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Twospoons
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Regarding OCXOs: Not a trivial project by any means - I spent 6 months working for Rakon on their double-oven oscillator. That's an ovenised
oscillator inside another oven!. One of the tricks is using an SC cut crystal, which has a cubic Tc, and operating it at a temperature which matches
one of the turning points of the Tc curve. Managed to get down to 0.2 ppb with the best one. To measure it we used a Trimble 10MHz GPS reference
oscillator - which makes use of the 30 odd atomic clocks circling overhead.
The crucial parameter for this project is going to be phase noise, as this will contribute directly to spreading of the demodulated signal, and
therefore its visibility amongst the noise. I suggest finding the lowest phase noise TCXO you can and make a small oven for it. You should be able
to get 0.1C stability on your oven without too much trouble, and let the TCXO take care of the rest. I'm thinking that as long as the short term
stability is very good, any absolute error can be compensated for later.
Helicopter: "helico" -> spiral, "pter" -> with wings
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aliced25
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Ok, found a processor & a half, have a look at this "FFT Implementation on the TMS320VC5505, TMS320C5505, and TMS320C5515 DSPs" (SPRABB6A) from Texas Instruments, the FFT is hardware optimized & it has some 500kB of memory on disk - with USB 2.0. Difficulty, the TMS320C5515 is a 196 NFBGA (14x14) chip, but that can be overcome (not quite trivial though) It is the one that the ECG Solution and the discussion vis-a-vis that is rather interesting to read. That uses the ADS1258 16-Channel, 24-bit ΔΣ ADC running at 23.7KSPS/Channel (so 11.8kHz would be the Nyquist boundary?), which is amplified by the INA128 Precision, Low Power Instrumentation Amp with the REF5025 Precision, Low-power, Low-drift, Voltage Ref (which looking at what else it is used for would seem to the the pick).
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Twospoons
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I'd be inclined to do all the processing in a PC, and focus the design effort on the analogue and data acquisition side. You have a lot more options
for processing software in a PC, as well as stuff all resource limitations. Looking at ECG solutions is a good idea - its the same sort of problem
finding a small signal amongst a lot of noise.
Helicopter: "helico" -> spiral, "pter" -> with wings
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not_important
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I think you'd best read that document yourself. ECG apps typically deal with nothing higher than around a KHz, and that on auxiliary signals; the
main signals top out around a tenth of that.
Yo're still going at this wrong way around. Effort in designing a system to satisfy unspecified requirements is mostly wasted. There's a load of
rather importan electronics ahead of the DSP section of FT-NMR.
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aliced25
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Yes, but the concept is similar enough to grab my attention - the need to attenuate a high-volume pulse, the need to filter out the majority of the
noise, all pretty much what we are dealing with in this application, the size of the bandwidth of interest is the only thing that really changes...
Actually, on that subject, I'm looking everywhere I've been told to look, I'm not seeing a whole lot on the actual breadth of the bandwidth come to
that, what is the area of interest? Failing that, how do I work it out?
The difference in specifications & price when dealing with 10's of KSPS to several hundred KSPS, to the MSPS range makes this kind of imperative.
I know (now) that I have to sample at at least twice the bandwidth, but I'm fucked if I can work out where the bandwidth ends.
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watson.fawkes
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Quote: Originally posted by aliced25  | | all pretty much what we are dealing with in this application, the size of the bandwidth of interest is the only thing that really changes...
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Except for the analog section, which is completely different, harder to get right than the digital one, and
where almost all of the stability, metrology, and calibration issues are.
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arsphenamine
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| Quote: Originally posted by aliced25 | | ... I'm not seeing a whole lot on the actual breadth of the bandwidth come to that, what is the area of interest? Failing that, how do I work it out?
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*facepalm!*
Understand the Larmor Frequency calculation.
Learn to perform ppm to Hz interconversions on NMR spectra.
Observe the practical ppm range of published 1H spectra.
Bandwidth figures will be incidental and obvious then.
I strongly urge that you gain practical experience with an NMR device.
That is to say that you must physically lay hands on the machine.
