You >>>>> then scan the phase of the reference to find the null, scan the >>>>> amplitude of the reference to optimize the null and then possibly >>>>> repeat? Otherwise I'm not sure how you get both phase and amplitude out >>>>> of this. >>>>> >>>> The phase detector will probably be a diode bridge type, e.g. a Mini >>>> Circuits MPD-1. It's approximately a multiplier. >>> >>> Why not use a real multiplier? Analog Devices have a couple of pretty >>> good analog multiplier chips. AD734 and AD834 come to mind. >>> >>> And if you are working at a fixed frequency, running the DDS staircase >>> approximation to a sine wave through an integrator (with the right >>> gain) turns it into a straight-line interpolation approximation to a >>> sine wave, which is a lot nicer, (and slightly easier to filter). >> >> an integrator is just a bad filter, why should a bad filter in front of >> a good filter suddenly make things better? > > An integrator converts the sawtooth error signal implicit in a staircase > approximation to a sine wave to a series of much smaller of continuous arcs. > > You've still got a high frequency error signal to filter out, but pretty > much all of the higher frequency content (at multiples of DDS update rate) has gone away. > > The integrator isn't functioning as a bad filter here - it's a device to > improve the quality of the approximation to the desired sine wave.But wouldn't any low-pass filter also do the same thing (ie: convert vertical parts of the waveform to less vertical parts)? Perhaps a low-pass filter could be chosen which gives superior performance over a straight integrator.
Reply by ●December 6, 20142014-12-06
On Saturday, 6 December 2014 17:11:43 UTC+11, Ralph Barone wrote:> Bill Sloman <bill.sloman@gmail.com> wrote: > > On Saturday, 6 December 2014 00:08:38 UTC+11, Lasse Langwadt Christense=n wrote:> >> Den fredag den 5. december 2014 13.54.58 UTC+1 skrev Bill Sloman: > >>> On Friday, 5 December 2014 14:27:37 UTC+11, Phil Hobbs wrote: > >>>> On 12/4/2014 7:10 PM, rickman wrote: > >>>>> On 12/4/2014 3:04 PM, Phil Hobbs wrote: > >>>>>> Hi, all, > >>>>>>=20 > >>>>>> I have a gig coming in that will have me revisiting my thesis rese=arch> >>>>>> from nearly 30 years ago, on interferometric laser microscopes. (=Fun.)> >>>>>>=20 > >>>>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz b=y> >>>>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping=a> >>>>>> 13-bit successive approximation loop round it (AM2904 with an extr=a> >>>>>> flipflop). With quite a lot of calibration, that got me a 13-bit,=2-pi,> >>>>>> 50 ks/s phase measurement that I was pretty happy with. (The extr=a bit> >>>>>> came from deciding which null to head for, which is why I needed t=he> >>>>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPI=O> >>>>>> card, and (eventually) worked great. I published one of my only t=wo> >>>>>> instruments papers on it (this was before I realized the total fut=ility> >>>>>> of almost all instruments papers). > >>>>>>=20 > >>>>>> The advantage of nulling detection is that you only need 1-D calib=ration> >>>>>> tables for phase shift and amplitude, whereas getting that sort of > >>>>>> accuracy with I/Q techniques requires a 2-D calibration table, whi=ch is> >>>>>> a gigantic pain. > >>>>>>=20 > >>>>>> I need to do this again, 2015 style. The speed requirements are s=et by> >>>>>> the acoustic delay in the AO scanner, so 50-100 ks/s is about all =I can> >>>>>> use. Rather than all that squishy analogue stuff, I'm planning to=do> >>>>>> the SAR in software and use a pair of AD9951 DDS chips, one to gen=erate> >>>>>> the desired signal and one to be the phase shifted comparison sign=al.> >>>>>>=20 > >>>>>> So far so straightforward. > >>>>>>=20 > >>>>>> What I'm less sure about is being able to keep the two channels > >>>>>> sufficiently isolated to be able to maintain 12 or ideally 14 bits=of> >>>>>> phase accuracy. Even with a full-scale input, I'll need 85 dB of > >>>>>> isolation to get 14 bits, and it gets harder with weaker signals. > >>>>>> (There'll be a DLVA/limiter ahead of the phase detector, which wil=l> >>>>>> help.) > >>>>>>=20 > >>>>>> I've never used DDSes before, and I'd appreciate some wisdom from =folks> >>>>>> who have. How hard is that likely to be, and what should I partic=ularly> >>>>>> watch out for? > >>>>>=20 > >>>>> I've read all the posts so far and it seems you are generating a VH=F> >>>>> sine wave to compare to a VHF signal you wish to measure the phase =and> >>>>> amplitude of. I think I get that. But it seems the modulation of =the> >>>>> VHF signal is pretty low rate so that 50 kSPS is good enough. > >>>>>=20 > >>>>> Then you ask about how to maintain enough isolation to preserve 14 =bits> >>>>> of phase measurement. I think the isolation you are worried about =it in> >>>>> the VHF range, no? That is the domain of RF design and not at all > >>>>> trivial. I think you will need to provide more info on design spec=ifics.