Joe Gwinn <joegwinn@comcast.net> wrote:> In article <54826A1A.9080101@electrooptical.net>, Phil Hobbs > <hobbs@electrooptical.net> wrote: > >> On 12/5/2014 7:44 PM, Joe Gwinn wrote:>> > 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.Don't give up so soon. I think you may be referring to Section 4, "The Effect of Truncating the Phase Accumulator on Spurious Performance," starting on Page 19 of http://www.analog.com/static/imported- files/tutorials/450968421DDS_Tutorial_rev12-2-99.pdf Fred Harris may have ways to help reduce it, discussed in "Ultra Low Phase Noise DSP Oscillator", at https://www.researchgate.net/profile/Fred_Harris2/publication/3321879 _Ultra_Low_Phase_Noise_DSP_Oscillator_DSP_Tips__Tricks/links/542425050cf2 38c6ea6e953a> 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.What did you use to solve the problem?> Joe Gwinn
DDS wisdom
Started by ●December 4, 2014
Reply by ●December 6, 20142014-12-06
Reply by ●December 6, 20142014-12-06
On 12/6/2014 2:47 PM, John Larkin wrote:> On Sat, 06 Dec 2014 14:24:50 -0500, Phil Hobbs > <hobbs@electrooptical.net> wrote: > >> On 12/6/2014 1:54 PM, John Larkin wrote: >>> On Sat, 06 Dec 2014 13:51:35 -0500, Phil Hobbs >>> <hobbs@electrooptical.net> wrote: >>> >>>> On 12/6/2014 1:42 PM, John Larkin wrote: >>>>> On Sat, 06 Dec 2014 18:00:28 GMT, Ralph Barone >>>>> <address_is@invalid.invalid> wrote: >>>>> >>>>>> 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? >>>>> >>>>> At 10s of MHz, passive LC filters work better. Inductors are not slew >>>>> rate limited. >>>>> >>>>> An integrator makes no sense here. Attenuation is a pitiful 6 >>>>> dB/octave at the cost of roughly 90 degrees of phase shift. >>>> >>>> Back in the early days of DSP, when it was more of a branch of numerical >>>> analysis, there were lots of arguments about slopes and derivative >>>> discontinuities. (Hamming and Lanczos are good reads on that.) >>>> >>>> When oscilloscopes became popular, people took a look at the actual >>>> waveforms they were arguing over, said "Never mind", and went back to >>>> work. ;) >>>> >>>> Cheers >>>> >>>> Phil Hobbs >>> >>> If you ever need a serious LC filter design, we have the NuHertz >>> dongled software. It makes all sorts of amazing exotic filters, using >>> standard part values. >>> >>> >> Cool. Got an example you could post? >> >> Cheers >> >> Phil Hobbs > > Here's one: > > https://dl.dropboxusercontent.com/u/53724080/Circuits/Filters/T346_filter.jpg > > It looks really dumb, but it isn't. That's the beauty of the NuHertz > thing. > > Used here: > > http://www.highlandtechnology.com/DSS/T346DS.shtmlSo roughly a 50 MHz LPF with a real zero at 100 MHz-ish? 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
On 12/5/2014 11:24 PM, rickman wrote:> 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.How is that different from saying that the hidden bits of the phase accumulator remain constant? It seems like we're in violent agreement, except that you haven't noticed yet. ;) If the hidden bits are always zero, then in each cycle, all the DAC codes repeat, so the waveform is ideally perfectly periodic. No? Cheers Phil Hobbs 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
On Sat, 06 Dec 2014 20:27:53 GMT, Tom Swift <spam@me.com> wrote:>John Larkin <jlarkin@highlandtechnology.com> wrote: > >> On Fri, 05 Dec 2014 06:26:40 GMT, Tom Swift <spam@me.com> wrote: > >>>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. > >As you say, you are not an RF guy and you don't do sine waves. These issues >are many orders of magnitude below your level of interest or ability to >measure.Not so.> >But obviously, they are real, measurable, and severely limit the use of DDS >in precision applications. > >But that's not your field.Our waveform generators are mostly used to test aircraft instrumentation and controls. Different requirements from RF. Our stuff is usually wideband: not, for example, 10 GHz but rather DC to 10 GHz. Just last week we were generating some 40 ps wide light pulses from a laser specified as a CW pump. We do have a good 3 GHz spectrum analyzer, and we do look at the spurs and harmonic distortion of all of our waveform generators, both sine generators and arbs. And we do a little tweaking to improve things. We have had more trouble with filters and opamps than we have had with the DDS math and the DACs. ARBS have more potential distortion products than sine generators. We care a lot about harmonic distortion; RF folks often don't. They work narrowband, and generally have bandpass stages downstream. -- John Larkin Highland Technology, Inc picosecond timing laser drivers and controllers jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
