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un0me2
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That MAX2306 Complete IF Subsystem, has dual synthesizers and dual output. If one were to run one input into the coil post-pulse as a carrier wave,
then it could be removed on demodulation, leaving noise + resonance. The sampling rates for the high-resolution sampling come down under the Nyquist
boundary if the only thing left in the sample is the resonance and noise.
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arsphenamine
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That MAX2306 is tuned by a RLC tank resonator circuit which
is inferior in accuracy and stability to an OCXO/TCO system.
Can it use an external clock source instead a tank circuit?
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not_important
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The MAX23xx uses an external reference, that's where the TCXO/OCXO come in to provide the stability needed. The PLL(s) in the chip take care of short
term noise; getting the desired offset for the IF desired may be a problem without using a switchable dividing PLL or DDS. Given that the FID is
acquired over 1/2 to 2 seconds, you need around 1 part in 10^8 stability over that interval to get good resolution.
Generally PWMs aren't that good for signal generation, being intended for driving motors and such. In the case of excitation for a FT-NMR it might
work, I've seen descriptions of such systems that used effectively narrow band noise for the excitation pulse.
Note that some of the devices you chose have LVPECL outputs.
Also "the simplest possible..." does not necessarily mean the fewest number of chips, particularly if those chips are being pressed into applications
on the end of their design space. Attempting to get away with only one coil makes the receiver + signal processing more demanding (hint - it's more
than just filtering, saturation and recovery time come into play).
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un0me2
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Microchip have dedicated DAC's and ADC's that will solve both resolution problems, high-speed with high-ish bit rates. I'm seriously looking at them
with the normal PIC32, the dsPIC's don't have enough benefits to justify getting rid of the memory & processing space (they can be run directly
from the PIC32 through the SPI/I2C interfaces as slaved devices).
I have to look further at the outputs, I am seriously going through at the present time looking at how to avoid having parts from too many
manufacturers, or the code to run this will be a Frankenstein-type monstrosity.
If I run dedicated high-speed DAC & ADC chips then the PWM channels on the PIC32 could be left alone waiting to be pressed into use as
electromagnetic coil drivers.
As to the utility of multiple circuits, etc. I am looking into designing a Quadrature Modulation - Demodulation circuit.
What I am thinking is that with multiple DAC outputs it should be possible to build the circuit that is needed. I was thinking that if the receiver
coil was running on the same rf signal as the larger rf pulse, it may cut down on echo & recovery time for the second coil. If the constant signal
is run through the output coil to the input coil, then both are running on the same frequency.
It is designing a switchable, much higher magnitude pulsed signal that is going to be the hard part. That said, if the pulsed signal is the same
frequency as the rf carrier signal, it should make it easier to filter out the echo from the pulse.
[Edited on 29-9-2010 by un0me2]
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watson.fawkes
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Quote: Originally posted by not_important  | | Attempting to get away with only one coil makes the receiver + signal processing more demanding (hint - it's more than just filtering, saturation and
recovery time come into play). |
There's also testing involved. I can't imagine how to a single-coil system
running without a second coil in the lab somewhere to test and exercise both the pulse generation and receiver circuits. And for a hobbyist who's
making exactly one of these, there's no particular reason to make a test coil that won't end up in the final device. I just can't see why using two
coils in the device is a downside in this scenario, where a certain amount of the test equipment cost should be considered part of the device cost.
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un0me2
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Coil design is a LOOOONG way down the track, getting hold of the electronic components, etc. is taking time enough. Let's finish the electronics
design argument, then we can get into the nitty gritty of the coil design.
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not_important
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Needed electronics depends a little on coil design. Coil impedance affects the transmitter and receiver designs, this isn't an electromagnet connected
to the power mains. A single coil needs much better protection of the receiver section.
Quadrature demod is often part of receiver chain chips, the shifted signal may be derived by that or generated in the clock generation section
depending on what you pick for each. Building it up from matched transistor such as Tim posted can give a bit better performance at the cost of more
components and more tweaking; more fully integrated ones are usually easier to use if you're not experienced in the art.
