Author Archive for Andy Murkin


Inductor Pickups 3 – a new instrument

In the first post in this series, I described experiments with different types of inductor pickups.  At the end of this, I had 3 types of pickup which I thought would be of further use: a mono guitar pickup; a stereo pickup using the insides of two Telephone Pickup Coils; and a stereo pickup using two 100mH inductors:

Later I added a similar pickup with two 200mH inductors I had been able to get hold of.

As can be heard in that earlier post, I had used my Macbook as the sound source, but I thought a better one would be a hard drive.

Some time ago I had bought a job lot of broken hard drives from eBay at a cost of about 50p each.  My original idea for these was to scavenge parts from them – magnets, discs, etc. – and to use the arms for a purpose which I may yet get the chance to write about.  (See this post for a description of how to dismantle them).

In this instance, however, I was looking for drives which would power up – many of them didn’t – and might emit interesting noises.  Not noises you could hear directly, but noises that could be picked up by an inductor.

I went through most of the drives – of which the above picture shows just one boxful – and tested them with the 3 inductors I had used in my last test on my laptop: the standard guitar pickup; the two Telephone Pickup Coils; and the two 100mH inductors.  The playing technique was simply a matter of powering up the drive and then passing the pickup slowly over the surface.

Surprisingly, the three different systems didn’t pick up exactly the same noises, and the range of sounds detected seemed sufficiently varied to make an instrument based on this approach worthwhile.  This recording is of the drive pictured, using the three pickups described above, in the order in which they are mentioned, the guitar pickup, Telephone Pickup Coils, and 100mH inductors:


It would have been perfectly possible just to use the hard drives spread out on the table, but I decided it would be much neater to make a proper instrument in a box, with its own power supply and preamp.  I was lucky enough to come across a supply of cheap wooden boxes which would be ideal for this project and, hopefully, some future ones.

There would be plenty of room inside for the power supply, circuitry and hard drives.

The first thing I added was an inexpensive handle to the outside (secured with nuts, bolts and washers, rather than screws, to ensure it won’t come off):

The second thing was to install a power source for the hard drives and circuitry.  The obvious thing to do was to use a typical hard drive power source, which would typically provide 12v and 5v DC – the 3.5″ drives would need 12v to spin the discs; 5v would be a suitable voltage for 2.5″ drives and for the electronic circuitry.  These were the main items I bought:

The 240v adaptor fitted into the corner of the box like this:

and an on/off switch was included inside the box lid:

Unfortunately, the mains power supply was too noisy to be used for the electronic circuits, so I used the mains adaptor for powering the hard drives and a PP3 battery for the electronics.

I used a box-type battery holder as it had an integral on/off switch and I was able to glue it securely to the inside of the instrument’s wooden case.


So far, so good; but for some reason the hard drives weren’t performing the way they were doing when I had been experimenting with them, and sometimes didn’t even appear to be powering up.  Only when the power supply exploded one day with a loud crack and the lights went off did the penny finally drop! . . .

. . . I went away and researched the amount of power required by a hard drive – and it’s much, much more than I imagined.  The ratings are normally found on the drive itself: the ones I had been using, when I finally read the labels, were rated up to 720mA @ 5v and up to 900mA @ 12v, although often somewhat less, but averaging out at about an amp/amp and a half.  Here’s a couple of examples of where you can find this information on a typical drive:

As you can see, there are big differences between the ratings for these two drives – and even the 5v consumption can be surprisingly high.   And that’s in normal use: when powering up, they can easily consume over 2A each during the first 2 or 3 seconds – no wonder my poor power supply couldn’t cope!  You can see from the label that it’s only rated at 2A at a time for each voltage, so is really only suitable for a single drive.

So, my first step was to buy two new power supplies, 5v and 12v, each rated at 8A – a bit of an expense I wasn’t expecting (about £8 – £9 per device)!   These were no longer to be incorporated into the box; they would be external, connected via two typical centre-positive power sockets on the rear of the box.

I was aiming to incorporate 5-6 hard drives in the box, so the two supplies would provide sufficient power for normal operaration – especially given that there would be no data input or output from the drives – but I had to arrange the power switches so that no more than 2 or 3 of the drives would come on at once.

I could have done this with just a row of power switches, of course; but a more interesting method was to purchase 4 of these ‘delay relays’ at only just over £1 each:

The large blue component is the relay.  It’s not completely in focus in this image, but you can just about see that it can handle 10A (10A of mains voltage, in fact), so was well up to the job in hand.

Item 1 in the above picture is a multi-turn preset, which allowed me to set the delay so the relay wouldn’t come on for at least 2 or 3 seconds – the length of time a high-amperage spike might be caused by a hard drive powering up.  This particular device could be adjusted for a delay of up to 10 seconds, so I set the 4 devices to work as follows:

When the on switch is operated, 12v and 5v power is immediately passed to the first one or two hard drives; after 3 or 4 seconds, power is passed to one or two more drives; and after 6-8 seconds to the one or two final drives.  In this way, overlapping spikes are avoided, and sufficient power is available for the drives to work properly after they are fully powered up.

A series of 6 LEDs, 3 for 12v, 3 for 5v, showed when the power connections were made.

Item 2 in the above picture is where the power lines are connected.  The centre connection is the ‘in’ or ‘common’ connection; either side of this are ‘normally open’ (normally disconnected) or ‘normally closed’ (normally connected) connections, which are then reversed by the operation of the relay.  I needed the normally open connections, so in two devices the centre connection was a 12v line, and in two the centre connection was a 5v line; the normally open connections were connected to the LEDs and the hard drives’ power connectors.

The 4-pin Molex connectors taking power to the drives are wired like this:

Pin 1 (yellow) = 12v;  Pin 2 (black) = Ground; Pin 3 (black) = Ground; Pin 4 (red) = 5v

So, the power section now looked like this (note: the on/off switch wasn’t quite in place when I took this picture):

For a short clip of the startup procedure, click here.

In this test I only used one hard drive connected to each of the three sections – the third one being unusually noisy!  They are, of course, all broken in some way, but you can hear that the first two, as they start up one by one, are not at all as rattly; but these are not the sounds the instrument is designed to create: as we will hear later, each drive creates its own interesting sounds when probed by the instrument’s inductors.


So much for the power connections.  Next, the electronics.

The first part of the circuitry was a preamp for the inductors.  For this I used the same transistor-based preamp I had used before for electrets and inductors, with the inductor connected on the left where the microphone is shown:

As this was a stereo instrument, with two inductors fitted side-by-side, as shown above, I used two of these preamps.


I tested this, and it worked fine with the mains adaptors for the hard drives and the battery for the preamp, so I turned next to adding a tone control.  I thought, as the instrument was based on inductors, that an inductor-based tone control would be the ideal thing, similar to the design that I had made before in the Bits & Pieces series, the ‘Active’ Tone Control  – which, in reality, is a passive tone control with a x10 amplifier in front of it to counteract the drastic loss of signal strength.

This is my design for the two-channel version:

In this case I used a TL072 instead of the 741 in the original: it was more up-to-date, less noisy, and neater, having two op-amps in one single 8-pin package; I also altered the resistors between the inputs and outputs (pins 2 and 1, and pins 6 and 7) from 100k to 1M to further increase the amplification.

