Taking the B&W 801Fs active

Part 1.

This is my first post on the Audiosmile Forum, although I’ve been quite active elsewhere. This series of posts was prompted by Simon who suggested that I post here as my experiences might be of interest to others. This is a long post so I’ve split it into several parts. Please feel free to comment.


I’ve been a user of active ‘speakers since the mid ‘80s as eliminating the high-level passive crossover and driving each loudspeaker with its own amplifier has always seems a far more sensible approach.

I had Meridian ‘speakers since the 80s, firstly the M2s, then DSP5000 supplemented by DSP1500 subwoofers. However, after some 13 years with the DSP5000s, I wanted to try something different, especially a bit of DIY. The loudspeaker I had always admired was the B&W 801. I heard these at various studios, and always thought the best sound I’d ever heard came from a pair of 801s in Danmark Radio’s IEC listening rooms.

To cut a long story short, I bought a pair of 801Fs, which were the second series, still with a sealed bass bin rather than the later reflex Matrix series, but with the improved Fibrecrete head rather than wood (hence the 801F). I planned from the start to make them active, given that if they’re that good passive, they should be rather better (or at least no worse) active. Using a DSP crossover with DSP parametric and graphic equalisation, I should be able to set them up to a high level of accuracy. Having had bass-reflex speakers for 20+ years, and Transmission Lines (IMF TLS50II) before that, I was quite keen to try sealed box bass for a change.

I got them home and listened to them as they were, driven by an old Yamaha CR1000 receiver, and compared them with the Meridians, but took no measurements yet. They did some things better than the Meridians, like vocal clarity, and the bass was different ( A single large sealed box rather than bass-reflex) but I couldn’t work out if any better, just different. Anyway, I was happy that they were both working correctly, given that they were 30 years old, and had come originally from Decca Studios so would have had some considerable usage.



I had previously decided to use the Behringer DCX2496 as a crossover, supplemented by the DEQ2496 Digital EQ to provide fine equalisation control. I got the DCX on order (I already had the DEQ) and looked for power amps. I needed 6 channels, and decided that to provide the necessary headroom, I would like 100 watts per channel as that would provide some 105dBSPL capability, given that the passive system was rated at 85dBSPL/watt. In fact, with active systems, the midrange and treble are often more efficient that the bass unit, and with music energy being predominately bass-heavy, I could have used smaller amplifiers for the midrange and top, but decided to keep all the amps the same. I looked around at new and second-hand, thinking in terms perhaps of three Quad 405s, but decided I could get three brand-new Behringer A500 power amps for rather less money than three used Quads, and get balanced ins, power metering and a warranty. I ordered the three A500s and started on the project.

The Behringer A500 is very cheap, around £150 for a 130+watt stereo amplifier, so it might be thought of as being hopelessly compromised or flawed. Not a bit of it! It’s well made inside, and whilst the performance is hardly state-of-the-art, and its published spec is somewhat optimistic, it is transparent by any objective measure. All three amplifiers measured identically, so consistency was good! I got close to 30 volts out into 8 ohms at the onset of clipping, which is 115watts with <0.06% distortion at both 1kHz and 10kHz. Of greater interest was the distortion at 1Watt, which is the level I’m much more likely to be listening at for most of the time. Here it was 0.07% at both frequencies, albeit with a bit of crossover distortion visible. Whilst I would have preferred no visible crossover distortion, at these levels it’s nothing to worry about and at the price, I couldn’t build myself anything better. As I have often said, adequate i.e. transparent, is all I need and the A500 is perfectly adequate. Due to a minor fault (low batteries), my generator was giving out 0.05% distortion, so I can expect the amplifier distortion to have been lower than noted above. One of these days, I might get round to remeasuring the amplifier distortion. Output impedance was a little higher than spec at 0.125 ohms, but that’s also low enough not to be of concern. All in all for around £150, I can’t fault the A500. Noise was within spec at -100dB A weighted, but only -90dB unweighted, more about this later.