I've worked on products that used processors from 4 different families, no code nightmare - each is a separate section, data and control signals going
between them are interface protocols tht you have to understand anyway no matter how many types of processors.
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un0me2
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Can someone please tell me I'm reading this wrong, I'm tired as hell and have been looking at this shit all day (so it is on the cards), but this
Application Note (#2009) "Performance of the MAX2395 PLL with 80kHz Comparison Frequency" seems to me, to be suggesting that with some careful, well, there really
isn't a better word for it, hacking of the external circuit - the [url=http://www.maxim-ic.com/datasheet/index.mvp/id/3974[/url] (rated @
1920-1980MHz) can be made to output at significantly lower frequencies (500Hz-~20MHz). If that is the case, then that really does provide a good
solution to the problem.
PS I have noted several online discussions about the utility of the Si5xx series of OX/VXCO's and am wondering if anyone has experience with the
Si401x series of programmable (27-960MHz) transmitter on a chip? From what I can see of the diagram, a DAC could provide a high-powered pulsed-tone
(for x-nsec) @ the excitation frequency & that would be that?
[Edited on 2-10-2010 by un0me2]
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not_important
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Why are you trying to run this at 500 Hz? I can't see a good reason to do so.
Likewise I'm puzzled by what you're trying to do with the oscillator. For the excitation coil you gate the power amp on for some number of cycles of
the excitation frequency, the pulse is a number of cycles long. I.m not fitting that to what you propossed.
What are you hoping to be able to do with the NMR system you want to build? The intended application sets some requirements for things such as
resolving power and stability of signals. Are you after measure water content in ice or hydrocarbons? Easy. Analytical and structural determination?
Harder. Doing 1D or 2D? You need to set out your target clearly before you start doing designs.
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un0me2
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No, not the point - they are achieving outputs several hundred MHz under what it is rated for.
As for the oscillator, I'm trying to time the DAC output, high powered tone @ the right frequency, added into the RF & then only the programmed
period of output from the DAC, gives the power & the pulse, no?
That said, the dual-DAC's and the MCP3901 dual 16-24 ΔΣ ADC's (with dual output), should really improve the signal quality (so should proper separation of the
Digital/Analog sections, grounding, etc.). The improvement in the Microchip Library means that the data can be dealt with (DFT/FFT/RADIXn) prior to sending it. Connecting the DAC to the PIC is dealt with in this
Application Note, while some of the design considerations are dealt with in this Application Note. The code for several processors, the Texas Instruments one, DSP, etc. are on the web, while these IC's look promising (damn, a discrete chip returning the FFT of signals virtually in real-time? That should change shit )
With the transmission/reception modulation/demodulation, is it possible to transmit a baseband just under the LF, ie.40MHz? Then add the excitation
pulse with 2.xxxMHz? That would allow for the same signal processing train...
[Edited on 4-10-2010 by un0me2]
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not_important
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| Quote: | | As for the oscillator, I'm trying to time the DAC output, high powered tone @ the right frequency, added into the RF & then only the programmed
period of output from the DAC, gives the power & the pulse, no? |
Still not making sense to me. If you have the RF, what is the DAC output tone for? If your thinking of using that as the pulse gate, it's something
you'd use a DAC for given the roughly 4 or 5 orders of magnitude of off vs on times.
And what you want the NMR to be capable of doing is _very_ important, as that drives the design.
1) what do you want it to be able to do?
2) determine needed specifications - field strength and uniformity, frequencies, noise levels, phase stability, ...