> >>>>>=20 > >>>>> I'm not clear on how you plan to do the phase detector. Is this ju=st> >>>>> subtracting the reference signal from the signal being measured? Y=ou> >>>>> then scan the phase of the reference to find the null, scan the > >>>>> amplitude of the reference to optimize the null and then possibly > >>>>> repeat? Otherwise I'm not sure how you get both phase and amplitud=e out> >>>>> of this. > >>>>>=20 > >>>> The phase detector will probably be a diode bridge type, e.g. a Mini==20> >>>> Circuits MPD-1. It's approximately a multiplier. > >>>=20 > >>> Why not use a real multiplier? Analog Devices have a couple of pretty > >>> good analog multiplier chips. AD734 and AD834 come to mind. > >>>=20 > >>> And if you are working at a fixed frequency, running the DDS staircas=e> >>> approximation to a sine wave through an integrator (with the right > >>> gain) turns it into a straight-line interpolation approximation to a > >>> sine wave, which is a lot nicer, (and slightly easier to filter). > >>=20 > >> an integrator is just a bad filter, why should a bad filter in front o=f> >> a good filter suddenly make things better?=20 > >=20 > > An integrator converts the sawtooth error signal implicit in a staircas=e> > approximation to a sine wave to a series of much smaller of continuous =arcs.> >=20 > > You've still got a high frequency error signal to filter out, but prett=y> > much all of the higher frequency content (at multiples of DDS update ra=te) > > has gone away.> >=20 > > The integrator isn't functioning as a bad filter here - it's a device t=o> > improve the quality of the approximation to the desired sine wave. > =20 > But wouldn't any low-pass filter also do the same thing (ie: convert > vertical parts of the waveform to less vertical parts)? Perhaps a low-pa=ss> filter could be chosen which gives superior performance over a straight > integrator.Interesting theoretical question. Part of engineering is explaining what yo= u are doing, and why you are doing it, in terms that even a manager can und= erstand. My contention would be that the integrator gets rid of the high-slew rate c= omponent of the error signal in the first stage of filtering, which makes a= difference to any active filter in a way that filter construction software= doesn't usually capture. LTSpice does, but making a stair-case approximati= on to a sine wave as a test signal might be tedious. --=20 Bill Sloman, Sydney
Reply by ●December 6, 20142014-12-06
On Fri, 5 Dec 2014 13:47:31 -0800, dplatt@coop.radagast.org (Dave Platt) wrote:>In article <m5t6j4$ur3$1@dont-email.me>, rickman <gnuarm@gmail.com> =wrote:> >>The rule is to use "an" when the following word starts with a vowel=20 >>*sound*, like honor and... istorical. lol While honor has an unsound =H=20>>and so starts with a vowel sound, historical starts with a sounded H.=20 >>But when used with "an" the H sound is truncated so it then fits the=20 >>rule. Rather a way of backing into it, eh? I'm not saying this is=20 >>"correct". But personally I don't give a rats ass about "correctness"=20 >>in this case. >> >>Language is alive and rules change. This is one that is already fuzzy=20 >>and using "an historical" is within the fuzz factor these days. > >One thing I read years ago, was that the English language has fewer, >and less rigid grammatical rules and cases than many of the languages >to which it is related (e.g. Latin, Greek, French, German) because >there was quite a long period of time during which it was essentially >a peasant's language - the common tongue of the common folk - and far >more a spoken language than a written one. =20 > >Scholars and rulers tended to do their business in one of the >languages I mentioned above - these