Reply by ●December 6, 20142014-12-06
On Sat, 06 Dec 2014 15:22:29 -0500, Phil Hobbs <hobbs@electrooptical.net> wrote:>On 12/6/2014 2:32 PM, Kevin Aylward wrote: >> "Tim Wescott" wrote in message >> news:i6udnTwKfZIUjR_JnZ2dnUU7-SednZ2d@giganews.com... >> >> On Fri, 05 Dec 2014 20:03:32 +0000, 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. >>> >> >>> 'H' is pronounced "ach" starts with an a, which is a vowel. >> >> Over here it's correctly pronounced "ech", with an e, but its still >> irrelevant. >> >> That's why you dudes make the mistake, but its still wrong. >> >> Sure, American English has bona-fide differences from UK English, but >> this is not one of them. >> >> http://www.oxforddictionaries.com/words/a-historic-event-or-an-historic-event >> >> >> Its actually how the following word sounds. Now.., >> >> "It's an Hewlett Packard 9816 computer" >> >> Does that actually sound ok to you? >> >> The Hewlett sound starts as "Hugh". That is not a vowel sound > >"HP" isn't prononuced "Hewlett Packard", it's pronounced "aitch pea". >Hence the 'an'. > >Cheers > >Phil HobbsIt's pronounced "Keysight" -- John Larkin Highland Technology, Inc picosecond timing laser drivers and controllers jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
Reply by ●December 6, 20142014-12-06
On 12/6/2014 4:15 PM, John Larkin wrote:> On Sat, 06 Dec 2014 15:22:29 -0500, Phil Hobbs > <hobbs@electrooptical.net> wrote: > >> On 12/6/2014 2:32 PM, Kevin Aylward wrote: >>> "Tim Wescott" wrote in message >>> news:i6udnTwKfZIUjR_JnZ2dnUU7-SednZ2d@giganews.com... >>> >>> On Fri, 05 Dec 2014 20:03:32 +0000, 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. >>>> >>> >>>> 'H' is pronounced "ach" starts with an a, which is a vowel. >>> >>> Over here it's correctly pronounced "ech", with an e, but its still >>> irrelevant. >>> >>> That's why you dudes make the mistake, but its still wrong. >>> >>> Sure, American English has bona-fide differences from UK English, but >>> this is not one of them. >>> >>> http://www.oxforddictionaries.com/words/a-historic-event-or-an-historic-event >>> >>> >>> Its actually how the following word sounds. Now.., >>> >>> "It's an Hewlett Packard 9816 computer" >>> >>> Does that actually sound ok to you? >>> >>> The Hewlett sound starts as "Hugh". That is not a vowel sound >> >> "HP" isn't prononuced "Hewlett Packard", it's pronounced "aitch pea". >> Hence the 'an'. >> >> Cheers >> >> Phil Hobbs > > It's pronounced "Keysight"Five years from now it'll be "FuzzyNuts". You heard it here first. Cheers Phil Hobbs
Reply by ●December 6, 20142014-12-06
On Sat, 06 Dec 2014 15:58:09 -0500, Phil Hobbs <pcdhSpamMeSenseless@electrooptical.net> wrote:>On 12/6/2014 2:47 PM, John Larkin wrote: >> On Sat, 06 Dec 2014 14:24:50 -0500, Phil Hobbs >> <hobbs@electrooptical.net> wrote: >> >>> On 12/6/2014 1:54 PM, John Larkin wrote: >>>> On Sat, 06 Dec 2014 13:51:35 -0500, Phil Hobbs >>>> <hobbs@electrooptical.net> wrote: >>>> >>>>> On 12/6/2014 1:42 PM, John Larkin wrote: >>>>>> On Sat, 06 Dec 2014 18:00:28 GMT, Ralph Barone >>>>>> <address_is@invalid.invalid> wrote: >>>>>> >>>>>>> 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? >>>>>> >>>>>> At 10s of MHz, passive LC filters work better. Inductors are not slew >>>>>> rate limited. >>>>>> >>>>>> An integrator makes no sense here. Attenuation is a pitiful 6 >>>>>> dB/octave at the cost of roughly 90 degrees of phase shift. >>>>> >>>>> Back in the early days of DSP, when it was more of a branch of numerical >>>>> analysis, there were lots of arguments about slopes and derivative >>>>> discontinuities. (Hamming and Lanczos are good reads on that.) >>>>> >>>>> When oscilloscopes became popular, people took a look at the actual >>>>> waveforms they were arguing over, said "Never mind", and went back to >>>>> work. ;) >>>>> >>>>> Cheers >>>>> >>>>> Phil Hobbs >>>> >>>> If you ever need a serious LC filter design, we have the NuHertz >>>> dongled software. It makes all sorts of amazing exotic filters, using >>>> standard part values. >>>> >>>> >>> Cool. Got an example you could post? >>> >>> Cheers >>> >>> Phil Hobbs >> >> Here's one: >> >> https://dl.dropboxusercontent.com/u/53724080/Circuits/Filters/T346_filter.jpg >> >> It looks really dumb, but it isn't. That's the beauty of the NuHertz >> thing. >> >> Used here: >> >> http://www.highlandtechnology.com/DSS/T346DS.shtml > >So roughly a 50 MHz LPF with a real zero at 100 MHz-ish? > >Cheers > >Phil HobbsYeah, this was designed to get us maximum bandwidth but maximally nuke aliases. That zero really helps. The DDS/ARB clock rate is 128 MHz. -- John Larkin Highland Technology, Inc picosecond timing laser drivers and controllers jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
