The Trouble with Hobbies

Have you ever suddenly been inspired to embark on a brand new hobby?

Maybe you’ve never owned a boat before, but having seen one chug by on the river you have thought “I’d love to do that!”. A quick browse in the classified ads shows lots of boats that look fine, and they don’t cost all that much. Basically any boat would be great, and you could gradually do it up, even if it is a bit shabby now. In your mind’s eye, your family will love you when you are able to take them on spur-of-the-moment, cheap weekends messing about on the water, starting in a few weeks’ time.

From this high point where the world is your oyster, you begin to take the advice of the magazines and other experienced hobbyists. Before you have even owned a boat, you become aware of the hierarchy of boat owners, and the boats that would render you a laughing stock if you owned them. You become aware of the general consensus on different types of bilge pump – not something you ever wanted to know. You begin to form an idea of the boat you should really go for – and it is not one of the bargain basement jobs you first saw. You might just about be able to stretch to a boat that would put you in the lower echelons of boat ownership but, importantly, not on the very lowest rung. You could always, perhaps, move up from there over time.

It now turns into an all-consuming hobby with the goal of having a boat on the river at the end of the year. In the end it costs thousands, and your children have grown up and left home before your boat finally takes to the water. You hit a bridge and rip the top off your boat the first time you take it out. You feel sick and abandon the whole hobby (a true story).

That’s the nature of male hobbies. They start out as wonderful, spontaneous ideas, but can turn into nightmares – mainly due to the existence of other hobbyists! Audio is one of those hobbies, I think. Ridiculously, the prices paid for bits of audio knickknackery rival the costs of boats.

A person could be seized one day by the idea of hi-fi as a way to improve their life, buy an amp and some secondhand speakers off Gumtree for £100, and plug their tablet or laptop headphone socket into the amp using a £2 cable. Hey presto, a hi-fi system that will sound much better than what they had before, and which has tinker-ability via the buying and selling of speakers and the audio streaming/library software options; there is no urgency in changing the amp and tablet hardware as they are pretty much perfect in what they do. The speakers are almost like pieces of furniture, so the person can indulge their tastes in how they look as well as how they sound, and they can be restored using standard DIY skills – a nice mini-hobby.

But what if the person does the natural male thing, and starts to read the magazines and forums? Immediately they will realise that their tablet’s headphone output is a joke in the audio world. They need to spend at least a few hundred pounds on a half-decent ‘DAC’, plus a couple of hundred on a budget cable. And of course, this is only for convenience: real audio quality can only be had if they own a decent turntable and a special vibration-free shelf to put it on. Where do they go from there? They need to make a decision on which turntable and which cartridge to go for. They need to take a view on cables, power conditioners, valve or solid state amps, accessories like cable lifters and record cleaning machines. Each decision, they are assured by their fellow hobbyists, will result in “night and day” differences in the sound.

After some months agonising over it, they assemble a beginner’s system for about £3,000 – they will upgrade as budget allows. It sounds OK, but they know that even though the brand is a highly recommended one, the particular model of valve amplifier they could afford has “hints of a slightly reticent mid range” – one of the magazines said so – and if they listen carefully, perhaps they can hear that… But the more powerful 18 Watt model cost £800 more and they decided against it. Perhaps they made the wrong decision. The nightmare unfolds…

Pop and click remover, old electronics magazines

Just saw a short article about a new product that aims to remove the pops and clicks from vinyl records. It…

…digitizes the signal at 192/24 bit resolution and then uses a “non-destructive” real time program that removes pops and clicks without, the company claims, damaging the music.

…In addition to real-time, non-destructive click & pop Removal the SC-1 features user controllable click & pop removal “strength”, a pushbutton audiophile-grade “bypass” that lets you hear non-digitized versus digitized signal (for when you don’t need pop and click removal), iOS and Android mobile app control and 192/24 bit hi-res digital processing.

Of course it is highly ironic that a vinyl enthusiast should need the services of the digital world to improve the sound of his recordings. And it is obvious (surely) that the digital stream could be stored for later replay without needing to further degrade the original vinyl or wear out the multi-thousand dollar stylus that is no doubt being used. (Omitting to mention the most obvious idea of just listening to a digital recording…)

The aim of the product reminded me of a certain project in an old electronics magazine, a huge number of which I still have in a set of bookshelves that I haven’t touched since 1990 – the date of the last magazine I seem to have bought. Sifting through them, it is amazing how familiar the front covers still are –  a measure of the intensity of youthful hobbies.


From Electronics Today International in April 1979, the project I remembered was a ‘Click Eliminator’ for vinyl records based on an analogue CCD delay line. The idea was to insert a few milliseconds of silence in place of the offensive click. Here’s how it worked:


Electronics Today International was the magazine I would go to WH Smiths for on a Saturday, being terribly disappointed if the latest issue wasn’t in. I would say more than 50% of issues featured an audio or hi-fi project: from 1982 an active speaker project for example, or from 1986 “Can Valves make a comeback?” with an accompanying valve amp project. There were any number of MOSFET amps, phono pre-amps, tape noise reduction units. Electronic music featured prominently with projects for effects pedals and synthesisers galore. I devoured this stuff.