3) figure out that hardware you want to use to satisfy 1) and 2)
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aliced25
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Sorry cannot access other account (for anyone who is reading this 2+2 generally does =5, there is always a part of the calculation that comes down to
common sense).
not_important,
What I am trying to realise is a high-speed, high accuracy digital-to-analog and analog-to-digital signal train. For the most part (except for one
offering from Silabs) there really isn't a whole lot that suggests this is feasible from one manufacturer, which is a bugger. That said I have decided
to "evaluate" the AD9852ASTZ for the signal generation and the AD7606 8-channel 16-bit, high-speed ADS (ADC by any other name) for the return signal.
Without using a carrier wave that means there is going to be a god-awful discrepancy between the initial signal strength and the end-strength, so I'm
thinking of looking at putting in a switch (to ground presumably), sending the overloaded signal generated immediately upon excitation (and a little
while thereafter) into the dirt (so to speak), then passing the lot through a low-pass filter with a cut-off immediately above the highest resonance
signal which is likely to be generated.
That would reduce the amplitude of the samples, thereby reducing the effect(s) of the echo/etc, while sacrificing some resonance data.
AD is coming out with a 400MHz Blackfin for ~$3-5 by the end of the year, which would see them squarely in the running for the MCU/MCP (at that speed
the FFT/Radix-n of the samples should be almost real time).
On the other hand, Microchip are still in with a very good chance, the PIC32MX795F512H (with integral USB 2.0 transceiver), 512kB Flash & 128kB RAM can be used (I2C/SPI) with the MCP3901 Dual-Channel 16/24-Bit ΔΣ ADC's (up to 64ksps, up to 5 on a SPI/I2C IIRC), while the DAC - MCP4728 is not only a DUAL 12-bit DAC, but it is cheap & fast as hell (and according to this paper, should be quite capable of handling the DDS, while they provide a program to design low-pass & stop-pass filters that work up to 10MHz (presumably anything above that is discarded).
Working out how to design a decent board that will allow me to separate the analog & digital ground planes while breaking out the 64-100 XXQFP
chips is going to be a problem, but I was thinking, once the digital end of this is set up to send analog signals to the analog end and to receive
& process the responses to the same, that is a digital spectrometer, right there (actually, given the ability to do the FFT/RADIX-4 on chip, an
FT-Spectrometer), which could then be utilized to control a laser (FT-Raman), a UV-Vis Light Source (FT-UV-Vis), an IR light source (FT-IR). The
spectrometer is essentially the same (the digital end anyway), digital to analog control signals, analog to digital response signals and digital
processing. With some thought, that could be taken way further, the pwm/dac outputs that aren't being used could be used to drive a moving stage, etc.
[Edited on 6-10-2010 by aliced25]
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not_important
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| Quote: Originally posted by aliced25 | ...
Without using a carrier wave that means there is going to be a god-awful discrepancy between the initial signal strength and the end-strength, so I'm
thinking of looking at putting in a switch (to ground presumably), sending the overloaded signal generated immediately upon excitation (and a little
while thereafter) into the dirt (so to speak), then passing the lot through a low-pass filter with a cut-off immediately above the highest resonance
signal which is likely to be generated.
That would reduce the amplitude of the samples, thereby reducing the effect(s) of the echo/etc, while sacrificing some resonance data.
... |
If by "initial signal" you mean the excitation bang, that's true. That's also why you isolate the receiver from the pick-up coil and have the pick-up
coil at right angles to the driver coil. Isolation is done by several methods, a simple one is to put a diode across the input and run DC through it
during the excitation pulse; the DC putting the diode into low-Z conduction mode, no DC and the low amplitude of the desired signal means the diode is
a hi-Z shunt that doesn't affect the signal.
But it's still not making much sense to me as to what is you are attempting to do. Possibly it's my never made it past systems engineer in the world
of electronics.
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aliced25
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OK
I'm going to have to do up a diagram showing the concept I've come up with...
Can anyone help point me in the direction of where arsphenamine came up with the 160MHz figure? IIRC it was the maximum resonance return that could be
expected from a 15ppm shift.