were the languages that were >written down, preserved, studied, prescribed, and criticized. >English? It's what those farmers and woodcutters speak, down at the >local pub... not a subject for serious study. So, without >school-marms cracking kids across the wrists with their rulers for >"mis-pronouncing", pronunciation and spelling did what they did based >on what seemed right to the speakers at the time. > >I suspect that early English started out almost as a pidgin, mixing >French and Scandinavian languages and grammers with Anglic and Saxon, >and went through the common process of developing into a creole and >then into a more standardized language. I've heard it described as >"the language which developed so that the sons of Norsemen who had >invaded and then settled down, could make dates with Saxon bar-maids >in town." > >And, as others have noted, English doesn't just borrow words and >meanings from other languages... it sneaks up on those languages in a >dark alley, clubs them into unconsciousness, and steals words out of >their pockets :-) > >So, pretty much by definition, "correct" pronunciation in English is >the way that a large fraction of the people pronounce. Pronunciation >rules are rather after-the-fact rationalizations.+2 Thanks for saying more eloquently what i wish i would have said. ?-) =20
Reply by ●December 6, 20142014-12-06
In article <54826A1A.9080101@electrooptical.net>, Phil Hobbs <hobbs@electrooptical.net> wrote:> On 12/5/2014 7:44 PM, Joe Gwinn wrote: > > In article <5481273A.6010107@electrooptical.net>, Phil Hobbs > > <hobbs@electrooptical.net> wrote: > > > >> On 12/4/2014 7:44 PM, Joe Gwinn wrote: > >>> In article <cYydnTNwFPGvIx3JnZ2dnUU7-W-dnZ2d@supernews.com>, Phil Hobbs > >>> <pcdhSpamMeSenseless@electrooptical.net> wrote: > >>> > >>>> Hi, all, > >>>> > >>>> I have a gig coming in that will have me revisiting my thesis research > >>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) > >>>> > >>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by > >>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a > >>>> 13-bit successive approximation loop round it (AM2904 with an extra > >>>> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, > >>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit > >>>> came from deciding which null to head for, which is why I needed the > >>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO > >>>> card, and (eventually) worked great. I published one of my only two > >>>> instruments papers on it (this was before I realized the total futility > >>>> of almost all instruments papers). > >>>> > >>>> The advantage of nulling detection is that you only need 1-D calibration > >>>> tables for phase shift and amplitude, whereas getting that sort of > >>>> accuracy with I/Q techniques requires a 2-D calibration table, which is > >>>> a gigantic pain. > >>>> > >>>> I need to do this again, 2015 style. The speed requirements are set by > >>>> the acoustic delay in the AO scanner, so 50-100 ks/s is about all I can > >>>> use. Rather than all that squishy analogue stuff, I'm planning to do > >>>> the SAR in software and use a pair of AD9951 DDS chips, one to generate > >>>> the desired signal and one to be the phase shifted comparison signal. > >>>> > >>>> So far so straightforward. > >>>> > >>>> What I'm less sure about is being able to keep the two channels > >>>> sufficiently isolated to be able to maintain 12 or ideally 14 bits of > >>>> phase accuracy. Even with a full-scale input, I'll need 85 dB of > >>>> isolation to get 14 bits, and it gets harder with weaker signals. > >>>> (There'll be a DLVA/limiter ahead of the phase detector, which will > >>>> help.) > >>>> > >>>> I've never used DDSes before, and I'd appreciate some wisdom from folks > >>>> who have. How hard is that likely to be, and what should I particularly > >>>> watch out for? > >>> > >>> DDSs have a forest of rational-multiple (but not necessarily harmonic) > >>> spurs, and it can be difficult to get them below -60 dBc unless you can > >>> place some restrictions on the frequency resolution. > >> > >> I can pick my IF to be anything I like, which I expect will help. > >>> > >>> Also beware phase jumps when the DDS phase wheel rolls over. > >> > >> Could you elaborate a bit? I thought the whole idea was to keep phase > >> continuity. > > > > Lots