The DCX crossover is so simple to use, it’s nice when a product has been well designed. All I needed to do was to dial-in the crossover frequencies, which I kept to B&Ws as I’m sure they got that right, and decide on what sort of crossover slopes and type. There’s no good reason I can think of not to use Linkwitz-Riley (Bessel and Butterworth are the other options) and the greatest slope available which is 48dB/octave. The original passive crossovers were 24dB/octave, and that’s about as steep as can be achieved with passive components, and the B&W crossover is pretty complicated at that. As I had the option of 48dB/octave, I decided to use it as it would give better defined bandwidths to each driver, so reducing the out-of-band signals they were sent and therefore reducing distortion, especially intermodulation distortion.

The DCX has both analogue and digital inputs, and 6 analogue outputs. I noticed that if a 0dBFS signal was put in, the analogue output clipped. Reducing the internal gain even by 0.2dB cleared the clipping, so I set the internal gain to -0.5dB. This problem I’ve found remarkably common with DAC products from several manufacturers. I rejected several “pro” ADC-DAC devices as the analogue outputs would clip on 0dBFS. It was irritating, but simple to get round, but I wouldn’t have noticed it if I hadn’t done the measurements.

So, with the amplifiers, crossovers and equalisers available, I could start on the project proper. The first thing was to measure what the 801s were doing passive. I had the original manuals with the individual factory-produced frequency response graphs, but whether it was differences in the measuring methods or 30 years of use, my measurements indicated that the frequency response variations were larger than the original graphs, with a pronounced extreme HF droop after 16kHz (not that I can hear that any more) which was also rather worse on one channel than the other.



The interchannel differences across the frequency range were also rather worse than the 0.25dB originally specced. My self-imposed task was to achieve a frequency response to within +-1dB (above 200Hz) and with a 0.5dB channel balance. Using ARTA impulse measuring software I can get pseudo-anechoic frequency response measurements, but only above 200Hz due to the limitations of my room. To get lower, it would need a much larger room (or better still, open air on a high platform) such that the early reflections take longer to reach the measuring microphone. In any event, below 200Hz the effect of the room will swamp whatever errors the loudspeaker has so as long as these errors are sensibly small, it’s not necessary to equalise LF to the same extent as mid and HF.

I was also expecting that the distortion would be less due to driving each loudspeaker directly rather than through a passive crossover. More on this later.


S.

 

Part 2. Time to start the modification.



The tweeter and midrange are housed in a separate head, which is attached to the base through a compliant coupling. Audio goes into the head from the main passive crossover through a Bulgin three pin connector, carrying Treble hot, mid hot and common cold. I wanted to replace this with a 4-pin XLR to separate the treble and mid cold sides. Not that this would have any sonic effect, but because communing the cold would give huge problems if anyone wanted to use amplifiers with bridged outputs. I wasn’t intending to do so, but for good practice, it was worth doing. I also bypassed the “environmental” controls which adjust the mid and treble level and sensitivity in relation to the bass. These controls, in effects part of the crossover, are unnecessary as I can do all that and more, in DSP.



The main bass bin is stuffed with wool waste. I removed three bags full so that I could have access to the internal crossover, which was removed.



I will keep the crossover in case I should even want to return the ‘speakers to original spec, or I might perhaps put them in a box so I can play active vs. Passive.



Here’s the bass unit. It is mounted with a rubber gasket and bolted through rubber shock mounts. So it’s more-or-less decoupled from the box whilst maintaining a “perfect” air seal. Any air leaks will affect the bass as it relies on the internal air spring as part of the suspension.



A view of the internal bracing of the bass box. B&W obviously thought the bracing was inadequate! As they then went on to Matrix construction, although they also abandoned the sealed box. The 801 Matrix had much better bass extension and sensitivity than my version, but I’ve had bass-reflex ‘speakers for many years and fancied trying a sealed box. You can see at the back the 6 holes for the new loudspeaker terminals.



Finally, a view of the top plate. The five white spots are where the old crossover mounted, and I’ve had to seal up the holes with dowels to maintain air tightness. The holes at the front are where the head mounts, and those are sealed with a rubber gasket which I’ve had to make as the old one was perished. This gasket, which overhangs the head mounting plate, seals the joins and at the same time decoupled the head from the bass box.

