40dB Stereo Pad + Balanced-to-Unbalanced

The 30/40dB pad from my previous post works fantastically well. I have been using it to bring the balanced line-level output from a mixing desk down to an acceptable level for the microphone input on my camera and the sound-quality is crystal-clear. 

But it is time for version 2.0.  The things I want to change:-

• Stereo. Up to now, the cable I have been using to connect the pad to the camera simply connects the camera's L and R inputs together. It works fine, but the camera records two audio channels and I want to use them. 
• The first version is nice and general-purpose: it provides balanced output, handy for feeding into a balanced microphone input, if I ever wanted to. But my camera doesn't have a balanced input, so a camera-specific version with unbalanced output to a more convenient connector (phono or 1/4" stereo) would make connecting up a little easier (although I do love how rock-solid XLR connectors are). I'm willing to sacrifice some generality for convenience here.
• I don't really need it to have a switchable attentuation level.  The mixer has an output level fader and the camera has variable sensitivity on its microphone input. I'm willing to sacrifice the flexibility of a switchable attenuation level to reduce the (slight) risk that the contacts in the switch will become oxidised and noisy at some point.

Here is the schematic for version 2.0

The pad part is mostly the same as the earlier version.  I included some extra resistors (R1, R3, R6 and R8) that I have bridged with little wire links to give myself some flexibility.  If I want to increase the attenuation, I can snip the wire links which will increase the attenuation to around 45dB.  I had the board space (and the resistors) so it seemed a shame not to use it.  At the output, I have given myself a choice of either 1/4" stereo jack output or 2 x phonos.

If I had thought of it, I might have included a way to bypass the pad completely so that I could use this to connect two dynamic microphones (with balanced XLR outputs) directly to the camera input without any attenuation, but I didn't think of it (until now).  That's what version 3.0 is for, I guess.

This is what it looks like:-

I haven't tried it in anger yet, but initial testing is positive.  I'll report back once it has had its first proper outing.

UPDATE January 2021: I have been using this gadget regularly since May and it works absolutely perfectly: I am very happy with the result.  The amount of attenuation is applies is just about perfect and I haven't felt the need to snip the wires bypassing R1, R3, R6 and R8.  I do have about 12dB of attenuation dialed in on the camera's microphone input, bringing total attenuation up to about 42dB, which sounds about right and - more importantly - lands the recorded audio levels just about where I want them.

30dB/40dB Switchable Balanced Pad

I had a requirement to connect the (line-level) output from a small mixing desk into the microphone input on my camera (mic-level, obviously). I was using a DI box with a 30dB pad in it, but that required a 9V battery which was sure to run out at the most inopportune moment.  Since the camera didn't take balanced input anyway, there was no real need for the DI (in fact, the cable from the DI into the camera was wired as a balanced-to-unbalanced cable).  A little bit of internet research led me to http://www.uneeda-audio.com/pads/ which has all of the relevant information (but does require careful study !).  Based on that, here's what I came up with:-

This gives me the option of either 30dB or 40dB of attenuation, depending on the position of the switch (I find that with the in-camera gain turned down as far as it will go, 30dB is enough, but the flexibility is nice to have).  The input impedance seen by the mixer is about 27K at 40dB and about 6.8K at 30dB.  The output impedance presented to the camera is 220Ω.  It works pretty well.  I built it into a small plastic project box lined with copper foil (originally bought with the intention of shielding a guitar cavity):-

The resistors are of the metal film type to minimise thermal noise.

This is the end-result:-

It is fairly compact and rugged, completely noiseless and doesn't require any power.  

The justification for the choice of resistor values is as follows.  The general equation for gain (which will be negative) of a resistor divider network is:-

...where R₂ is the lower resistor in the circuit diagram and R₁ is the sum of the resistors labelled R_1.n in the circuit diagram above.  So, for 30dB of attenuation, the switch is closed and the 10K resistors are bypassed and the equation becomes:-

With the switch open, the 10K resistors are included in series and the equation becomes:-

I could (possibly should?) have used a slightly lower resistor value than 10K to land a little closer to -40dB.  7.5K would have done it, but it had already been a long night.  a 30K resistor in parallel with each of the 10Ks would do the trick.