I would just like to confirm this (as I am using TI excellent FilterPro Program to design the filter for the analog return signal, I'm thinking put
the passband @/~ 160MHz and the stop band @/- 240MHz (that, with a gain of 50v/V or 33.98dB (and a negative gain of -50dB for those wavelengths
outside the band), returns as a 4-Stage (8th Order) Chebychev Filter (with a straight line @/~160MHz), so I'd like to make sure that I have it right
before I source the components & build it).
I then have to determine how to synthesize an accurate frequency (the AD985x series aren't filtered).
I also then have to work out which amplifier fits with which ADC combination, the trick is going to be the bandwidth 160MHz is still pretty chunky,
especially if the majority of the signals are 1/10th of that.
I'm looking toward the fast Delta-Sigma ADC's interfaced to the MCU (whichever variety), with several small (1/2 channel) ADC's running on an
I2C/SPI/Parallel bus, sending the data into the processor. As the Delta-Sigma ADC's send, effectively, a noise-reduced, averaged signal, then
collecting that will give a truer representation of what is in the mix.
But I'll quickly explain the system as I see it, purely the spectrometer end - the analogue end will have to be worked out later.
Interface + MCU
USB 2.0 Port <==> MCU (with USB Transceiver + SPI/I2C/Parallel buses).
Output
The output will consist of either dual interleaved DAC's with a sine table & oscillator or a Programmable Integer-N Synthesizer + at least 1 PLL
with onboard clock, etc. That should really give a fairly accurate signal (based upon what I've read). As the FGPA's have really only turned the
output on and off, I'm confident I can achieve that with the MCU. The output can then be amplified until it meets the needs of the coil. The output
will then be passed to a 50 Ohm RF Connector Socket.
Input
The analogue input will be taken from a 50 Ohm RF Connector Socket & filtered down to 200MHz (max - see attached). The filtered analogue signal
will then be further amplified. It will then be passed to the multiple Dual ADC's (either high-speed, high-resolution SAR/Delta-Sigma). The ADC's will
then transmit the 16/24 bit signals to the MCU via the relevant bus (all the ADC's will be slaved).
Digital Signal Processing
The data stream from the ADC's will then be further processed, with 2 filter algorithms, one for echo cancellation, the other for noise cancellation.
They will then be averaged across the time domain and the average will be kept. They will also be, best case scenario (according to the application
notes it is well within the MCU's capacity), an FFT/DFT will be performed on the entirety of the sample/data-set, giving an accurate FT-NMR result
which, with the sample average, can be sent to the PC/Laptop/SD Card/etc.
Other Issues
I will not know until I can test the board & then try it with a coil in the magnet assembly, whether the level of homogeneity is adequate. If it
is not, Golay Coils will be needed, but as no use has been made of the MCU's onboard DAC/PWM, then it should be possible to utilize these controls for
the Golay Coils.
Attachment: FilterPro.Design.Report.Schematic.Version.1.0.pdf (269kB)
This file has been downloaded 780 times
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arsphenamine
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| Quote: Originally posted by aliced25 | | Can anyone help point me in the direction of where arsphenamine came up with the 160MHz figure? IIRC it was the maximum resonance return that could be
expected from a 15ppm shift. |
No, because I never wrote that.
The signal return is estimated by the ratio of up-down proton spin states
and which, at its heart, contains a magnetic field term over a Boltzmann thermal distribution.
The ratio is very small, but the great number of protons in
organic matter means that it is sufficiently detectable.
The NMR wiki has links to good pedagogical material.
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watson.fawkes
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Yeah,
that's kind of a problem, isn't it. You're trying to measure frequency shifts in the range of a few parts per million, and you need a frequency
reference with stability of less than that. One bad stage anywhere in your transmit/receive chain and you'll lose it all. Since I haven't seen it
mentioned before, only hinted at, drink deeply the concept of "error budget". I did a brief search and this introductory article from Electronic Design came up first. It's by Bob Pease, and it's reasonably motivational. Now with this project it's not just a single
circuit, but an entire instrumentation chain. Nowhere in this whole thread have I seen any component choice constrained by error analysis. And I don't
mean anything about bit lengths or bandwidth queenery. I'm talking about the analog stages. It doesn't matter if you're sampling 24 bits at 1 Ghz if
your analog noise is up at 100 ppm.