of people have elaborated on the point, so I won't recite it. > > > > It's true that choosing tuning words with the lower k (one chooses a > > suitable value such that nothing is truncated in lookup tables) bits > > zero will greatly reduce the number of spurs, and get rid of the phase > > bump when the phase wheel rolls over, but there will still be lots of > > spurs from the limited width of the lookup tables and DACs. > > If the 'hidden' bits in the phase register are always zero, then the > output of the DAC should be strictly periodic at f_out. That means that > all, and I mean *all*, of the artifacts will be harmonics of f_out. > Isn't that so?Almost. It helps a lot, but there are various non-linearities and jitter effects. Much of this is chip design specific, and vendor app notes are the best source on the oddities of their designs. It's a complex area, and there are many approaches to reducing the spurs. Some are built into the chips, and some can be applied in the surrounding circuitry and/or controlling firmware and/or software. Which methods are applicable will as always depend of application details. As a rule, chips are designed to a specific spur level, and all the various sources of spurs are balanced by design to yield roughly the same peak spur levels. In other words, no specific spur cause dominates.> > So, the question is if your application is bothered by a bunch of > > spurs, some near in, at about -60 dBc. This is the key analysis to > > perform. If the answer is no problem, then life is simple. If it is a > > problem, there is a longer discussion in store. > > -60 dBc is pretty crappy.Yes, and it causes much trouble in radar applications. But is it a problem for your application? And in what way? Given this, approaches can be suggested.> > ADI has a very good tutorial on DDS theory, "MT-085: Fundamentals of > > Direct Digital Synthesis (DDS)". I'd read it. > > Thanks. I did read it, but it didn't say what you said.Hmm. I must be misremembering where I saw that. I think they mention the wheel, but they may not go into what happens at rollover. I discovered the effect when analyzing why a prototype DMTD (Dual Mixer Time Difference) instrument was suffering periodic phase bumps. When one is measuring the performance of Rubidium clocks (10^-11 fractional frequency error), it doesn't take much. Anyway, the relevant data trove is at work, so it'l be a few days before I can reconstruct where I saw the business about bumps. Joe Gwinn
Reply by ●December 6, 20142014-12-06
On Friday, December 5, 2014 9:37:20 PM UTC-5, Phil Hobbs wrote:> On 12/5/2014 7:12 PM, George Herold wrote: > > On Thursday, December 4, 2014 3:04:06 PM UTC-5, Phil Hobbs wrote: > >> Hi, all, > >> > >> I have a gig coming in that will have me revisiting my thesis research > >> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) > >> > >> Back in the day, I made a nulling-type phase digitizer at 60 MHz by > >> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a > >> 13-bit successive approximation loop round it (AM2904 with an extra > >> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, > >> 50 ks/s phase measurement that I was pretty happy with. (The extra bit > >> came from deciding which null to head for, which is why I needed the > >> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO > >> card, and (eventually) worked great. I published one of my only two > >> instruments papers on it (this was before I realized the total futility > >> of almost all instruments papers). > > > > Hi Phil, I've been sorta half following this thread, > > and I wonder if you could tell me what a nulling type phase digitizer is? > > (I "turn" the phase knob of a lockin type mixer/detector till the signal goes to zero?) > > Maybe just a reference to your instrument paper...? > > > > George H. > > > Hi, George, > > The idea is to use a phase detector wrapped in a successive > approximation loop. Like other SAR ADCs, you run the register to null > out the error signal to N bits' accuracy, and read off the value from > the DAC control word corresponding to the null. In this case, the 'DAC' > is a phase shifter. > > I looked around for a copy of the instruments paper, but couldn't find > it--it predates my earliest digital archives, having been published in > 1987. It's at http://dx.doi.org/10.1063/1.1139391 . > > Cheers > > Phil HobbsGot it, and by 2-D you mean fixing the phase and measuring the amplitude and phase in both I/Q channels. I'll check out your paper on Monday. We have the whole series of RSI that a library was pitching. Makes great fodder for the "reading room." George H.> > > > -- > Dr Philip C D Hobbs > Principal Consultant > ElectroOptical Innovations LLC > Optics, Electro-optics, Photonics, Analog Electronics > > 160 North State Road #203 > Briarcliff Manor NY 10510 > > hobbs at electrooptical dot net > http://electrooptical.net