Other magazines included: Practical Electronics, Wireless World, Everyday Electronics, Elektor, Electronics and Music Maker, and one I didn’t recall Hobby Electronics. I also bought any number of computer magazines. I have never thrown any away, so I have hundreds of them gathering dust.

Thoughts on creating stuff


The mysterious driver at the bottom is the original tweeter left in place to avoid having to plug the hole

I just spent an enjoyable evening tuning my converted KEF Concord III speakers. Faced with three drivers in a box, I was able to do the following:

  • Make impulse response measurements of the drivers – near and far field as appropriate to the size and frequency ranges of the drivers (although it’s not a great room for making the far field measurements in)
  • Apply linear phase crossovers at 500Hz/3100Hz with a 4th order slope. Much scope for changing these later.
  • Correct the drivers’ phase based on the measurements.
  • Apply baffle step compensation using a formula based on baffle width.
  • Trim the gain of each driver.
  • Adjust delays by ear to get the ‘fullest’ pink noise sound over several positions around the listening position.
  • ‘Overwrite’ the woofer’s natural response to obtain a new corner frequency at 40 Hz with 12dB per octave roll off.

The KEFs are now sounding beautiful although I didn’t do any room measurements as such – maybe later. Instead, I have been using more of a ‘feedforward’ technique i.e. trust the polypropylene drivers to behave over the narrow frequency ranges we’re using, and don’t mess about with them too much.

The benefits of good imaging

There is lovely deep bass, and the imaging is spectacular – even better than my bigger system. There really is no way to tell that a voice from the middle of the ‘soundstage’ is coming from anywhere but straight ahead and not from the two speakers at the sides. As a result, not only are the individual acoustic sources well separated, but the acoustic surroundings are also reproduced better. These aspects, I think, may be responsible for more than just the enjoyment of hearing voices and instruments coming from different places: I think that imaging, when done well, may trump other aspects of the system. Poorly implemented stereo is probably more confusing to the ear/brain than mono, leaving the listener in no doubt that they are listening to an artificial system. With good stereo, it becomes possible to simply listen to music without thinking about anything else.

Build a four way?

In conjunction with the standard expectation bias warning, I would say the overall sound of the KEFs (so far) is subtly different from my big system and I suspect the baffle widths will have something to do with this – as well as the obvious fact that the 8 inch woofers have got half the area of 12 inch drivers, and the enclosures are one third the volume.

A truly terrible thought is taking shape, however: what would it sound like if I combined these speakers with the 12 inch woofers and enclosures from my large system, to make a huge four way system..? No, I must put the thought out of my head…

The passive alternative

How could all this be done with passive crossovers? How many iterations of the settings did it take me to get to here? Fifty maybe? Surely it would be impossible to do anything like this with soldering irons and bits of wire and passive components. I suppose some people would say that with a comprehensive set of measurements, it would be possible to push a button on a computer and get it to calculate the optimum configuration of resistors, capacitors and inductors to match the target response. Possibly, but (a) it can never work as well as an active system (literally, it can’t – no point in pretending that the two systems are equivalent), and (b) you have to know what your target response is in the first place. It must surely always be a bit of an art, with multiple iterations needed to home in on a really good ‘envelope’ of settings – I am not saying that there is some unique golden combination that is best in every way.

In developing a passive system, every iteration would take between minutes and hours to complete and I don’t think you would get anywhere near the accuracy of matching of responses between adjacent drivers and so on. I wouldn’t even attempt such a thing without first building a computerised box of relays and passive components that could automatically implement the crossover from a SPICE model or whatever output my software produced – it would be quite big box, I think. (A new product idea?)

Something real

With these KEFs, I feel that I have achieved something real which, I think, contrasts strongly with the preoccupations of many technically-oriented audio enthusiasts. In forums I see threads lasting tens or even hundreds of pages concerning the efficacy of USB “re-clockers” or similar. Theory says they don’t do anything; measurements show they don’t do anything (or even make things worse with added ground noise); enthusiasts claim they make a night and day improvement to the sound -> let’s have a listening test; it shows there is no improvement; there must have been something wrong with the test -> let’s do it again.

Or investigations of which lossless file format sounds best. Or which type of ethernet cable is the most musical.

Then there’s MQA and the idea that we must use higher sample rates and ‘de-blurring’ because timing is critical. Then the result is played through passive speakers with massive timing errors between the drivers.

All of these people have far more expertise than me in everything to do with audio, yet they spend their precious time on stuff that produces, literally, nothing.

New bass drivers for KEF Concords

Finally got round to ordering some better bass drivers for the KEF Concord III conversion at the very high end price of £19 each.

They’re Skytronic 902.208 8″ polypropylene drivers, and as you can see, they’re quite a bit beefier magnet-wise than the Peerless SKO200.

There seems to be some confusion about the Thiele Small parameters for this driver. As far as I can tell, the ones here are correct. It probably works out that the 30l KEF cabinets are too small, and we end up with a Q of 0.97. No matter.

I have measured the driver in the cabinet in the nearfield, and attempted to correct it for phase and amplitude, and then modified the filter to give me a driver with 38 Hz corner frequency and a roll-off at 12dB per octave. The cones move quite a lot sometimes, but the sound is good.