I would heartily recommend doing this project in stages, and to make the first stage be a CW transmit/receive pair at the frequency of interest,
paying special attention to frequency error issues. What's your calibration standard going to be for frequency stability? It's a pretty basic
question, because it's relatively easy to make an oscillator that looks stable to the naked eye, which really means that it's at least as stable as
your oscilloscope. Every oscilloscope has some internal timebase error, so you can't determine stability better than the capability of your scope. So,
what's your calibration standard?
And I'm not really referred to absolute frequency measurements, because you can eventually determine that with known samples. I'm speaking of
calibration for oscillator stability, such things as jitter, phase noise, frequency drift, and the like. You'll need a stable time base somewhere in
your lab, or access to one in someone else's lab. You could, for example, build an OCXO unit or buy one. I did a little looking around, and the
cheapest one Digi-Key sells near your band, AOCJY-100.000MHZ-F, is $144 retail. It's got +/- 30 ppb stability (that's "b" as in "billion"), so that'd be adequate for measuring. Or you could
build one. An OCXO unit all by itself would be a solid little open hardware project. It'd be particularly useful because you could more easily get
custom crystal frequencies. But back to the question: what's going to be your metrology standard for frequency stability?
After you've got CW working at some known stability, then you can worry about high power and pulsed switching. This isn't trivial either, since the
transients induced by switching definitely have the capacity to wreak havoc with your frequency stability. But you can't possibly know if they are or
not in the absence of a measurement and calibration regime.
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not_important
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Seconding watson.fawkes It's worth getting a highly stable frequency reference. Use a PLL+dividers to get other frequencies
(f<sub>r</sub>*M/N ), use a DDS chip to get fine control of frequency and phase and to quickly shift between several pre-selected
frequencies. Filter those outputs with bandpass or low pass filters to get good clean sine without higher frequency stuff.
Remember that for many of the applications you want to do, your talking about resolving sub-part-per-million values and collecting data over hundreds
of milliseconds; if your base time/frequency reference isn't up to it, it matters not how good the analogue chain is or how fast your ADC and DSP are.
A good OCXO is worth it, pick a useful frequency that conveniently supports the other frequencies and time intervals you need (20 MHz gives 50 nsec
resolution and all that). This is especially true if you do spin-echo to compensate for inhomogeneity in the magnetic field.
I'd separate the ADC+FFT/DSP section, basically the digital stages, from the analogue front end - put them on separate PCBs with low-Z analogue
signals plus SPI/I2C controls and maybe some other simple digital signals; USB goes on the digital board. This makes it easier to get low noise in
the front end, keeps all the noise from DSPs and RAM away from it. Also there is fair to considerable differences in the analogue for the
applications you listed, it would be simpler to do an application specific front end. FT-NMR needs strong signal generation plus low level
amplification and translation to lower frequencies. FT-Raman sees a fairly narrow spectrum, input signals of under a MHz, and ramped or stepped power
output; Ft-IR is similar except the spectrum is wider and the signal band of lower frequency due to slower detectors. Doing FT-UV-Vis is somewhat
demanding on the optics including sensor, you're talking at least 300 to 800 nm, 200 to 1000 for a good system and sensors for that much range are
likely rather pricey - the FT techniques often used for wideband UV-Vis typically use 2D detectors. For the optical stuff you often have a reference
channel that is processed the same way as the sample one, twinning the front end. No offsetting from some reference frequency, no IF stages, oftn no
I/Q signals.
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arsphenamine
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re: ppm vs. Hz calculations
Assumption: anything you want to see in 1H NMR is within 15 ppm of the Larmor Frequency,
although most of it is within 8 ppm.
ppm = parts per million from which you infer a 1e-6 multiplier.
