DDS Selection

I am designing designing a system that uses changes in the transmissivity light in a detection chemistry to detect gas. The light source is a sine wave modulated LED source. After the light passes through the detection chemistry it will be measured by a photodiode and a transimpedance amplifier circuit. The output of the transimpedance amplifier is measured by a 16 bit AD7606 ADC which does a DMA data transfer into a Blackfin DSP where a Fourier Transform is done to extract an amplitude at the LED's modulation frequency.

There will be six detection channels. Each channel is to be modulated at a different frequency to reduce crosstalk. The length of time the Fourier Transform is done over will be in multiples of 100ms for maximum rejection of both 50Hz and 60Hz. The modulation frequencies are chosen in multiples of the inverse of this length of time for maximum crosstalk rejection.

Between the output of the transimpedance amplifier and the ADC is an anti-aliasing filter. Its cutoff frequency and number of polls are chosen so the attenuation at the ADC's sampling frequency is equal or less than the ADC's LSB divided by the ADC's total number of states (2^16).

The system needs an option for battery power so current draw is an issue.

I need to choose a DDS chip to generate the sine wave reference for the LED's modulation. I have been looking at 14 Bit DDS chips at:

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I notice there is a dramatic increase in power requirements the more bits the DAC part of the DDS has. I would like to choose a 10 bit one such as AD9838 that uses little power but I am concerned about the quantization noise that appears as a spurs in the DDS output beginning at twice the output frequency. That spur would be within the anti- alias filter's passband. Theoretically this spur would appear right in one of the notches of the sync function that is the frequency response of a Fourier Transform and so would be taken out by it. In practice can I actually count on it doing that? The DDS has the same frequency reference as the ADC's sample rate so the spur would be precisely located there.

Or do I need a reconstruction filter for the DDS that would have a cut off between the modulation frequency and the first spur at twice this frequency? I am sure it would need many poles. If so could the anti- alias filter, although it is at the output of the transimpedance amplifier instead of the DDS, double as a reconstruction filter if it would have the same cutoff frequency and poles?

Reply to
spflanze
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Why not use a synchronous detector (lockin) architecture?

John

Reply to
John Larkin

If we did a mental exercise and reduced the bits in the DDS, eventually it would be a square wave. So wouldn't the first noise appear at 3x the modulation frequency if the modulation frequency and clock rate have an integer relationship. If so, then your post filter is substantially easier to build.

Reply to
miso

All that just to modulate a LED with sine wave output? Sounds a bit over kill to me.

One item on the list you need to consider, the LED is not linear in its transition to current verses emissions. The first item on the list should be a driving circuit using a photo feed back to ensure you have a linear representation of photons emitting through your chemistry.

Generating the SINE wave in short, can be done with out the use of DDS technology, which just adds to the pile of over kill.

Using an analog method of generating that sine wave with a synchronous signal for the receiver would be more to what I would expect to use.

Jamie

Reply to
Jamie

You designing a homework.

Reply to
Vladimir Vassilevsky

Probably not. A student following a sensibly designed course wouldn't be thinking of using DDS to get a sine wave, nor of applying a fourier transform to the data collected to find out the amplitude of the sine wave getting through.

This is almost certainly a graduate student who has done some reading, but of the wrong kinds of books. They need to read Horowitz and Hill's "The art of electronics" which was designed as an electronics text for bright physics students (at Harvard) and covers useful subjects - like lock-in and phase sensitive detection - in some detail.

God only knows why spflanze thinks you need to use sine-wave modulation on a signal being used to measure an extinction coefficient relatively slowly.

We could usefully know more about what is actually being measured - what are the six separate detection channels actually doing?

-- Bill Sloman, Nijmegen

Reply to
Bill Sloman

Right. Then instead of doing an FFT, just synchronously demodulate (digitally, if you want) and lowpass filter or boxcar average. You don't need a DSP source or a Blackfin for that.

John

Reply to
John Larkin

Nah, LEDs are pretty linear above about 10 uA. I second the suggestion to use square wave drive and a lock-in per channel, i.e. an AM radio with synchronous detection. (AM radios are pretty good at adjacent channel rejection, for such simple devices.) For instance, the OP can pick drive frequencies of, say, 12, 13, 14, 15, 16, and 17 kHz, with 100 Hz-ish lowpass filters after the demodulator. One wants the low order harmonics of each modulation frequency to miss each other, because that helps a lot with the spurious signal reduction. If he doesn't want to lose all that nice software work, he can write a program to find the optimum channel assignments that minimize crosstalk.

Square wave lock-ins are sensitive to odd harmonics, but a reasonably good lowpass filter after the TIA will get rid of those, as well as noise contributions from their neighbourhoods.