Coming next, reassembly and testing, with more pictures and some graphs.

S.

 

Part 3

Part 3.
With the terminals fitted into the holes in the back panel, I wired up the XLR for treble and mid, and wires for the bass unit. I used 15 amp mains cable for the bass unit, but wanted something less stiff for the head unit so used some CAT5 cable. Each twisted pair was soldered together to provide 4 wires with very low resistance, and all in one reasonably flexible sheath. Simple and cheap.

Before refitting the wool and bass unit, I polished one cabinet but it didn’t look a lot better after I finished it, so decided not to do the second one. If they sound good once I’d finished and they were keepers, I would have the boxes reveneered professionally.

Reassembly was the reverse of the strip-down, so I was now ready to fire them up.



With nothing done to the crossover other than dialling in the crossover frequencies and slopes, (380Hz and 3.6kHz, 24dB/Octave Linkwitz-Riley (More about the crossover slopes later), I connected up the three power amps with 16dB attenuators on their inputs so that with the +22dBu maximum output of the DCX when the input is at 0dBFS, the amplifiers can’t clip. 16dB attenuation gives me 110 watts per amp for 0dBFS input, and the amps are good for around 120 watts , so there’s no risk of the amps clipping. Later, as part of the final adjustments, I can set limiters on the treble to around 40 watts as I don’t want to risk a blown tweeter if playing “enthusiastically”...Scalford for example.
The original passive crossover had protection that switched off the ‘speaker if any drive unit was overloaded, and it came on at around 36 watts for the tweeter, much higher for the bass and midrange, but limiting the amps to 110 watts I won’t be exceeding these drivers’ capabilities.

So, how did it sound? Pretty horrible! There was no attempt made yet to level the crossover or match sensitivities of the drivers, so they were squawking away wonderfully. Still, it proved they worked, so the fun of measuring, adjusting and listening begins.
One further thing to address was that of a volume control. Using my analogue pre-amp (originally Meridian 201, now 501) the volume control reduces the analogue level into the ADC. Consequently, most of the time the digital level is somewhere between -40dBFS and -20dBFS. This makes the noise level somewhat higher than ideally I would have liked, but it isn’t audible from the listening position, and the alternative is finding a 6-channel remote volume control that tracks better than 0.1dB.
It turned out that pre-amp noise wasn’t any issue, but power amp noise was. More about this later.
The Measurements:-
All pseudo-anechoic measurements are done by gating the impulse response to remove everything from the first reflection onwards. This means that there are very few individual measurements at low frequencies, and below a certain level, 200Hz in my case, any measurement is meaningless. The maths for this is very heavy, and although I’ve tried to understand it, I can’t say with any honesty that I could explain it, and therefore it’s clear I don’t understand it! Nevertheless, with 2.4m ceilings, the furthest I can get the microphone from a boundary is 1.2m (either the ceiling or the floor) and that limits measurements to above 200Hz.
Below 200Hz, one can do “near field” measurements where the microphone is right up against the diaphragm (within 2mm at most) and the measurements are stitched to the far-field pseudo-anechoic measurements. These near-field measurements only hold for frequencies below a certain value dependent on the size of the driver cone, and by stitching together the far-field and near-field measurements, one can derive the frequency response of the whole system. However, given that especially at low frequencies, the room has a major effect on the perceived frequency response of the loudspeaker , I prefer to get the high bass, mid and top anechoically flat, then adjust the low bass by ear, or use one of the in-room measuring softwares like REW (Room EQ Wizard).
As well as adjusting the frequency response, the DEQ/DCX can correct for time-alignment of the various drive units. Although the 801s have staggered drive units, the measurements mad by “Stereophile” in the 1980s indicate that time-alignment wasn’t that accurate. However, I’m not that convinced that time-alignment is audible. My own tests didn’t show any audible difference when the tweeter was moved back and forth by around 150mm relative to the bass/mid unit of a two-way ‘speaker. Obviously, if the tweeter was a mile away, even accounting for the level loss, that sort of time delay would of course be very audible. OK, so a mile (5 seconds delay) is too long, what about 1 second, or 100mS or 1mS? At what point does it stop being audible? I don’t know, but it doesn’t seem to me to be that important. It might affect a large PA system when the separation between bass bins and tweeter hors is large, but I don’t see it as an issue with domestic HiFi loudspeakers. Anyway, the DCX has an automated time-align function that takes a few seconds to do