Reply by ●December 6, 20142014-12-06
On 12/6/2014 11:27 AM, George Herold wrote:> On Friday, December 5, 2014 9:37:20 PM UTC-5, Phil Hobbs wrote: >> On 12/5/2014 7:12 PM, George Herold wrote: >>> On Thursday, December 4, 2014 3:04:06 PM UTC-5, Phil Hobbs wrote: >>>> Hi, all, >>>> >>>> I have a gig coming in that will have me revisiting my thesis research >>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) >>>> >>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, >>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit >>>> came from deciding which null to head for, which is why I needed the >>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>> card, and (eventually) worked great. I published one of my only two >>>> instruments papers on it (this was before I realized the total futility >>>> of almost all instruments papers). >>> >>> Hi Phil, I've been sorta half following this thread, >>> and I wonder if you could tell me what a nulling type phase digitizer is? >>> (I "turn" the phase knob of a lockin type mixer/detector till the signal goes to zero?) >>> Maybe just a reference to your instrument paper...? >>> >>> George H. >> >> >> Hi, George, >> >> The idea is to use a phase detector wrapped in a successive >> approximation loop. Like other SAR ADCs, you run the register to null >> out the error signal to N bits' accuracy, and read off the value from >> the DAC control word corresponding to the null. In this case, the 'DAC' >> is a phase shifter. >> >> I looked around for a copy of the instruments paper, but couldn't find >> it--it predates my earliest digital archives, having been published in >> 1987. It's at http://dx.doi.org/10.1063/1.1139391 . >> >> Cheers >> >> Phil Hobbs > Got it, and by 2-D you mean fixing the phase > and measuring the amplitude and phase in both I/Q channels. > > I'll check out your paper on Monday. > We have the whole series of RSI that a library > was pitching. > Makes great fodder for the "reading room."Youch, that smarts. ;) (But not too inaccurate for the most part.) Cheers Phil Hobbs -- Dr Philip C D Hobbs Principal Consultant ElectroOptical Innovations LLC Optics, Electro-optics, Photonics, Analog Electronics 160 North State Road #203 Briarcliff Manor NY 10510 hobbs at electrooptical dot net http://electrooptical.net
Reply by ●December 6, 20142014-12-06
Bill Sloman <bill.sloman@gmail.com> wrote:> On Saturday, 6 December 2014 17:11:43 UTC+11, Ralph Barone wrote: >> Bill Sloman <bill.sloman@gmail.com> wrote: >>> On Saturday, 6 December 2014 00:08:38 UTC+11, Lasse Langwadt Christensen wrote: >>>> Den fredag den 5. december 2014 13.54.58 UTC+1 skrev Bill Sloman: >>>>> On Friday, 5 December 2014 14:27:37 UTC+11, Phil Hobbs wrote: >>>>>> On 12/4/2014 7:10 PM, rickman wrote: >>>>>>> On 12/4/2014 3:04 PM, Phil Hobbs wrote: >>>>>>>> Hi, all, >>>>>>>> >>>>>>>> I have a gig coming in that will have me revisiting my thesis research >>>>>>>> from nearly 30 years ago, on interferometric laser microscopes. (Fun.) >>>>>>>> >>>>>>>> Back in the day, I made a nulling-type phase digitizer at 60 MHz by >>>>>>>> driving a phase shifter with a 12-bit DAC (AD-DAC80), and wrapping a >>>>>>>> 13-bit successive approximation loop round it (AM2904 with an extra >>>>>>>> flipflop). With quite a lot of calibration, that got me a 13-bit, 2-pi, >>>>>>>> 50 ks/s phase measurement that I was pretty happy with. (The extra bit >>>>>>>> came from deciding which null to head for, which is why I needed the >>>>>>>> extra FF.) It was all interfaced to an HP 9816 computer via a GPIO >>>>>>>> card, and (eventually) worked great. I published one of my only two >>>>>>>> instruments papers on it (this was before I realized the total futility >>>>>>>> of almost all instruments papers). >>>>>>>> >>>>>>>> The advantage of nulling detection is