KEF Concord in print

I just noticed that Ken Kessler’s lavish book on the history of KEF contains several pages on the Concord – the speaker I have been re-building in active form. He makes it sound like a much better speaker than I found it to be prior to conversion, but maybe I just had a bad pair – I’m being diplomatic.

The mark IV version looked subtly cheaper and less sophisticated than the III due to small details like the badge, base plinth which was now plastic(?) and the texture of the all-round fabric. It had a removable plastic cap on the top of the enclosure, and it seems that this was to allow users to change the ‘sock’ for different colours, although no one ever bought anything but black and brown, leaving warehouses full of the other colours – how I would love to have some of them now!

There’s also a story of one of the bosses getting his wife to try one on as a boob tube…

Vinyl in Space

jack white

Jack White aims to play the first vinyl record in space.

From The Guardian:

With the aid of a ‘space-proof’ turntable and high-altitude balloon, the singer’s Third Man Records will try to beam Carl Sagan’s A Glorious Dawn from orbit…

The Guardian link writers came up with

The Vinyl Frontier

Now that’s quite good…!

The musical ‘observer effect’

In scientific audio circles, it is believed that if you are aware (or think you are aware) of what hardware you are listening to, then you are incapable of any sort of objective assessment of its quality. This leads to the blind listening test being held up as the Gold Standard for audio science.

But here’s an irony: almost everything of value that man creates comes into being through a process of ‘sighted’ creation and refinement – and it seems to work. Bridges are designed by architects who refine CAD models on a screen, but the finished products don’t fall down, and are admired by ordinary people for their appearance. Car bodies are designed by engineers and stylists in full sight, yet the holes line up with the rest of the car, and they achieve great measurements for aerodynamics and the cars look good as well. Pianos are tuned by people who know which way they are turning the lever as they listen.

So if ‘sighted-ness’ leads to a completely fictitious, imaginary perception, then presumably our pianos are not really in tune, but we imagine they are? Maybe everyone but the piano tuner would hear an out-of-tune cacophony when the piano is played? But no, it turns out that everyone, including the piano tuner, can tell consistently when a piano is in tune without resorting to blind tests, and this can be confirmed with measurements.

So how come ‘sightedness’ is so problematic for the creation or assessment of audio equipment? I think that the question is “not even wrong”. The faulty logic lies in the erroneous idea that audio equipment is being listened to, as opposed to through, and that the human brain when listening to music is similar to a microphone. There is no reason to believe this at all; to me, it is just as likely that the brain is acting as an acquirer and interpreter of symbols. The quality of the sound is part of the symbol’s meaning, but cannot be examined in isolation.

As a result, it may just be that there is no way for a human listener to reliably discern anything but the most obvious audio differences in A/B/X listening tests. Using real music, the listener may be perceiving sound quality differences as changes in the perceived meaning of the symbols, but repeated listenings (like reading a phrase over and over), or listening to extracts out of context, kills all meaning and therefore kills any discernment of sound quality. Consciously listening for differences as opposed to listening to the music, pressing buttons while listening, breaking the flow of the music in any way, all have a similar effect. Alternatively, using electronic bleeps, or randomised snippets as the ‘test signal’, the listener is effectively hearing a stream of noise without any context or meaning, so the brain has nothing to attach the sound quality to at all.

In effect, the act of listening for sound quality in scientific trials may kill our ability to discern sound quality. Can this be proved either way? No.

I don’t see this as a problem to be ‘solved’; it is simply the kind of paradox that pops up when you start thinking about consciousness. Music has no evolutionary survival value, but we enjoy listening to it anyway – so we are in Weirdsville already. The extreme ‘objectivists’ who hold up ABX testing as science are extremely unimaginative if they think their naïve experiments and dull statistical formulae are a match for human consciousness.

Within the limitations of their chosen technology, most hi-fi systems are created with the aim of being ‘transparent’ to levels that exceed the known limitations of the physiology of the ear, and people seem keen to buy them. Without referring to scientific listening test data, the customers know that, in normal use, proper hi-fi does sound better than an iPod dock with 2″ speaker. But, as their own preference for the sound can’t be proved scientifically because of ‘the observer effect’, and because a human is bound to be influenced by factors other than the sound, then at some level they have to buy their hi-fi equipment ‘on faith’; maybe being influenced by the look of it, or because they believe the meme that vinyl is superior to digital. So be it. But they may find that, later, the system fails to meet their expectations and they are on a ruinous treadmill of “tweaks” and “upgrades”.