Phase shift is another thing that seems to exercise audiophiles, but again, this is not something I think is audible. Different crossovers have 90, 180, 270 or 360 degrees of phase shift at the crossover frequencies depending on type of crossover, and yet that doesn’t appear audible. Furthermore, using an all-pass phase shift circuit doesn’t seem audible.

Before doing any serious measurements, I wanted to get the gain structure right for the whole system, from 201 pre-amp to power amp outputs. I want to have the pre-amp output drive the DEQ’s analogue input as fully as possible so the following ADC doesn’t “waste bits”.

Plugging the output of my 201 (later changed to 501) pre-amp into the DEQ, I played a 0dBFS tone from my Squeezebox Touch and adjusted the pre-amp’s analogue volume control until the DEQ indicated 0dBFS output. This was at a volume level of 59 out of 64, so my usual listening volume 20dB below that would put the volume control comfortably in the middle of the range. The output of the DCX followed by 16dB of attenuation into the power amps gave approximately 30 volts output, so pretty much correct.

As previously mentioned, the frequency response of the system was very middy, indicating that the mid-range unit was a lot more sensitive than the bass or treble. Looking at the old analogue crossover values, it seemed that I would need to back off the mid by 7dB. This was remarkably close. The treble needed to be backed off by 2dB.

I use a Behringer ECM8000 measuring microphone which I have calibrated against a known reference microphone. The ECM8000 is a very inexpensive microphone, and as standard is fine for use in SPL meters or setting up subwoofers in an AV system but not good enough for the sort of accurate measurements I was taking. However, by calibrating it against a known reference microphone, I got the accuracy I need.



I’ll skip all the detail of the measurements, as they were pretty tedious, but by using ARTA software, and tweaking the parametric and graphic equalisers step by step, I got some pretty good results.
I measured the passive ‘speaker first, to get some idea of what my target was, and could I get it better. I struggled for a couple of days using different softwares, ARTA, REW, HolmInpulse and RMAA. Succeeded in thoroughly confusing myself , as all these packages do different things, in different ways, and trying to get consistent answers that make sense just wasn’t happening. In the end, the software that proved most useful was ARTA, once I got the hang of it. I found a tutorial, which helped enormously. I have the manual, but to understand the manual one needs to know how the software works. At first, I wanted to modify one loudspeaker only and keep a standard passive one as a reference, but decided that I might as well do both, and only think about going back to passive if I find I really don’t like the actives.
Before getting the hang of ARTA, I used it incorrectly, and using the EQ, I got some very nice flat graphs. Unfortunately, listening to the ‘speakers was horrible. There was a screaming top. Clearly something was wrong. This is what I mean by using listening as a reality check on measurements. If the measurements are good, then the sound will be right. If the sound is wrong, then clearly there’s something wrong with the measurements. It also showed the practical impossibility of setting up Active ‘speakers (certainly three way) by ear. There are just too many variables. Even sorting out the polarity of the drive units compared with what different crossovers need was hard work. The first frequency sweeps I did showed notches at the crossovers. This could be fixed by reversing the polarity, only Linkwitz-Riley filters don’t invert polarity, so it shouldn’t be necessary. This then got me going as to what the polarity of the drivers really was. Had B&W labelled the connections to make life simpler for production, or are they really labelled Red + and Black – as one would expect. I made a simple polarity tester from one diode and one resistor. This half-wave rectified a sine-wave from my generator, so I could tell on a ‘scope which way up it was. After a while fiddling with the equalisers on the microphone amplifier to sharpen up the display, I worked out that B&W had labelled the ‘speakers correctly for their polarity. So now, why was I getting nulls at the crossover points? Found that if I used pink noise rather than swept sines, I wasn’t getting the nulls, so I needed to redo the frequency response measurements to check.
I decided to use 8th order (48dB/octave) L-R filters as these have twice the cutoff that the original passive filters had, and when setting up the equalisers, there were smaller ripples to equalise with the 8th order as opposed to the 4th order. I didn’t try any other filter types as each one required different EQ settings, and life’s too short!