You need an experiential skeleton to flesh out with theoretical concepts,
otherwise this DIY NMR is only intellectual masturbation -- feels good, but
nobody else cares.
Many universities and community colleges have ageing functional NMR's
with which they train students. Arrange for a session or two.
Come back when you can explain the splittings and integration line on the 1H-NMR spectrogram of ethanol below.
Notes:
Sample was 5% in CCl4 from which you infer minimal intErmolecular hydrogen bonding from the -OH moeity.
0.0 ppm is the tetramethylsilane calibration standard;
1.2 ppm is a triplet,
3.7 ppm is a quartet,
4.8 ppm is a singlet.
Don't tell us about it.
Just get the skill and figure it out.
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arsphenamine
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@aliced25
You need this.
It interconverts field strength and Larmor Frequency.
[code]<script language="JavaScript"> function btof() { var bfield = document.ppfcalc.bfield.value; var freq = 267522205 * bfield / (2 *
Math.PI * 1e6); document.ppfcalc.freq.value = freq.toFixed(9); } function ftob() { var freq = document.ppfcalc.freq.value; var bfield = 2 *
Math.PI * freq * 1e6 / 267522205; document.ppfcalc.bfield.value = bfield.toFixed(9); } </script> <form name="ppfcalc"> <table
style="text-align: left; margin-left: auto; margin-right: auto;" border="2" cellspacing="1"> <br> <tbody> <tr> <td
style="text-align: center; vertical-align: middle;"> <input name="bfield" size="11" type="text"> Tesla</td> <td style="text-align:
center;"> <input size="2" value=">>" onclick="btof();" type="button"> </td> <td style="text-align: center;">
<input size="2" value="<<" onclick="ftob();" type="button"> </td> <td style="text-align: center;"> <input
name="freq" size="11" onchange="ftob();" type="text"> MHz<br> </td> </tr> </tbody> </table>
</form>[/code]
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Twospoons
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That's interesting. It shows a shift of several hundred kHz with just 0.01T field change at 0.8T. So that magnetic field has to be exceptionally flat,
otherwise the signals are going to be spread all over the band, and will be invisible amongst the noise. In fact, I would hazard a guess that
flatness is two or three orders of magnitude more important than peak field strength.
Helicopter: "helico" -> spiral, "pter" -> with wings
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arsphenamine
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| Quote: Originally posted by Twospoons | | ... So that magnetic field has to be exceptionally flat, otherwise the signals are going to be spread all over the band, and will be invisible amongst
the noise.. |
Ayup.
Magritek has a pleasant video on inhomogeneity.
It is part of a series of NMR/MRI video tutorials.
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not_important
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| Quote: | | So that magnetic field has to be exceptionally flat |
Thus shimming, twisting screws and inserting brass shims to get the magnet pole pieces as close to parallel as possible. Also thus the spinning of
the sample to expose it to in effect an averaged magnetic field in the X and Y axis; really useful only after shimming out larger field imperfections.
The crystal structure of permanent magnets create some inhomogeneities in the field, one problem with some designs using NbBFe magnets.
For some applications there's also a mid- to long- term stability issue, thus systems that maintain a lock on the <sup>2</sup>H in
deuterated solvents, using an axillary coil to tweak the field strength in real time - sensing lock via CW NMR methods.
-----------------------------------------------------------------
You need to sample (at least) a bit more than twice the highest frequency of interest; both to reduce distortion and because real filters are not
brick walls.
If you sample a sine wave at a rate exactly equal to the wave's frequency, you get a DC level. For signal frequencies slightly lower or higher than
that you will see a low frequency - the difference between the sample rate and the signal being sampled; as you move away from the sample rate the
apparent frequency increases. This means that you want the anal signal to have as little energy as possible above (or even at) the Nyquist frequency
(1/2 the Nyquist rate). Rather than designing complex analogue filters, it's easier to oversample quite a bit with reasonable analogue filtering to
take out frequencies a ways about the highest frequency of interest but still well below the Nyquist, then use digital filtering to improving the
filtering of the frequencies about those of interest. End result is elimination of wrap around, generation of spurious signals in the result.
