Showing posts with label receiver. Show all posts
Showing posts with label receiver. Show all posts

Wednesday, March 4, 2026

Modifying the MFJ-5008 parabolic ultrasonic receiver for better sensitivity and wider frequency response

Figure 1:
Front view MFJ-5008 parabolic dish with
integrated microphone and receiver
(located on the back side).
Click on the image for a larger version

The MFJ-5008 Parabolic Ultrasonic receiver

Note:

Since the MFJ-5008 is no longer being sold it can be found only on the "used" market.  A future posting in this blog will (hopefully) show how to construct a similar unit using readily-available kits and parts.

The MFJ-5008 was marketed primarily for detecting arcing on failing power line hardware, but there are other reasons why you might use such a device:

  • Listen to Bats' echolocation.  The "clicks" emitted by bats are well above human hearing.
  • Listen to other animals and insects.  Other animals and insects also emit ultrasonic sounds - both for echolocation and communication.
  • Find high pressure leaks.  Leaks in high-pressure systems (water, gas, engines, compressors) often make a lot of noise at these frequencies.
  • Locate switching power supplies.  These devices often make noise due to magnetostriction of devices (transformers, coils.)

As I find this topic to be interesting, I've written about the detection of ultrasonic signals on two previous occasions in this blog:

  • Improving my ultrasonic sniffer for finding power line arcing by using MEMs microphones - Link
  • An ultrasonic superheterodyne receive converter (e.g. "Bat Listener") - Link

* * * * *

While there are several devices out there that you can buy to enable listening at these frequencies, the landscape has changed in the past few years when it comes to how one might do this on a budget:

  • In years past, the MFJ-5008 was available - its primary purpose being to locate and identify arcing on power lines and related infrastructure.  As MFJ is no longer in business, this device is available only on the used market.
  • Some "bat listeners" have used electret microphones.  These inexpensive capsule microphones - while having good response across the human hearing range - lose sensitivity rapidly above this, limiting their usefulness above 20-30 kHz.  In doing A/B testing with a MEMS and an Electret ("capsule") microphone, the MEMS appears to be superior in every way when it comes to ultrasonic response.
  • Many ultrasonic detectors - including "bat listeners" - have used ceramic transducers.  Most often found for the 40 kHz range (and some were made at lower frequencies) these can be fairly sensitive.  Their frequency range is quite limited and they are only usable within a few kHz above and below their design frequency at best.  As different types of ultrasonic noise sources tend to occur at various frequencies, being able to detect such energy at various points across the spectrum can improve the usability of the device.
  • MEMs-based microphones have become cheap and available.  These devices - based on microscopic elements - can operate over a frequency range from a few 10s of Hz to over 100 kHz making the excellent replacements for the (increasingly hard-to-find) ceramic transducers.  Having a wide frequency range allows the user to tune to the peak frequency of the noise source rather than being limited to the immediate vicinity of 40 kHz.

The problem

Both the MFJ-5008 and the device described in the April 2006 QST article (noted below) utilize the then-available 40 kHz ceramic transducers.  These devices seemed to be a reasonable choice as they were relatively inexpensive and sensitive - but they are quite narrowbanded, unable to detect much once you get more than a kHz or two away from their 40 kHz center frequency.  (Note:  Similar devices were made for other frequencies - including 25 kHz - but these were not as readily availble.)

From my experience, the strongest peak of ultrasonic energy from arcing power line hardware is in the area of 25-32 kHz - wholly outside the frequency range of not only the passband of the tuning of the onboard potentiometer, but also well below the 40 kHz peak frequency at which the 40 kHz transducer has usable sensitivity.  What this means is that for these types of noises, the MFJ-5008 is quite "deaf":  Some of the arc's energy is around 40 kHz, but little enough that the sensitivity of the unit is really quite poor.  As noted above, an electric microphone would have been a better choice at this frequency range, despite its response typically dropping off rapidly above the audible range.

I can't help but think that the designers of the MFJ-5008 and the advice used to inform the design of the ultrasonic receiver described in the April 2006 QST article was misplaced:  For the reason most amateurs would have purchased or built a device - to find noisy power lines - the 40 kHz transducer is a very poor choice!

* * * * *

How the MFJ-5008 works

Made by (the now defunct) MFJ Enterprises, this includes a 18" (46cm) diameter vacuum-formed plastic parabolic dish with a 40 kHz ceramic transducer at its focus.  Mounted on the back of the dish is a direct-conversion receiver centered at about 40 kHz that converts energy around this frequency to the audible range.   As can be seen in Figure 1 there is a bar across the front in which the ceramic transducer is mounted - but it also has holes that - along with one located behind it in the plastic dish - form a crude sighting system that works quite well to determine from where detected noises might be emanating.

If one disassembles the electronics of the MFJ-5008 they will discover a small circuit board with rather common components - namely a 555 timer used as the oscillator, an LM386 audio amplifier to drive the headphones and a few common transistors to amplify and convert the ultrasonic signals to audible.  There is a "tuning" control on board consisting of a 10k trimmer potentiometer, but it is not accessible from the outside - and it has a range of about 38-48 kHz:  A slight modification will be necessary to allow us to take advantage of the wider frequency response of the MEMS microphone.

Consider the (annotated) schematic of the MFJ-5008, below: 

Figure 2:
Schematic of the MFJ-5008 ultrasonic receiver.  The circuitry is straightforward - a simple, run-of-the-mill direct-conversion design that is very similar to the one described in the April, 2006 QST article.  Changes to C2/C8 and the added inductor are noted on the diagram.
Click on the image for a larger version.
 

If we compare the above schematic with that from the April, 2006 QST article, A Home-made Ultrasonic Power Line Arc Detector - link) we see some very striking similarities:  Both use a 555 timer for the local oscillator, both use a series of bipolar transistors for signal amplification, and both use a single JFET for the frequency conversion mixer.  There are some differences, but these are pretty much superficial when you consider that the same goal is accomplished with the same types of components.

A cursory analysis of the above diagram shows that the first two amplifier stages are coupled with 1uF capacitors allowing the full audio frequency range to pass:  This mystified me at first, but in looking at the circuit board and noting some unpopulated parts locations I realized that there may have been plans to allow this circuit to be used at audio frequencies - and, perhaps, have a switch to select audible or ultrasonic ranges as well.

For the original 40 kHz ceramic transducer, this wide frequency range isn't a problem, but for a MEMS microphone - which can hear equally well over a 100Hz through at least 60 kHz, this would be:  As the mixer (Q3) is just single-ended, it will happily amplify the original input as well as do a frequency conversion meaning that you are likely to hear audio-frequency "bleedthrough" on the audio output - and indeed, when I retrofitted it with a MEMS microphone (to be described shortly) I did.

Figure 3:
Picture of the MFJ-5008 with location of the various
various components and board locations involved in the
modifications annotated.
Click on the image for a larger version.

Adding "proper" high-pass filtering to the MFJ-5008

The only sort of "high-pass" filtering present are capacitors C10 and C11 which are conspicuous by their being in series:  Why use two capacitors (1000pF and 220pF) rather than just a single 180pF capacitor?  The answer lies on the circuit board where there are unpopulated locations marked "L1" and "L2" (see Figure 2) which correspond with an (uninstalled) pair of inductors between the junction of C10 and C11 and ground.

To make the unit much less sensitive to audio frequencies - and to make it more compatible with a MEMs microphone, several changes should be made:

  • Change C2 and C8 to 0.01uF (e.g. 10nF) capacitors.  This will prevent the first two amplifier stages from being overloaded by audio frequencies and go a long ways in prevent "bleedthrough".
  • Install inductance at the positions of L1 and L2.  I suspect that two inductors were in mind when they designed the board as high-inductance, surface-mount devices are comparatively rare and expensive, so they could use a pair of lower-value coils in series to get the desired value.  See the footnotes on the bottom of this blog for suggested inductors.
  • Figure 4:
    Apparently designed to be used in several ways, the MFJ-
    5008's board has several unused parts locations, including
    positions for inductors that could be used for improved
    high-pass filtering as shown here.
    Click on the image for a larger version.
    Connect a 4.7k resistor between the center pin of the RCA connector (to the microphone) and the "V+" pad near the un-populated switch.  This inserts a current-limited 9 volt supply on the microphone lead.