The only thing I found is that I sourced the parts for the original about 20 or 30 years ago, when it was evidently much easier to obtain a 1H inductor – this is a very large value, rarely seen nowadays, and I couldn’t find one.

However, inductors are like resistors, you can put them in series to obtain larger values, so I bought 10 @ 200mH, which only cost about £1, enabling me to create two inductors of 1H.  I spaced them out on the circuit board, hoping to minimise interaction between them, and connected the outputs of the preamps directly, without a switch.  This tone control varies the sound quite a bit over its full range, and I was fairly sure there would be one position which would be very similar to the unaltered sound of the inductors picking up the sounds of the hard drive in action.

(In the event, I had a problem with the circuit around the TL072, so the amplifier and the tone control parts of the circuit ended up on separate boards, as can be seen in later pictures).

The 8-way phono socket panel in the bottom left is where the pairs of inductors plug in, and allows 4 separate pairs to be connected at the same time.  Multiple Molex power connectors like the ones illustrated above allow a number of different drives to be running at the same time, giving the possibility of more complex, multi-layered sounds.


The following picture shows the electronics in a more or less finished state in the lid of the box:

On the left, from top to bottom are Panel 1: 9v Battery Power on indicator light, 3.5mm stereo audio out socket, Tone and Volume controls;

Panel 2: 4 x Stereo Inductor inputs;

Panel 3: An LED on the left for 12v power on, and an LED on the right for 5v power on – the third hard drive or pair of drives.

On the right, from top to bottom, are the circuit boards for the tone control, the transistor-based buffer/pre-amp, and the op-amp-based pre-amp; and at the bottom, the 9v battery box with integral on/off switch.


To finish the instrument off, I just needed to arrange for the hard drives to be secured in the main part of the box.

I began by putting in a layer of foam rubber, mainly with a view to deadining the sound of the spinning drives.  Some time ago I had purchased a roll of foam, advertised as a yoga mat or sleeping mat.  It only cost about £4.50 and was quite big – about 2 metres by half a metre (perhaps rather narrow for sleeping!), and seemed the ideal thing for this purpose.  I lined the box with the foam, sticking it down with hot glue.  (Again, this is before I replaced the power supply).

To keep the hard drives in place, I used blocks of polystyrene, and cut more squares of foam to insulate drives which would have to sit on top of others.

Cutting polystyrene is messy and rarely successful, so I used an electric polystyrene cutting kit with a heated blade, like the one shown below.  This cost under £10 and proved considerably easier and neater in this and other projects – and elsewhere in the house.

Taking care not to set light to anything, or breathe in fumes from burning polystyrene, I trimmed the pieces without causing any mess.

Turning to the case, I decided a companion box was needed to transport the leads, power supplies and spare hard drives; so I fixed another of the carrying handles to the new box, and stuck on two small engraved plates to indicate which one was the instrument and which one carried the parts.

Using the two boxes, the instrument and accessories could easily be transported together.

A small length of yellow plastic from a cable tie was fixed to the right-hand side of the lid of the instrument box to ensure that it stayed open at the best angle.

I recorded the instrument using Audacity on one of my old MacBooks.  The pair of 100mH inductors were very noisy.  I didn’t have time to find out why, so I unplugged them; but I found that the most productive technique was to use two sets of inductors – the 200mH and the telephone coils – one in each hand.  This enabled me to search for interesting sounds in two places at once, to balance these sounds, and also on occasion to create interactions between them.

It was also possible to lay one set carefully on a drive, to continue picking up sounds, and leaving one hand free to operate the tone crontrol.

The following sound file illustrates some of the typical electrical/mechanical/drone sounds I was able to get from the drives:


Electret microphones and a parabolic reflector

One final – well, maybe not final, we’ll see how it goes! – type of microphone I wanted to try while out field recording was a parabolic dish or reflector.  I planned to use electret microphones in the way described in the series of articles beginning here.

Strictly speaking, the three-dimensional shape of the parabolic reflector is called a paraboloid, and the adjective is paraboloidal. A parabola is the two-dimensional shape and the distinction between this and a parabaloid is like that between a sphere and a circle, according to the Wikipedia.  However, in informal language, the word parabola and its associated adjective parabolic are usually used in place of paraboloid and paraboloidal.

So, this is the shape of the dish.  Note that there is a point marked ‘focus’.

Diagram by Melikamp – Own work, CC BY-SA 3.0,

So, now we know exactly what we’re talking about!


As with some of my other recent experiments, it’s not so much the microphone itself as the way it’s mounted that’s significant; and the significance of the particular shape of the parabolic dish is that all the sound captured within it is reflected back and focused on a single point a few centimetres from the centre of the dish. The effect of this is to naturally amplify the sound captured – and amplify it by quite a lot.

This diagram illustrated how the sounds coming into the dish are all focused on the same spot – the spot where the microphone is placed, facing back into the dish.

Own work assumed (based on copyright claims)., Public Domain,

In addition to this, the captured sound is from a restricted area, directly in front of the dish, so what it allows you to do is pick out an individual sound source – a person, bird or animal, machine or natural feature – some distance away and record it without having to get too close, which may cause disturbance, or resort to extreme amplification, which may cause noise or instability.

This is basically the audio equivalent of using a telescope – and, indeed, astronomical telescopes – not just optical, but also radio – use parabolic reflectors to focus light or electromagnetic waves, as do satellite TV dishes.

This photograph from the Wikipedia, showing the receiver from the MERLIN array at the Mullard Radio Astronomy Observatory, Cambridgeshire, is essentially a giant version of the parabolic reflector microphone, and illustrates the reflector’s features: the shape of the dish and the focus point – usually in the centre (although typically on the edge of a TV satellite dish).

Photograph y Cmglee – Own work, CC BY-SA 3.0,

The idea of using a parabolic reflector to gather sound from a distance has been going for a long time – since classical antiquity, in fact, as the Wikipedia points out, when the mathematician Diocles described them in his book On Burning Mirrors, and it has been claimed (although probably wrongly) that Archimedes used parabolic reflectors to set the Roman fleet alight during the Siege of Syracuse in 213–212 BCE.

In the UK, as far back as the First World War, giant concrete ‘sound mirrors’ were erected on the south and east coasts. Before the invention of radar, using these structures to listen for the sound of their engines was the most effective way of detecting the approach of enemy aircraft.

The caption to the above photograph – also from the Wikipedia – says: ‘On the pipe in front of the acoustic mirror was a trumpet-shaped ‘collector head’, a microphone which could pick up the reflected engine sound of Zeppelins approaching from the sea. Wires passed down the pipe to a listener seated in a trench nearby with a stethoscope headset, who would try to determine the distance and bearing of any enemy airships.’

[Photograph by Paul Glazzard, CC BY-SA 2.0, – ‘WW1 Acoustic Mirror, Kilnsea, East Riding of Yorkshire, England. Rare 4.5 metre high concrete structure near Kilnsea Grange, northwest of Godwin Battery, a relic of the First World War.’]

This photograph from the same source shows 3 ‘Listening Ears’ together, near Greatstone-on-Sea, Kent.