Note that each loudspeaker is now +-1dB or the variations are constrained within a 2dB envelope. This is much closer than the 4dB envelope of the original passives, and much better than the 30 year old passives, which had a significant droop in the extreme HF above 16kHz.
Distortion was next. I measured at 31 1/3rd octave spot frequencies at 1m, and at a level of 90dB SPL. This level makes sense as firstly it’s about as loud as I’m likely to play and still be concerned about quality, and it’s the level often quoted in reviews or specs so I can make some comparisons.
I measured the 2nd and 3rd harmonic separately, but didn’t go to higher harmonics as they were all significantly lower with one exception as marked below.
Here’s the chart. I’ve also included the Meridian DSP5000 as a comparison.





Note 1: There was a spike in distortion at 8kHz to 2%, but this dropped to 0.3% immediately either side of 8kHz.
Note 2: There was a spike in distortion to 3% at 250Hz, but dropped either side to be 0.5% at 200Hz and 0.45% at 315Hz.
As these were both on the same loudspeaker cabinet and drive units, and I can’t find any reason for them, I can only put it down to a measurement error of some sort which I may recheck next time I have the test equipment set up. See * below.
I'm not sure what to make of the distortion figures I don't know how much better they should be when active compared to passive. The active 6143 is significantly better than 6143 passive, but unfortunately I didn't measure 6144 passive before modifying it. 6144 is still better than spec at LF, if no better elsewhere. They are, however significantly better than the Meridians (without subwoofer) at LF. This is probably a function of the much larger cones of the 801. How much difference the bass loading makes, I don’t know.
At above 100Hz, the distortion doesn’t seem to benefit from active operation which makes me question whether at very low frequencies, the crossover is the limiting factor, whilst above 100Hz, the drive units themselves and their linearity are the factors that affect distortion most, These are 30 year old drivers, and looking at distortion plots of loudspeakers of that era, these are pretty good numbers, but bettered by modern designs. I wonder how the latest B&W 802D would fare on a similar test.
*One final comment about the distortion measurement:- This was done using 500mS tone-bursts, measured at 1m. This therefore includes the room’s response. If, for example, the room had a 6dB peak at 2fHz compared with f, then the 2nd harmonic would be boosted by 6dB (or cut, if the room had a dip at 2fHz. Ditto at 3f compared to f. Consequently, these distortion measurements can only be taken as a broad indication rather than as an accurate figure. However, they were done in more or less the same position for each loudspeaker, so comparative figures should still be usable. Nevertheless, a future measurement is needed, as the effect of the room may explain the two anomalous readings mentioned above. Anyone have a spare Anechoic Chamber I could borrow?

To come, some subjective impressions and some thoughts on future changes.

S.

 

Part 4

Listening tests:-

This is where I get out of my comfort-zone in trying to put into words what I hear, so apologies in advance if it’s not as articulate as I would like.

Firstly, the precision of image struck me as staggering. Mono sounded almost as if it were coming from a dedicated loudspeaker rather than being a phantom image. Stereo is of a precision I hadn’t previously heard. I can put this down to a combination of the narrow almost diffraction-free head and accurate pair-matching. Level-matching between the two ‘speakers is typically within 0.5dB, and everywhere within 1dB above 200Hz (I haven’t measured below 200Hz, so don’t know.) One of Beethoven’s Razumovsky quartets on Radio 3 had the four musicians arranged in an arc, with the cello clearly forward and to the right of one violin, with the Viola to the left of the cello. It was slightly irritating that one of the violinists was swaying about as they played. I had not appreciated image precision of that nature before.