902.208 mounted in place on the KEF III. The diameter of this driver rim is smaller than both the originals and the previous Peerless replacements, hence the need to clamp the driver as there isn’t sufficient wood to screw into.

Software: the future of audio

Last night, on a whim, I decided that I would like my active crossover software to display some sort of indication of the output levels being sent to the DACs. This is quite important, and something that I should have tackled quite a while ago. Basically, we should be worried about clipping, and also ‘overs’ i.e. those interpolated samples that are generated by DAC reconstruction filters in between the recorded samples and which have the potential to clip even though the recording does not, directly. By messing around with various types of driver correction and so on, am I running the risk of clipping? Or, am I wasting DAC resolution by needlessly attenuating my DAC outputs too much?

Here is how easy it was to display the information in a useful and aesthetically pleasing way:

  • I created six vertical rectangular areas on the active crossover app’s screen – one bargraph for each DAC output.
  • I decided upon a linear percentage display (not dB) and an update rate of 10 Hz
  • A timer was set to trigger at 10 Hz (the timer is provided by the GTK GUI library) and call the function to draw the six bargraphs
  • In the output function for the DACs, I take the absolute value of each sample as I write it to the DAC and compare it to the maximum recorded so far for that channel (out of six channels). I overwrite the maximum if it is exceeded. There is a ‘mutex’ interlock around the maximum value to prevent the bargraph drawing function from accessing it at the same moment.
  • The bargraph drawing function for each channel accesses that maximum recorded value and saves it. The maximum value for that channel is then reset to zero. The saved value is compared against that bargraph’s previous displayed value. If it is greater, a coloured rectangle is drawn directly proportional in length to the value. If it is less, the previous value is multiplied by 0.9, and the rectangle drawn to that height, instead. With this simple system, we have a PPM-style display that shows signal peaks that slowly decay.
  • The bargraph display function also records an absolute maximum for that channel, which doesn’t get reset. This value is displayed as a red horizontal line, thus showing the maximum output level for that particular listening session.

The result is one of those attractive arrays of VU meters that dances in response to the incoming signal levels. The results were interesting, and will alert me to any future mis-steps with regard to clipping – it still doesn’t tackle the issue of ‘overs’ directly, however.

But the reason for mentioning it, is to show the power and simplicity of engineering with software. To build a PPM meter in hardware and wire it all up, would not be trivial, and would take days, weeks or months for a commercial product. In software, it takes less than an hour and a half to construct it from scratch. Audio processing functions are equally simple to create and integrate within the system. It seems clear that once the basic DSP ‘engine’ is in place, complex audio systems can be put together like Lego. A perfectly capable three-way speaker can be built in days. It is not too hard to see how a three-way, six channel DSP system could simply be scaled up to create something like the Beolab 90.

Is this an exciting trend, or the end of everything that makes audio interesting? I think it is the former, but I can see that many traditionalists might disagree.

KEF Concord III conversion

kef badge

Recently, I thought I might try to combine modern technology with the styling of 70s hi-fi by converting a pair of KEF Concord IIIs to work with DSP active crossovers, and also upgrade them from 2.5-way to 3-way with all-new drivers. The scheme is based on the same software and DAC that I used for my earlier DIY effort.


Some KEF Concord IIIs (not my particular pair) []

I bought the KEFs a few years ago because I thought they looked fabulous. I thought they would sound OK because they’re not tiny and contain two 8 inch drivers. I was wrong: to me they sounded weak and ‘boxy’, so it required no soul-searching for me to decide to modify them irreversibly. Who knows: maybe they had bad capacitors or something, but as you might have guessed, I probably wasn’t going to be keeping them in their original form, anyway.

I didn’t give my conversion project much planning. I already had some Peerless 8″ polypropylene drivers bought very cheap, which WinISD indicated were perfect for the enclosures, and I thought I could cross these over to 3″ drivers rather than the 4″ I used for my big speakers; I duly bought some Monacor SPH75/8 polypropylene mid-bass drivers. I thought about using 19mm tweeters, but in the end plumped for the same Monacor DT25 as I used in my main system because of their small size, particularly the front flange. All pretty cheap.

The KEFs are stylishly covered in a fabric ‘sock’ that was no doubt very cheap to make, but I think looks good. (There is even the possibility of commissioning the very talented mother-in-law to make new ones in funky colours).

I removed the small plinth at the base of the speaker (four long wood screws) and peeled back the ‘sock’ from there to reveal a rounded chipboard enclosure and the three drivers – the Concord is a 2.5-way system. I decided that I would replace one of the 8″ drivers with my mid and tweeter, and that I should therefore invert the enclosure in order to keep all three drivers close together with the tweeter close to the top of the enclosure. I removed the two 8″ drivers but left the original tweeter in position as a ‘plug’ for its hole.

I dusted off the router and made two 18mm MDF flanged discs to replace the 8″ drivers. I should have made the flanges wider because they’re not quite wide enough to take a screw head and clear the necessary foam gasket underneath, meaning I’ll have to clamp them externally. I painted them to seal in the sawdust.