The amount of inductance to install at L1 and L2 isn't too critical, but finding such components may be awkward - but the total amount of inductance to use may be anything between 27mH (that's milliHenries!) and 68mH with 47mH being optimal - a relatively huge amount for an SMD device.  In perusing my collection of inductors, I found a through-hole 27mH inductor that I tacked into place, securing it with glue:  Note that it gets soldered across the two pads of L1 and L2 closest to the socketed 555 IC as Figure 4 depicts.

When modifying the MFJ-5008, the MEMS microphone was fitted first and it became clear that audio-frequency energy sailed right through the system, significantly reducing its efficacy at the detection of ultrasonic energy.  It is my opinion that both the changing of C2 and C8 to 0.01uF capacitors and the addition of the inductor are necessary modifications for good performance. 

Note:

If you don't have a suitable inductor for the above modification, the receiver will still work, but you will hear a bit of audible frequency bleedthrough:  In a location with high ambient noise, this may be a problem, but in an otherwise quiet location, it probably won't be an issue:  Changing C2 and C8 do a reasonable job of reducing audio-frequency response and should be considered to be mandatory if you use a MEMS microphone.

In other words, if you don't install the inductor, don't let that stop you from making the modification to the MFJ-5008 and using it with a MEMS microphone - just be aware of the audio frequency "bleedthrough" issue.

Extending the tuning range of the MFJ-5008

Figure 5:
Potentiometer R10 - originally 10k - was replaced with a
50k miniature potentiometer to allow tuning.  A 4.7k resistor
paralleling R22 can be seen in this fuzzy photo.
Click on the image for a larger version.
As the tuning control isn't readily accessible - unless you drill a hole in the box and use a screwdriver - a modification is required to both make the tuning accessible and increase the range.  To do this, I found a small 50k potentiometer and soldered it into place where the original 10k trimmer (R20) was:  Note that two of the potentiometer's leads are connected together, so the "new" device would go between Pin 7 of the 555 in the schematic and resistor R22.  While doing this, R22 should be changed from its original value of 10k to 3.3k (or you could tack a 4.7 or 5.1k resistor in parallel with it).  Increasing the value of R20 from 10k to 50k allows the frequency to be tuned down to 20-22 kHz while lowering the value of R22 allows it to be tuned above 50kHz, all of this encompassing the frequency range where noisy, arcing connections (and bats!) are likely to be found.

Figure 6:
The modified MFJ-5008 with the (barely visible)
tuning knob sticking out on the left.  The blue
label indicates the approximate tuning frequency.
Click on the image for a larger version.
While I was able to cram the (very small) potentiometer onto the board (Figure 5), you may need to be creative - possibly mounting the potentiometer on the cover or side of the box using (very short!) flying leads:  If you use a metal potentiometer, I suggest connecting is body to the "ground" of the circuit (e.g. the outside shell of the microphone's phono plug) to prevent pick-up of nearby electric fields that might affect tuning.

The final result of the modification can be seen in Figure 6:  The cut-off shaft of the potentiometer protrudes slightly out of the left side of the enclosure and there is a label depicting the approximate frequency of the oscillator (and the center of the converted range) with respect to the adjustment of the potentiometer and its white paint mark.

What potentiometer to use?

To fit in the location of the original 10k trimmer, one needs to use a small potentiometer:  A suitably small potentiometer is the Bourns 3310C-001-503L which is available from DigiKey HERE and from Mouser Electronics HERE.  With a bit of care, it can be mounted to the board and the case modified to allow the shaft to protrude out the side - but it would be a good idea to use something (e.g. "hot melt" glue) to make it more rigid and prevent fatiguing/breaking the potentiometer's leads.  If you are creative, a larger potentiometer might be usable, attached with flying leads, but if it's metal, be sure to connect its body to the V- (battery negative - the shell of the phono plug will work) to minimize noise pick-up.

Note:

If you don't make the (highly recommended!) "tuning" modification, the MEMS microphone is still useful in that its sensitivity extends over a wide frequency range:  You may be able to adjust the original potentiometer (which can be adjusted between 35 and 48 kHz) to a frequency that is more suited for the types of noises that you are seeking.

Using a MEMS microphone


Note:  

In this section, I refer to a "homebrew" MEMS microphone carrier board - but there are "breakout" boards available that are already assembled:  This next section describes how either a "breakout" board or a homebrew board like this may be mounted in the focus of the dish.

Figure 7:
The original 40 kHz ceramic transducer and
carrier board (top) and the homebrew version with
the MEMS microphone (bottom) both mounted using
the pairs of screws on stand-offs in the front bracket.
Click on the image for a larger version.
Farther down this page you will find a description of a commercially-available MEMS break-out board (from SparkFun) and how it may be used, should you be unwilling to assemble your own!

                    * * * * *

For the specific MFJ-5008 depicted in this article I used an already-prepared MEMS microphone module:  This was described in a previous article linked HERE.  This circuit was designed to accept a wide range of voltages (3.5-10) to be imposed onto the same conductor as the audio, making it easy to interface on a single cable as we did here.

In the MFJ-5008, there is an aluminum "U" channel across the front in which the ceramic transducer is mounted and its location places it at the focus of the parabolic dish.  What this means is that when we replace this device with something else - a MEMS microphone in this case - it must not only be located at the same axial position (left, right, up, down) as the original, but the sensing element must also be at the same distance from the surface of the dish.

Behind the nesting cover (accessible via the removal of four screws - two at each end) there is a circuit board mounted on two stand-offs and the focus of this dish is precisely midway between the two.  Removing this and peering inside the original ceramic transducer, you can see the element located inside, recessed slightly from the front grille:  The distance of that element from the circuit board is that which should be replicated with the replacement microphone.

Figure 8:
Homebrew carrier board with MEMS
microphone installed, facing the surface of the
dish.  The microphone's "sound hole" - facing
from the camera in this photo - is located
precisely between the two mounting screws.
Click on the image for a larger version.
As can be seen in Figures 7 and 8, I mounted the homebrew MEMS modules on the "front" side of a piece of PCB prototype board, taking care of placing the center of the microphone (not visible in the photo) on the center line between the two screws and equidistant between them.  Once this was done, the "new" microphone was mounted back in the "U" channel and the wires soldered as seen in Figure 8.  As it turned out, the thickness of the homebrew board placed the MEMS element at the same distance from the dish as the original element - a fact later verified by noting that the "sharpness" and accuracy of the pointing with the new element seemed to be the same as before.

Using a Sparkfun MEMS microphone "breakout" board

Soldering a tiny microphone module successfully to a circuit board requires a bit of skill - but there are "breakout" boards that already have the microphone and some of the needed components already on them - and one of these is available from SparkFun (the "BOB-19389") for about US$9.00 at the time of writing. While it is possible to order from SparkFun directly, I ordered it via Amazon for the same price - plus shipping was "free".  Detailed information on this board may be found here:

https://www.sparkfun.com/sparkfun-analog-mems-microphone-breakout-sph8878lr5h-1.html

This breakout board contains both a microphone and an operational amplifier and here are their respective data sheets:

  • Microphone element data sheet - LINK
  • Op Amp data sheet - LINK

As originally designed, the SparkFun board "sort of" works for ultrasonic detection, but there are a few circuit elements that require attention before we use it.  Consider the schematic, below:

Figure 9:
Diagram of the SparkFun BOB-19389 MEMS breakout board.  As can be seen,
there's nothing special about this design:  A microphone coupled to a single op-amp section - but
but there's a problem with this circuit in our application:  The gain set by R4 is unnecessarily high
for our needs and this - along with C3 - reduce the useful frequency response to less than 30-sh kHz.
Click on the image for a larger version.