that you only need 1-D calibration >>>>>>>> tables for phase shift and amplitude, whereas getting that sort of >>>>>>>> accuracy with I/Q techniques requires a 2-D calibration table, which is >>>>>>>> a gigantic pain. >>>>>>>> >>>>>>>> I need to do this again, 2015 style. The speed requirements are set by >>>>>>>> the acoustic delay in the AO scanner, so 50-100 ks/s is about all I can >>>>>>>> use. Rather than all that squishy analogue stuff, I'm planning to do >>>>>>>> the SAR in software and use a pair of AD9951 DDS chips, one to generate >>>>>>>> the desired signal and one to be the phase shifted comparison signal. >>>>>>>> >>>>>>>> So far so straightforward. >>>>>>>> >>>>>>>> What I'm less sure about is being able to keep the two channels >>>>>>>> sufficiently isolated to be able to maintain 12 or ideally 14 bits of >>>>>>>> phase accuracy. Even with a full-scale input, I'll need 85 dB of >>>>>>>> isolation to get 14 bits, and it gets harder with weaker signals. >>>>>>>> (There'll be a DLVA/limiter ahead of the phase detector, which will >>>>>>>> help.) >>>>>>>> >>>>>>>> I've never used DDSes before, and I'd appreciate some wisdom from folks >>>>>>>> who have. How hard is that likely to be, and what should I particularly >>>>>>>> watch out for? >>>>>>> >>>>>>> I've read all the posts so far and it seems you are generating a VHF >>>>>>> sine wave to compare to a VHF signal you wish to measure the phase and >>>>>>> amplitude of. I think I get that. But it seems the modulation of the >>>>>>> VHF signal is pretty low rate so that 50 kSPS is good enough. >>>>>>> >>>>>>> Then you ask about how to maintain enough isolation to preserve 14 bits >>>>>>> of phase measurement. I think the isolation you are worried about it in >>>>>>> the VHF range, no? That is the domain of RF design and not at all >>>>>>> trivial. I think you will need to provide more info on design specifics. >>>>>>> >>>>>>> I'm not clear on how you plan to do the phase detector. Is this just >>>>>>> subtracting the reference signal from the signal being measured? You >>>>>>> then scan the phase of the reference to find the null, scan the >>>>>>> amplitude of the reference to optimize the null and then possibly >>>>>>> repeat? Otherwise I'm not sure how you get both phase and amplitude out >>>>>>> of this. >>>>>>> >>>>>> The phase detector will probably be a diode bridge type, e.g. a Mini >>>>>> Circuits MPD-1. It's approximately a multiplier. >>>>> >>>>> Why not use a real multiplier? Analog Devices have a couple of pretty >>>>> good analog multiplier chips. AD734 and AD834 come to mind. >>>>> >>>>> And if you are working at a fixed frequency, running the DDS staircase >>>>> approximation to a sine wave through an integrator (with the right >>>>> gain) turns it into a straight-line interpolation approximation to a >>>>> sine wave, which is a lot nicer, (and slightly easier to filter). >>>> >>>> an integrator is just a bad filter, why should a bad filter in front of >>>> a good filter suddenly make things better? >>> >>> An integrator converts the sawtooth error signal implicit in a staircase >>> approximation to a sine wave to a series of much smaller of continuous arcs. >>> >>> You've still got a high frequency error signal to filter out, but pretty >>> much all of the higher frequency content (at multiples of DDS update >>> rate) > > has gone away. >>> >>> The integrator isn't functioning as a bad filter here - it's a device to >>> improve the quality of the approximation to the desired sine wave. >> >> But wouldn't any low-pass filter also do the same thing (ie: convert >> vertical parts of the waveform to less vertical parts)? Perhaps a low-pass >> filter could be chosen which gives superior performance over a straight >> integrator. > > Interesting theoretical question. Part of engineering is explaining what > you are doing, and why you are doing it, in terms that even a manager can understand. > > My contention would be that the integrator gets rid of the high-slew rate > component of the error signal in the first stage of filtering, which > makes a difference to any active filter in a way that filter construction > software doesn't usually capture. LTSpice does, but making a stair-case > approximation to a sine wave as a test signal might be tedious.And if my low-pass filter was a biquad, how would that compare?