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
845-480-2058

hobbs at electrooptical dot net
http://electrooptical.net
Reply to
Phil Hobbs

Why do a simple analog solution when you can throw a computer at the problem. ;-)

I'm probably not following the need for a lock-in since he has all the oscillator sources at hand. You don't have to lock, just demodulate the photo signal using the modulation source. Quite possible I'm missing something here.

If the modulation circuit also produced a quadrature clock source, then the photo signal could be IQ demodulated. One path would be the signal level, and I suppose the other path would be a figure of merit of the noise. [All assuming no significant phase shift in the photo signal.]

Reply to
miso

"Lock-in amplifier" refers to syncronous demodulation followed by low-pass filtering.

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John

Reply to
John Larkin

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Strictly speaking, you've got to have a modulation on the signal being detected before you can lock-in onto that modulation and synchronously detect it.

Normally you excite the system under test with some sort of alternating signal - sine wave are often handy, square wave are easier to make, but repeated narrow pulses at regular intervals can be more appropriate for some problems.

-- Bill Sloman, Nijmegen

Reply to
Bill Sloman

Reading the description, lock-in sounds like more than just sync demod plus LPF. But we are in agreement that is what would be done to recover the signal. Generating I and Q with the modulation source would make this task relatively simple, plus yield a figure of merit signal.

Reply to
miso

This is actually a better description. There is really nothing to lock. You have the source already, so just feed it to the demod.

Reply to
miso

Lock-ins don't phase-lock to the input--otherwise you couldn't measure the static phase. You supply them a reference, which in this simple situation is the same as the drive signal, phase shifted to match the pre-detection filter (e.g. the RF filter in an AM radio). Alternatively, since it's only the amplitude he cares about, he could phase-lock to the detected signal. That can get exciting in a multichannel setup, though.

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
845-480-2058

hobbs at electrooptical dot net
http://electrooptical.net
Reply to
Phil Hobbs

But you don't really care about the static phase shift, other than if it is significant, then there is an error in the amplitude if the demod is not quadrature.

I think we're in agreement. You have the reference and demod the photo signal, beat down to DC. With quadrature demod, you could be phase independent presuming the I and Q are demodulated.

Reply to
miso

The point about the static phase is to distinguish between a lock-in and a PLL, which you seem to have been conflating.

And in the present case, you don't want to measure the phase, but you do care about it--you want it to be zero, ideally in an architecture where the temperature drift cancels to leading order.

You don't want to use I and Q, because if the phase is adjusted right, Q contains precisely zero information, and if you let the phase be just anything, you have to reconstruct the amplitude from I and Q, which costs hardware and extra engineering time, and gets you nothing whatever. It also puts a serious premium on the linearity of the phase detector, and on the accuracy of the waveform. Switching at the zero-crossings is about the most accurate way to use CMOS transmission gates for this job.

On a custom chip, I'd want to use a PLL, because the amplitude channel is insensitive to jitter, and making performance independent of normal process variations in RC products and GBW would improve yield. You'd need a Q channel for that, of course.

In a built-up circuit, though, complexity costs money, and accurate RCs are cheap, so I'd make the TIA wideband, and use a filter architecture that has zero phase shift at the desired operating frequency, such as cascaded 2-pole HP and LP filters.

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
845-480-2058

hobbs at electrooptical dot net
http://electrooptical.net
Reply to
Phil Hobbs

Fun discussion, Sorry, but I can't help you with the DDS selection. I'd fourth the lockin/ demod approach. Are your six LED's all different wavlenghts? (and if not why six?)

To the group... is there some point where it is advantageous to use the FFT technique. (what if he had more sources) (I'm thinking of FIR spectroscopy and the Fellgett advantage.) At least with one detection channel you don't have to worry about channel to channel variation.

George H.

Reply to
George Herold

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At some point making all those separate filter channels and PSDs becomes more of a pain than doing it all digitally. A windowed FFT gives you a whole bunch of identically-shaped filter channels centred on each frequency sample, which is a very nice feature.

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
845-480-2058

hobbs at electrooptical dot net
http://electrooptical.net
Reply to
Phil Hobbs

That would be true if it were a perfect DAC with infinite bandwidth. But there is a finite bandwidth and finite slew rate so its output is other than a succession of perfect squares. Triangular elements are involved and those have even harmonics.

Also there are other spurs that can be present the time length of each square will vary slightly depending on the lower order tuning bits. The variations happen when the bits that are discarded between the phase accumulator and the phase to amplitude converter vary. And that happens when any of the corresponding bits in the tuning word are set.

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Reply to
spflanze

An FFT is a digital demodulation with a box car average. An FFT multiplies by a sine wave and extracts the 0 Hz difference frequency by averaging.

Reply to
spflanze

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