The implementation of this breakout board is nothing special - and it's worth noting that even without the gain of the op-amp, the MEMS microphone itself would have a suitable amount of drive for the MFJ-5008.

As part of our circuit analysis, I will call the reader's attention to R4 and C3 (300k and 27pF, respectively) which form a simple low-pass filter - but these components, along with the unity-gain bandwidth product of this op amp being 1 MHz - conspire to cause the frequency response to roll off rather dramatically above 15-20 kHz or so:  It will still detect lower-frequency ultrasonic signals, but sensitivity is reduced at higher frequencies while the signals that we don't want (e.g. audio-range frequencies) are not attenuated - and even if the frequency response was flat into the ultrasonic range, it would have way too much gain for our application, anyway:  Unless the SparkFun board were modified, the gain would be so high that the unit would be completely unusable!

Figure 10:
A close-up of the SparkFun BOB-19389 MEMS microphone
break-out board.  The location of C3 - now replaced by a
resistor.  Not also that the "sound hole" of the microphone
is on the bottom of the board, facing down in this photo.
Click on the image for a larger version.

The "fix" is to replace C3 with a resistor.  For the MFJ-5008 I would suggest using a 10k 4.7k resistor in this location and by lowering the gain, the op amp's bandwidth product isn't going to get in the way of the needed frequency response.  While it doesn't really matter if one removes the capacitor or not when using a 4.7k resistor (the -3dB point for a 4.7k resistor and 27pF capacitor is somewhere north of 700 kHz) it's pretty easy to remove just the capacitor and replace it with the resistor if you have SMD parts on hand.  If you have only through-hole parts, it should be possible to tack a 1/4 or 1/8 watt 4.7k resistor across them.  (Note:  I used the MEMS board in Figure 10 for a different project which is why there's a 47k resistor at the position of C3:  A 4.7k resistor is appropriate for the MFJ-5008.)

The other issue is that of the voltage range of the breakout board's components.  In testing, the board worked "OK" at just 1.8 volts - below the "official" specifications of the the Op Amp - but it worked "better" in the specified 2.3-3.6 volt range.  In the modification for the MFJ-5008 described above, the addition of the 4.7k resistor across the "audio in" phono plug put the full 9 volts battery voltage (minus resistive drop) on this line so we need to do two things to make this work:

  • Limit the voltage to the 2.3-3.6 volt range.
  • Combine split the audio signal from the voltage at the microphone breakout board.

Fortunately, this is quite easy, requiring just a small number of components and the following diagram shows:

Figure 11:
Powering the SparkFun MEMS break-out board from the audio cable with DC bias on it as depicted in the MFJ-5008 modifications, above.  Capacitor C1 blocks the DC from the Op Amp,
resistor R1 isolates the audio and DC lines while LED1 is used as a voltage shunt to limit the
voltage to somewhere between 2.3 and 3.6 volts:  An ordinary white or blue LED is perfect for this
as they are readily available and provide a voltage in the middle of this range.
Click on the image for a larger version.

Note:  I could have simply run a separate DC line from the circuit board to the detector, but this would have still required regulating the voltage down to the voltage needed for the MEMS device:  Putting DC on the signal line is easy to do and it requires only a few, inexpensive components.

Capacitor C1 has two functions:  Block the DC from the "Audio Out" terminal and to offer a bit of a high-pass frequency response to filter audio-range energy.  Resistor R1 extracts the voltage from the "DC + Audio" line and sends it to the "VCC" terminal on the breakout board and across this, the LED acts as a voltage limiter.  As noted in the diagram above, one can use a blue or white LED as the voltage limiter:  These will "turn on" at between 2.8 and 3.2 volts which is right in the range that we need.  Alternatively, if you have some "old fashioned" red LEDs that operate from about 1.7-1.8 volts, two of these in series will do the job.

Figure 12:
The SparkFun MEMS microphone break-out board with
the circuitry in Figure 11.  These components could be
"dead bug" mounted like shown in the photo or they
could be incorporated on the "carrier" board used to hold
it at the focus of the dish - either method works!  The
"sound hole" can be seen in the lower-right portion of the
board, just above the letter "H".  Note that it is not
centered on the board - something to note when mounting.
Click on the image for a larger version.

It is recommended that you use the "diode test" function of an volt-ohm meter to verify the turn-on voltage of your LEDs and to make sure that they are connected correctly.  If you have a variable-voltage bench power supply, connect it across the two leads and, starting out at less than 2 volts, slowly increase it while measuring the voltage across the "GND" and "VCC" connections:  The voltage should limit in the 2.3-3.6 volt range and you should see the LED(s) dimly illuminate.  In testing I haven't found that light falling on the LED causes any effects in the audio, but if you are, for some reason, worried about that, feel free to cover the LED with black paint, put it in some black heat-shrink tube or shield it from light in some other way.  (Note that in the MFJ-5008, the carrier board is contained within the "C" channel aluminum pieces and mostly shielded from light, anyway.)

These three components may be mounted either as shown in Figure 12 with the components' "flying leads" holding things together, or on a piece of prototype board to function as the "carrier" board of the same type shown in Figures 7 and 8.  Note that the "sound hole" on the breakout board is on the "back" (non-component) side of the circuit board (visible in Figure 12) and that it is NOT in the center of the board and take this into account when you are mounting it to the "carrier" board.

Mounting the SparkFun MEMS microphone in the MFJ-5008 ultrasonic dish

 As with the homebrew MEMS module shown in Figure 7, a small "carrier" board - made from a small piece of prototype "perf" board - was used to mount the SparkFun MEMS module at the focus of the dish.  As the "sound hole" of the microphone is on the bottom side of the circuit board, the MEMS boar's "component" side is against the carrier board as shown in Figure 13.

Figure 13:
The SparkFun MEMS microphone board attached to the
"carrier" board using UV-cured epoxy.  The notch near the
GND connection was filed prior to mounting.
Click on the image for a larger version.

The same external components - the LED, resistor and 0.01uF capacitor - were used, but mounted to the carrier board - but there is one minor complication:  The SparkFun MEMS board itself.

For whatever reason, the designers of the SparkFun MEMS board chose to locate the "sound hole" for the microphone NOT on either the "X" or "Y" axis, but at some random location in the lower-right of the the PC board as can be seen.  For the focusing and sighting/aiming of the MFJ-5008 dish to be correct with its existing view holes, the "sound hole" of the microphone must be located at exactly the center point between the two mounting posts and in line with them - but if that is attempted, the SparkFun MEMS board gets in the way.

One option is to simply move one of the mounting posts slightly, but I chose another option:  Make slight modifications.  As can be seen in Figure 13, a small "notch" was filed in the SparkFun MEMS board using a needle file in line with the "GND" connection:  As there are no components near this connection, it was fairly safe to do so - but be absolutely sure to cover the "sound hole" with a piece of tape when filing to prevent dust from getting inside the microphone and, possibly, ruining it.  As seen in Figure 13, a bit of the red coating on the SparkFun MEMS board was scraped off to provide a new GND connection to it.

Figure 14:
The left-hand mounting post, the top filed away
to provide clearance to the SparkFun MEMS
circuit board.  An alternative solution would
be to have moved the post 1/4" (5mm) farther
apart.
Click on the image for a lager version.
A bit of additional work is required to make the board fit and the mounting post itself was attacked with a file as seen in Figure 14 to gain additional room to provide clearance to the SparkFun MEMS board.

The SparkFun MEMS board itself was mounted to the homebrew carrier board using some UV-cured epoxy - although some "hot melt" glue would probably have worked just as well.  As seen in Figure 13, the board is rotated slightly counter-clockwise so that the "sound hole" would alight with the center line between the two screw mounts and also so that the divot that was filed out at the GND connection would line up with where the mounting post would be.

When all was finished, the result looked very similar to that in Figure 8:  The MEMS board was facing toward the surface of the dish and the original cable carrying audio - and now power - was attached to the connections shown in Figure 11.