The SPH75/8 is troublingly difficult to mount for a one-off hand-made ‘rapid prototype’: a virtually non-existent flange from the front or behind, and a magnet that is almost as wide as the driver, meaning that if you mount it from the front, there’s almost no gap for the air to flow around unless you widen out the area around the driver from behind. It’s squarish, so if you mount it from behind but don’t want the full thickness of the baffle in the way, you end up having to accommodate the corners, which is fiddly without machining a complex-shaped recess. I ended up mounting the driver from behind, shaping the corners with a chisel. Next time, I will definitely find a woodworking expert to make the ‘plugs’ to my CAD designs!

I needed to make a chamber for the SPH75/8. WinISD told me it should ideally be 3 litres or so – but probably not all that critical for the mid range. I figured the easiest way to do it would be some 110mm plastic piping from the local DIY shop which is quite thick and fairly ‘dead’ if you knock it. I could even buy a ready-made fitting to allow me to plug the end. I duly made an assembly and fastened it to the rear of the MDF ‘plug’ using some bent aluminium brackets. I stuffed it with speaker wadding. The volume works out at about 2 litres, so not far off ideal.

IMG_0488 cropped

Mid range chamber made from 110 mm plastic pipe and end cap. Hopefully airtight by virtue of neoprene foam gaskets. It is stuffed with wadding .

Using self-adhesive neoprene foam and P-section draught excluder (this really does make a great seal), and plugging various holes, I rendered the mid range and bass enclosures pretty airtight. A top tip: hot melt glue is your friend. It plugs holes and gaps perfectly, and I have found that with a quick application it doesn’t seem to melt PVC cable insulation or ABS, so it’s ideal when you just want to feed cables through a hole in wood or plastic and seal the hole.

Crudely fastening it all together, I fired up one speaker to have a quick listen using slightly modified settings from my big system. I found it really interesting and encouraging, but when the bass drivers were played in isolation there was audible distortion. I worried that it might be the enclosures (they are made from mere 15mm chipboard), but I eventually narrowed it down to the drivers.


A modified KEF Concord. Those particular pieces of foam are just a temporary experiment, and would be too thick to fit under the fabric cover, anyway.

The mid and tweeter sounded sound spot on.

After building up the second speaker, the next stage was to set them up slightly more scientifically than before. I measured them (woofer near field, and mid and tweeter far field ‘pseudo-anechoically’) and applied roughly the appropriate correction to each driver (phase and frequency response, delay, gain). I also implemented bass extending EQ to aim for the same response as my big system(!) i.e. a 38Hz -3dB point. It sounded pretty reasonable, but I knew the bass drivers were not very good.

Next, I replaced the bass drivers with some cheap but much better Skytronic units (the same brand as my larger speakers). I made the appropriate measurements and compensations in the DSP, and raised the -3dB point to 40 Hz for the sake of reducing the power into the bass drivers.

I added some bracing to the most obviously flappy bits of the KEF Concord enclosures. Broom handle was much cheaper than dowel of the same diameter! The black square between dowel and enclosure is 1mm neoprene sheet. Dowel held in with countersunk wood screws from outside the enclosure.

Yes, the photos make it all look very ‘agricultural’, and the wide angle iPhone lens makes this bit of it look anything but square and perpendicular, but it is actually about right, and the speakers are solid, airtight, etc. where it matters.


Did the bracing changed the sound? Can’t say, but it had to be done. I measured the driver in the near field again, and it hadn’t changed at all.

I re-fitted the fabric ‘socks’ – which I managed to get wrinkle-free much to my surprise.

finished KEF

A KEF Concord III with its fabric covering restored

As mentioned before, I ended up inverting the enclosures which meant that I had to remove the fabric ‘socks’ which were stapled very close to the ‘lip’ that is formed at the top of the enclosure. I was worried that I couldn’t find a staple gun that could get right into the corner of this lip, but in the end I found that an ordinary office stapler could do the job, which was fine. At the bottom of the enclosure, there are drawstrings which are pulled tight and tied off. The fabric stretches, so it forms a very flat covering.

The coverings are in pretty good condition for speakers over 37 years old, with just a couple of snags and small holes. They have faded from black to a very dark blue over the years which is only obvious if any of the non-faded material becomes visible through any slight misalignment. New coverings could be made in a variety of colours, but I think it would be preferable to retain the moderately coarse texture of the original material if possible.

Something that seems to have been an irritation to the previous owner is that the tops of the speakers are capped off with a square of hardboard covered with fabric, and over time these have warped, with the corners rising slightly. These have been re-applied by the previous owner using No-More-Nails or similar, to no avail.

In the end, I restored the fabric caps by carefully removing the material from the original hardboard, and stretching them over sheets of black-sprayed 2mm aluminium, fastening them to it with carpet tape. These now fasten to the tops of the speakers using Velcro. They look really good. I gave the pieces of cloth a rinse in warm water because they were looking a bit grotty, having taken the brunt of spilled drinks at parties over the years I imagine. It looks to me that it would be possible to give the whole fabric sock a proper wash, and it would survive OK.

I resprayed the wooden plinths with black satin paint, and the same for the 1970s speaker stands with casters.

I also decided to try the option of some of the original ‘inverted mushroom’ stands. I bought some of Mk IV ‘donor’ speakers, but unfortunately had to do a bit of metalwork to make these work with the Mk III. They now look ‘the business’, and I am very much enjoying their sound.