[RAF Denge photograph by Paul Russon, CC BY-SA 2.0,]

A great collection of photographs of a whole range of these sound mirrors from Selsey to Sunderland by Joe Pettet-Smith is featured on this page from the BBC website.

Concrete Blocks


Normally, commercial parabolic microphones are extremely expensive, although excellent ones are available from companies such as Telinga and Wildtronics.

As usual, I tried to do things on a budget, but finding a suitable parabolic dish proved difficult – bearing in mind that the parabolic shape itself is the important thing, as explained above, and a plastic bowl of some other type wouldn’t work as well.

Other factors included size and weight. The reason for the large size of the coastal ‘sound mirrors’ was not just the aim of collecting sound over a large distance; the size of the dish also determines how easy it is to detect low-frequency sounds. In the case of the sound mirrors, the low frequencies of aircraft and airship engines were a priority. This also has to be borne in mind with the portable reflector, which will inevitably be more suited to higher frequency sounds.  This partly explains its popularity amongst those who go out to record birdsong.

As far as weight is concerned, you have to take into account that the dish might have to be carried for quite a while in the field. Wildtronics, in particular, make a point of stating the weights of their dishes, to the extent of naming their thinnest variety Feather Light, and emphasising that it can be folded or even rolled for transportation. There’s heaps of information online about satellite TV dishes – and you’d think a second hand one of these would be a good bet, cost-wise – but nothing about how much they weigh.  However, they look heavy to me, and their particular design style, with the focus point well outside the rim of the dish, makes it seems as if they’d be difficult to wind-proof.

At the other end of the scale, I almost went for this hand-held item below.  However, although it’s much bigger than it looks – some 25cm (10″) diameter – and despite more positive than negative reviews on Amazon, it really did seem a little too expensive (around £25) and a little too small to me, and would almost certainly not be that effective – I’m looking out for a cheaper second-hand one on eBay to give it a try, though!So, in the end, I went with a UK firm I found, who make a decent reflector at a very reasonable price – OK, more expensive than most of my other projects, but reasonable indeed in the world of commercial parabolics. This was Innercore; or Parabolic Microphone, who make a 50cm ABS plastic reflector for about £65 with an integral stem for microphone mounting, a rubberized hand grip and a standard tripod mounting thread. I also bought their spandex wind shield for an extra £10, as I know from experience that wind can be a real destroyer of decent recordings in the field. As the dish is white, a black cover would, in any case, be a good thing from the point of view of concealing the dish – avoiding disturbing wildlife, and so on.


When the dish arrived, it was exactly as described, and, as well as the windshield, even included a microphone, User Guide and cable ties to attach the microphone to the central stem.

The central stem had the focal point clearly marked.  As the picture shows, a handle was attached on the back.  The dish, made of ABS plastic, as I said, was surprisingly light and could comfortably be carried for some time; but in the base of the handle is a hole with a standard 1/4″ thread in it, which would fit a photographic tripod.  I have a couple of handheld devices with 1/4″ threads on the top, capable of folding into a small tripod, which could prove useful.

I also acquired a light, but full-sized tripod, which could be used in the same way.


The main task, however, was to attach the microphone inside the dish – or, in this case, microphones, as I wanted something of a stereo effect.

This was unlikely to be pronounced, as the only sound entering the microphones would be that captured by the dish.  I have seen 3-microphone systems where two of them are forward-facing, recording ambient sound in Left/Right stereo and one is facing into the dish, recording the sound on which the disc is focused; but I decided to go with my standard 2-microphone set-up and not worry too much about creating mixers for extra microphones or how different the left and right recordings were.

So, to this end, I used a standard twin-phono socket lead, cut the plugs off one end, drilled a suitable diameter hole in the dish, threaded it through and soldered two small electret capsules to the end.

I wanted some small ones, and the only ones I had left after the two binaural projects, – the dummy head and the dummy ears – were a type called WM-61A.  The Panasonic WM-61A was a very popular and often-used quality electret, now no longer manufactured and consequently becoming more expensive; these were not advertised as ‘Panasonic’, and were not expensive, so their quality was not guaranteed . . .

To fix them to the central stem I used an old Allen key and a jubilee clip – the Allen key only because it had a right-angle shape with some straight sides, and would therefore fix fairly solidly in place. You can also see in this picture the blue band which marks the focal point of the parabolic dish.

I attached the lead and capsules to the stem, close to the focal point marker, with cable ties.

The small felt pad on the end of the stem was to protect the wind shield, which was quite thin, and, I thought, could be damaged by the pointed stem.

The final thing to be done was a little wind-proofing.  Firstly, I took a spare microphone windshield, cut a small hole in the end, and pulled it over the Allen key mount, covering the two capsules.

Finally, I pulled the spandex cover over the front of the dish, covering the whiteness of the plastic as well as helping to keep wind out of the dish.

The parabolic reflector was now ready for testing.

I went to a local nature reserve and made recordings in different areas: woodlands, a river path and a small lake with wildfowl.  The following extracts are typical of the results.  The first recording illustrates the difference in what is picked up when the dish is turned in a different direction.

By and large, though, I had to turn the recording levels up too high, and there was too much noise.  The first and third recordings are just as they came out; the second and fourth are the same recordings with noise reduction applied.

The noise reduction makes them just about acceptable to use, but this is contrary to the purpose of the reflector disc, which is supposed to amplify the sound naturally, without the need for noise-making electronics.

So I’m going to have to do more research and find out the cause of this: are the electret capsules at fault?  Are they badly placed within the dish?  Are the sounds I’m trying to capture too faint?  I’ll report back on any improvements I manage to make.


Binaural Recording, Pt 3

After my second experiment in binaural recording, described in Part 2 of this series, the third area of my research focused on the importance of the ears, rather than the head, to the quality of the recorded sound.  The pdf article I had orginally read on the subject had stressed the importance of the shape of ears, rather than the presence of the head, and at least one range of well-known commercial binaural microphones, the 3dio, consists of ears without a head.

So, I set about sourcing a pair of realistic ears.  Some of these can be prohibitively expensive, but I found a type that were more reasonably-priced (a little under £6 for the pair, from China), moulded in silicone, and intended for acupuncture, it said, or medical study.

When they arrived I was surprised, as they were heavier, softer and more flexible than I was expecting – nothing like the hard, fixed ears of the dummy head.  Much more realistic, and softer even than real human ears.

Unfortunately, as you can see here, the right one – on the left in the picture – had been rather crushed in the post, with the ear lobe bent right over:

The insides of the plastic bags seemed quite moist, so I was hoping that if I pulled the ear lobe back and stored the ear upside down, with a bit of weight on it, it would still be malleable enough to return to something like its original shape.  Perhaps when taken out of the bags, the ears would then harden up in the right shape.

In the  end I had to use a dab of hot glue to pin the ear back to its proper place.  The ears became less moist, but not really less floppy after being out of their bags for a few days.


The next part of the plan was a pair of ear muffs, readily available from Chinese sources on eBay at about £1.  They come in all sorts of colours and finishes, but the ones I picked on had a fur interior and a faux-leather finish on the outside:

The reason I chose this type was that the fur would provide a certain amount of sound deadening, and the faux-leather would take hot glue.