On a strictly rational basis, bypassing all that anguish, the new generation of DSP-based speakers gets even closer to the ideal of transparency by virtue of superior design – no listening tests required. I am confident they will sound great when being used for their intended purpose.

[Last edited 06/08/16]

Does hi-fi end here?

kii transparent

Reports are coming in that hi-fi may, after a century of development, have actually reached its logical conclusion. It is beginning to look as though the Kii Three may be the technology beyond which it simply wouldn’t be worth going, for the vast majority of people. If so, this is quite a significant moment.

Everything up to this point has been a flawed, intermediate step.

It all started in the 19th century with the stunningly simple observation that sound is nothing more than variations of air pressure and that these can be picked up by a diaphragm and reproduced by another diaphragm. The hi-fi story has been one of how best to store the information encoded within the vibrations, and how to get the vibrations back out into the world at some time later.

First, we had purely mechanical systems which had to contend with the imbalance between the tiny amount of energy that can be picked up when making a recording versus the large amount of energy that is needed to play the recording back.

Then, with the introduction of electronics into the equation, the path towards the truly linear system was opened up. We had recording on magnetic tape, distributed to the listeners via vinyl LPs. Amplification with valves, then transistors, Class A, AB and now Class D. Horn speakers, multi-way speakers, direct radiators, acoustic suspension, and detours into panel speakers, electrostatics and even plasma. Interestingly, active crossovers are not new: they were used in cinemas in the 1930s, and there was at least one well-heeled enthusiast using them in a domestic system in the 1950s.

A major disruption occurred with the development of digital audio in the 1980s which, at a stroke, propelled performance in terms of noise, distortion and linearity to the point of practical perfection and slashed the size, weight and price of audio storage and playback equipment.

(At this point, ‘high end’ audio as a hobby left the rails and, for many, became an exercise in masochism, superstition and nostalgia).

The next part of the puzzle was solved when computing power became available. Using a computer it is possible to perform digital signal processing (DSP), allowing precise tailoring of crossovers and EQ, and for the characteristics of mechanical transducers (the speaker drivers in their boxes) to be modified.

The linear system

Now, all the pieces were in place to build a linear reproduction system using the following building blocks:

  • Digital storage of stereo or multichannel recording
  • DSP to process the signal for crossover, time alignment between drivers, driver amplitude and phase correction, EQ, woofer distortion correction using voice coil current or motion feedback
  • One DAC per driver
  • One solid state amplifier per driver
  • Loudspeaker comprising several dynamic drivers each allocated to a narrow frequency range, including sealed woofer whose bass can, if necessary, be extended using DSP EQ.

This is all perfectly realisable at low cost using physically small electronics. The advent of Class D amplification makes it even smaller and cheaper. Such a system is virtually noiseless, has extremely low levels of distortion and covers the entire human hearing frequency range.

The final part of the puzzle

There has been a lag in the acceptance of such systems even though they are spectacularly good. The recent development of a system to tackle directly the issue of the speaker’s interaction with the room at bass frequencies may be the final part of the puzzle that means these systems take off. I think the Kii Three is the first speaker to do this using DSP, followed closely behind by the huge and expensive Beolab 90.

There is some confusion over why DSP-based ‘room correction’ is needed, and what it is capable of. Although the room appears to mangle the signal terribly in terms of frequency response and phase when measured, the listener hears the direct sound from the speaker first, and an average room just adds agreeable ‘ambience’ that blends the immediate surroundings with the recording and helps to cement a convincing illusion of ‘being there’. Trying to ‘correct’ the effects of the room will make the system sound worse.

The one area where genuine problems may occur, however, is in the bass, and people attempt to solve this with DSP (not very successfully), and with room treatments (not particularly effective for the bass). The Kii Three and Beolab 90 both take the approach of using extra drivers driven by DSP to make the speaker more directional at low frequencies by cancelling out some of the almost omnidirectional bass that comes from the main driver, at the sides and rear. This effectively provides the same directionality as a huge baffle, but from a compact speaker.

Intuitively, it seems obvious that in a highly reflective, echoey room, this technique would improve the clarity of what was heard. It would also tackle problems of speaker placement near walls and corners. The amount of bass bouncing around the room is being reduced at source, rather than trying to catch it afterwards with bass traps etc. The result, apparently, is spectacularly good.

By all accounts, the Kii Three is a compact, good looking speaker with a moderate (OK, not outrageous) price, that simply disappears acoustically, leaving the music as a solid 3D image. It is loud enough and goes deep enough to satisfy the vast majority of people. No other equipment is needed other than a digital source, which could be a PC, streamer or network.