Scalford Hall 2017


I took my KEF Concords to the HiFiWigwam show in March. The room may have been smaller than last year’s, and I didn’t think the speakers sounded as good as they might. Nevertheless I found a few nice comments on the web.

Generally this year all the best sounding systems were active, with a system by “looper” which cost less than £300 with cheap drivers and FIR filters playing through a AV amp – leaving most high end speakers systems at the show in its wake.

Big shout out to Looper in room 232 who proved you don’t have to spend thousands to enjoy decent hifi. 

Looper in 232 had a similar home made set of filters driving the rebuilt Concords.  Better sounding than they were in 1975 I’m sure.  It was great talking to him about his thoughts and design ideas, and he managed it on a very tight budget.

All rooms were great but the things that got me going back for more –

…- Looper’s lovely KEF concord III

I was able to do the same ‘party trick’ as my other DIY system, where I showed that changing the crossover frequencies and slopes in real time – even quite drastically – had virtually no audible effect on the sound. If, however, we listened to, say, the mid range driver in isolation, the change was plainly obvious. I would say that this was what you would expect from a correctly set up system with not-too-bad dispersion characteristics, but most people had never encountered it before. The ‘trick’ depends on all the filters being calculated and implemented on-the-fly, and the fixed driver correction and in-room EQ being implemented as separate layers that are overlaid on the crossover filters. I think this demonstration is a kind of sanity check that your setup is somewhere near where it needs to be.

I can’t say why this is considered so unusual, but the fact that it is could go part of the way in explaining why most speakers sound a bit ‘odd’, and why speaker design is considered to be a mysterious art rather than a very straightforward procedure. It seems probable that the results would not be as transparent with a two-way speaker as a three-way because this introduces several new factors into the equation: significant driver beaming – a phenomenon that cannot really be corrected or neutralised – and issues with the drivers having to cover wider frequency ranges. Also, non-linear phase crossovers introduce “phase rotations” through the crossover.

[Last edited 30/06/17]



What should we be listening for?

As you may have seen, I built my own audio system because I had never before heard a system which took the seemingly obvious steps of using large sealed woofers, time alignment, DSP crossovers, driver correction etc., and I wanted to know what it sounded like. I also wanted to experiment with various aspects of crossover design (although I ended up doing less of this than I expected), and to understand what is important versus what is myth. Maybe DEQX, Kii, B&O or Meridian could have sold me something that sounded good, but it would have been an expensive black box that I couldn’t tinker with.

What have I learned from listening to my DIY system? I think the following:

There is a superficial ‘hi-fi’ sound that is achieved by many conventional systems – and I have owned some of these systems. The frequency response is balanced. Harmonic distortion is low – it sounds ‘clean’ at moderate volumes – it has bass and top end in seemingly generous amounts. It is certainly ‘stereo’ as you can clearly hear different things coming from the left, right and middle. If you start to turn up the volume, it does begin to sound somewhat ‘loud’ and ragged, and it can also feel as though the sound is being extruded through an opening that’s slightly too small. After a long listening session at high volumes your ears feel quite ‘sore’, but you are reasonably satisfied with the sound. This, presumably, is how recordings must really sound – you certainly can’t put your finger on anything that isn’t a reasonable facsimile of what it is supposed to be.

And then there are other, more specialised systems which cost much more to buy, and take us into the realms of audiophilia. They’re often impressive to look at, but I think they can sometimes sacrifice “high fidelity” in order to indulge their creator’s interest in a particular material or ‘retro’ technology. You may disagree.

There should be the possibility of a system that just implements the obvious pragmatic steps necessary to get the recorded waveform out of the speakers reasonably accurately (however we define that). There aren’t many of these about, it seems to me. I have heard only one such system – the one I put together using cheap off-the-shelf parts and DSP ‘glue’ – and I find its sound to be different and, if I may say so, better than conventional systems. Here’s what I think it sounds like:

The first thing that strikes you is that although it is ‘clean’, ‘sweeter’ and less ‘edgy’ than the conventional system, it also has ‘flavour’ and ‘body’ – it sounds just like real music. If it’s a double bass being plucked, you hear the fingers releasing the string, and the sound hits you in the chest; if it’s a hi-hat cymbal being struck, you hear the stick meeting the cymbal in delicate detail; if the sound is being made by something heavy and solid, or wooden and hollow, you picture the object making the sound. A part of the dynamism of the sound is how quickly it stops, as well as how quickly it starts. Bass is ‘real’ and not just a ‘rumble generator’; there is no arbitrary limit on how low the bass can go – not that you analyse the sound in those terms. It is simply ‘real’.

Next, you notice the imaging, the clear separation of the instruments, and the acoustics. The person singing is in front of you, located at some position in space between or behind the speakers – you feel as if you could reach forward and touch them. If it’s a live acoustic recording, they are in an acoustic space, and you are there too. There is separation between the singer and other instruments spread around the space – if that is how they were recorded. In a studio recording, maybe the vocalist is in a smaller space than your listening room, and you picture them close to the mic in a booth, perhaps, singing towards you. Or if the sound is coming from within a cool, stone cathedral, you picture the cathedral extending beyond the walls of your room. This is definitely the ‘party trick’ of stereo – a compelling, coherent acoustic space that appears as if by magic in thin air.