More hot glue?  The glue, of course, was needed to attach the silicone ears to the ear muffs!

The purpose of this was to be able to use the ears in different contexts, either attached to a dummy head, or used on their own, relying on the ear-shape for the binaural effect, rather than the whole head.

Accordingly, I first obtained a cheap polystyrene dummy head and inserted some long nails to act as a support for the ear muffs:

It took a few tries to get the exact locations for the nails, but I fairly soon had them in the right place to hold the ears at the correct angle:

I then cut a small block of polystyrene to a suitable size to fit the ears on as a free-standing unit.  I estimated that 14-15cm would be a suitable separation distance for the two ears and trimmed the polystyrene accordingly:

The final and most important stage was to add the electret capsules inside the ears and attach a twin phono lead for the output to the preamplifier.

I used smaller capsules than the ones I had used in the dummy head, as it was difficult to make holes in the silicon ears.  In the end I used my polystyrene cutter, which has a small heated blade, inserted the cable between the silicon block and the ear muff backing, attached the capsules and then pushed them down into the ear canal towards where the ear drum would be.

These photographs illustrate this process:

You can also see a certain amount of melted silicone debris, which I had to remove.

After tidying this up and testing the sound was OK, I superglued the cables along the back of the ear muffs to keep the microphones from moving, and the project was finished.  As soon as there is some better weather, I’ll be able to go out and test the three different binaural systems together.


Ultrasonic Field Recording

I had been looking for ways of recording sound from scenes which was not immediately obvious.  I had developed an inductor-based recording device which could pick up electrical noise, and the next area I wanted to explore was ultrasonic sound – that is, sounds which are too high for our ears to hear.

A recording of sounds which are too high for our ears to hear, of course, would not be too interesting – you wouldn’t be able to hear anything on the recording, either; but I got the idea of what to do with them from a type of device available commercially in finished or even kit form – a Bat Detector.

Bats make noises which are mostly ultrasonic – too high for our ears to hear – when they hunt and communicate.  What the classic bat detector does is pick up these ultrasonic sounds, amplify them and lower their pitch by an octave or two, so we can hear them.

But not only bats make noises in the ultrasonic region, many noises around us contain ultrasonic elements which we can’t hear as well as elements which we can hear.  Lowering the pitch of the ultrasonic elements would enable us to appreciate the fullness of sounds which at the moment we only hear part of.

There are several ways of changing the pitch of ultrasonic sounds, and there are many circuits available on the internet utilising these methods for producing a bat detector.  The three main ways are:

1. Frequency Division, in which the very high-pitched bat sounds are converted into square waves which can be digitally divided by – typically – 10, to produce a much lower sound, within the range of human hearing. For example, a bat making calls at 50kHz, when divided by 10, will sound at 5kHz – quite high, but well within our normal hearing range of 20Kz to 20kHz.

2. Time Expansion, where sounds are recorded digitally in the bat detector at a high sampling rate, then played back at a slower rate.  This is a common method of pitch-changing in the world of digital sampling (and the method I used in the computer-based Black Widow sample manipulator).

3. Heterodyne, which works on the same principle as the electronic musical instrument, the Theremin: the high-pitched sounds are mixed with equally high-pitched sounds produced by an oscillator inside the bat detector. The aim is to tune the bat detector to produce a slightly different pitch to the bat, as the output is designed to be at a frequency which is the difference between the two pitches. In this case, if the bat is making calls at 50kHz, for example, and the bat detector is tuned to 45Hz, sounds will again be heard at 5kHz.

This latter turned out to be the method I used – in fact, I bought a bat detector kit.  The best commercially available bat detectors are very expensive, providing a wide frequency range to work within, and often providing more than one way to hear the bats.  I should make it clear at this point that I wasn’t particularly focusing on bats – I wanted to listen to any ultrasonic sounds that might be in the vicinity; but I wanted as wide a variety of sounds as possible to be detected and brought into hearing range.

So I found a neat-looking and very reasonably-priced kit, the Franzis Bat Detector, which was available from a number of sources, all at around £20-£25.  It comes in an attractive and quite sturdy cardboard box, which can serve as the container for the circuit when made up, with art work for the two potentiometers required – for frequency and volume – and holes behind which the speaker is attached.

Inside the box is a very small PCB, no more than 3″ long, already populated with about 25 tiny SMD (Surface Mount Device) components:

Together with the board is a plastic bag containing a handful of non-SMD components, which you solder in place yourself.  These include a battery clip (not shown), a 5v voltage regulator, a few capacitors, an LM386 i.c. amplifier, the two potentiometers and an ultrasonic receiver – in appearance rather like a large electret microphone capsule.

Together with the components, there was a nicely-produced and informative booklet, containing general information about bat detection, an explanation of how the circuit works, a circuit diagram and comprehensive instructions for assembly and testing.


Beginning at the front end of the circuit, there are many different types of ultrasonic receiver available, and the majority are rather expensive.  The one that came with the kit is the most popular of the more reasonably-priced ones, but is designed to work best at 40kHz, with quite a narrow band of frequencies in which it works at its greatest efficiency.  This is because it is designed to work in precisely that way, usually being paired with an identical-looking 40kHz ultrasonic transmitter, and commonly used together in detection or distance measuring applications.  I was concerned that this frequency restriction might limit the sounds the detector was able to pick up, bats or otherwise, but there is a certain amount of pickup outside the intended operating range.


This particular circuit would probably not suit it, but a possible alternative to the ultrasonic receiver in some circumstances would – surprisingly – be an electret capsule.  Although these are sold for use in ordinary microphones, some have good ultrasonic capabilities.  It’s hard to know which ones, and to what extent they might be useful in this regard, as figures are not normally released for frequencies above 20kHz, the normal extent of human hearing.

However, one capsule known to have a good response even in the environs of 100kHz is the Panasonic WM-61A, one which has been tested and used in this way.  Unfortunately, this particular capsule was discontinued more than a decade ago, and remaining ones are getting more and more expensive, even if they can be found.  Some are still advertised on, for example eBay, but I was put off by warnings of possible fakes, whose frequency response would not necessarily be the same.

A good currently available alternative is the Primo EM258 from FEL Communications.  At over £5 each, these were 20 times as expensive as the unbranded electret capsules I’d bought before for other more conventional microphone projects, but they are not much more expensive than genuine Panasonic WM-61As these days, and have been tested and shown to have a good ultrasonic response (better, in fact than the Panasonics, according to FEL).

JLI Electronics manufacture the JLI-61A, which is intended as a direct replacement for the WM-61A, but it wasn’t clear that this was available at a reasonable price in the UK.  In the US this would be a good  alternative, at half the price of the Primo.

FEL, incidentally, also advertise a potentially excellent alternative, a tiny SMD-style MEMS microphone.  Normally these things are practically invisible to the naked eye, but FEL have installed one on a breakout board like this:

The microphone itself is a Knowles SPU0410LR5H-QB, as the legend on the PCB suggests, with a sensitivity to ultrasonic frequencies up to 200kHz and beyond.  It was almost twice as expensive as the Primo electret, but would, no doubt work very well, and that price, just under £10, is not at all unreasonable compared to current good quality alternatives.