The search is, apparently, over. While it would be possible to build a bigger system, with bigger drivers, higher powered amps and so on, this would just be scaling the same fundamental design. This has already been done in the form of the Beolab 90. The system could be further scaled to provide more channels than just stereo, and more precise control of dispersion in the vertical as well as the horizontal – if anyone thought it necessary.

Conclusion

In the end, it turned out that the ‘objectivists’ were basically right: you really do just need perfect linearity to build the perfect hi-fi system (but you also have to have accuracy in the time domain, which most audio objectivists ignore).

According to reviews, and based on my own experience of not completely dissimilar DIY systems, the Kii Three is the only hi-fi system anyone will ever need. Valves, vinyl and passive crossovers seem positively quaint in comparison; ‘high tech’ passive speaker systems seem almost perverse. No doubt the Kii Three will be copied, and cheaper versions will appear, but there is no need to fundamentally change the design from now on. It should be game over for other forms of hi-fi. (It won’t be, of course!)

Thoughts on creating stuff

IMG_0206

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 35 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.

IMG_0208

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.

KEF_ConcordIII_lf1_610x449_pixels

Some KEF Concord IIIs (not my particular pair) [nrpavs.co.nz]

I bought the KEFs a few years ago because I thought they looked fabulous, and 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.

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 (it won’t matter how it looks when covered with the ‘sock’), I fired up one speaker this evening to have a quick listen using slightly modified settings from my big system. I found it really interesting and encouraging.

concord

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 are going to sound spot on. The bass is not as deep as my big system (obviously) but reasonably adequate. Nevertheless, as an experiment, I tweaked the code to boost the bass in order to emulate a bigger driver and box. I am able to mute the individual drivers and so was able to listen to the woofer in isolation. The driver is clearly struggling! And so is the box (or is it a nearby piece of furniture ratting?). At (excessively?) higher levels I can hear some distortion. These bass drivers really were the cheapest possible, though. The enclosures are made from mere 15 mm chipboard and need bracing. So more work needed, but I will probably persevere because I really like the look of them (when covered). I imagine some reasonable woofers won’t cost much.

But the experiment possibly reinforces my notion that reducing the size of a speaker by a few percent really does make the job of getting it to work properly that much harder. If you want something that sounds astounding, without having to spend much money or effort, make it big.


Day 2

Perhaps I was being a little too ambitious yesterday – trying to get too much bass out of the speaker. Also, I found a couple more existing holes in the box, and have plugged them with hot melt glue. I am pretty sure the seal is now good.

I have removed any bass EQ, and it sounds pretty good. Even though the settings are just guesswork variations on the ones for my big system, it is proper ‘hi-fi’. If I listen to the woofer in isolation, I can still tell it’s not absolutely top notch, but I may just not worry about it. Once I have the other speaker, I will be putting less power into it anyway.

The frustrating thing is that it will be a few days until I can do the other speaker, and hear them together…


Day 3 (elapsed time)

The second speaker has now been built up and I have endeavoured to set them up slightly more scientifically than before. I have measured them (woofer near field, and mid and tweeter far field ‘pseudo-anechoically’) and am applying roughly the appropriate correction to each driver (phase and frequency response, delay, gain) – but I still have ideas for refining this further. I have also re-implemented the EQ for the bass to aim for the same response as my big system(!). And you know what? It sounds fine! The bass really is there – not as ‘authoritative’ as my big system, but very deep nonetheless – I wish you could hear it. If the idea is that sealed woofers don’t go as deep as bass reflex or that an 8″ driver isn’t all that impressive, this might confound that notion. The cones do move quite a lot, though, and I know there is scope for better drivers.

‘Imaging’ is good, but I just don’t think that is all that difficult for an active three-way system. If I hadn’t heard my bigger system, I would think that these speakers were excellent – which is kind of interesting if you consider the way they have come into being.

[I just roughly re-fitted the fabric covers and put the speakers on their plinths. In dim light, they look like they will when finished properly. Of course they sound even better – you know it’s true…]

There is still scope for adding a foam anti-diffraction surround for the tweeters such as were fitted to the original speaker but have long since crumbled to dust (frequent hoovering was necessary when dismantling the speakers!). The enclosures could certainly do with some bracing, too.

And I still may try some different/better 8″ woofers.