Typically the filtered digital data is decimated before performing the FFT, as this reduces memory and processor speed requirements.
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aliced25
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That is what I was looking for not_important, the lower end, medium speed ADC cited above is capable of operation well and truly above the Nyquist
envelope, but I'm looking at the trade-off in speed:accuracy in ENOBs, how far above the Nyquist envelope is too much of a good thing?
I'm looking at the filter on that ECG - the one that is there to grab the over-voltage from a single-shot pulse and take it to ground, plus the
low-pass filter implementation (such as I can generate with FilterPro). As I make clear (to arsphenamine below), all I need to know is the maximum ppm
that anyone would want out of this unit - I'm presuming that noone is likely to want to try and source 13C at home? (then again, presumption would be
the bastard-brother of all fuckups). If I only have to design for 1H-NMR, then 10 or 15ppm as a baseline reference bandwidth?
I was looking very closely at the digital filter specifications, that with the decimation and the ability to average the result set in almost real
time (thereby saving each cycle as one averaged, decimated & noise reduced 32-bit word for later processing).
Any experience with digital isolators? This would appear to be precisely the sort of application that they would be targeted at? Here is an App. Note
from TI (SLLA198). There are two reasons I ask, firstly, the suggestion that 100+ gauss within any manufactured device (fail-point) is unlikely (not here
it ain't - there is a magnet not too far away that will be producing about 10 times that) is unlikely, but that this IC will damp EMF & EMI - ie.
noise. I'm presuming it probably should straddle bare (copper free) board in the middle in order to maximize the isolation, but I noted one figure in
the cited paper where they use precisely this IC to separate the Analog input from the Digital processor.
As for magnetic homogeneity, you'll note in the original paper from Danieli, et al, on the small magnet they designed, they were discussing how the
design allowed them to get rid of inhomogeneity in the order of 20K ppm, well according to the model, this design can get down to 1K or even 500ppm.
From that point, the only way forward (without designing a motor to spin the fuck out of the sample) is shimming. The major difficulty I'm facing is
the 1% (minimum, generally well above that) rating on field sensors, Hall effect sensors, etc. It would appear that the best - indeed possibly the
only - way to test the magnet assembly and get it down to the minimal ppb range would be to actually test a water sample (or a solution) as used by
Dogan, et al. Jachmann, et al, describe an extremely interesting solution, a small Halbach array within a Halbach array which would be interesting to
model, but I've promised myself that I'll work on the basic electronic component design area before I fuck around with anymore modeling (with too much
open this PC does not like FEMM, tends to freeze up).
I actually suggested that the pwm/dac drivers on the DSP/MCU could be used to drive small shim coils, in fact I cannot see why it would be impossible
(difficult yes, impossible we'll see) to design the system so that it senses inhomogeneity itself and auto-shims to correct it.
@ arsphenamine,
Thank you indeed for illustrating the various ppm shifts of two methylene groups and the OH, not precisely what I had in mind, but a nice start. What
I was after is more of an indication of precisely what is the maximum ppm shifts one would expect OF a portable NMR device? Quite frankly given the
time & effort it is going to take to design this sucker (and the board it goes on), I might as well look at shifting some populated/unpopulated
boards for anyone who is interested in order to keep my costs down (and theirs). Fuck all point of that if the unit doesn't perform at the minima one
would expect is there?
[Edited on 12-10-2010 by aliced25]
Attachment: Danieli.etal.Mobile.Sensor.for.High.Resolution.NMR.Spectroscopy.and.Imaging.pdf (746kB)
This file has been downloaded 1139 times
Attachment: Danieli.etal.Small.Magnets.for.Portable.NMR.Spectrometers.pdf (107kB)
This file has been downloaded 643 times
Attachment: Dogan.etal.Development.of.Halbach.Magnet.for.Portable.NMR.Device.pdf (783kB)
This file has been downloaded 815 times
Attachment: Blumich.Casanova.Appelt.NMR.at.Low.Magnetic.Fields.pdf (800kB)
This file has been downloaded 1057 times
Attachment: Jachmann.etal.Quadupolar.Order.Shimming.of.Permanent.Magnets.Using.Harmonic.Corrector.Rings.pdf (457kB)
This file has been downloaded 738 times
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arsphenamine
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| Quote: | | ...all I need to know is the maximum ppm that anyone would want out of this unit - I'm presuming that no one is likely to want to try and source 13C
at home? (then again, presumption would be the bastard-brother of all fuckups). If I only have to design for 1H-NMR, then 10 or 15ppm as a baseline
reference bandwidth? |
Pay. Attention. This. Time.