Final words on the MFJ-5008 modifications

The above modifications should allow the MFJ-5008 to work over a wider variety of frequencies to allow optimum detection of energy from electric arcs, high-pressure gas leaks, bats, insects, switch-mode power supplies and many other things.

Prior to modification of the first unit, a "test range" was set up in my back yard:  A 40 kHz transducer was driven with a sweep/function generator (an old Wavetek Model 180) and the output level at its lowest-possible setting.  From about 33 feet (10 meters) away the "warble" from the swept output was easily audible - but not particularly strong.

After the modification, the subjective impression was that the sensitivity was equal or better than the original 40 kHz ceramic transducer - but a quick walk around the house revealed the ringing presence of several switch-mode power supplies, each producing low-level noises of their own due to magnetostriction of components within - something that was totally inaudible prior to the modification, made possible only by the broad-range response of the MEMS microphone and the added ability to tune the center frequency.

* * * * *


In-field comparisons

Recently, a modified MFJ-5008 was put to test in the field alongside an unmodified MFJ-5008 to locate a noisy power pole.  As it turns out, the pole itself had been identified - from a distance on HF and a small magnetic loop and then up-close using a small Moxon on 2 meters - both using AM on a Yaesu FTX-1.  The role of the ultrasonic dish in this case was to try to identify the specific hardware on the pole.

With the modified MFJ-5008, with the center receive frequency tuned in the vicinity of 28 kHz or so, one could easily hear the arcing from several 10s of feet (5-10 meters) away and as such, it was possible, using the "iron sights" (holes) in the MFJ-5008 zero in specific hardware on the pole.  When the modified MFJ-5008 was tuned above 35 kHz or so - outside the majority of the noise energy of the arcing - it became nearly inaudible as this shifted the noise peak in the headphones from a few kHz to over 10 kHz - likely too high for the ears of anyone but a young person:  This further reinforces the need for a tunable, broad-band detection system.

With the unmodified MFJ-5008, one had to stand as close as possible to the noise source - right under the pole - to even hear the ultrasonic noise from the arcing hardware and this vantage point did not lend itself to trying to identify the failing hardware since it was all straight above one's head.  Unlike the MEMS microphone - which had a broad response - the narrow response of the 40 kHz transducer meant that there really wasn't any hope of even hearing the peak around 28 kHz, even if one were able to retune the receiver.

Shortly after this comparison, the owner of the unmodified MFJ-5008 came over to my house with the parts necessary for modification (e.g. SparkFun MEMs board, 50k potentiometer, inductor for the high-pass filter modification, etc.) and it, too, was retrofitted.  In comparing the two modified MFJ-5008 units side-by-side, they performed identically with the modification opening up an entire new "world" in ultrasonic sounds from power supplies, insects, birds, vehicles, etc. that had previously been all but inaudible with the original fix-tuned 40kHz-only MFJ-5008!

 * * * * *

Footnote:

  • Here are a few suggested parts for the inductor in the modification of the MFJ-5008 - all 47mH:
    • https://www.mouser.com/ProductDetail/Murata-Power-Solutions/17476C?qs=5CKLVr1iF0nvNdEM16T%2F2A%3D%3D
    • https://www.mouser.com/ProductDetail/EPCOS-TDK/B82144A2476J?qs=v4Mlc8l4PHmthTExsnwGmg%3D%3D
    • https://www.digikey.com/en/products/detail/bourns-inc/RLB1014-473KL/2561378
    • https://www.digikey.com/en/products/detail/murata-power-solutions-inc/22R476C/1924732
    • https://www.digikey.com/en/products/detail/central-technologies/CTS4HTF-473J/16048522

 * * * * *

This page stolen from ka7oei.blogspot.com


[END]


Friday, June 13, 2025

A 15 (and 10) meter high-pass filter for Field Day

QRM from a transmitter to receivers on lower bands

[Go to the end of this article for an "After-Field-Day follow-up" about this filter] 

A friend of mine belongs to a club in a town north of me and he was describing an issue that they've been having for the past several years during ARRL Field Day:  A station on an upper band (e.g. 15 or 10 meters) degrading reception on 20 or even 40 meters when transmitting.  What was needed was something that could be used on both 15 and 10 meters and protect the lower bands (e.g. 20, 40 and 80) meters - and this protection would go the other way, preventing the 15/10 meter station's receiver from being overloaded by transmissions on the lower bands.

Figure 1:
Exterior of the 15 Meter high-pass filter,
built into a die-cast aluminum box with single-hole
UHF connectors on the sides.
Click on the image for a larger version.

First, a bit of background.

The ARRL Field Day event is held on the fourth (but not last) full weekend every June.  During this event thousands of clubs and individuals go forth into the wilds to set up and operate an event where they attempt to contact as many stations as they can in a 24 (or 27) hour period.  In the case of club stations - or where multiple individuals are involved - it's very common to have more than one transmitter at a given site.

As the Field Day rules stipulate that all antennas/radios be within a 1000 foot (305 meter) circle it isn't possible to provide much geographical separation between different transmitters.  This separation is important because a transmitter produces a very strong signal and the received signals are very weak by comparison:  Receivers can be easily overloaded by these nearby strong signal sources and transmitters can produce low-level signals on frequencies other than those on which they are operation - ones that are too weak to cause problems under normal situations but when placed in close proximity to a receiver these weak emissions can block out/interfere with other receivers - even on different frequency bands.

Other-band signals can cause problems

The degree to which a transmitter radiates these low-level spurious signals - and that to which a receiver is able to tolerate a very strong signal - depends considerably on the transmitter/receiver itself.  Some high end makes of radios (e.g. Elecraft, Flex Footnote 1) can be very clean in terms of transmitted spectra and higher-end receivers of all makes may be capable of tolerating a very strong signals - perhaps even in the same band - and this strategy works as long as the potentially-interfering transmitter itself is clean:  If that "other" transmitter is producing noise at any frequency of reception, there's nothing that can be done at that receiver to fix the problem other than to quiet or clean up the errant transmitter.  Meanwhile, even a "good" radio - such as an Icom IC-706MK2G or IC-7300 Footnote 2 or a Yaesu FT-757 - which works well by itself - may not "play nice with others" when immersed in an environment with multiple transmitters and receivers in very close quarters for reasons largely related to their design architecture.

What this means is that if one uses directional antennas (e.g. Yagis or beams) they are often placed north-south of each other and pointed parallel  Footnote 3 so that there is some isolation off to the sides of these antennas - and it goes without saying that antennas of any sort are separated as far as the rules - or the operating space (e.g. park, yard, forest clearing) allows.

Sometimes, this isn't enough:  Interference can result despite the precautions (e.g. transmit/receiver separation) so additional filtering may be necessary.

The use of Band-Pass filters

Barring the ability to separate antennas or place them in each others' nulls, there are other options:  From a number of manufacturers Footnote 4 there are available band-pass filters that - as the name implies - are designed to pass one specific amateur HF band with low attenuation (loss) while offering significant rejection of other bands above and below.  By placing one of these filters inline with the radio and the antenna, it not only will reduce the probability that a very strong signal from another band might overload the receiver (a particular problem with a radio like the Icom IC-7300 and certain models of other radios from other manufacturers) but it also attenuates the broad-band noise Footnote 5 that almost all HF radios produce that can encompass frequencies other than the band on which they are operating.

This low-level interference - often in the form of a white noise (or hiss) is produced by the amplifier stages in the transmitter itself.  Most modern HF transceivers - while equipped with low-pass filters that attenuate harmonics at multiples of the transmitted signal and generally prevent this noise from being emitted on the next-higher non-WARC band - do NOT have an equivalent high-pass filter in them that prevents low-level spurious signals or broadband noise from being output to the antenna on frequencies below that on which it is operating.  What this means is that a transmitter operating on, say, 15 meters, can produce a "hiss" that may degrade reception on 20, 40 or even 80 meters whenever it is keyed up - in this example, depending on how well that 15 meter antenna can radiate such signals and how close the two antennas are to each other.  (I discussed this very problem in an early article of this blog:  Getting the rigs ready for Field Day - Link).