Then, you realise that you can listen to the recording at its intended volume. The ‘natural’ volume setting for each recording is usually quite apparent, but the system doesn’t mind what volume you set it at. The copious, dynamic bass means that you are not tempted to turn up the level excessively to compensate for a missing part of the spectrum. Your ears don’t ring afterwards, and even after listening for long periods at what would normally be considered high volume, they don’t feel sore. And there is a physical element to loud, natural, dynamic music that generates an excitement all its own.

It doesn’t need training or experience to appreciate these aspects of the sound, but at the same time there is just so much more to hear in the recording. You become more engaged with it; involvement rather than passively observing a superficially pleasant sound wafting over you.

Thinking about it some more, it is obvious that what I am describing is the sound of the recording, not the system. Some would describe this as a “neutral” system, but the mistake would then be to say that the resulting sound is neutral; I think recordings are astonishing if only we get to hear them without an intervening interpretation.

Light entertainment

Here’s a little controversy from the archives of Stereophile magazine.

Stereophile has an interesting policy whereby an equipment reviewer writes up his subjective experience of testing a device, and only then is it measured for distortion, frequency response and so on. It seems that the magazine has the integrity to publish the two reports whatever the outcome.

Have you ever seen a more polarised review than this one from 2005?

The reviewer says:

The CyberLights represent one of the greatest technological breakthroughs in high-performance audio that I have experienced in my audiophile lifetime….

…for the first time in your life you’ll hear no cables whatsoever. When you switch back to any brand of metal conductors, you’ll know you’re hearing cables—because what’s transmitted via CyberLight will be the most gloriously open, coherent, delicate, extended, transparent, pristine sound you’ve ever heard from your system…

The measurements person says:

If this review were of a conventional product, I would dismiss it as being broken. …I really don’t see how the CyberLight P2A and Wave cables can be recommended. I am puzzled that Harmonic Technology, which makes good-sounding, reasonably priced conventional cables, would risk their reputation with something as technically flawed as the CyberLight.

You’ll have to read the full review for yourself, because the contrast between the two opinions is almost comical. The measurements are quite something to behold.

You see, I sometimes worry that perhaps I just don’t ‘get’ this hi-fi business. £80,000 analogue systems don’t sound anything special to me. Vinyl doesn’t sound as good as digital to my ears but everyone else says it is much better. Designing and building my own system was really quite straightforward, yet the internet is full of intense discussion about how difficult it is; people spend their entire lives building their own speakers and are never happy with them yet it’s almost three years in and counting, and I haven’t felt motivated to modify mine yet. Are the experts hearing something I am not? Perhaps this review sheds some light on the answer.

Analogue enthusiasts often claim that the signal-modifying effects of whatever product they are listening to actually improve the sound. The usual line is that the indefinable magic of valves and vinyl is down to what those devices add: they are serendipitously restoring something that is supposedly missing from the recording. ‘Poor’ measurements are simply an indication of an harmonious combination of factors that enable the leap from clinical, neutral signal to real music. There is no argument possible against this assertion.

However, in the above review, the writer cannot make that claim. Clearly he has confused high levels of distortion and noise plus extreme frequency response variations as an absence of colouration. For him, replacing metal cables with “light” was all about removing “grunge” and other “well-known problems”. Because of his extreme analogo-philia, I don’t think he actually knew what ‘neutral’ sounded like. When he heard something that was different from anything he had heard before, he automatically assumed that it must be because cables really are the sonic quagmire he thought they were and that the product was doing what he assumed it was designed to do. For once, it actually was a “night and day” difference but his understanding of what he was hearing was 180 degrees wrong. In the scheme of things, it doesn’t really matter, but it reassures me that 99% of the ‘expert’ opinion based on listening is very dubious indeed – I do think there are people out there who would find much to like in a pair of yoghurt pots linked with string as long as they cost enough.

Stereophile, it appears, doesn’t normally measure cables when they are reviewed. I think we can guess why: there is nothing to measure. Each and every review would feature the same distortion and noise measurements at the very lowest depths of the test equipment’s range, plus a ruler-flat frequency response when using the cable in normal circumstances. It wouldn’t matter if the cable cost £1 or £10,000 – which, absurdly, they sometimes do. To arrange anything different would actually be quite difficult. It is this complete, boring neutrality that Michael Fremer and other cable mythologisers are convinced is plagued with “grunge” and other problems. The justification for the Cyberlight product, so appealing to Fremer, is that it replaces a short section of metal with light and fibre optics, and is analogue – you still connect to the input and output with those awful grungy wires. It is no different from becoming excited about the audio quality of headphones that use an analogue wireless link rather than a cable. Just as with those headphones, there is a little “background hiss” but this is a small price to pay, apparently. And just like those headphones, the signal goes through a link of dubious quality. Very dubious. At least there is a valid justification for wireless headphones, though.

If you gave me about £20 to buy a few parts, I could build you this device in an afternoon, probably. But if I did, I would try to make it work properly. I would certainly try to convince you that the whole product was unnecessary and was corrupting the signal, and that if we really had to use fibre optics we should digitise the signal and send it as pulses. I might also point out that the commercial product is a mess: various “wall warts”, $400 battery packs and “pigtails” that could, depending on what equipment you’re using, destroy your speakers.