U Wisconsin has a good page on 1H-NMR chemical shifts.
Look at the shift tables at the top.
For ordinary organics (C,H,O), everything of interest is between 0 and 12
ppm. Don't take my word for it. Gloss over the charts.
15 ppm is easier for estimation purposes. So is 20 ppm.
Anything outside 0-12 ppm is usually a hetero (B,P,N) or metallic/semi-metallic that
you are unlikely to encounter or must not encounter outside of a dry box.
| Quote: | | I was looking very closely at the digital filter specifications, that with the decimation and the ability to average the result set in almost real
time (thereby saving each cycle as one averaged, decimated & noise reduced 32-bit word for later processing). |
In practice, it often reduces to something like this:
unsigned int sum = 0;
for (int i=0 ; i<16 ; i++)
{ sum += readADC() ;}
sum >>= 4 ;
return (sum) ;
| Quote: | | As for magnetic homogeneity, you'll note in the original paper from Danieli, et al, on the small magnet they designed, they were discussing how the
design allowed them to get rid of inhomogeneity in the order of 20K ppm, well according to the model, this design can get down to 1K or even 500ppm.
|
Homogeneity coils for a small Halbach or Stelter array
will be small but easier to execute as printed circuits,
perhaps copper on Kapton.
First you make a Z axis coil...
Halbach Arrays are sexy, but I would consider designing a Stelter Array
since it uses alloy pole faces which, incidentally, improve field homogeneity.
The Stelter is also easier to design, uses block parts, and it, too,
is proven in NMR/MRI devices.
| Quote: | | From that point, the only way forward (without designing a motor to spin the fuck out of the sample) is shimming. |
To this day, liquid samples spin under air pressure. See the original
patent for drawings, US 2,960,649 "Line narrowing gyromagnetic apparatus".
The practice is to spin the sample around 10 rps, fast enough to time average
out some field inhomogeneity, but not so fast that sideband peaks appear from
liquid turbulence. The sample tube mounts through the center of a plastic sawtooth
gear that spins from tangential air flow -- essentially a small turbine.
| Quote: | | It would appear that the best - indeed possibly the only - way to test the magnet assembly and get it down to the minimal ppb range would be to
actually test a water sample (or a solution) as used by Dogan, et al. |
Hold that thought.
Testing is your friend.
| Quote: | | Jachmann, et al, describe an extremely interesting solution, a small Halbach array within a Halbach array which would be interesting to model...
|
Figure that the Jachmann,et.al. design of static correction rings placed
them in the hands their machinist when it came time for executing the design.
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not_important
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| Quote: Originally posted by aliced25 | ...- I'm presuming that noone is likely to want to try and source 13C at home? (then again, presumption would be the bastard-brother of all fuckups).
...
It would appear that the best - indeed possibly the only - way to test the magnet assembly and get it down to the minimal ppb range would be to
actually test a water sample (or a solution) ... |
<sup>13</sup>C shifts are very roughly an order of magnitude larger then <sup>1</sup>H shifts. Both
<sup>13</sup>C and fluorine could be done on the same unit as ordinary proton NMR, provided the electronic - coils and RF stage amps - can
be tuned for them. The low concentration of <sup>13</sup>C does make it more difficult, but FT-NMR with good frequency/time stability can
do it.
See the attached for a discussion of shimming, as well as other things that impact homogeneity, including the makeup of the coils themselves, dioxygen
dissolved in the sample, and so on.