Experiments with a Smartphone Barometer

Modern smartphones have a wealth of sensors that provide all sorts of avenues for amusement for modern geeks.  One of them is a barometric pressure sensor, which is the topic of this post.  I have an app called Phyphox which provides direct access to the data from those sensors.

I was in London a while ago, working on the 18th floor of an office-block. There is an express lift which gets from the ground to the 18th floor (60m up) in 15 seconds non-stop. The locals tell me that when the lifts went in first, they used to be even faster but they hurt peoples' ears and had to be slowed down. I have no trouble believing that.

These graphs were produced using the Phyphox app on my phone, measuring the changes in air pressure with altitude (like the altimeter in a plane) as well as vertical acceleration.  This is from ground straight up to the 18th floor.

 

This from the 18th down, but with stops on the 17th and 16th (other people wanting to use the lift and messing with my experiment...no consideration !) - hence the "staircase" effect for the first 45 seconds or so.

 

A bit later on, I managed to capture a "clean" descent all the way from the 18th down to the ground:-

 

The cabin of an aeroplane is another interesting place to try this.  These were taken on the flight home during take-off and climb-out:

...and coming in to land:-

(I'm probably on a no-fly list now). You can see the cabin-pressure gradually drop from ground-level pressure to around 760hPa as the plane climbs and gradually increase from that back to ground-level pressure again as the plane comes in to land. The change is made gradually over the course of about 14 minutes each time.

 

Of course the cabin of the plane is pressurised and outside air-pressure at the cruising altitude of 36,000ft is far lower than 760hPa. The cabin pressure is equivalent to being at an altitude of around 2,300-2,400m (a good bit less than ⅓ of the way up Mount Everest).

 

The Phyphox app really is an awful lot of fun. It gives you direct access to all of the sensors on a modern smartphone (and there are lots of them) and lets you run experiments like these with them.

Experiments with an AD633 Multiplier IC

Something I have always found a little confusing was the idea of signal mixing.  The source of my puzzlement probably has its roots in the audio world of mixing desks for combining the sound from instruments and singers for music recording or live performance.  These mixing desks are (or are supposed to be) additive: their output should be the simple arithmetic sum of all of the inputs, scaled according to how far the sound engineer pushes the fader.

In the RF world, mixing means something quite different.  When we mix RF signals, we (usually) specifically want them to interfere with each other to create frequencies that weren't present in either (because we're usually talking about two) of the original signals.  That is usually the last thing in the world you want to happen in the audio world, but it how a heterodyne radio works (more on this a little later).  For this to happen, the RF signals need to be mixed in a non-linear way, usually by multiplying them together.  It took a while before this difference dawned on me.

The maths behind mixing in the RF sense of the word is pretty straightforward.  It relies on the trignometric relationship:-

sin(a) ✕ sin(b) = ½sin(a+b) + ½sin(a-b)

In plain English, if two sinusoidal signals are multiplied together, the resulting signal will consist of two sinusoidal signals at the sum and difference of the two original frequencies. 

That has to be tried.

Inspired by some of the many great W2AEW videos (Alan is my hero!) - specifically this one and this one - I did some some experiments with multiplying a sine wave by a lower-frequency square wave using the diode-switching technique and got some interesting results.  However, the real fun was had with an Analog Devices AD633 multiplier IC.  Its a little beauty: it takes in two signals (differential) and outputs the result of multiplying them together (÷10).

Here is a 1KHz sine wave being multipled by a 800Hz square wave (the yellow trace is the result):-

Its pretty easy to see what's going on here - the inversions of the sine wave when the square wave changes state are pretty evident.

Here's 10KHz and 800Hz:-

Its a little less obvious here, but if you look carefully, you can still see the sine wave change phase by 180° when the polarity of the square wave changes. 

We'll come back to this in a bit.  For now, let's try multiplying two sine waves together, one at 10KHz and one at 800Hz.  In theory, the result will be a signal which is the sum of two new sine waves, one at 9.2KHz and one at 10.8KHz.

Bingo !!  The purple trace is the FFT and - sure enough - there are two (and only two) peaks at exactly the frequencies that the maths predicts.  Note also that there is none of the original 10KHz signal present (there is no 800Hz signal there either, although it would be off-screen to the left even if there was).