15 ppm of a 42.5MHz Larmor Frequency = 15e-6 * 42.5e6 = 637.5 Hz.
The phrase downfield shift was current in pre-FFT (viz., continuous wave) NMR spectroscopy.
Back then, it was easier to decrementally sweep the magnetic field than with an RF oscillator.
A 15 ppm shift in 60MHz 1H NMR is equivalent to a 0.2 Gauss field change.
0.2 Gauss.
A miniscule 0.2 Gauss.
An amount slightly less than the minimum observed Earth magnetic field.
In a 14092 Gauss field.
You need a frequency standard that is at worst 100x as stable over
the measurement interval, i.e., no more than 150 parts per billion.
The fineness of field and frequency control required is extreme
even by today's technological standards.
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not_important
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Note that you don't need the high quality reference to start with, you can use a simple crystal oscillator to get things debugged, and even see some
NMR results. But for any sort of analytical work you'll need to sub in a hig-stab source.
| Quote: | | analog return signal, I'm thinking put the passband @/~ 160MHz and the stop band @/- 240MHz |
'The results in your PDF are for KHZ, not MHZ ...
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arsphenamine
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+1
A search on "crystal oven" turns up some delightful projects ranging from
a pure analog design that heat the crystal with a power transistor and sense by thermistor,
to microcontroller PID systems using the venerable LM35 linear temp sensor.
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aliced25
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| Quote: Originally posted by not_important | Note that you don't need the high quality reference to start with, you can use a simple crystal oscillator to get things debugged, and even see some
NMR results. But for any sort of analytical work you'll need to sub in a hig-stab source.
| Quote: | | analog return signal, I'm thinking put the passband @/~ 160MHz and the stop band @/- 240MHz |
'The results in your PDF are for KHZ, not MHZ ... |
Um, yeah sorry about that - I was thinking something else entirely and wrote the wrong thing (My bad), the actual filter design was intended to
cut-off from 240kHz with the 0-160kHz pass. I do seem to "recall" reading that some of the expected return was in the 150kHz region, if not that makes
life easier, the smaller the bandwidth of the signal fed to the ADC, the smaller the reference bandwidth and the better the result.
I'm actually looking for a way to narrow the generated signal - ie a lowpass filter, at the requisite band, which would also help to ensure stability
in the bandwidth (but not necessarily the strength transmitted) of interest.
As to physically separating the digital & analog, I was thinking that if separate ground planes were used it should be possible to keep the noise
down (especially if the power source is separated), these have everything on one board and I'm presuming the signal integrity would be at least as high. It's also good to see things like that, it
helps me to realise that I'm not "inventing the wheel" (for all that I AM reinventing it somewhat)
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not_important
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Separate grounds is a requirement given the low level signals. But NMR will need a bunch of more-or-less analogue functionality that the other
instruments won't, makes more sense to build that into a head for NMR and send the AF-ish signals to the digital section over low-Z twisted pair.
Also note the phase shift for the amp design, that might be important in some applications.
Note that crystal oscillators may have short term jitter and hase instability, syncing a PLL with a rather low pass feedback can improve those
factors.
As those pages note, good insulation surrounding the OCXO is helpful. Enclosing that in a container with high volumetric heat capacity (Cu or Fe) can
also be helpful as it reduces the impact of sudden changes in external temperature.
| Quote: | | I'm actually looking for a way to narrow the generated signal - ie a lowpass filter, at the requisite band, which would also help to ensure stability
in the bandwidth (but not necessarily the strength transmitted) of interest. |
Which generated signal? The received one? That's what one of more IFs help with, followed by low pass filtering of the 'audio', high rate
oversampling of the signal (means that a wider range of frequencies above the band of interest get properly digitised without folding over) followed
by digital low-pass filter.
[Edited on 10-10-2010 by not_important]
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aliced25
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I was actually talking about the 43-44MHz signal, the use of a filter on that to keep the output within a specific narrow bandwidth is what I'm
looking for.