Attachment: shimming.pdf (639kB)
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aliced25
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<sup>13</sup>C shifts are an order of magnitude larger than the <sup>1</sup>H shifts, then aren't they going to be rather
abruptly cut off by the multi-part active filter on the analog input? I'm not sure how I'd go about having a programmable active filter, at least on
the analog side.
By the way, when I had another look at the ECG Solution (just down the page, under signal acquisition challenges (dot point #4), one of the design issues was that the design had to allow
for the potential use of 0.5-1,000Hz in diagnostic mode. As the signals we want are well under the 1kHz range (even @20 ppm), then the question is,
wouldn't the solution for that apply here? If that is the case, presumably we could design an active filter with the bandpass so it wouldn't cut off
the <sup>13</sup>C resonance, incorporate the gross-overvoltage filter, and allow the 24-bit ΔΣ ADC to do the same job it does
there, getting rid of any remaining noise with the digital filter & the noise & echo reduction algorithms?
I'm just trying to work out why so many separate shims are needed X,Y,Z I get, why multiples?
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arsphenamine
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13C resonance signals?
Don't worry about it.
Since there is usually less carbon than 1H in a compound,
and 13C's natural abundance is only 1.1% that of 12C,
and 13C's relative sensitivity is 1.6% that of 1H,
and its signal bandwidth is >10X that of 1H ...
13C signals appear as noise in the 1H spectrum of interest,
in the absence of clever excitation tricks and expensive low-noise amps
on the front-end of the receiver coil.
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arsphenamine
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Simplified shim coils on printed circuit boards:
Attachment: PCBshims_pat3735306.pdf (412kB)
This file has been downloaded 606 times
Note that this patent references several others:
US 3.469,180
US 3,488,561
US 3,515,979
The first two show the progress of simple magnet shim coils and suggest where the
current 18-30+ coil shim systems came from.
The last is by the venerable Golay and has diagrams of second and third order effects
from the initial XYZ homogeneity corrections. It shows you _why_.
You don't need to address them all (or any) so much as be aware of the limitations.
Lacking a sophisticated shim coil implementation, solid sample NMR is a waste of time.
Absent any shim coils, only low resolution spectra of physically spun liquid samples
are practical. Not too shabby, actually.
In accord with the Principle of Maximum Laziness, I'd go with a PCB version of one
of the simpler versions having only 2-3 coil types.
Finally, all these patents antedate superconducting solenoid magnets which use
more coils and more complicated shapes whose patents don't seem directly relevant to this project.
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not_important
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| Quote: Originally posted by aliced25 | | <sup>13</sup>C shifts are an order of magnitude larger than the <sup>1</sup>H shifts, then aren't they going to be rather
abruptly cut off by the multi-part active filter on the analog input? I'm not sure how I'd go about having a programmable active filter, at least on
the analog side. ... |
Ignoring the <sup>13</sup>C shifts temporarily, consider a <sup>1</sup>H system that yields a Larmor frequency of around 30
MHz. If you were to RF amplify the signal, then mix it against a 30 MHz reference, the desired data would be in that audio range on up around 500 Hz
(17 ppm * 30 Hz-ppm = 510 Hz).
But remember that when you mix (same as multiply) that reference against the input RF you get say 75 Hz from both a signal 57 Hz above the reference
and 75 Hz below the reference. You just don't get filters narrow enough up in that part of the RF to dump the undesired frequencies while passing the
desired ones.
So generally you don't mix against the zero-shift Larmor frequency of 30 MHz, but rather an offset frequency of perhaps 40.7 MHz, giving an
intermediary frequency of 10.7 (used by broadcast FM receivers). For this the unwanted image frequencies would be 10.7 MHz on the other side of te
reference, 40+10.7 MHz, or 21.4 MHz different than the NMR ones. That's plenty far away that simple RF filtering will reject the undesired frequencies
in the RF section _before_ they get mixed with the reference - they just don't show up in the IF.