For this reason it is often preferable to use a band-pass filter on every transmitter that is used, for the specific band on which it will be operated:  This will not only protect that receiver from the other bands' signals but also prevent the low-level energy from being emitted on bands other than that on which it is being used.

Note:  If you are experiencing interference from another transmitter that is actually producing noise on your receive frequency (this, given the presumption that your receiver isn't being overloaded) the addition of a filter on the station receiving interference will do no good at all since it cannot possibly filter out interference that is already there, on-frequency.  In this case, the only appropriate remedy would be to filter the offending transmitter.

As an aside:  In some cases simply enabling the radio's built-in antenna tuner - or using an external tuner - may significantly reduce the amount of out-of-band energy that the transmitter emits as well as adding to the attenuation from "other-band" signals during receive. Footnote 6 

A high-pass filter

While band-specific filters are preferred, my friend presented a case where a high-pass filter (one that blocks signals below a certain frequency) may be appropriate.  In his Field Day environment there has always been a station operating on 20 meters and usually another operating on 40 meters as well - but a third station was available to operate on 15 or 10 meters - depending on propagation conditions.  The problem was that when this third station transmitted, 20 and 40 meters were often degraded - likely by the broadband noise mentioned earlier.

While it would be possible to obtain separate 15 and 10 meter band-pass filters at some expense, I decided on a different approach:  A 15 meter high-pass filter.  This filter - which could be made to strongly attenuate frequencies on the non-WARC amateur bands below 15 meters (e.g. 20, 40 and 80 meters) - it would have the advantage of also being usable on both 15 and 10 meters.  Since it was unlikely that they would have stations on both 15 and 10 meters this strategy seemed sound for their application.

Using the ELSIE program (from Tonne software - link), I first calculated an "N=5" pro-forma high-pass filter using the "Elliptical" (e.g. "Cauer") circuit topology observing that I could get low attenuation at 15 meters and above while achieving more than 40dB on 20 meters and below.  Using the ability of the ELSIE program to do Monte-Carlo type optimizations, I then tweaked the filter topology from a pure Elliptical filter to a hybrid one and this resulted in even better attenuation at 40 meters than the original:  The schematic diagram of this filter is shown below:

Figure 2:
The 15 meter high-pass filter as iterated by the ELSIE program.  The circuit topology - originally
Cauer (Ecliptic) was modified by using simple inductors in sections 1 and 5 and a capacitor at
position 6 and then re-iterated to optimize performance.
Click on the image for a larger version.

As can be seen from the diagram, there are three capacitors in series with the signal path with two inductors directly to ground:  The center inductor is in series with another capacitor, forming one of the "notches" typical of the Elliptical filter topology - and it so-happens that it's possible to tweak the filter so that this notch just happens to land in the middle of the 20 meter band to maximize attenuation there.

Rummaging around in my junk box I found several 500 volt silver-mica capacitors:  For some reason I have a lot of 160pF units, so that was placed at section #2 and three of them were put in parallel for the series capacitor in section #3.  I found a 200pF capacitor for section #6 and I paralleled a 120pF silver mica and an NP0 disc ceramic for that in section #4.

Many people doing homebrew construction seem to intensely dislike toroidal inductors - but while they would be more compact, there is no need to use them here, so large-ish air-core inductors were used.  As the inductors are all "about" the same value (in the 225-300nH range) I wound 7 turns of 17 AWG (but anything 14-18 AWG would do) wire on a 13/32" drill bit for each of them, the precise value being unimportant as their turns would be stretched/compressed while using a VNA to "dial in" the filter response.  As mentioned earlier, I'd added one more inductor from the initial design because the inductors were the cheapest of all of the components (they are just wire!) and they are very adjustable - simply by compressing/spreading the turns which meant that by picking capacitor values that were just "pretty close" to those called out by ELSIE, the coils could be used to tweak the filter's response.  Note in Figure 3 that the inductors that are close-ish to each other are placed at right-angles, or in parallel with each other:  Avoid placing two adjacent coils "end to end" with each other to minimize coupling between them.

Figure 3:
Inside the 15 meter high-pass filter.  Copper-clad PC board
material is used as the backplane (ground) with small pieces
used as "islands" for connection and support points.  The
added capacitor and inductor are those on the right-hand side.
Click on the image for a larger version.

The filter was built on a piece of copper-clad PC board material as a back-plane and ground and small pieces of that circuit board material were cut out to form "islands" - the so-called "Manhattan" construction:  These islands would allow the junctions of the various circuit components to be connected together and with these islands glued to the back-plane and mechanically support the components soldered to them.

The piece of circuit board used as the backplane was sized to fit in the bottom of a die-case aluminum box that I had handy (about 6" x 3.25" x 2" or approx. 15 x 8.3 x 5 cm - but it could have been a bit smaller) onto which I'd installed two chassis-mount UHF connectors:  These connectors were placed rather close to the bottom of the box so that their ground lugs could be soldered to the backplane, providing both the "ground" connection to the copper clad and for mechanical support

Using a VNA, I first adjusted the inductor in section 3 to provide a notch at about 14.24 MHz and then iteratively tweaked the inductors in sections 1 and 5 to provide the lowest insertion loss and lowest VSWR at 15 and 10 meters.  When I was done, the insertion loss was just fine - less than 0.5dB - but the VSWR was about 1.45:1 at 15 and 10 meter so I added two more components (the 30pF capacitor in section 7 and the inductor in section 8 of the diagram) to act as a bit of a "tuner" to improve the match:  In the figure above you can see an inductor (in section 8) that goes to the right-hand UHF connector (6 turns of the same wire as the other coils on a 13/32" drill bit) and a 30pF disk-ceramic capacitor (section 7) between the PC board "island" to which it connects and ground:  With a bit more adjustment of all four inductors this brought the VSWR at 15 and (most of) 10 meters down to about 1.25:1 or better - plenty good enough!  The response of this filter is shown below:

Figure 4:
Insertion loss and VSWR plot of the 15 MHz high-pass filter as plotted by a VNA.
As can be seen, attenuation at 40, 30 and 20 meters is well over 45dB with less than 0.5dB
at 15, 12 and 10 meters:  The VSWR is also acceptably low on these band as well.
Click on the image for a larger version.

Not shown in Figure 3, I later used RTV (silicone) adhesive to stabilize the coils and add support - after tuning, of course:  This reduces the probability of the coils being detuned by the filter being jarred or dropped.  RTV is fairly low loss (at least at HF) and far superior to "hot melt glue" in this case (it's lighter - and it won't melt!) and unlike hot glue or cyanoacrylate (e.g. "Super") glue, it can withstand mechanical shock without breaking loose - even when cold.

This filter should easily handle 100 watts - and the low loss is largely due to the use of silver-mica capacitors:  After all, 500 volt silver mica capacitors - such as those used here - may be found in wide-range antenna tuners made by LDG and the like where they would be exposed to more stress than in the filter.  If you are wondering about the use of the small, disc-ceramic capacitors, they are used in "low stress" parts of the circuit - to "trim" the capacitance to the needed value (e.g. a NP0 ceramic in parallel with a 120pF silver mica to get about 130pF) or used to "tune" the filter as in the case of the 30pF capacitor on the output.  While it might seem risky to use these tiny ceramic capacitors at 100 watts, a quick look at almost any Japanese-made amateur HF transceiver - particularly those made up until fairly recently - you'll find them sprinkled with these capacitors in the low-pass filters and even for matching in the final amplifiers - both at HF and VHF/UHF - for matching:  If it works for them, I'll not worry about using them here in the right places.

Don't forget to change the filter when you change bands!

