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. -- John Larkin Highland Technology, Inc picosecond timing laser drivers and controllers jlarkin att highlandtechnology dott com http://www.highlandtechnology.com
DDS wisdom
Started by ●December 4, 2014
Reply by ●December 6, 20142014-12-06
Reply by ●December 6, 20142014-12-06
On Friday, December 5, 2014 1:15:20 PM UTC-8, whit3rd wrote:> It's possible, too, to mix down to an 'easy' frequency (maybe 1 MHz) > and use software-defined-radio type digitizers....> Then, it's just a matter of doing a couple of FFTs, and doing a weighted > curve fit of the phase*groan* no, that should be a weighted average of the phase, and the weight is the inverse square of the amplitude, weight = W(f) = I(f) **2 + Q(f)**2 and, the average is average_phase =sum_over_f( W(f) * atan2(I(f), Q(f)) ) / sum_over_f (W(f) ) Calibration consists of switching the input from 'received signal' to 'reference signal' to get a phase zero point.
Reply by ●December 6, 20142014-12-06
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 -- 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 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 HobbsIf 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. -- 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 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 -- 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
"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 Kevin Aylward www.kevinaylward.co.uk www.anasoft.co.uk - SuperSpice
Reply by ●December 6, 20142014-12-06
"John Larkin" wrote in message news:sk748a1osa1tatpve11envmt0g8jnubnqh@4ax.com... 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.>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.Yes. Kevin Aylward www.kevinaylward.co.uk www.anasoft.co.uk - SuperSpice
Reply by ●December 6, 20142014-12-06
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 HobbsHere'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 -- 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 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 -- 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
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. But obviously, they are real, measurable, and severely limit the use of DDS in precision applications. But that's not your field.