Then after a bit of amplification and filtering, you might mix against another reference to give an output in the range of 455 KHz - commonly used by
AM broadcast and various 2-way communication systems. Why that range? First the image will be 910 KHz away from the desired signals in the 10.7 MHz
IF, again fairl easy to reject by filtering; filters that are fairly low in cost can be used. And second, that's low enough that you'll find it easy
to have a fairly narrow bandpass at 455 KHz, with low cost commercial filters common.
Now finally you convert the 2nd IF to 'audio. But once again you don't want to end up with the no-shift Larmor at 0 HZ, signals near DC are more
difficult to deal with and best avoided if practical. And then there's the image problem, 2 Hz below and 2 Hz about the Larmor frequency (or its 455
KHz substitute) will both give 'audio' of 2 Hz.
Instead once again pick a reference that's offset from the Larmor, Let's pick a frequency 5 KHz away from the Larmor stand-in frequency, meaning NMR
data in the 5 KHz plus/minus a few tens of Hz.
So now your analog audio filter is centered around 5 KHz. what does its bandpass need to look like? Let us assume a 16 X oversampling, meaning we
sample at about 80 K samples per second. We want 'zero' energy up at that point, and as little as possible at the Nyquist - say around 7 KHz. It's
not too difficult to design an analogue filter gives good performance to fit. But we really don't need to cut off that quickly, the analogue filter
could have a high rolloff starting at say 10 KHz and still give good suppression at 40 KHz. Low side blocking is less critical, as the DSP side can
deal with it.
Ru in through the ADC and you've a 80 K samples/sec data stream. Pass it through some digital filtering to isolate the NMR range of interest - around
5 KHz. Decimate, FFT, only grab the bins corresponding to the frequencies of interest, then renumber them to actual NMR frequencies or shifts in ppm.
In this fashion you can pick the final frequencies you want, easily changing them by tweaking the digital filtering while leaving the analogue filters
fixed.
Having the analogue bandwidth greater than needed for <sup>1</sup>H NMR has advantages in that early set-up of the instrument is easier,
handy for mapping the magnet field where field strength varies enough to give shifts much greater than those for actual NMR work, and so on. For
that reason you might wish to pick as high an ADC rate as economic for the desired number of bits, allowing a wide band look at the
<sup>1</sup>H signals for development and shimming, then 'zoom in' for actual NMR work.
Example filters
10.7 MHz http://search.digikey.com/scripts/DkSearch/dksus.dll?Detail&...
455 KHz the CFL455G3 on this page http://www.surplussales.com/filters/filters-1.html
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12AX7
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Downside: two conversions adds two mixers' worth of noise, plus filters. I wonder how noisy ceramic resonators are.
The only signal below 0ppm is noise, so you're weighing converter noise versus double thermal noise over the same bandwidth. There is no external
noise (interference), since the whole system shall be very nicely shielded.
Tim
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not_important
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True, and I'd use just one IF myself, with a really good RF stage. It can be difficult to avoid image problems without an IF, unless you offset the
signals and do a fairly high speed ADC
During shimmy you can get shifts below apparent zero, it's handy to see them. Plus offsetting zero to see a bit of the negative shift range gives
you some feedback that the system is functioning propperly - lack of anything but noise in the negative zone is reassuring.
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12AX7
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Which is another good reason to set 0ppm to a few kHz, so you have the bandwidth below to see any shifts.
Tim
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densest
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If you have a "good enough" bandpass filter on your input, there is no energy in the images produced by subsampling. I've been sampling a bit of wine
and beer tonight so please forgive any stupid mistakes:
A technique to avoid multiple conversions is subsampling: deliberately sample at a submultiple of the Nyquist limit of your carrier frequency.
Current audio converters give 100 dB s/n up to 96KHz and a few op amps can work there without degrading things below 90 dB or so.
30MHz +- 12 ppm in (signal) or 30MHz +- 360 Hz
Nothing outside of 30 MHz +- 1 KHz is of interest, so throw it away!
Filter to pass 30MHz +- 250 KHz - bandwidth is wide to accommodate magnet variation. Each magnet set would have to be calibrated to set the stimulus
pulse and the filters trimmed to match.