One hazard with outboard filters of any type:  Be sure that the filter is removed if you attempt to transmit on a frequency for which it is not designed!  After nearly every Field Day I hear/read reports where someone - say, originally on 20 meters - then tries to QSY to another band with the 20 meter filter still inline using the radio's built in antenna tuner or an outboard tuner:  The result is is often that the filter is damaged!  Footnote 7 

Final comments

As can be seen from the response plot of Figure 4 this filter will attenuate signals on the bands 20 meters and below by more than 45dB and this should be enough to quash to inaudibility any low-level noise produced by the transceiver at these lower frequencies that might degrade reception on these bands.  Similarly, energy from transmissions on 20 meters and lower from other stations will be at a much lower level prior to reaching the front end of the radio using this filter, further reducing the probability that they could overload/cause noise.

One thing that has not been discussed thus far is the fact that harmonically-related frequencies (e.g. a transmitter on 7.05 MHz would have harmonics at 14.10 and 21.15 MHz) are likely to be audible on other receivers, despite heroic attempts to fully-filter them.  The reason for this is that these harmonically-related signals will be fairly strong compared to the noise floor of the amateur bands and, unlike the low-level noise discussed earlier, would have their energy concentrated into a small bandwidth.  

Such signals are also likely to be radiated not only from the antenna ports, but from other cables connected to the radios themselves - namely the power cables, audio/microphone connections, data and PTT lines which means that a filter on the output won't suppress those other leakage sources.  Other than wide-spaced separation (e.g. not placing radios in the same location and moving them as far apart as possible) there's no way to completely prevent harmonically-related QRM other than to coordinate efforts and simply avoid operations that could result in harmonically-related interference.

As it is not yet Field Day, I don't know if this filter will "fix" the problem that my friend was reporting, but  should help, and it was quick, cheap and easy to throw together. Footnote 8

* * * * *

After-Field-Day follow-up

A few days after 2025 Field Day I spoke again with the friend for whom I constructed the filter described above. While 10 meters was mostly dead, 15 meters was reasonably productive at times and operation on that band was successfully carried out - mostly using digital modes.

There was time to do a bit of A/B testing and it was noted that without the high-pass filter, the 15 meter transmitter did produce a very audible "hiss" on lower bands - most notably 40 meters - when it keyed up, but this was totally absent with the filter in placeAdditionally, a slight amount of QRM from the lower-band stations (on 80, 40 and 20 meters) transmitting was noted in 15 meter reception without the filter - likely due to strong signals impinging on the radio's internal switching diodes - but this was also absent with the high-pass filter installed.

It's also worth noting that several other tactics were employed to minimize the possibility of QRM (interference) between stations, including:

  • Separating antennas as much as practical.  Depending on the site - not to mention the "1000 foot" rule - you can do only so much, but it helps to carefully consider the layout of the site to maximize distance between antennas.
  • Using band pass filters.  These go a long way toward preventing interference to and from radios on other bands. 
  • Use of mains filters.  L/C filtering (e.g. filters using inductors/capacitors) were installed on the long extension cords that fed the individual stations to prevent RF from being conducted on the electrical power leads and for this, "Isobar" plug strips were used.  Alternatively, putting a half-dozen or so turns of the extension cord through an FT-240 or "Monster" toroid (using 31 or 43 material for either one) would work as well - but it's recommended that the toroid be protected from damage by putting it in a box (e.g. a plastic electrical box with notches to allow the cord to pass).
  • The use of "balanced" 4:1 baluns.  As it turns out, most "4:1" baluns are NOT very balanced - and as such they will put significant RF current onto the radio and feedline, not only reducing antenna efficacy but also potentially improving susceptibility of RFI (Radio Frequency Interference) both TO the radio and devices connected to it (e.g. computers, other radios) but also allow RF interference (from generators, switching supplies, lighting, computers, etc.) to find their way onto the feedline.  This "un-balanced" nature of most baluns can be proven by noting signal strength of a consistent off-air signal and disconnecting only one side of the feed to the balun:  With a truly balanced balun you should see signals reduce by 20dB (3-5 S-units, depending on the radio) or more, but with most baluns you will see only a small (6-10 dB - 1-2 S-units) at most.  Of all of the commercial offerings, one of the very few truly "balanced" baluns is the Balun Designs "Hybrid" balun (Model 4116 - link) - but most 4:1 baluns can be made to be balanced by immediately preceding it with a common-mode current choke (e,g, 8-14 turns of coax on an FT240-31 or FT-240-43 toroid).

 * * * * *

This page stolen from ka7oei.com

[END]

Footnotes:

  1. Unlike most radios, Flex radios do include filtering to prevent low-level noise from being emitted on bands lower than the one on which it's being operated:  Specific models of other manufacturers may also include this - although most do not.
  2. Direct-sampling receivers such as that of the IC-7300 have "different" problems in the presence of very strong signals compared to more conventional superheterodyne receivers:  Any signal that hits the analog-to-digital converter can cause overload, no matter the frequency.  While a conventional receiver can have a very "strong" mixer and some "roofing" filters in its IF (Intermediate Frequency) stages, this is not possible on a direct-sampling receiver.  Instead, it must rely on a rather large number of individual, overlapping band-pass filters to cover its intended frequency range and the ultimate attenuation of these filters may not be "strong" enough to prevent a nearby transmitter on another band from adding to the already-strong melee of signals on the crowded bands during Field Day and causing overload - or, at least, significant de-sensing (e.g. reduction in sensitivity).  This property is also what almost certainly makes them very poor candidates for being able to tolerate another local transmitter on the same band (e.g. a 20 phone and a 20 CW/digital station at the same Field Day site).  There are strategies that can improve the probability of two stations co-habitating on the same band - mostly having to do with picking the "right" radios (e.g. Elecraft K3S or the K4HD are known to work in this environment as are a few others) - as can the use of parallel-pointed Yagi antennas (see the next section, below) - or very "sharp" band-pass and notch filters can be constructed as described in two articles on this blog, namely:  A 100 watt "Helical" resonator bandpass/notch filters to increase isolation of 20 meter stations during Field Day (link) and Revisiting the 20 meter "helical resonator" band-pass/notch filters (link).
  3. Being able to point beams parallel to each other is at least partly a matter of geography.  A station on the east coast is likely pointing their antennas west while the situation would be reversed on the west cost:  A station in the middle of the country - with signals coming from potentially all directions - would be less-likely to be able to use this tactic, at least not without a degree of coordination among the individual transmitters/stations.
  4. A number of different manufacturers make band-specific filters for HF.  Depending on the design, these can offer modest (>=30dB) adjacent-band suppression - which is usually enough to solve most interference problems - or much higher degrees of filtering, even more than 50dB.  In addition, individual-band "Notch" filters are available from some suppliers that reject a specific band of frequencies which can be used several ways - on a transmitter to suppress any low-level noise that it might be generated on a specific band, or on another station to reduce the levels from a transmitter on that other band to prevent overload - and it can also be used to further-improve performance of a band-pass filter and increase attenuation on that specific band.  One of the companies that supplies such filters is Morgan Manufacturing (link)Full disclosure - I know the person that runs this company and am quite familiar with the products.  Other manufacturers also make similar, excellent products as well.
  5. This "hiss" can usually be detected without any sort of special equipment.  To do this, one would set up two transceivers in a relatively RF-quiet location (perhaps NOT a suburban home), each on its own antenna spaced within a few hundred feet/meters of each other.  On the radio doing the transmitting turn down the microphone gain all of the way after verifying that the RF power output would otherwise yield 100 watts peak when talking.  On the receiver, tune in the next band lower than the transmitter and note the noise floor with and without the transmitter keyed up.  In many cases, a "hiss" that can mask weak signals can be observed - particularly if using a resonant antenna on the transmitter without an antenna tuner.  If you couple carefully into the transmitter (using attenuators or directional couplers) this noise floor can be measured directly with a spectrum analyzer - even the $50-ish "TinySA" is up to the task!  It also goes without saying that a transmitter that IS outputting full power will also be prone to producing such hiss as well - not only above and below its actual transmit frequency, but on the "lower" bands as well.
  6. Testing to determine the efficacy of the built-in tuner as a band-pass filters was done using a Kenwood TS-450SAT, a radio from the 1990s.  When the tuner was switched in and "tuned" - even if the load was already matched - it functioned as a low-Q band-pass filter that reduced the broadband noise and adjacent band signals by at least 8dB - and typically 20dB or so.  Whether or not this strategy is likely to work on specific radios (e.g. some may switch out the tuner if there is already a good match) would require testing as described above.  (These measurements were discussed in a very early entry of this blog - link).
  7. The most likely components to be damaged when trying to "force feed" RF on the "wrong" band are the capacitors, followed by toroidal inductors being somewhat less-likely - and this will often happen when transmitting is attempted at full power (100 or more watts) rather than at the low power level used for tuning.  Usually, the operator realizes the mistake after the tuner fails to find a match, or it does find a match but signals are weak or absent.  For this reason, if you are using a filter with a radio and an external tuner it's strongly recommended that you place the filter between the radio and the tuner:  This will prevent damage to the filter as the radio will protect itself if it's used on the wrong band, presumably alerting the operator to the problem!
  8. It took far longer to put together this article than it did to design, gather parts, assemble and tune the filter!

