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DDS wisdom

Started by Phil Hobbs December 4, 2014
On Fri, 05 Dec 2014 06:26:40 GMT, Tom Swift <spam@me.com> wrote:

>Phil Hobbs <hobbs@electrooptical.net> wrote: > >> On 12/4/2014 7:44 PM, Joe Gwinn wrote: > >>> 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. > >> Thanks > >> Phil Hobbs > >Joe is right. Most DDS are 32 bits. As I understand it, when you program >in a frequency, there are some bits left over. When the counter rolls >over, these do not align with the starting phase. Depending on the clock >and output frequencies, there will be a phase bump every several seconds >or so.
A binary-radix DDS, the only kind you can buy, forces the frequency output to be quantized to Fclk/2^n, which means you can't generally exactly hit nice decimal-expressed frequencies. But nothing special happens when the phase accumulator rolls over; there are no periodic phase bumps, other than the inherent jitter associated with quantizing the output waveform to the clock frequency and the phase accumulator math. There are some 48-bit DDSs, too. -- John Larkin Highland Technology, Inc picosecond timing precision measurement jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
On 12/5/2014 4:03 PM, rickman wrote:
> On 12/5/2014 3:03 PM, Kevin Aylward wrote: >>> "Phil Hobbs" wrote in message >>> news:cYydnTNwFPGvIx3JnZ2dnUU7-W-dnZ2d@supernews.com... >> >>> It was all interfaced to an HP 9816 computer >> >> it is "to a HP 9816" >> >> You yanks should learn that it is an "a" before any word that does not >> start with a vowel. It is an "an" for words starting with vowels. >> >> This seems to be a pretty universal error that you guys make. It is >> teeth gritting to hear these persistent "an historical event" instead of >> "a historical event". >> >> That's my wisdom for the day. > > Except that rule is wrong. Using "an" is about making speech easier. > Using "a" in front of a vowel sound is awkward, so "a" is changed to > "an" which flows from the tongue more easily. > > The rule is to use "an" when the following word starts with a vowel > *sound*, like honor and... istorical. lol While honor has an unsound H > and so starts with a vowel sound, historical starts with a sounded H. > But when used with "an" the H sound is truncated so it then fits the > rule. Rather a way of backing into it, eh? I'm not saying this is > "correct". But personally I don't give a rats ass about "correctness" > in this case. > > Language is alive and rules change. This is one that is already fuzzy > and using "an historical" is within the fuzz factor these days. If your > teeth grit, you should talk to your dentist about bruxism. > > That's *my* wisdom for the day.
Oh yeah, while you are at it, look up the rule for using "a" vs. "an" in front of a "y" or a u" pronounced as a "y" like "union". Not only will your teeth grit, but the hair will stand up on your neck. ;) -- Rick
On Fri, 5 Dec 2014 20:03:32 -0000, "Kevin Aylward"
<ExtractkevinRemove@kevinaylward.co.uk> wrote:

>>"Phil Hobbs" wrote in message >>news:cYydnTNwFPGvIx3JnZ2dnUU7-W-dnZ2d@supernews.com... > >> It was all interfaced to an HP 9816 computer > >it is "to a HP 9816" > >You yanks should learn that it is an "a" before any word that does not >start with a vowel. It is an "an" for words starting with vowels. > >This seems to be a pretty universal error that you guys make. It is teeth >gritting to hear these persistent "an historical event" instead of "a >historical event". > >That's my wisdom for the day. > >Kevin Aylward >www.kevinaylward.co.uk >www.anasoft.co.uk - SuperSpice
I was told that a/an can be applied to the sound of the following word, not strictly to the next letter. So, do what sounds right. -- John Larkin Highland Technology, Inc picosecond timing precision measurement jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
On Friday, December 5, 2014 6:48:22 AM UTC-8, Phil Hobbs wrote:
[about a phase meter]
> Yes, it's basically a double-balanced mixer run with its IF at DC, with > some hacks to increase output voltage and reduce DC offset. > > There's also a DLVA (detector & log video amplifier) in there, which > gets rid of the amplitude information and produces a logarithmic AM > output.