And don’t ever unplug or plug in the power to the cables with the amplifier turned on or you’ll send a horrendous THUMP through your system.

For people who might dismiss active speakers and DSP as too complex, there are no limits to the Heath Robinson-esqueness that they can tolerate in the name of ‘analogue’.

Handmade electronics

Seen on a forum elsewhere: someone promoting a new ‘DAC’ based on multiple, now-obsolete, consumer-grade integrated circuits cobbled together in series and parallel (I think) using a large circuit board that strongly resembles the sort of thing I used to make in the 1980s. Double-sided rather than using power planes, you can see the multiple power and ground busses running around the board as relatively puny tracks. To any experienced printed circuit board designer, its appearance is literally offensive, presumably designed using a computer but looking as though created with self-adhesive tape and transfers. Integrated circuits in sockets, which is what people do when they’re not quite sure if they might need to replace blown-up chips or are worried about their soldering skills, resulting in multiple cheap contacts in the signal path which is kind of ridiculous when the purchasers are then going to be using multi-thousand dollar ‘audiophile interconnects’.

You may think I am being very unkind, but get this: the manufacturer wants in excess of £50,000 for it!

I always find it interesting when the producers of these devices provide close-up photographs of their efforts. This again takes me back to my teenage years, where I used to be very proud of my own early electronic assemblies and would photograph them in great detail. It means that sceptical people like me can pore over the photos thinking “Oh yes, I used those terminal blocks in that burglar alarm I once made because they were so cheap” and “Look how he has spliced two wires together and covered it with heatshrink sleeving. And what is that extra wire for?” It also brings back memories of the times when, in my ignorance, I ran into trouble with this kind of construction and would probe around with ground wires attempting to reduce hum loops or noise caused by cross-contamination between the digital and analogue sections. Occasionally I found a connection that would reduce the noise a bit. When this happened I would solder the wire in place!

The asking price is, according to contributors to the forum, justified because not many of this particular product will be made. This is one of my pet hates: assuming that because something is “handmade” it must therefore be better than something churned out by the thousands. I am not even sure it is true for things like furniture or musical instruments, but it is most assuredly untrue for electronics where instead of “hand made” we should be thinking “prototype” or “cobbled together”. On whether the electronic design itself is sound… I couldn’t possibly comment. All I know is that my teenage ‘wannabe’ designs were pretty atrocious and they looked remarkably similar to these photographs. It is very easy to knock something together that ‘works’, but how immune is it to radio frequency interference? Would electrostatic discharge (ESD) damage it, or make it go haywire? Does it produce an almighty ‘thump’ when powering on or a horrible squeal when powering down?  What happens if one of the flimsy wires breaks off? What if there’s a mains ‘brown-out’ – will it blow up the speakers?

And that price. It looks kind of typical in the context of certain audio forums, but just consider what it means. If I were considering having an extension built onto my house, it would probably be of that order of cost. It would involve professional architects, planners, builders. Lots of equipment would be needed, and a lot of materials. And a heck of a lot of labour. Or, I could splurge the cash on some cruddy circuit boards of sub-hobby level quality. Please tell me that no one would ever dream of doing that.

Active crossover running on fanless PC

sumvision cyclone

I bought a Sumvision Cyclone Mini PC for experimenting with running my active crossover software on a fanless PC. It’s no more than a tablet in a box, but it’s quad core and runs 64 bit Linux on an Intel Atom Bay Trail chipset and, presumably, can perform GFLOPS without dissipating more than few watts – that’s really quite amazing but it’s so easy to take such things for granted these days! It comes pre-loaded with Windows 8.1 and it was a pain to make it work with Linux. I relied heavily on a guide on the internet – thanks to the person who provided it.

Undoubtedly it will be much easier to install Ubuntu on one of these PCs in the future when the Linux people have caught up with the hardware. The WiFi doesn’t work yet so I am using a USB dongle, nor does the on-board audio but I am using the Asus Xonar U7 for that, anyway. Interestingly (to me anyway) I was able to remove Pulse Audio (I think) from this version of Ubuntu without affecting the other system settings [I think this was a fluke: removing pulseaudio properly is impossible, and what I really should do is merely set “autospawn=no” in /etc/pulse/client/conf and reboot].

Absurdly, once Linux was installed following the guide, it worked straight away with the Xonar U7 and my crossover software, plus 64 bit Spotify.

I am assuming I could plug in a common or garden USB DVD drive for playing CDs without a problem [tried this and it works fine].

While running the active crossover software and streaming from Spotify, according to psensor the core temperatures are stabilised at about 56-60 degrees C in an ambient room temperature of 22 degrees C, and overall CPU usage is about 18%.

UPDATE 13/09/15

Things haven’t worked out quite as smoothly as I thought: on the fanless PC I have been getting occasional glitches in the audio, in the form of a click audible within the music, perhaps once every 10 minutes on average. These don’t occur in silent sections of the music so I am assuming that it is a case of missing, or extra, samples rather than corrupted samples. I didn’t notice this when running the code on a Pentium IV based desktop machine.