 

Sunday, October 30, 2022

An ultrasonic superheterodyne receive converter (e.g. "Bat Listener")

In the mid 90s I decided to throw together what I called a "Bat Listener" - a simple receiver used to convert ultrasonic sound down to the audible range.

Figure 1:
The exterior of the ultrasonic receiver, complete with fancy
labeling!
Click on the image for a larger version.

Two types of circuits:

There are two common ways to convert a higher (ultrasonic) signal to the audible range, whether this is done using analog or DSP (Digital Signal Processing) techniques.

Frequency division

There are several ways to do this, the simplest being the "divider" type which digitally converts ultrasonic frequencies to audible by integer division of the input to a lower frequency.

The problem with this simple approach is that it does not preserve the amplitude (loudness) of the original sound since it must take the input signal, amplify/convert it to a series of logic-level pulses - which loses any amplitude reference - and do a brute-force digital division.  Additionally, if there are multiple signals present, for the most part only the strongest one will be converted down.

Clearly, one cannot "tune" this type of circuit:  A signal at 40 kHz will always be divided down by a fixed integer amount,  Let's say that the circuit digitally divides by 32:  That 40 kHz signal will be at 1.25 kHz.

Additionally, the direct "A-B" frequency differences between ultrasonic signals is lost, instead being "(A-B)/N" where "N" is the number of divisions.  In other words, the relative frequency differences between signals is not preserved.

Heterodyne conversion

The other way to do this is to convert the frequency.  In this technique, two signals - the ultrasonic to be converted - and another generated by the device (the "local" oscillator) are mixed together.  The result is an arithmetic shift in frequency.

The biggest advantages of this method are the fact that that not only are the differences in frequency preserved (e.g. two tones 1 kHz apart at ultrasonic will appear as two tones 1 kHz apart at audio) but the relative amplitudes (loudnesses) of the received signals are preserved as well.

Frequency conversion:

I chose to build a heterodyning receiver to convert the input frequency to a lower one.  This can preserve the amplitude and frequency relationships  - plus it is fully tunable, allowing one to choose the frequency range to convert to audible sounds - and since it is a simple conversion, multiple signals present will also be preserved.

When it comes to frequency conversion, there are two ways:  The simplest - direct conversion - would involve mixing a variable oscillator with the incoming signal and filtering/amplifying the resulting audio.  This has the advantage of being the easiest, and it is the method described in this article:

     April, 2006 QST article, A Home-made Ultrasonic Power Line Arc Detector - link)

A similar device - the MFJ-5008 - was made by MFJ Enterprises, but as this company is no longer doing business, it is found only on the used market.

While I could have easily built something like this a decade before the above article was published, as I'm sometimes wont to do I decided to make it a bit more complicated, constructing a superheterodyne converter.

While a direct-conversion receive simply mixes an oscillator with the desired signal to cause a frequency conversion, a superheterodyne receiver operates like a conventional AM or FM radio:  The desired signal is first converted to an IF (Intermediate Frequency) - and this IF is then converted to audio.  The advantage of the superheterodyne scheme is that filtering may be applied at the IF to limit the receive bandwidth - and since the IF is fixed, its width remains constant over the tuning range, just like that in a conventional radio/receiver.

Circuit description

Figure 2:
Schematic diagram of the superheterodyne ultrasonic receiver.
See text for a circuit description.
Click on image for a larger version.
 

As noted above, this circuit is more complicated than it needs to be, so make of it what you will!

VCO:

The heart of the unit is U1, the VCO (Voltage Controlled Oscillator) which uses the venerable CD4046 PLL chip.  Often used for frequency synthesis, we are using (only) the oscillator portion, which provides a linearly-tuned and fairly stable frequency source, adjusted by the voltage applied via R101 (and scaling resistor R102).  The values were chosen to provide an approximate frequency range of 125 to 185 kHz (more on this later) to allow tuning of audio signals from (ostensibly) 0 to about 60 kHz.  The actual tuning range is closer to 115-190 kHz as a bit of extra margin for the frequency range.

The only critical component here is C101 which should be a frequency-stable capacitor.  I used a polystyrene capacitor, but an NP0 (a.k.a. C0G) or silver-mica could be used, instead.  When I reverse-engineered this device, I noted that the marked capacitance value was unreadable, but back-of-the-envelope calculates indicate that a value of "about 150pf" should be in the ballpark.

R103, connected to the "R1" pin of U1, sets the approximate center frequency range while R104, connected to the "R2" pin - sets the lowest frequency - which important, since we want to constrain the tuning to 125-185 kHz.  Additionally, the low end of the tuning range was further refined by R102 on the "ground" side of the tuning potentiometer, which sets the minimum voltage that may be applied to the "VCOIN" pin.

The VCO output, a square wave, is buffered by U2, a hex inverter, and several sections are used to provide both a VCO signal and its inverted version to drive the mixer.

While the 4000 series CMOS chips throughout this receiver will happily run from 3-15 volts, they are operated from a regulated 5 volt supply - mainly to improve frequency stability and to provide a nice, stable voltage for a few other low-level circuits and to provide isolation from the main battery supply which will vary a bit, particularly at higher receive volumes:  This variance, if it gets back into some earlier stages, could cause instability of the receiver in the form of "motorboating" (low-frequency oscillation or pulsing) or some other type of feedback.

BFO:

Another circuit is the BFO (Beat Frequency Oscillator) which is used to convert the IF signal back down to audio - both being processes that we'll discuss shortly.  This uses an inexpensive 500 kHz ceramic resonator to form an oscillator using one of the sections of U2C, the signal being buffered by U2B.  This signal is divided-by-two using U3A, one half of a 4013 dual flip-flop - and then divided by two again using U3B, yielding a stable 125 kHz signal.  As with the VCO, two phases of this signal (normal and inverse) are available, this time using the "Q" and "!Q" outputs of the 4013.

Input signal path:

J1, a disconnect-type 3.5mm stereo jack is wired so that an internally-mounted electret "capsule" microphone is connected by default.  This microphone element (M301) is of the "2 wire" type or electret microphone in which a bias voltage is applied to the same pin from which audio is drawn - this voltage being applied via R301 from the 5 volt regulated supply.  The specific make/model of this electret element is unknown as it was selected from a small collection to find the best performer at ultrasonic.

Using a MEMs microphone

Since this article was first written, I replaced the electret with a more modern MEMs microphone as described in Another article:  Improving my ultrasonic sniffer for finding power line arcing by using MEMs microphones - link.  These types of microphones are capable of detecting from low audio frequency (a few 10s of Hz) and much higher - often through 100 kHz.