Although it’s been written before, even to the point of cliché, this stuff about “removal of veils”. I never thought I’d be writing anything like this, but I can’t think of a better way of describing the improvement in solo singers. I suspect that the dedicated mid-range driver, carrying 380Hz to 3.5kHz, is contributing to this extra clarity. That and the superior imaging. I don’t think it’s anything to do with better distortion performance, as distortion is pretty unexceptional, although the very flat frequency response must help.

Finally, the bass end:- Here, bass isn’t as extended as with the Meridians and subs as it starts rolling off above 50Hz, but being a sealed box, it will roll off at 6dB/octave, i.e. pretty slowly, rather than the 12 or 24dB/octave of the Meridians. Although positioned more or less exactly where the Meridians were, the 801s also don’t seem to set off the room boom in the same way the Meridians did, and the bass does have a more “gentle” character than the seemingly more aggressive bass of the Meridians. Rock music isn’t as “up-front” in the bass with the 801s, but nor is it particularly lacking or weedy, just less prominent. It does seem easier to follow complex bass lines, but I’m still not sure which I prefer, but I can play with EQ once I’m happy about every other aspect.

In summary, I’m very happy indeed with what I now have. Musically it is more satisfying than before, and technically, given the age of the drivers, I don’t think I can do better. Whether changing the drivers for modern lower distortion units would be worthwhile, I don’t know. The chances of getting something modern that fits in the same mounting holes without butchery is slight, I think, so perhaps I won’t bother unless I have to.

The original passive measurements I made showed that the tweeters had lost some HF output, albeit it was easily corrected with the equalisers. Nevertheless, I was concerned that the tweeters might be failing as the common reason for old tweeters to lose output is the Ferrofluid going a bit dry and sticky. This wasn’t the case as my tweeters were made before B&W started using Ferrofluid. As B&W could still make 801 tweeters available, I bought a new pair. Congratulations to B&W for their support. Superb service.

Anyway, these new tweeters were physically the same dimensions as mine, but were for the 801M and so metal domed and of slightly different sensitivity. Changing the tweeters was easy, but that necessitated setting up the crossover and equaliser again. I decided to do it again from scratch, more or less for the practice, and got it back to the 2dB envelope of the original tweeters, but without having to apply so much EQ, and both tweeters were far better matched. The result isn’t greatly different, but I’m happy having new tweeters that don’t need so much correction.

One further thing, DC protection. Although my amplifiers are AC coupled, the energy available in case of an output fault would still put a lot of DC through the tweeter and mind-range. I put some modest value capacitors in series with the mid and top drivers to protect them against an amplifier fault. I reckoned that the bass unit is sufficiently well protected by its sheer size, and the fuses in the amplifier’s DC lines. Anyway, putting a capacitor in series with the bass unit would rather detract from the direct drive which is one benefit of active operation in terms of the low output impedance at LF. However, B&W’s passive crossover did put a 1000uF capacitor in series with the bass unit, this was as much protection as paradoxically trying to get a bit more bass extension due to resonance with the main inductance. I prefer to do this by EQ if necessary.

So, now I have the ‘speakers fully active and working really well. I have no regrets about replacing the Meridians which my son was very grateful for.

I mentioned above about system noise. The Behringer A500 amplifiers are decently quiet, having noise below 100dB A weighted. However, the A weighting hid some power-supply buzz which was just about audible in my quiet rural location. With normal passive ‘speakers, the crossover filters off noise from drive units outside their frequency range, and -100dB A Weighted is normally quite sufficient for silence unless the ‘speakers are very efficient, in which case one wouldn’t use a 120watt amplifier! With Active ‘speakers, each power amp feeds each driver directly, so any noise that’s out of band for the driver is still there, and in my case the small amount of mains buzz was audible (just) coming through the mid unit. I poked around with my scope and found that the main reservoir capacitors were somewhat undersized, so increased them from 3500uF to 13500uF per supply (although the A500 has a floating supply) and that dropped the noise a further 10dB so now silence! Just for completeness, I modified all 6 channels although it wasn’t necessary for the bass and HF.