Returning to the 10KHz sine wave and the 800Hz square wave: a square wave consists of the sum of the odd harmonics of the fundamental.  So our 800Hz square wave has components at 800Hz and 2400Hz and 4000Hz etc.  So multiplying this by our 10KHz sine wave should produce an output with tones at 10KHz +/- (800Hz, 2400Hz, 4000Hz...).  Let's see if it does:-

Yep.  The cursors show the fundamental.  The horizontal divisions are 1.25KHz each, so you can see that the other tones are all about 1600Hz above and below the fundamentals.  Again, note that there is no sign of the original 10KHz signal at all...just at the theory predicts. 

"Why?", you might be tempted to ask.

The plan for the AD633 is to use it as the basis of a bat detector.  Bats' echo-location sounds are somewhere around 48KHz (at least the Pipestrelle bats that are common in Ireland) - far above the range of human hearing.  The idea is to mix the signal from an ultrasonic transducer (hopefully from lots of hungry bats) with - say - a 40KHz sine wave, thus hopefully shifting the bats' sounds into the human-audible range.  This is the same basic principle on which a heterodyne radio receiver works, with a local oscillator being mixed with the target RF signal to produce a intermediate frequency which is then detected with a high-Q filter.  

555 Timer IC

I have been studying the classic and venerable 555 Timer IC recently and have created an animation showing the various signals on it as it moves from state to state.

http://bogpeople.com/555/

Also a video...

Wireshark Traps For Young Players

I ran into an interesting problem in work today.  An Ethernet switch appeared not to be behaving correctly: frames that I know were being received on an 802.1q trunk (coming in from a WAN carrier) weren't being sent out an access port for no reason I could see.  The switch configuration was fine...there was no obvious reason it shouldn't have been working.

A bit of background.  

The IEEE 802.1q header has a bit in it call the CFI bit.  CFI stands for "Canonical Format Indicator". It is (or was) used to distinguish between Ethernet-format MAC addresses ("canonical") and Token Ring-format MAC addresses ("non-canonical").  Basically, this bit is set to 0 on Ethernet frames and 1 on Token Ring frames.  Since Token Ring is (mercifully) now a thing of the dim-and-distant past, this bit is now always set to 0 (which is why most people - including me - have never noticed it and may be oblivious to its existence).  The 802.1q standard specifies that frames received with the CFI bit set (indicating Token Ring frames) should not be transmitted on untagged Ethernet interfaces (because the MAC address format is different and won't make any sense).  This will become important later.

The more recent 802.1ad specification which - among other things - codifies "Q-in-Q" (a.k.a. double-tagging) renames and repurposes this bit so it is now called the DEI bit.  This is where my problems started, but more on that later.  DEI stands for "Discard Eligible Indicator" and is a hint that this packet can be discarded in the event of congestion on a link.  It is in the same bit position as the CFI bit was...it just has a completely different meaning now.  

This gets me to my first problem.  I was told that my carrier was sending me tagged frames with the DEI bit set (there was a legitimate reason for this).  I figured that my Ethernet switch might interpret the DEI bit as a CFI bit (remember, same bit position in the 802.1q header) and might (correctly, from its point of view) refuse to send the frame on an untagged Ethernet interface.  Sure enough, once the carrier stopped setting the DEI bit, everything started working OK.

That would be the end of the story except that when I ran packet-captures to see what was what was going on, the frames coming in from the carrier had the DEI bit clear.  I can think of a few possibilities:-

• My tip-off that the carrier was setting the DEI was incorrect and there was something else going on. Not likely since the problem went away as soon as the carrier changed this (and my theory around this seemed pretty sound).
• The interface-mirroring on the switch was "lying" about the CFI/DEI bit being set (which didn't seem hugely likely either)
• There was a problem with my packet-captures.

After some research, I found this (short but very informative) thread https://lists.linuxfoundation.org/pipermail/bridge/2016-March/009940.html.  The TL;DR version is that the Linux kernel "abuses" the CFI/DEI bit for its own internal purposes and always clears it !  This was fine when the bit was CFI (and therefore always 0), but it is distinctly not fine now that the bit should be interpreted as DEI.  As it happened, I had done all of my packet-captures on Linux machines 😞

So I set up an experiment.  I have a machine transmitting Ethernet frames (using this https://gist.github.com/austinmarton/1922600) with with the following characteristics:-

• IEEE 802.1q encapsulation (Ethertype 0x8100)
• VLAN tag 120
• CFI bit in the 802.1q header set to 1

Then I used Wireshark to capture these frames on a few platforms.  The results were eye-opening.