The schematic in Figure 2 now shows how an ANALOG MEMs microphone (see the link above for more information) may be integrated into the unit.  As these microphones are not tolerant of 5 volts, I used some "normal" (not "high brightness") GaAsFET Green LEDs:  D401 limits the voltage to 1.8-2.0 volts on the microphone's power supply while D402 prevents "glitches" from damaging the MEMS device that might get through via coupling capacitor C402 when an external microphone is connected/disconnected.  When everything was done, there was a very dramatic improvement in sensitivity above 15 kHz and several switching supplies near the workbench - which I couldn't hear before - suddenly became very audible.

Microphone amp and filtering

The signal from the microphone is applied to U4A which is wired as a unity-gain buffer.  For this, an LM833 is used, an inexpensive, low-noise dual op amp:  An LM358 or many other types may be used here as well - just make sure that it is is fairly low noise:  I'd avoid the use of the LM1458 here as it is quite noisy by comparison!

Section U4B amplifies the signal voltage by 10 (20 dB of power gain) and this signal is applied via R305 to a simple L/C high-pass filter consisting of C303, C304, L301 and L302 the latter two components being inexpensive 18 milliHenry inductors.  Certainly, an R/C-based high-pass filter could have been constructed using U4B, but I chose not to do that for some reason.

Figure 3:
Inside the ultrasonic receiver, constructed on
prototype board and having been modified
several times over the years, prior to the
installation of the MEMS microphone.
Click on the image for a larger version.

In simulation, the C303/C304/L301/L302 filter has a -3dB roll-off of about  23 kHz, it's down by 10dB at about 19.5 kHz, by 20dB at about 16 kHz and by 40 dB at 9 kHz and with the values shown, it's flat to within 1 dB between about 24 and 100 kHz.

The output of the filter is amplified by U5B - and then even more by U5A (which has a bit of roll-off from C307) to yield a whole lot of gain.  It's very possible that I over-did the gain here, but unless the signal source is quite close, there is no clipping observed on the output of U5A.

Its worth noting that a mid-supply voltage is created using R309/R310 to provide a "virtual ground" for the op amps and to maintain stability, it is heavily filtered by C306 and C302, each located near the respective op amp shown on the diagram.

Mixer and band-pass filter:

It is this next section that may seem unfamiliar to some - the use of a CMOS analog switch as a signal mixer.  For this, a CD4066 is used which consists of four separate analog switches.  The filtered and amplified ultrasonic input signal from U5A via C308 is applied to pins 2 and 10 of U6A/U6D.  When the respective signals on the control pins "VCO_A" and "VCO_B" go high, the switches are activated, and because VCO_A and VCO_B are inverts of each other, each of these switches is closed in turn.  The result of this is that the inputted signal is chopped up at the rate of the 125-185 kHz VCO and this produces two mixing products.  

For example, let's assume that there is a 40 kHz signal is present on the input that we wish to hear.  If the VCO is tuned 40 kHz above the 125 kHz IF (again, more on that momentarily) - to a frequency of 165 kHz - the switching action of U6A and U6D produces both the sum (165 + 40 = 205 kHz) and the difference (165 - 40 = 125 kHz).

T301 is a filter/transformer that passes only the 125 kHz signal - the difference signal in this case.  This transformer consists of two separate windings, each resonated using its internal capacitors and the externally-added 820 pF capacitors on each winding (e.g. C309/C310) to "pad" it down to 125 kHz.  This forms a fairly wide (8-10 kHz) filter that rejects signals outside the immediate vicinity of its 125 kHz frequency.  Because this filtering is at a fixed frequency, it does not vary with input tuning which means that its bandwidth is constant over frequency.

Of all of the components in this device, this transformer is unique:  It was originally a 262.5 kHz IF transformer from a 1970s/1980s Philco (Ford) AM-only car radio.  While I could have certainly used the original 262.5 kHz frequency - or even 250 kHz, when I built this I decided to pad it down to 125 kHz using C309/C310  - a frequency that is conveniently 1/4th of the 500 kHz resonator.

It's been so long since I built this, I don't recall why I didn't simply divide the 500 kHz by two and readjust that transformer to 250 kHz.  Practically speaking, I could have also up-converted to 455 kHz and used either transformers or ceramic filters from a modern AM radio as 455 kHz ceramic resonators were certainly available at the time - but I didn't do that.

Each half of T301 has a center tap and to this, a bias voltage is applied via R315 to assure that the voltage on these switches was in the middle of the supply range, away from the protection diodes on the 4066's I/O pins, which could cause clipping/distortion should they be allowed to conduct if the signal voltage got too near the ground or supply rails.  To prevent coupling between the two halves of the transformer via the center tap, R314/C311 was added, the resistor adding isolation with the capacitor bypassing the remainder of the signal.  Practically speaking, being able to adjust the bias voltage was unnecessary as a simple resistive voltage divider to set the bias at 2.5 volts (1/2 the supply voltage) would have been just fine.

On the "other" side of the transformer is the other half of U6 (e.g. U6B/U6C) - this time, clocked from the fixed 125 kHz oscillator.  From this, the signal - previously converted up to 125 kHz is now converted back down to audio.

Post-mixer amp/LPF:

The output of the down-converting mixer is applied to U7B via R316, a 1k resistor and a 0.001uF capacitor, both of which form a simple R/C low-pass filter to attenuate any high-frequency leakage signals from the mixer.  Because the mixing process itself is a bit lossy (about 25% efficient) as is transformer/filter T301, U7B boosts the signal by a factor of 10 (20dB) and then applies it to U7A, which is configured as a variable gain amplifier section.  The output of this is then boosted again by U8, an LM386 which is capable of driving headphones or even a small speaker.

A few comments about the design:

Originally, the circuit lacked U7 at all, but it was added when the gain of U8 (the audio amplifier), by itself, was found to be inadequate.  Since U7 was "patched" into place, this explains the odd gain distribution:  If I were rebuilding this from scratch, I'd certainly not need two post-mixer amplifier sections and I could have likely eliminated one full dual op-amp package.  As it is, I may add a "high/low" gain switch somewhere around U5 to allow reduction of the gain somewhat when in the presence of possibly-high ultrasonic signal levels to prevent clipping prior to the band-pass filter which would surely degrade overall performance.

If I were to build this again I would likely use a 455 kHz IF, instead as noted.  While not as plentiful, 455 kHz ceramic resonators are available to use for the BFO as are either transformer or ceramic-based band-pass filters.  I would also likely reconfigure U4B or U5 to perform the high-pass filter function rather than using harder-to-find inductors.

Again, I built this unit in the mid 1990s and have since lost my original notes, but I do recall that I modified it a few times since, simply tacking changes onto the old circuit rather than completely revising it.

Use as a longwave receiver:

While primarily intended to "hear" ultrasonic sounds such as those produced by bats, insects, leaking pipes, arcing power lines, etc., it is just a longwave radio receiver connected to a microphone:  If one connects a few 10s of feet/meters of wire to to J1 - and provides an Earth/ground reference to its shield connection - one can easily tune in the high-power transmitters used for submarine communications (around 20-30 kHz) plus the WWVB time signal at 60 kHz.  This must, of course, be done away from man-made noise sources such as power lines.

Alternatively, I have used a loop of about 1 foot (25cm) diameter of a dozen or so turns of wire along with a 10uF capacitor in series (to optionally block DC from R301) and been able to hear such signals - even in suburbia - but with this arrangement you'll also likely hear plenty of similar signals from the myriad switching supplies that likely inhabit your house as well!

Final comments:

The reader should be under no illusion that this is an optimized circuit or that I would do it this way again:  It was assembled fairly quickly to suit a need and to test a few random ideas, just to see if they would work.  Will I rebuild it at some point?  I don't know - it works as it should, so I don't plan to re-make something that is currently fit for purpose.

While I've heard very few bats with this - probably due to the deficiencies of the electret microphone at ultrasonic frequencies (which explains the future switch to MEMS-type microphones) - I've used it to find powerline noise (arcs are noisy at ultrasonic) and to test longwave receive antennas.

This page stolen from ka7oei.blogspot.com

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