Alternatively, the cheapest way to obtain a second ultrasonic detector – other than ordering direct from China – might be to purchase a module like this:

Its purpose is distance measurement – the device on the left, marked ‘T’, is an ultrasonic transmitter, the one on the right, marked ‘R’ is a receiver.  It would be a few moments work to detach the receiver from the board, and attach it at the beginning of the circuit.


As for the construction, I planned to fit the circuit inside one of the small boxes I had previously used for microphone preamps; so I connected the small PCB to sockets in the box for power and audio out – I intended to use headphones instead of the speaker supplied.  I also added an extra socket for a line out from the wiper of the volume control.  Together with an appropriate preamp (for example, the one I use for my contact mics), this would enable me to record the ultrasonic sounds I was picking up.


I attached the ultrasonic detector to the front of the box with hot glue, and attached a pair of 2.5M bolts for the two aerials, which had threaded bases (in the end I only used one aerial).  I also added an extra socket for attaching an external aerial or detector; a plug here would disconnect the internal ones.

The remaining parts of the circuit were: a buffer/amplifier for the ultrasonic detector; a high-frequency oscillator, based on 555 integrated circuit; a mixing/heterodyne circuit, and an audio amplifier, based on an LM386 (the only part of the circuit which uses the full 9v available from the PP3 battery).


The difficulties with the first part of the circuit  – the buffer/amplifier – if you were to build it yourself, are to do with size.  The specified transistor for this amplifier, the BC849C, is a tiny, tiny 3-pin SMD device.  I didn’t have a 5p handy – physically the smallest coin currently in use – but I did have a 1p, which is only a little bigger, and a BC849C, and this is how they compared:

It would be quite a task trying to attach wires to this miniscule component – but at least they’ve separated the 3 pins onto separate sides.  I obtained this one just as an example – the actual one used in this circuit was already happily soldered to the PCB by the makers!


The mixer section is based on a CD2003 chip, ‘originally developed for radio receivers’, as the kit booklet says, ‘the core of an AM/FM radio with oscillators, mixer stages, intermediate frequency amplifiers and demodulators for the two ranges’.  In this design, only the AM preamp and AM mixer stage are used.  According to the booklet, the i.c. ‘offers a total amplification of 40 dB and a suppression of the input signal of -20 dB. The output of the mixer provides a low pass filter for an additional damping of the input signal.’ – a very handy chip for the task in hand.  It is possible to get hold of these, but they are not nowadays common or cheap – except from China, it appeared.


After connecting everything together, I plugged the headphones in and tested the detector out, using the preamp I had constructed for the piezo contact mics.

The device worked well: I detected ultrasonic sounds from jangling a bunch of keys and from rubbing my finger and thumb together – everyday sounds known to have a significant ultrasonic component – and outside in the evening I was pretty sure I detected some genuine bats.

This view of the front of the device shows the two potentiometers which need to be accessed when the device is in use: the tuning control on the left, which adjects the frequency of the internal oscillator and effectively ‘tunes in’ the high frequency noise picked up by the ultrasonic detector; and on the right the volume control.

This view of the back shows, on the left, the unit with the 9v battery attached (with velcro, in my usual way); and on the right, without the battery, but showing the caption for the switch, indicating that either the ultrasonic ‘mic’ or the attached aerial can be selected; a 3.5mm plug in the ‘EXT AERIAL’ socket on the front disconnects the switch, so a different size or kind of aerial can be used.

The following sound file gives examples of some quick recordings I made with the ultrasonic detector.  Unfortunately, it’s now late November – much later than when I first tested the circuit out – so no bats flying around.  Instead, I just went round the house for 10 minutes, picking out a few promising locations.

So, you can hear fingers rubbing together, the laptop, TV set, some unexplained radio-tuning type sounds, a low-energy light-bulb, jingling keys, and water streaming slowly into the sink from a tap. The laptop, TV set, radio-tuning and low-energy light-bulb were recorded with the aerial, the others with the ultrasonic microphone-type detector.

The keys are particularly interesting, and I look forward to trying this on some natural sounds outside – especially bats when they emerge from hibernation next spring.



Making music with the BBC Micro:bit, Pt 1

Recently I was lucky enough to be given a BBC Micro:bit as a present.

What is a Micro:bit? – It’s a micro-sized computer, half the size of a credit card, with a row of input and output pins along one edge, Bluetooth capability, a display consisting of 25 LEDs in a a 5×5 pattern, and able to be powered and programmed via USB.  For ultimate simplicity, it can be programmed directly from the website, via WebUSB using the Chrome web browser.

When it was introduced, its purpose was to ‘encourage children to get actively involved in writing software for computers and building new things‘, rather than being merely consumers of media, and around a million of them were given away to schoolchildren by the BBC.

Once I found out that it was considered suitable for primary school children, I was greatly encouraged that I would be able to write programs for it; and, in fact, there is a particularly straightforward way of doing this (as well as Javascript and a version of the Python programming language): that is, the Microsoft MakeCode system, which involves arranging and sequencing interlocking ‘blocks’.  You still have to have the programmer’s mentality, but without having to remember the precise wording and syntax.

As shown in the pictures above, The Micro:bit has a row of about 20 pins on its bottom edge, which have different digital or analog functions, and 5 of them, in particular, are easily accessed with crocodile clips or 4mm banana plugs.  These are +3v (actually 3.3v) and Ground, and analog pins P0, P1 and P2.  As a matter of fact, these last 3 pins can be operated simply by touching them!  You may have to touch Ground with your other hand to make sure they work, though.

Two tactile switch buttons are also available for incorporation into simple programs, and there is a tactile ‘Reset’ button on the back, which causes the Micro:bit to reboot.

For more complex applications – which can even include operating external devices such as servos – a range of edge connectors is available, which can aid permanent connections or take advantage of the smaller input/output pins.

There seemed to be several different ways in which I could make use of the Micro:bit’s capabilities.  First of all, it can, with a speaker attached directly to one of its analogue pins (typically pin P0), make monophonic square-wave sounds of its own, including playing a number of tunes stored in its own memory; secondly, it can be programmed to output MIDI messages; and thirdly, it can utilise and output data from its onboard detectors, which include light and temperature sensing, a three-dimensional accelerometer, and a compass/magnetic sensor.

I decided to start by using the Micro:bit’s internal sound-generating capabilities.

For this, I decided to make a module into which the Micro:bit could be inserted, and which would contain a power supply, speaker and various input and output sockets – all the hardware needed for a free-standing Micro:bit-controlled musical instrument.  That is the subject of this post.


I began with an empty Stylophone case, left over from an earlier project.  I left the speaker in, and the two slide switches on the front left.

In order to incorporate the Micro:bit, I bought a ‘proto-board’ – an edge connector into which the Micro:bit slots at a right-angle, with a section of stripboard attached for ancillary circuits to be built on.  I chose the right-angle one because I needed the Micro:bit to be visible when in use, and to have access to the front and back of the Micro:bit where its sensors are located.

I used this board for a couple of things: a simple op-amp based buffer/voltage follower, and an audio output transformer to be used with an alternative to the speaker, a piezo element.