Convert to 455 KHz
Filter +- 8 KHz (using AM radio technology)
Sample at (waving my hands) 48 KHz.
Digital filtering removes anything over 360 Hz.
This approach is commonly used in "software defined radios", removing one or all of the frequency conversions. If your input bandpass filter is narrow
enough and linear phase enough, sampling in the audio range yields good data.
Waving hands again: since we control the carrier phase and frequency, it is possible to null it out during one of the early conversions,
drastically improving the signal to noise (calling the carrier noise) ratio.
It isn't necessarily easy to do all of this well, but I've played with the math some and have played with circuitry and filters in the
required ranges enough to think it's possible. Phase shifts are deadly if not compensated for; luckily, our required bandwidth is so narrow
that simple filters will work.
Actually, 360 Hz around 30 MHz is narrow enough that crystal filters (see amateur radio circa 1960) would knock out everything but the useful signal
and provide excellent S/N - a single conversion afterwards would suffice. The tricky part would be adjusting the magnets to hit the bandpass of the
crystals. The advantage would be removing a great deal of the noise early in the chain. Such a filter is very very stable. Crystals ground to exact
frequency used to be amazingly inexpensive - a few dollars each.
Ceramic filters are good (60+ dB s/n) or better. I believe that there are significant differences between manufacturers and styles of filters: SAW
filters are very good at higher frequencies and have good S/N but are expensive unless you're using one designed for a big demand. L/C filters are way
less fun (many stages, critical tuning, and awful tempco even with T/C capacitors (N750, anyone?) but can be quieter if one invests in really
stable coil forms and lots of mechanical, electrical, and thermal shielding.
I am drifting in and out of this discussion and haven't read enough lately. Has anyone quantified the likely signal amplitude????? femtovolts,
nanovolts, or microvolts? Anything above 10 microvolts can be considered to be trivial to work with in this context, since we are looking at
multi-millisecond to fractional second data acquisition intervals.
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arsphenamine
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N52 Halbach segments for sale, 1.125T field claimed.
Plan B -- Stelter Array using 8 N52 1" cubes
Homogeneity is comparable at the 5mm gap core, the diameter of a standard NMR liquid sample tube.
FEMM4.2 file for stelter array:
Attachment: STELTER.FEM (4kB)
This file has been downloaded 846 times
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aliced25
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Ok, for the frequency synthesis side of things, here is something very interesting - the use of 2 PLL's in the one IC (here) and another paper on the use of 2 PLL's and DDS (here). Now both claim serious reductions in phase/spurious noise, while having read both there is an obvious road to reducing complexity by adding
in just a high speed DAC (instead of the DDS chip) between the 2 PLL sections of the one IC. The filters that are recommended aren't negotiable,
although I suspect we could work around the need for the mixer (from Paper 1 - NB It isn't used in Paper 2). If the DAC were simply spitting out a
monotonic output (of pretty much any frequency, wouldn't that translate to a monotone at the mixed frequency (ie. mixed via the PLL)?, which after
filtration should give a nice sine wave?
I'm also wondering, instead of fucking around with multiple filters, why aren't SAW filters used in this regard? They appear to have a smaller
bandwidth, so I'm obviously in the dark on something...
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arsphenamine
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Frequency locked to B field
ΔB/ΔT is important.
NdBFe temperature coefficient is -0.0011% T/°C,
or 11 gauss per degree for a 1T magnet which
corresponds to a -47 kHz shift in the Larmor Frequency.
Back when NMR electromagnets weighed a ton, it was easier to adjust the field directly. Varian described this in their "field-lock" patent
3,109,138 in which they determined the frequency drift from a built-in water sample and adjusted the field accordingly.
Given today's DDS oscillators, it is easier to frob the frequency knowing
that pure water's chemical shift is 4.8 ppm.
Modern NMR devices use a deuterium reference and call it 0 ppm.
Varian "Gyromagnetic Methods and Apparatus" patent
Attachment: 3109138__VARIAN_.pdf (433kB)
This file has been downloaded 636 times
[Edited on 17-10-2010 by arsphenamine]
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