New data acquisition moduleMost importantly, I replaced the spectrum analyzer by a self-made data acquisition unit which is basically a Software-defined radio (SDR). Its purpose is to take the 1.4 GHz signal, digitize it, and send it to a computer.
Functionality descriptionThe board contains various submodules, most of which are labelled in the following image.
|Some of the components of the acquisition board.
Mixing works by multiplying the actual signal with a reference signal, the so-called Local Oscillator (LO) [data sheet of part used here]. You can imagine the LO as a component which generates a sine wave at a specific, but programmable, frequency (in this case around 1.27 GHz); then the mixer basically subtracts this frequency from the frequency of the original signal (so in this case, with 1.42 GHz and 1.27 GHz you end up at 0.15 GHz = 150 MHz). The lower-frequency signal (the 150 MHz in this case) is called the Intermediate Frequency (IF) signal.
Next, there are various filters and amplifiers which are very simple to understand; they just make the signal amplitude large enough so that it can comforatbly be detected by the analog-digital-converter later on.
This part of the circuit ends with a so-called Surface Acoustic Wave (SAW) filter [data sheet of part used here], which is a quite modern kind of filter with the special property that it can select a relatively small part of the signal very accurately, while blocking everything else almost completely. It only lets the frequency components of the signal pass which have frequencies between about 148 and 151 MHz. Which part of the original signal can be found in this range (remember, we move the original signal to a different frequency) is determined by the mixing step: through programming the local oscillator (LO) for a different reference frequency, one can look at the input signal's spectrum one part at a time.
The bridge between the analog and the digital world is built by the Analog-Digital-Converter (ADC) [data sheet of part used here]. It gets an input clock from the microcontroller, which is just a signal which toggles up and down exactly 11.25 million times per second; on each tick of that clock, it measures the voltage at its input pin, and converts it into an 8-bit digital value. Each of the bits of that value is then provided as a voltage level (either +3.3V for "1" or 0V for "0") on one of the output pins. They are read in by the microcontroller [data sheet of microcontroller], packed into chunks of a few kB in size in the controller's RAM, and then sent out via USB.
Getting the microcontroller firmware to do this as fast as required was surprisingly nontrivial, but that's such a broad topic that I do not want to get started on it here.
The computer receives the data, calculates the fourier transform to get the frequency spectrum, and averages the resulting spectra (see end of post for software screenshots).
(If you ever heard of the Nyquist frequency, you might wonder how it is possible to digitize a 150 MHz signal with just 11.25 MSa/s sample rate. This is why we need the narrow SAW filter in front of the ADC: it limits the bandwidth of the signal to 3 MHz and thus allows to use the undersampling technique. In this case, the ADC samples the 26th alias of the original signal.)
Below is an illustration of the paths the signals described above take through the board. Note that almost all of the blue path is at 150 MHz; just the first centimeter up to the small black mixer IC is at 1.4 GHz.
|Signal paths in the data acquisition board.
Manufacturing the boardThis is a 4-layer board; that means, it has tracks on the top and bottom side, and two metal layers in between (you cannot see them, they are glued inside the board) which also have tracks. Connections between the layers are established by metal-filled drill holes (in this case about 1000 of them, each 0.3mm in diameter).
Having 4 layers makes it much easier (or possible at all) to arrange tracks in a way that they do not cross, and also has some important advantages for high-frequency signals. Unfortunately, it also makes fabrication of the prototypes more expensive. There are various chinese vendors which will manufacture a small amount of pieces of such a board for an affordable price; including customs and shipping it will end up costing about 100 € for 5 pieces (less pieces doesn't get cheaper though). Components are extra and you have to put them on the board yourself.
There is specific software to design such circuit boards; first, you usually draw a schematic, and then you use the layout editor to place components and tracks in accordance with the schematic. The layout itself quickly becomes hard to overlook; the software guarantees electrical equivalence between the easy-to-read schematic and the layout (e.g. it guarantees you don't have wrong connections, missing connections, or tracks which cross or are too close together -- of course only if you did it right in the schematic).
The software exports vector-graphics like files for each layer and a list where holes must be drilled, which is then sent to the manufacturer.
|This board's layout loaded in kicad. Only the top (red) and bottom (green) layers are shown.
|A raw board without components (but with soldering paste on it) and the stencil for the board. Also an example for how not to use a stencil.
|Board with solder paste, ready for component placement.
|Board with components placed, after heating it up and letting it cool down again (and fixing a few things by hand). Most of the solder joints look pretty good.
Mirror filterWhen mixing a signal to a different frequency as described above, there is one problem I did not mention so far: if I mix 1.27 GHz with a broadband signal (as it comes from the antenna, it doesn't only contain the 1.42 GHz part but lots of uninteresting noise at basically all other frequencies as well) of which I am interested in the 1.42 GHz part only, there are actually two frequencies of the broadband signal which will end up at 150 MHz in the intermediate frequency (IF) signal. This is the 1.42 GHz = 1.27 GHz + 0.15 GHz part I want, but also the 1.15 GHz = 1.27 - 0.15 GHz part, which I have no interest in. After the mixing step, those two signals are added together and cannot be separated again. It is thus necessary to remove the 1.15 GHz part of the signal before it reaches the mixer.
To do this, a bandpass filter is needed. I did some simulation with Sonnet (give it a try if you want to do stuff like this, it has a quite functional free version), which basically allows you to simulate electromagnetic waves in planar (i.e. "mostly 2D") arrangements of metal, and ended up with this magic-looking design:
|Geometry for a microstrip bandpass filter at 1.42 GHz.
|Simulated filter response for the optimized design.
For a prototype, I then proceeded to cut this geometry into a copper board by hand ...
|Hand-made prototype (with a cutter knife).
... and measured it to compare with the simulation. As expected, it has more loss in the passband than in the simulation (reality always has more loss than expected) but not much, and the shape of the response is much more irregular, probably mostly because of manufacturing inaccuracies. Still, the important parameters agree really well with the simulation, which I find quite impressive.
AstronomyNow, finally all the important components are in place! I set up my 1.2m dish and pointed it at the sky, and tadaa -- the familiar HI signature of our galaxy showed up:
The signal is calibrated by a method called frequency switching: one periodically shifts the observation frequency by a few MHz, and thus observes just the background noise. This background noise is then subtracted from the actual measurement. This might sound like it reduces noise (it does not, but that's not trivial to understand), but the real gain is that the frequency dependence of the IF part of the acquisition board is removed from the signal. Why this is important may become evident from the following image:
None of the peaks shown there is an actual astronomical signal; the signal shown in the plot is far too weak to see it here. The non-flatness of the band is created by the bandpass filter (the sharp spikes are interference); this non-flatness should be calibrated away in order to reliably distinguish real signals from the bandpass shape.
Another interesting thing is to look at the waterfall plot of the record, i.e. encode signal intensity by color, and use the y-axis for time instead:
|Waterfall diagram of the acquired signal.
The next thing I will try to do is build another of those boards, and do a bit of interferometry. Stay tuned!