I don’t actually know what I’m going to do with these speakers. If we ever build a house extension, I quite fancy a ’70s aesthetic’ and these speakers might even be allowed into the living room by the missus. But as an experiment they are very interesting. With my first ever speakers I think I hit the jackpot simply by making them big enough, with DSP active linear phase crossovers, three-way configuration and sealed woofers i.e. all the obvious, pragmatic features. Maybe the combination of wide baffle for the bass and narrower baffle for mid and tweeter also had something to do with the great sound. With these speakers I have made things slightly more difficult to start with which makes it more interesting perhaps.


Day 4

Have raised the bass roll-off a bit (aiming for -3dB at 45 Hz, -12dB per octave) and temporarily added some pieces of foam around the baffle surround around the tweeter and mid. Crossover frequencies are 325 Hz and 4 kHz, both 4th order (actual filters are modified with driver correction). Now just listening to them, attempting to forget why I’m listening to them. And they’re pretty good!


At the next opportunity, I will maybe try replacing the woofers with the original driver as an experiment. An irritation is that I have filled up the original drivers’ mounting holes with glue.

[01/06/16] Just tried it, and found that they weren’t very impressive.


29/06/16 New bass drivers added. These are much better than the Peerless drivers. Corner frequency dropped to 38 Hz, and sounding good.


01/07/16 They’re sounding great.


15/07/16 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.

IMG_0201

Has 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.


25/07/16 Restored the fabric ‘socks’ and they look good and wrinkle-free, much to my surprise. I really am quite excited by the fact that I may actually finish something for once.

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 36 years old, with just a couple of snags and small holes. The main problem is that 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. It was presumably an irritation to KEF as well because the Mark IV used plastic caps instead, but I am not convinced they looked as good. If I’m feeling creative I might try something different with black painted MDF caps, or aluminium sheet covered in fabric.

[Last edited 30/07/16]

An audio breakthrough

It would appear that there is a particular audiophile DAC with a cult following that gets rave reviews and costs over $2000, and is based on a non-audio DAC chip.

Why would they do that? Well, I think it is so they can run it “NOS” (not New Old Stock, but “non-oversampled”) and add their own “proprietary” filtering – plus it’s different from what the hoi polloi uses so it must be better. But, it would appear that someone has found a glitch, literally.

I am no expert, but I think that because this chip is a non-audio DAC, the output comes directly from a R-2R ladder, or similar. Small capacitive charges are transferred whenever the ladder switches operate, and sometimes the switches don’t all operate at the same speed. This means there is a glitch at the output whenever the DAC value changes, and it is worst when all the switches operate simultaneously i.e. when the most significant bit changes – around the mid range in other words (hmm…). Presumably there are other significant glitches at multiples of 1/4 full scale and 1/8 full scale too.

Low pass filtering the output can reduce the amplitude of the glitch at the expense of increasing the settling time. There are better techniques using a further piece of circuitry (a sample-and-hold) but, apparently, for the designers this was regarded as unacceptable for some reason (why?), and at audio frequencies still wouldn’t be as good as a typical $1 audio DAC in a mobile phone.

The evidence is all in the DAC chip’s data sheet:

glitch

I don’t know whether the glitch energy scales with the the VREF (i.e. the full scale signal amplitude), but this glitch is huge compared to the smallest signals that we might generate with the DAC.

An owner of this product now thinks he is hearing a certain harshness in the sound, and seems to have found that when reproducing a sine wave at -90dBFS, the output of the $2000 DAC contains significant glitches at the zero crossings. It would be interesting to know if there are detectable glitches at 1/4 and 1/8 full scale, too. This could be the phenomenon shown in the data sheet, or a by-product of whatever mechanism is being used, unsuccessfully, to suppress the glitches – they are rumoured to be using a combination of two DAC chips. Scrutiny of other reviews and measurements of the device seems to reveal distortion and noise figures that suggest something strange is going on – apparently.

An aspect of integrated circuit DACs is that because they are very small and constructed on a single chip, they have fantastic performance relative to themselves i.e. they remain monotonic and linear at all times. However, their absolute gain and offset may drift slightly with temperature. These temperature coefficients vary from chip to chip and can even be positive for one chip and negative for another (this appears to be the case for this particular DAC chip according to the data sheet). This means that any attempt to blend the outputs of two DAC chips externally using a combination of scaling, offsetting, inverting, mixing and interleaving would be most unlikely to succeed down at the lowest levels.

If these suppositions are correct, then this product is a great example of where the basic engineering of a basic product appears to have been sacrificed in the interests of just making something ‘different’ and supposedly ‘simpler’ – although as usual it ends up being more complicated.

[Last edited 04/05/16]