One further modification is to the DCX. The output ICs used by Behringer have decently low distortion, both THD and IMD, but these could be lower, especially IMD which is surprisingly high for a modern IC, albeit still likely to be well below thresholds of audibility. I’m having the mid and HF outputs modified by Simon to change the IC type such that the THD and IMD are vanishingly small. I doubt whether this will be audible, but from an engineering point of view, there is no harm in reducing distortion.

There is also a modification available for the DCX to create a 6 channel remote control so that I can run the inputs flat-out and do the volume control at the analogue outputs. This is interesting, and had noise been any sort of a problem, I might have considered it. It would also allow me to use the Squeezebox Touch’s digital output and do the volume analogue.

Something perhaps for the future.

Right now, I’m enjoying my 801s immensely, definitely keepers!.

S.

 

Brilliant stuff Serge.
I'll read it in detail over the next day or so.

 

No, I paid no attention at all to phase other than polarity. I'm not convinced that phase is at all audible as passing a square-wave through an all-pass phase-shift network will show. The waveform changes radically, but the sound doesn't. Also, as you said, using 48dB/octave filters, the overlap is quite short. I would have used 96dB/octave if they had been available! I also haven't been able to hear any difference with the drivers time-aligned or left with the mechanical alignment of the 801 boxes. According to Audiophile magazine's measurements, the 801s aren't accurately time-aligned, and indeed when I time-aligned them using the DCX, it introduced some very short time differences, but I couldn't hear any difference before/after.

S.

 

Unfortunately, if I change the filter slopes, then I will have to reequalise the 'speaker back to +- 1dB or what I'm likely to hear is the frequency response difference not any possible difference due to phase. As this takes about an hour to do, and I would have to do it for every combination of slopes, it's a fairly onerous amount of work. Fortunately, the DCX and DEQ have memories, so switching between them will be easy once the work is done. Perhaps next winter, when I'm getting bored, I might spend a day doing this and listening.

I try and avoid coming into London as much as possible, but your invitation could tempt me into the Metropolis.

S.

 

No, I did the measurements as single channel, and I didn't do them for each individual driver, just together as a system.

One thing I dont understand, though, is with individual drivers, phase relative to what?

Maybe I will try and get to London for a visit, as a chat might be instructive.

S.

 

Yes, thanks, that makes sense. Next time I get the measuring kit out, I'll try dual channel mode, see what the differences are. My main problem with getting the kit out, so to speak, is that I've got to lift the 'speakers onto a stand so that the measuring axis is far enough off the floor to get down to 200Hz. At my age, lifting those 'speakers isn't easy!

I can see a day or even two of measurements coming up.

S

 

The story continues.

Further to the above, I have now taken some photos of the 'speakers after having had them professionally reveneered by a local furniture restorer. They now look like this:-



I've also remeasured the Behringer amplifier's distortion having overcome a small problem with my generator's distortion. It's now down to 0.012%, and the A500 measures 0.02% at 120 watts into 8 ohms at 1kHz. Much closer to the 0.01% spec now.

I've also received back from Simon my DCX2496 modified to remove much of the IMD from the MF and HF outputs. Simon left the LF output unchanged as that's unlikely to be audible in any event. I haven't yet had long enough to be able to tell of any improvement, but having looked at the audio files Simon sent me, the modified outputs are vastly better than the unmodified ones by an order of magnitude or more.

However, given that I was without the box for 2 weeks whilst away on holiday during the modifications, I very much doubt my audio memory would be good enough to tell what at best would, audibly, be a subtle improvement. Nevertheless, from an engineering point of view, its got to be in the right direction.


S.

 

Serge, what's the veneer?

Looks like zebrano.

Lovely whatever it is

 

Yes that's right. The brown stripes sets off the brown grilles (or vice-versa) very well.

The restorer did superb job. Very pleased with them.

S.

 

He charged me £500 inc VAT. More than I'd hoped but less than I feared, so probably quite fair.

www.reandjblebbon.com

S.