As a result, I have made major changes to the software, reducing the number of threads from three to one (plus a default thread for the GUI – which is currently just a mute checkbox for each driver). There are suggestions on the web that the ALSA functions are not ‘thread safe’. So now, all the ALSA audio and DSP processing runs in a single thread and all ALSA calls are non-blocking. This arrangement dispenses with the necessity to lock various circular buffer pointers with mutexes when accessing them, so the code is now more stripped back and simpler to understand.

The main motivation for multi-threading originally was that I assumed that the OS would assign threads to different cores, so for coolest running it would be best to share the computation load across several threads. Therefore I expected to see the CPU load on one of the cores go up as a result of combining three threads into one, but it doesn’t seem to have greatly affected the CPU load traces, nor the core temperatures.

No glitches so far.

[UPDATE 29/10/15]

Still had the glitch problem! It wasn’t happening on a P4 desktop minitower PC, but on the Sumvision I might get a glitch once in ten minutes. Nothing drastic, but I found myself on edge waiting for it. Really, any glitches are unacceptable, even if only one every three hours.

Is it related to input or output? As an experiment I modified the code to stream the incoming audio to a file while playing music. When I heard a glitch I noted down the time it occurred. Examining the data in the audio editor app Audacity I found a discontinuity in the waveform. Gotcha! In order to test for this problem reliably I created an audio file containing a continuously-repeating ramp waveform. In my program I added a check on consecutive samples to flag up any discontinuities. Sure enough, the problem only occurred occasionally, but it always happened eventually. Playing with threads etc. didn’t get rid of it.

In desperation I started to look at the open source code for the snd-aloop driver I am using as my bridge between audio player apps and my code. I found a mysterious system whereby there are separate ‘rate shifts’ (the programmable sample rate I am relying on in my code) for playback and capture. I don’t really understand this: unless playback and capture are locked together (at least on average), it seems to me that they must eventually diverge and cause audio discontinuities. I bodged the snd-aloop source code in order to precisely lock together the playback and capture ‘deltas’. This sort of thing is outside my comfort zone. I had to re-compile the snd-aloop driver and use the Linux command insmod to load it into the system.

It worked. I now get zero errors no matter how long the system is running. The difference between the Sumvision and the old P4 may be explained by the fact that the PCs’ clocks were quite a bit different, and much more rate shift was necessary in the Sumvision in order to synchronise with the sound card.

I still think I am making this harder than it needs to be. Do I need the snd-aloop driver at all? Can it all be done with ALSA plugins? One Linux guru said I should write my program as a plugin itself. It has occurred to me that at least I can now modify snd-aloop in order to make it work as I want: not with its own sample rate at all, but merely as a relay from the capture demand to the playback demand.

But the bottom line is that the system is now working perfectly on the Sumvision Cyclone.

02/12/16: It seems that there will be no ‘official’ version of Ubuntu for the Bay Trail and Cherry Trail chipsets, but someone called Ian Morrison (a.k.a. “Linuxium”) has created an installer and very kindly made it available. I found that his version of Ubuntu 16.04 seemed not to boot on the Sumvision Cyclone, but 16.10 appears to be fine. I haven’t transferred my software over to it yet but my second Sumvision Cyclone appears to be working fine, with Wi-Fi. Many thanks to Ian for this. I wouldn’t know where to start in creating such a thing.

UPDATE 30/01/16: I just spent quite a large proportion of my Christmas break worrying about, and trying to fix, an issue that arose after I put the latest version of Ubuntu (mentioned above) onto a Sumvision Cyclone and installed my active crossover software. Glitches were back!

I cannot tell you how many fruitless attempts I made to solve it. I narrowed it down to the DAC output, where some zero-value samples were being substituted in the analogue output, but with the overall timing remaining correct. There were no EPIPE (buffer underrun) errors.

I created a ‘glitch detector’ where I generated a waveform from the DAC’s analogue output and fed it via a cable into the microphone input, looking for excessive sample-to-sample amplitude changes. Glitches would always occur, but it seemed worse when loading web pages etc.

Finally, I think I hit upon the solution in this forum topic:

Problem with Bay Trail and new kernels

It seems that recent versions of the Linux kernel have changed something regarding ‘C-states’, related to the way the processor cores are dynamically put into low power modes when idle in order to reduce average power consumption. With the new, more aggressive, power saving, they take longer to start up again (flushing pipelines etc.), and this has been causing Bay Trail setups to freeze completely. It is still being discussed as a live issue on Intel forums. I think I have been suffering from another side effect of this misguided change.

There is a workaround, which is to specify a boot option to keep the cores relatively ‘alive’ at all times. (There may also be a BIOS setting that I could have changed, too). It seems to have fixed my problem completely.

If this turns out to be the issue, it highlights the fragile nature of any IT-based product. An innocuous update in the operating system can kill your product because of real time issues; there is no amount of testing that can be done by the OS people that can eliminate the potential for problems in users’ own applications.

At one time, I naively thought that it was possible to put together an embedded PC-based system that could be ‘frozen’ and would always work, and could always be duplicated, but I have long given up hope on that. Embarking on any digital audio scheme based on a PC implies a commitment to constant maintenance, in a way no different from the constant maintenance you commit to when using, say, reel-to-reel tape recorders.