Reply by ●December 6, 20142014-12-06
On Sat, 06 Dec 2014 16:17:49 -0500, Phil Hobbs <hobbs@electrooptical.net> wrote:>On 12/6/2014 4:15 PM, John Larkin wrote: >> On Sat, 06 Dec 2014 15:22:29 -0500, Phil Hobbs >> <hobbs@electrooptical.net> wrote: >> >>> On 12/6/2014 2:32 PM, Kevin Aylward wrote: >>>> "Tim Wescott" wrote in message >>>> news:i6udnTwKfZIUjR_JnZ2dnUU7-SednZ2d@giganews.com... >>>> >>>> On Fri, 05 Dec 2014 20:03:32 +0000, 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. >>>>> >>>> >>>>> 'H' is pronounced "ach" starts with an a, which is a vowel. >>>> >>>> Over here it's correctly pronounced "ech", with an e, but its still >>>> irrelevant. >>>> >>>> That's why you dudes make the mistake, but its still wrong. >>>> >>>> Sure, American English has bona-fide differences from UK English, but >>>> this is not one of them. >>>> >>>> http://www.oxforddictionaries.com/words/a-historic-event-or-an-historic-event >>>> >>>> >>>> Its actually how the following word sounds. Now.., >>>> >>>> "It's an Hewlett Packard 9816 computer" >>>> >>>> Does that actually sound ok to you? >>>> >>>> The Hewlett sound starts as "Hugh". That is not a vowel sound >>> >>> "HP" isn't prononuced "Hewlett Packard", it's pronounced "aitch pea". >>> Hence the 'an'. >>> >>> Cheers >>> >>> Phil Hobbs >> >> It's pronounced "Keysight" > >Five years from now it'll be "FuzzyNuts". You heard it here first. >Maybe they'll buy Rigol and put them out of business. Ditto! -- John Larkin Highland Technology, Inc picosecond timing laser drivers and controllers jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
Reply by ●December 6, 20142014-12-06
On Sunday, 7 December 2014 04:52:34 UTC+11, Phil Hobbs wrote:> 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.)The paper *is* a bit hard. But you need to read RSI with Sturgeon's Law in mind - 90% of everything is rubbish. There's gold in the gravel, but it can take some panning to extract it. Larsen N T 1968 Rev. Sci. Instrum. 39 1-12 on microdegree thermostats is the first example of that that comes to my mind. -- Bill Sloman, Sydney
Reply by ●December 6, 20142014-12-06
[[ This message was both posted and mailed: see the "To," "Cc," and "Newsgroups" headers for details. ]] In article <XnsA3FBA89E133D1idtokenpost@69.16.179.23>, Tom Swift <spam@me.com> wrote:> Joe Gwinn <joegwinn@comcast.net> wrote: > > > In article <54826A1A.9080101@electrooptical.net>, Phil Hobbs > > <hobbs@electrooptical.net> wrote: > > > >> On 12/5/2014 7:44 PM, Joe Gwinn wrote: > > >> > 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. > > Don't give up so soon. I think you may be referring to Section 4, "The > Effect of Truncating the Phase Accumulator on Spurious Performance," > starting on Page 19 of > > <http://www.analog.com/static/imported-files/tutorials/450968421DDS_Tutorial_rev12-2-99.pdf>Ahh. Perhaps so.> Fred Harris may have ways to help reduce it, discussed in "Ultra Low > Phase Noise DSP Oscillator", at > > <https://www.researchgate.net/profile/Fred_Harris2/publication/3321879_Ultra_Low_Phase_Noise_DSP_Oscillator_DSP_Tips__Tricks/links/542425050cf238c6ea6e953a>I may remember this. I'll look into my trove at work.> > 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. > > What did you use to solve the problem?I didn't. The prototype belonged to one of our vendors (who shall remain nameless), and they had developed it first for use in their own lab, with distant theories about maybe selling it as well. But no such product has appeared in their catalog. I bet they bought a Symmetricom 5115a, which would cost a lot less than the effort to develop a good enough unit. The vendor was trying to use their prototype DMTD unit for testing of a product being developed for my employer, and were getting unexpected results. My contribution was to figure out why. The way Symmetricom (actually Timing Solutions, acquired by Symmetricom) solved the whole DDS spur and bump problem is by computing the sine wave amplitudes directly (no DDS chip) and loading it into a clock-indexed RAM unit. These numbers are fed directly to a digital multiplier (replacing an analog mixer). The actual frequency is tweaked such that there is no glitch when the memory rolls over. I got this from a Timing Solutions patent. That too is in the trove. Joe Gwinn