As for filtering and amplifying the return signal, if the resonance signal is in the "audio" range, then the number of options in the way of low-cost,
high-precision filters, amplifiers, etc. grows exponentially (some go to 100-120kHz). High speed ADC's, including MSPS ΔΣ ADC's with
oversampling, are much easier to match to the audio bandwidth than any other (obviously, that is the purpose behind the manufacture thereof). High-end
audio components cost next to nothing when dealing with IC's, they are high-volume products.
I'm looking at Digital Interpolating Filters for the DAC's and/or integer-n PLL's (all digital PLL's are available) but fully integrated circuits
aren't available at the 40-50MHz range unfortunately (well some are, but they still need cleaning up). High speed interleaved 12-bit Dual DAC's with
PLL & a filter would allow for much easier pulse-generation & timing than would be possible with other components. Just pre-program the table
look up and the pulse length & subsequent wait period, with the number of pulses also set by programming.
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12AX7
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http://wingzero.ath.cx/index.php?page=xtal-osc-varactors
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not_important
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| Quote: Originally posted by aliced25 | | I was actually talking about the 43-44MHz signal, the use of a filter on that to keep the output within a specific narrow bandwidth is what I'm
looking for. |
It's called a tuned circuit, or a bandpass filter - both done with inductors and capacitors. With most M/N PLL and DDS chips, the leftovers from
generating the target frequency are much higher than the desired output, simply filters take car of them.
| Quote: | | As for filtering and amplifying the return signal, if the resonance signal is in the "audio" range, then the number of options in the way of low-cost,
high-precision filters, amplifiers, etc. grows exponentially (some go to 100-120kHz). High speed ADC's, including MSPS ΔΣ ADC's with
oversampling, are much easier to match to the audio bandwidth than any other (obviously, that is the purpose behind the manufacture thereof). High-end
audio components cost next to nothing when dealing with IC's, they are high-volume products. |
The FID signals are standard NMR, plug in your magnetic field and see what the value will be - 10s of MHz for the fields you've been talking about.
You need to mix the received signal with the high stability reference to generate the lower frequency stuff. You still need to worry about the image
on the other side of the reference, which is why there's generally at least one IF stage - take a, say, 80 MHz FID signal, mix it down to 10.7 MHz,
the image frequency is 21.4 MHz away, 10.7 MHz on the other side of the reference, so your RF front end just needs to block that range well (in this
case 1/8 the reference or desired signal, no prob). Then demod the IF to AF, or use a second lower frequency IF followed by down-conversion to AF.
As noted earlier, generation a reference + 200 Hz (for example) frequency for the down conversion means that the signals you want will all be in the
audio range with no DC or really low frequencies to worry about.
| Quote: |
I'm looking at Digital Interpolating Filters for the DAC's and/or integer-n PLL's (all digital PLL's are available) but fully integrated circuits
aren't available at the 40-50MHz range unfortunately (well some are, but they still need cleaning up). High speed interleaved 12-bit Dual DAC's with
PLL & a filter would allow for much easier pulse-generation & timing than would be possible with other components. Just pre-program the table
look up and the pulse length & subsequent wait period, with the number of pulses also set by programming. |
I still don't see this, it sounds quite complex for just generating pulses of RF; DACs are just not making sense. Just use a diode bridge or balanced
mixer to gate the RF. The RF should be derived from your master reference, and you don't turn oscillators on and off if you're expecting any sort of
solid, stable signal out of them. The control pulses come from timers counting at a high enough frequency to give the needed resolution, also derived
from the master reference. Some signal generator (PLL/DDS) chip include smart gating that allows near zero-crossing gating; or you can Schmitt
trigger the RF and count cycles to generate gating signals locked to the RF.
BTH - the DACs you referenced in a previous message won't hack it - check the settling times, and how long it takes to get a setting into the DAC even
at max control line rate.
[Edited on 10-10-2010 by not_important]
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