I also bought a sturdy protective cover for the Micro:bit to ensure that it suffered no damage while being plugged in and out.  This particular type of cover leaves the edge with the pins exposed for just this sort of application.

This picture shows the board in situ, kept in place with matchsticks used as locating pins.  The transformer and op amp are on the right-hand side; on the left are a 4-pin i.c. socket and capacitor left over from an experiment with an audio amplifier, which I decided not to keep.

The voltage follower is the simplest possible op-amp circuit:

I used one half of an NE5532, as that’s what I happened to have available, but any op-amp can perform this task of ensuring the output has a suitably low impedence level.

I’m not sure how useful the transformer is in this exact context, but I learned of its use with piezos from this Nic Collins video on YouTube and had bought a handful of them for this kind of use.

These pictures show how the piezo was fixed into the former Stylophone case, from the inside (L) and outside (R) in the place previously occupied by the tuning knob, which protruded from the bottom of the case.

The Micro:bit can also be powered through thie proto board: mine had a terminal block already installed in it for +3v and Ground; I powered the unit with a PP3 battery, as I am wont to do, so added a small adjustable voltage regulator before the terminal block, set at 3.3v.  This meant that the Micro:bit was receiving its correct voltage, but more volts (up to 9v) would be available for any additional circuitry required.  Two LEDs show whether the 9v and 3.3v parts of the circuit are powered up.

While working on the inputs and outputs, I added the following: an external 9v power in socket; send and return sockets so that external tone controls (such as these) could be used; sockets for an external speaker or headphones and an external piezo; a line out socket for easier mixing or recording; and two sockets connected to analog input pins P1 and P2.

Pin P1 was connected to a 10k lin potentiometer, one end of which was connected to 3.3v, the supply voltage for this part of the circuit, the other to 0v; in this way a continuously variable voltage could be available at pin P1 for control purposes: the Micro:bit would interpret this as a number between 0 and 1023.  If something was plugged into the P1 socket, the potentiometer would be disconnected and the external signal read by the pin instead.

Pin P2 was not connected, unless something was plugged into the P2 socket.  A second 10k lin potentiometer, set up exactly like the one for P1, was added, terminating in a 3.5mm mono plug.  This was available to be plugged into the socket and used for the same purpose if a program were to require it.

In use, I found I had problems with the circuit if the PP3 battery wasn’t new or, if rechargeable, well charged; fortunately, with the external socket it would be possible to use a mains adapter.


I set the mIcro:bit up to make some trial tones, but the sound was rather quiet, through both speakers and piezo, so I added a small amplifier.  This was a very economical (99p) module from a UK eBay seller, based on the PAM8403 chip.  This is a 3W stereo amplifier chip, but I would only be needing one of the channels.

It required a 5v supply, so I powered it from the 9v source and added a small 7805 5v regulator.


I put a 10k log volume control before the amplifier input, with the line out socket between the volume control and the amplifier input.  If a plug is inserted in the line out socket, the signal to the amplifier is cut.

After the amplifier there is a 3-way switch to choose between internal speaker, external speaker or headphones, and piezo element.

The following picture shows the majority of the interior at the end of phase 1 of construction (but before I added the two LEDs):

Features marked: 1 = the original Stylophone switch PCB, repurposed for 9v on/off and 3v on/off; 2 = volume control; 3 = the original Stylophone speaker; 4 = the 5v voltage regulator and amplifier PCB; 5 = the 3.3v voltage regulator PCB; and 6 = the 3-way audio output switch (internal speaker/external speaker or headphones/piezo element).

On the exterior I added three tactile switches to replace the two switches A and B on the front of the Micro:bit.  The first two switches simply duplicate the effect of the two switches on the Micro:bit; with the aid of a couple of diodes and a 4066 digital switch i.c., the third tactile duplicates the effect of pressing A and B at the same time, an action recognised by the Micro:bit.

I didn’t add any pull-up resistors on the outputs of the 4066 as the Micro:bit already has these internally.


After this I had a working module, so it was time for some programming!


I started with a couple of simple programs to make use of the light sensitivity and magnetometer functions.

While experimenting, I connected the Micro:bit to my Macbook with a USB cable, opened the Chrome web browser and wrote the programs using MakeCode on the website.

The following picture shows the main features of the coding page:

1 = the simulator.  Most – but not always all – of the coding you write is automatically simulated here.  As you can see, the first instruction of my program, to display the number 1, is shown, as is the indication that I have chosen instructions which require the speaker to be attached;

2 = the button you press to load the program into your USB-connected Micro:bit.  WebUSB has not yet been implemented in most web browsers, which is why I use Chrome to do this.  Incidentally, if you’ve made any syntactical errors in the program, you will be told when you click this button;

3 = the area to name and save your program as a hex file.  The alternative way to program the Micro:bit is to open its window and drag and drop a hex file onto it;

4 = where you choose to program using the blocks, as I have done, or to write the code in Javascript;

5 = the area where the program is written.


As for my first program: the light control on the Micro:bit takes advantage of the fact that if the current to an LED is inverted, it becomes (slightly) sensitive to light.   The Micro:bit has an array of 25 LEDs on it, and it’s possible to use 9 of these together to record light levels on and around the device.

I found that the most useful way to use this feature was to set the Micro:bit up to sense the ambient light level, and then react to brighter lights being shone onto the LEDs, rather than the other way round (i.e. reacting to being shaded from the ambient light).

In this case, the reaction I programmed was to sound a note, then increase the pitch of that note as the light level increased.  Initially I just used the torch in my mobile phone, but I had already made 2 devices which could be used to control the Micro:bit in Light Reactive mode, the UFO and the Shuttlecraft.

As I said, the Micro:bit was programmed to start up in a waiting state and display the number ‘1’:

Then, when its Button ‘B’ was pressed – or the Button ‘B’ on the top of my module – it would run the part of the program related to the Light Reactive Instrument and display the number ‘3’:

The Micro:bit outputs a number between 0 and 255 to represent how bright the light level is.  0=dark and 255=bright.  The variable lightlevel in the program is set to be 20 times this number, i.e. between 0 and about 5,000.  The opening pitch chosen, 220Hz (the note A below middle C) and the range of values represented by lightlevel is designed to allow the Micro:bit to output pitches between 220Hz – a low to medium note – and about 5kHz – a very high note.

The variable lightambient is used to sample and remember the normal light level around the instrument on startup.  If the light level is no greater than this, the instrument makes no sound, the idea being that it should remain silent until a light is deliberately shone on it.


The second simple instrument I programmed was, in essence, the same thing, but reacting in this case to the presence of a magnet.

The Micro:bit measures magnetic force, as it says above, in microteslas (µT).  It turned out that the range of readings it gave did not need scaling or multiplying like the light level readings did, but produced a useful range of pitches without any changes.  So in this case the variable magforce was equal to the microtesla reading and gave rise to an output at the same Hz as the reading.

In this case, a magnet in the vicinity of the Micro:bit caused a pleasing arpeggiated effect, increasing in pitch the closer the magnet came to the Micro:bit, and decreasing as it was moved further away.

Again, the variable magambient was inserted in order to stop the instrument from sounding until the activating magnet was intentionally brought close to it.