Linux

Sure enough, the 802.1q header and the VLAN tag are there, but the CFI/DEI bit is set to 0.  Bingo !!

Windows

The second packet-capture is done on the same NIC on the same laptop (it dual-boots Windows and Linux)

Even worse !!  Under Windows, I don't even get to the the 802.1q header.  I tried this on two different Windows machines (both Windows 10, admittedly) with the same results.  This could really send you off on a wild goose-chase if you were trying to troubleshoot a real problem.

Mac

Perfect: I can see the 802.1q header and the CFI/DEI bit is still set to 1, just as it should be.

The message thread indicates that this only happens with if the Ethertype value is 0x8100 (802.1q) or 0x88a8 (802.1ad) but not if another Ethertype is used.  Let's try that:-

Look at that !  Changing the Ethertype from 0x8100 to 0x8101 and now the Linux laptop leaves the CFI/DEI bit untouched (I had to twist Wireshark's arm to get it to parse 0x8101 as an 802.1q frame).  That makes sense: the Linux kernel no longer sees the frame as an 802.1q/802.1ad frame and therefore doesn't see the following two bytes as an 802.1q/802.1ad header that it can mess with.

The first lesson is that one needs to be very circumspect about packet-captures unless you are very sure about the behaviour of the platform you are doing the capturing on.

Second, if you have a WAN provider setting that DEI bit (e.g. on a "best-effort" WAN service), be aware that your switch may be refuse to pass those frames through.

ATX Bench Power Supply Status LED

Like lots of electronics hobbyists, I have an old ATX power supply salvaged from a PC that I use as a bench power supply.  It cost practically nothing and it works well enough.

However, like any self-respecting geek, I can't look at anything for any length of time without thinking of ways it could be improved. The target of my ruminations today is that LED in the centre of the picture. It was originally wired to that it would glow dimly when the PSU was in "standby" mode (i.e. connected to the mains, but not supplying output to the front panel) and brightly when the PSU was turned on.  This was easy to do (see below), but not a good idea.  It was practically impossible to tell at a glance whether the LED was "bright" or "dim".  It was just crying out to be changed.

Among other things, the ATX power supply outputs two useful signals:-

• 5VSB : 5V standby, a low-current 5V source that is available when the rest of the power-supply is in standby mode.  This is on the purple wire of the ATX connector.
• PWR_OK: A logic-level signal that goes high when the power supply has turned on and stabilised. This is on the grey wire of the ATX connector.

In my original version, I simply wired both of these to a red LED via two different resistors: 4.7K on the 5VSB line and 470Ω+1N4148 diode on the PWR_OK line.  If only the 5VSB line was powered, the LED glowed dimly (through the 4.7K resistor) but if both were high then the LED was powered via the 470Ω resistor and glowed more brightly.  The 1N4148 diode on the PWR_OK line was there to stop it sinking the current from the 5VSB line if it went properly low.

I decided I wanted to use a multicolour LED instead, red for "standby" and green for "on".  Here is what I came up with:-

When the power supply is in "standby" mode (5VSB on, PWR_OK off), current flows from 5VSB, through the red LED and R3 and hits the drain of Q2.  Because PWR_OK is low Q1 is switched off and its drain (and therefore the gate of Q2) is sitting at about 2-3V (the 5V from 5VSB minus the voltage drop through the green LED.  This is enough to turn on Q2 (which has a typical gate threshold voltage of 2.1V), so the red LED lights.  Virtually no current flows through the green LED to the gate of Q2, so it stays off.

When the power supply is turned on, the PWR_OK signal is asserted.  This turns on Q1 causing its drain voltage to drop to near ground.  This has the effect of turning on the green LED while simultaneously pulling the voltage at the gate of Q2 to ground, causing it (and therefore the RED LED) to turn off.

The 100K resistors are really precautionary: the gates of the MOSFETs draw no current at all.  They are only there to limit current if a MOSFET dies in a way that shorts its gate to ground.  They could be omitted and the circuit would still work just fine.

The whole thing fitted onto a bit of perfboard around 25mm² which was encased in some transparent shrink-wrap. Its not pretty but it works well:-

There may well be better ways of doing this (suggestions welcome in the comments below).