It's possible, too, to mix down to an 'easy' frequency (maybe 1 MHz) and use software-defined-radio type digitizers. If you had simultaneous mix-down of reference and unknown signals, against the same LO, the phase shift of the IF signals is the same as the RF ones. Then, it's just a matter of doing a couple of FFTs, and doing a weighted curve fit of the phase (determined from I and Q) with the weight being determined by the amplitude (because phase is indeterminate if sqrt(I**2 + Q**2 is zero. I think the weight is I**2 Q**2 /(I**2 + Q **2)
On 12/5/2014 4:15 PM, whit3rd wrote:
> On Friday, December 5, 2014 6:48:22 AM UTC-8, Phil Hobbs wrote: > [about a phase meter] >> Yes, it's basically a double-balanced mixer run with its IF at DC, with >> some hacks to increase output voltage and reduce DC offset. >> >> There's also a DLVA (detector & log video amplifier) in there, which >> gets rid of the amplitude information and produces a logarithmic AM >> output. > > It's possible, too, to mix down to an 'easy' frequency (maybe 1 MHz) > and use software-defined-radio type digitizers. If you > had simultaneous mix-down of reference and unknown signals, > against the same LO, the phase shift of the IF signals is the > same as the RF ones. > > Then, it's just a matter of doing a couple of FFTs, and doing a weighted > curve fit of the phase (determined from I and Q) with the weight > being determined by the amplitude (because phase is indeterminate > if sqrt(I**2 + Q**2 is zero. I think the weight is I**2 Q**2 /(I**2 + Q **2) >
I/Q at 14 bits requires one of two things: (1) very accurately known waveforms, or (2) a 2-D calibration table. All the upstream signal processing stuff will have some amplitude dependence of its phase shifts, for a start. A nulling measurement gets rid of both of these requirements, which is why I like it. 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
On Fri, 05 Dec 2014 16:03:50 -0500, rickman wrote:

> On 12/5/2014 3:03 PM, Kevin Aylward wrote: >>> "Phil Hobbs" wrote in message >>> news:cYydnTNwFPGvIx3JnZ2dnUU7-W-dnZ2d@supernews.com... >> >>> It was all interfaced to an HP 9816 computer >> >> it is "to a HP 9816" >> >> You yanks should learn that it is an "a" before any word that does not >> start with a vowel. It is an "an" for words starting with vowels. >> >> This seems to be a pretty universal error that you guys make. It is >> teeth gritting to hear these persistent "an historical event" instead >> of "a historical event". >> >> That's my wisdom for the day. > > Except that rule is wrong. Using "an" is about making speech easier. > Using "a" in front of a vowel sound is awkward, so "a" is changed to > "an" which flows from the tongue more easily. > > The rule is to use "an" when the following word starts with a vowel > *sound*, like honor and... istorical. lol While honor has an unsound H > and so starts with a vowel sound, historical starts with a sounded H. > But when used with "an" the H sound is truncated so it then fits the > rule. Rather a way of backing into it, eh? I'm not saying this is > "correct". But personally I don't give a rats ass about "correctness" > in this case. > > Language is alive and rules change. This is one that is already fuzzy > and using "an historical" is within the fuzz factor these days. If your > teeth grit, you should talk to your dentist about bruxism. > > That's *my* wisdom for the day.