I bought a neodymium (NdFeB) magnet especially for this purpose, as these are particularly powerful – up to 20 times as powerful as conventional ferrite (iron) magnets, in fact.

Neodymium magnets are graded from 28-52 according to their strength.  This one was an N52 (highest power) type.  It works very effectively, but has to be kept well away from any metal parts of the instrument – indeed metal parts of anything – as it will stick very easily and very strongly to them.  I wasn’t sure, but it even seemed to be affecting the speaker in some instances – a speaker, of couse, being driven by a magnet.

The only negative thing about this second instrument is that every time you reboot the Micro:bit, the compass/magnetometer has to be reset if you come to use it.  This is not, unfortunately, a simple matter, as you have to turn the Micro:bit every which way, with your progress shown by the LEDs lighting up one by one.  Only when all 25 of them are lit can you proceed.  This is obviously critical, if you genuinely want an accurate compass direction, but not so critical if all you want to do is make entertaining noises . . .


As I mentioned earlier, I had added two potentiometers connected to analog input pins P1 and P2.   I didn’t have an immediate use for these, but I thought it would be handy to have two variable inputs – for tuning or transposition, for example.

In preparation for Phase 2 of the instrument, I made use of the input to pin P1, and programmed it in the following way:

At the beginning of the program the potential input from pin 1, 0 – 1023, is divided into 21 equal sections, and scaled down to 0 – 20.  Each of the 21 divisions is numbered. ‘notenumber 0’, ‘notenumber’ 1′, ‘notenumber 2’, etc.

There are 20 notes on the former Stylophone keyboard, so there is one number for ‘off’, and one number for each note on the keyboard, ranging from 220Hz (‘Low A’) up to 659Hz (‘High E’).  When the potentiometer connected to pin 1 is turned fully anti-clockwise, the instrument is silent; as it is slowly turned clockwise, each note is stepped through in turn, until the highest note, ‘High E’ is reached.

This experiment proved that it should be possible to connect the keyboard to pin 1 and use the Micro:bit like a Stylophone.  This what I intend to do in Part 2 of this series.

Meanwhile, here are some recordings of these two instruments.  There are 8 short recordings, following this pattern: first the Light Reactive instrument through the internal speaker, external speaker, internal piezo element amd external piezo element; then the Magnet Reactive instrument through the same four media.

The recordings with the external piezo, glued to the bottom of the tin, are particularly interesting.  My experiments with piezos (beginning here), have so far only been in using them as microphonic elements, in particular as contact microphones, and as part of custom-made percussion instruments, and although the introductory article refers to their use as speakers, this is the first time I had actually used one that way.  Hopefully, there will be an opportunity to complete my survey of piezos by looking into this in more detail.

The tin, in fact, had been prepared for use as a kind of drum – and could easily function that way if attached to a suitable preamp.  It is quite large in size – 22cm (about 8.5″) in diameter and 14cm (about 5.5″) deep – and would make a very effective drum.  In this instance, using it as a speaker adds a noticeable reverb effect to the sound of the Micro:bit instrument.




Binaural Recording, Pt 2

After trying the commercial microphones in Part 1 of this series on Binaural Recording, I thought I should try something more home-made – although this also involved a large-ish initial purchase.

What I bought was this handsome life-size mannequin head, intended for work in a hat shop:

The idea, of course, was to install microphones in the ears of the dummy head.  It was made of a fairly hard, but not too brittle, plastic (PVC, I believe it said in the eBay listing), which seemed to be a couple of millimetres thick.

There were several things I particularly liked about this style of head: first of all, the realistic appearance – the whole point of binaural recording is realism, so the closer the recording device resembled the human head, the better.  Unlike some mannequin heads, however, this one wasn’t painted to look like a real person – that would be too spooky! . . .

In particular, the ear was quite well-fashioned:

A big part of the way we hear things is because of the size and shape of our ears, so the accuracy of the ears of the dummy head would have an effect on the quality of the recordings.  For similar reasons, some dummy heads for recording include shoulders, as sound will bounce off these into the ears.

Finally, the underside of the base had a socket which would make it possible for the head to be mounted on a pole or stand, so as to be set at an appropriate sitting or standing height,  whichever was required for a particular recording situation.


My task in this case was essentially to drill suitable holes in the mannequin’s ears, and insert a pair of electret capsules.  I began by soldering a pair of capsules to the cut ends of the twin phono lead I had left after removing 10cm of one end for the previous project.

The capsules were like these:

The lead with the three connections to the side of the capsule is the Ground lead, the other is the signal.  I connected the two capsules this way, with some shrink tubing to make the joints stronger and stop them short-circuiting.

Turning to the mannequin head, the base was only attached by a dab of glue on one side, so came off easily with a little twisting and a cut with a craft knife.

It was a fairly quick procedure to drill 3 holes in the head: one in each ear, slightly smaller than the size of the electret capsules, and another, larger one at the back for the leads to exit from:

I pushed the lead in through the hole in the back, ran a big blob of hot glue round the front edge of the electret capsules and stuck them just behind the ear holes.  I chose hot glue as it’s easy to remove in case the capsules need replacing at some point in the future; it didn’t matter if a bit of the glue came over the front edge of the capsule as the actual hole on the front which the sound goes in through is very small, just a millimetre or so, right in the middle of the capsule.

Looking inside through the base, you can see how the electret capsules are stuck inside the ear, and the cables are held in place with more hot glue:

This took only a matter of minutes, and the final result looked like this:

As you can see, the microphones are held discreetly in the ear holes, and the twin phono lead, which connects to the preamp, exits from the large hole in the back of the neck, where it’s held in place by further hot glue.


The cost of this project was £10.50 – about half as much as the first one.  The mannequin head was £9.50; the electret microphone capsules about 25p; and the half phono lead was 75p.

The only other thing to consider is whether the head should be filled – and, if so, what with – to more accurately reflect the fact that human ears are separated by more than air.  The human brain is about three-quarters water and has the consistency of jelly or tofu; it’s quite heavy, but soft and squishy, and you can’t really pick it up until it’s been preserved in some way, which most brains we see pictured have been.

So what the best thing would be to fill the head is difficult to decide, given that jelly or tofu would soon go off.  In one article that I read the dummy head maker installed the microphones then filled the head with liquid silicone, which gradually set solid.  That seemed to be a good plan, although there’d be no way of getting to the microphones again if there were a problem with the capsules or the wiring.  My thinking is it would be sufficient to use something sound deadening, like wool or felt, to ensure that the microphones would only be picking up sound from outside.

[Edit: This is what I did, filling the empty head with a pyjama jacket, which I had been bought but had never worn].


In the third part of this series, I’ll complete the final project, do some recording and compare the results.

In the meantime, here’s a short extract from the first recording I made with the head:


Binaural Recording, Pt 1

I’d done some field recording with conventional microphones, and after recording with contact microphones and hydrophones, as described in this post, I decided it would be interesting to try binaural recording.

Binaural recording is an attempt to record in as likelike a way as possible.  Since both the size and shape of our ears and the fact that they are placed on opposite sides of our head are important factors in establishing the quality of the sound we naturally hear, binaural recording attempts to replicate this by, most usually, placing microphones within the ears of a dummy head.