 

excellent article. I also have some 801's that I'm swapping out the passive filters for active.
You mentioned phase shifts and not being able to hear them. They are hard to hear. any phase shift is usually associated with non linearities of the drivers. these will effect more the polar response of the drivers around cross over frequency, and will cause things like movement of sounds you are hearing in the stereo field, and this can be masked by the bleed between the speakers and your ears. Those FFT's that displays waterfall plots often can be spotted with a train eye. as far as time alignment between drives - this will tilt the polar response quite a bit, and the combination of filters and driver placement can be more or less a guessing game unless you actually time them. I usually have the speaker in the open - I try to place them in a hole pointing up - that way there is no reflections, plus the bass roll-off can be calculated via using 180 degree loading. it seems to work better then hoisting way up in the air. For timing I usually create a coded signal that I apply separately to both drivers, and record them using a mic that is level to what would be the plane of where you would be hearing it. once recorded I FFT the signals and from there I can measure the group delays between the drivers around the crossover frequencies. But you need to first make sure that your D/A's are in sync - such that the signals are in sync. its a lot better these days but the older A/D/D/A can be quite sloppy in their syncing. Also check multiple levels of power as well. drivers in the most part are not very linear (ribbons and AMT's are much better at being linear in their audio range). and that's where active filters are better, they take much of the non linearities out of the equation that are hard to compensate in passive filters.

As far as Behringer equipment, change the filtering caps in the power supply. Behringer is notorious for under designing their PSU using really low rated caps - part of the designed obsolescence, they can't cheap out on any of the audio components because you can hear that. But once the cap goes, the power control chip will fry, and Behringers are too cheap to repair in most places, where the repair cost is more than the unit's price.

I am using a chip amps for the mids and tweeters (50watts @ 8ohms) while I am using a D class amp (180watts) for the bass units. digital amps work better with low frequency drivers, they have far more flexibility with matching low impedance drivers, and woofers in general will have quite a bit change in impedance around Fs. while the chip amps are very clean sounding and very fast as well, making for a good amp for mids/tweeters.

My units came from a recording studio, such that they are dark gray colour, and don't have the speaker protection included. I think that the protection circuity did alter some sonic details being they were non active protection. the later units had active (powered) protection, but from what I read it still has clarity issues.

My active cross over is yet to be decided. I do have Elliott Sound Products (ESP) active crossover filter board (3 way http://sound-au.com/project09.htm.) But I not sure if I want to use them with this project. I'm also looking at digital filters - FIR types, and there is a growing list of software venders getting into the action. But I'm still in research mode on these. I have one of Behringer's audio interfaces which will give my separate outputs - and I have replaced those inferior caps in the power supply.
I do have a Yamaha DME which can do everything digitally up to 24db. its got the power to cross over about 40 units - used in PA systems mostly. I use it for testing/simulating crossover points, group delays and slopes. I think the newest ones will do 192khz at 32bit resolution, but I dont know if they can run FIR filters in their software

I'm not sure about using 48db mainly because there may be (just guessing) a fair amount of phase shifting close to the crossover point even though its past the F3 point. At least with the Linkwitz -Riley, it quite linear across and a bit beyond the F3 if you use the right op amps. Sony for example, early in the digital game had some 48db filters in their D/A converters and they sounded horrible. The sigma delta modulation AD/DA chip sets(Philips) are much much better these days.

My 801's are just a recent purchase so it will be awhile before that will be completed. I have some old fostex nearfields co-axials that I'm fixing up as well. They had a great printed ribbon tweeter but were easily blown. they use a passive crossover - and the reason why they blew. But one of the best nearfields for vocals I have used. The woofers are very nice as well (still working) so I don't want to just chuck them. another project for another day. After they blew, I purchased some Fluid XP7's to use when doing mix downs. They are also co-axials with an AMT tweeter. Very nice units as well. Final mastering will use the 801's, I had mastered using them 30 yrs ago and they were great, but very expensive at the time.

anyway nice to have someone else doing something similar to me.

cheers