And no typos! -- www.wescottdesign.com
On 12/5/2014 2:33 PM, John Larkin wrote:
> On Fri, 05 Dec 2014 13:42:04 -0500, Phil Hobbs > <hobbs@electrooptical.net> wrote: > >> On 12/5/2014 12:43 PM, John Larkin wrote: >>> On Fri, 05 Dec 2014 10:32:04 -0500, Phil Hobbs >>> <pcdhSpamMeSenseless@electrooptical.net> wrote: >>> >>>> On 12/5/2014 7:54 AM, Bill Sloman wrote: >>>>> 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. >>>> >>>> The beauty of the nulling technique is that you don't depend on the >>>> accuracy of the phase detector characteristic--it just has to have >>>> stable nulls, which the MPD-1 has. I'll always be winding up at almost >>>> exactly the same relative phase, i.e. at the null. >>>> >>>>> >>>>> 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). >>>> >>>> I've been reading the references everybody's been citing, and they're >>>> pretty illuminating. >>>> >>>> Re: subharmonics due to residual phase accumulator values >>>> >>>> Since I can pick the LO any way I like, I'll just use an integer >>>> multiple of f_clock / 2**14, which ought to get rid of the subharmonic >>>> problem, since the value in the phase accumulator will be the same at >>>> the beginning of each cycle, to the bit width of the lookup table. With >>>> a 400-MHz DDS, those are only 24 kHz apart, so there will be no problem >>>> finding one inside whatever filter passband I wind up with. >>> >>> Right. That amounts to a counter/lookup-table/DAC. Seweet spots in the >>> DDS frequency list. No squirmies. >>> >>> >>>> >>>> I've often used 10.7 MHz ceramic IF filters (as Rick Karlquist suggests >>>> in that excellent paper referenced upthread), so I might do that again, >>>> although I'll need a wider one after the variable phase oscillator, >>>> because a 14-bit conversion at 100 kHz would leave me only about 7 >>>> cycles per bit. >>>> >>>> I'd probably need to resynchronize the comparator output anyway, to make >>>> sure I'm always sampling at the null of the residual phase detector >>>> ripple. >>>> >>>> Or maybe I should just use one of JL's ECLiPS Lite D-flops for the phase >>>> detector instead. With a 1-ps decision time, that would be good for >>>> >>>> N = -log_2(10.7 MHz * 1 ps) = 16.5 bits >>> >>> Even better with averaging! >>> >>> You can make a sampling oscilloscope, too, with a minor variant of the >>> bang-bang phase detector, using very few parts. >> >> Averaging would be pretty simple--use its output to gate the 10.7 MHz >> into a counter input of the MCU. With 7 cycles per bit, I'd >> theoretically get 19 bits. Awesome. >> >> Which leads to the next question, namely layout and clock distribution. >> >> In my original setup, I built a calibrator that was much more >> complicated than the digitizer itself. It had two separate 60 MHz >> synthesizers, running at a reference frequency of 600 kHz >> (divide-by-100). One of them used a 10/11 dual modulus prescaler, so >> that I could swallow individual pulses and so make the phase walk around >> by 3.6-degree steps. Before using the microscope, I ran the phase >> shifter calibration and did a normal cubic spline fit to the results, >> which worked fine as long as everything was well warmed up. >> >> That strategy lives and dies by the isolation between channels. I used >> a bunch of isolation amps, based on MRF966 dual-gate GaAs FETs, which >> were good enough for 70-dB isolation at 100 MHz in one stage--and the >> whole thing was a dead-bug proto. It used a lot of bolted-together >> aluminum boxes and double-shielded coax, and was about 18 inches square. >> But the isolation was very good--when I leaned on the "pulse swallow" >> button, which moved one synthesizer off by 600 kHz, I could see on the >> spectrum analyzer that the leakage was at the -90 dBc level. >> >> This one is going to be on the same board, talking to the same >> processor, and so on. I'm planning to run the two synths on their own >> power supplies, with buffers on the digital lines that are also run from >> those supplies, and run all their traces on interior levels between two >> ground pours with via stitching. >> >> It's really the clock distribution and especially the local grounding >> and decoupling at the DDSes that I'm worried about. You're generally a >> fan of just connecting all the flavours of ground to a single >> featureless plane, with an equally featureless supply plane next to it. >> Is that the right answer here as well, or do I risk extra spurs that way? >> >> Cheers >> >> Phil Hobbs > > If my comparator idea is of any interest, the isolation situation is > vastly improved. Differential PECL signals with 35 ps rise/fall times > don't crosstalk much. The two DDSs could be kept far apart and > converted to PECL before converging into the phase comparators. > > We have a sample kit from Autosplice of really cool surface-mount > shield clips. A standard (or etched, custom folded Fotofab) shield box > would plug onto them, as needed.
Doing it that way sounds pretty plausible. I'll think about it, and I'll certainly take you up on your offer of a second set of eyes, thanks. 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
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 >*sound*, like honor and... istorical. lol While honor has an unsound H >and so starts with a vowel sound, historical starts with a sounded H. >But when used with "an" the H sound is truncated so it then fits the >rule. Rather a way of backing into it, eh? I'm not saying this is >"correct". But personally I don't give a rats ass about "correctness" >in this case. > >Language is alive and rules change. This is one that is already fuzzy >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. 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.
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.
> > 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? > > Thanks > > 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
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. 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. ADI has a very good tutorial on DDS theory, "MT-085: Fundamentals of Direct Digital Synthesis (DDS)". I'd read it. Joe Gwinn