‘Lifelike’ aspects which could be captured by binaural or dummy head recording include time differences in the arrival of sounds at one ear or the other, and types of frequency-dependent level differences and distortions which vary with the direction of the sound source.  These would allow a listener using headphones to gain extra information about the precise location and distance of sounds they were listening to; information which would not as readily be apparent if the recordings were made with conventional microphones or played back via loudspeakers.

This article [note: it’s a pdf] refers to three ways in which binaural recordings preserve ‘cues’ as to sounds’ direction and location.  These are, in order of importance: the shape of the ear; the time difference between sounds arriving at one ear, then the other; and, least significantly, it says, the presence of the bulk of the head between the two ears.

(This isn’t a universally-held view: a number of binaural recording devices feature two microphones, side by side, and separated by a sound-absorbing panel.  A notable example of this is the Jecklin disc, which has a diameter of 35cm, is covered with sound-absorbing foam or fleece, and placed between two omnidirectional microphones).

I experimented with three ways of creating binaural recording devices, comparing the results in terms of quality, practicality and cost.  I intended to use the pre-amp I had originally designed for use with electret capsules, and which I had built into a handy case when developing an inductor pickup, so I didn’t include this in the cost.  The pre-amp – which in any case was very cheap and simple – looked like this:

and the case like this:

The preamp is stereo, so inside there are two of the circuits above.  The inputs are phono sockets; the output, a 5-pin XLR, is compatible with the connecting lead of my recording device, a Marantz PMD-660.  The 3.5mm mono socket was for a 9v power source; the velcro on the top of the case was to mount the holder for a PP3 battery.


The first, and simplest, method involved purchasing the microphones!  I found that Roland made a very useful-looking pair of microphones resembling ear-buds, the idea being that you would wear them while recording – no need for a dummy head when your own head could do the job!   This set (CS10-EM) sounded as if it would be particularly effective, as the microphone earpieces also contained earphones, enabling recordings to be monitored while being made.  The downside, however, was the cost: over £70 on Amazon – a reasonable price, I suppose, for a potentially very useful pair of microphones, but not in the price-range for projects in this blog.

However, I noticed that Roland also made a similar pair of microphones for use with GoPro cameras, the WPM-10 WearPro:

Although this set didn’t have actual earphones for monitoring, the cost was considerably less: just under £20, including postage; so I invested in a set.  When it arrived, it contained a choice of different-sized earpieces, so it was possible to select the best fit for your ears; and, as can be seen from these drawings, it also included a pair of foam covers that might have an effect – albeit a small one – on pick up of wind noise.  Having fitted them, it didn’t look to me as if they would stay on for very long, though, so I didn’t plan on using them.

Because of their intended purpose, the plug on these microphones is not an audio plug, it’s a mini USB.

Except that it isn’t a simple mini USB plug at all.  A standard sized USB plug has 4 pins; a mini- or micro-USB has 5; this one is the size and shape of a mini-USB, but has 10.  I suppose it could be called a proprietary connector, as a number of manufacturers use them, but not always in the same way.  GoPro uses them like this:

The connections in the row along the bottom are the standard mini- or micro-USB set: +5v, Data-, Data+, ID and Ground.  ‘ID’ is the one omitted from the standard-sized USB connection: using standard-sized USB connections, the ‘in’ socket on a host device (e.g. a laptop) should be the narrow, rectangular one, Type A; the ‘out’ socket on a peripheral device (e.g. a printer) should be the square one, Type B.  With mini- and micro-sized sockets, there is no distinction in shape between these ‘in’ and ‘out’ sockets, so the job of distinguishing them is done by the ‘ID’ pin.  If the socket is performing the job as a peripheral – e.g. a camera, the ID pin is not connected; if as a host, e.g. a laptop, it is connected to Ground.

In the case of the GoPro, the device responds differently to different resistances between the ID pin and Ground.  As shown in the diagram, a resistance of 100k between ID and ground causes the device to function as a video and audio source, and it can be plugged into an external video/audio receiver; a resistance of 330k, and it will receive signals from a microphone.  In both cases, the presence of a resistance between ID and Ground allows the upper 5 pins to come into operation; their specific use is shown in the diagram above.

(I have seen it suggested that a resistance of 33k allows both these functions at the same time, but that was not confirmed by experiment in the article where I read it).

The reason for the resistor shown with a dotted line is that the conventional place to connect the 33k/100k/330k resistor would be the case of the plug, but one experimenter who posted a video on YouTube had difficulty with this, and used the Ground pin in the centre of the upper level instead and confirmed that this worked fine.


However, I don’t have a GoPro, and this is not the way I intended to use the microphones – in fact, it was the exact opposite.  What I needed to do was remove the mini-USB plug altogether and attach instead two phono plugs, so that the microphones could be used with the preamp shown above, for recording on my Marantz machine.

I was banking on the fact that these microphones would be a pair of electret capsules, and would receive sufficient power from the preamp to operate correctly.

So, as I had done before, instead of buying a pair of separate in-line phono plugs, I bought a 2m twin phono lead – it was only about £1.50 and would, when chopped in half, make two single-ended twin leads.

I took one of the leads and the WearPro microphones:

and cut off the mini-USB plug from the microphone lead.  As expected, this left two signal leads, Red for right, Yellow for left, and two unenclosed ground connections.

I didn’t need a whole metre of extra cable, and my next project would require as much of the 2m as could be spared; so I cut off one pair of phono leads with about 10cm of cable.  In this case the two signal leads were red and white, and the two grounds were black.

All I had to do was connect these 4 leads together, red to red, yellow to white and black to copper, and test the microphones with my recorder.

The test was fine, the microphone in my left ear was recording to the left channel of the recorder, the microphone in my right ear was recording to the right channel; so I sealed the wiring with shrink tubing and duct tape.

I used duct tape because the quality of the electrical insulating tape I’ve been coming axross recently has been terrible: neither flexible enough or sticky enough.  After I took the lower picture I added an overall layer of duct tape to bind the two wires securely together.


The cost of this project was £19.75 – the microphones were £19 and half the phono lead was 75p. Afterwards I felt slightly guilty at being lazy and buying the microphones.  It would have been possible to buy a pair of headphones or earphones and adapt them by gluing electret capsules on the outsides and transferring the wiring from the speakers to the microphones.  The electret capsules would have cost no more than about 20p each, and the price of a pair of not-very-good headphones or earphones would have been minimal, so it could have been done for a quarter or a third of the price.

However, this was a neat solution, took only a few minutes to finish, and the resulting set up looks good.  The quality comparison of my different binaural systems – and sound files – will come later, after I’ve finished all three projects, but this one is going to have the advantage when it comes to practicality, as it’s very simple and discreet, certainly the best solution for situations in which I don’t want to be advertising the fact that I’m recording.

This was my first recording with these microphones. I started in the hallway, walked out of the house into the shed, out of the shed, brushing past some bushes, picked up and started filling the watering can from the water butt, then turned and walked back towards the house. No noise reduction has been applied to the recording, which shows you can get a decent clear sound from the WPM-10s.

As soon as I had completed the project, I took the remaining part of the phono lead and moved on to the next one, which is described in Part 2 of this series, here.


February 2020
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