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jim henderson
10-08-2003, 02:25 PM
Does anyone have any experience with the BSS FDS 388 or 366?

My main concern is the trade-off between the benefits of time-alignment that a 388/366 gives you vs. the superior audio quality of a passive x-over (on the mid/hi transition).

Oldmics
10-08-2003, 03:24 PM
O.K.,I"ll go first and I know that Scott will be chimin in right behind me.If your only concern is for the time alignment,then depending upon budget,application and many other criteria its a good unit.More information would need to be forthcoming about your particular audio useage before a full answer could be formulated.Also the two units that you have mentioned are compleatly different in both features and sonic interpetation.Are you looking for an analogue to digital device?Do you want a crossover with limiting?How many bands of crossover points do you need?Is stereo a requirement?What are the chores that you need for a processer to do in relation to what sort of system you have and the application of your system?Do you really even need a processer?You have addressed the need for time alignment which they both do.If you only need a time alignment unit,a time delay unit of some kind may fit your requirements at a lower cost than the BSS equipment.If you have an unlimited budget look at Lake Technology Limited equipment for some unbelieveable capeabilitys.Oldmics

scott fitlin
10-08-2003, 03:36 PM
Whether you use the 388 or the 366 the time alignment features allow you to precisely dial in the correct alignment between the ranges! It can be done in feet or Milliseconds! With the 366 it now has an additional feature which automatically computes the proper settings for each frequency range, and this requires an additional calibrated mic, I believe!

Properly set up, time aligned speakers offer very good imaging! Very clear, and stable images with noticeable lack of smear. But, with the BSS units you are always aware that you are hearing music through the 388 or 366! This was what I objected to! My midbass through analog crossovers has a much fuller sound! The mids through the BSS units seem to have a white noise type of sound in the background! faint, but noticeable! They just dont disappear like really good analog crossovers can! So, in my opinion, you would be hearing perfectly time aligned signals, but with the distinct sonic character these units have! Even with all the features the BSS units offer, there was something in the midrange, and you just CANT EQ it out! The 366 is superior to the 388, 24 bits vs 16 bits! But, a really good analog crossover still has a more natural quality! I see it this way, you listen to a CD, the CD player or external DAC converts the binary code to an analog waveform, now you go into a BSS 366 and convert the analog signal back into binary code, process the signal with all the features, then re-convert the signal back into analog signal. Too many conversions, and something gets lost in the process!

For me, since Im horn loaded the mids and midbass are already in pretty close alignment as the voice coils of the mid horn drivers are within 1/2 inch vertical plane with the 15 in woofers voice coils!

Time alignment can also be done in certain analog crossovers as well! For example, Marchand offers his crossovers with built in time delay for a specific range, but you will have to know how many feet or ms you need to delay! he will also make crossovers with baffle step correction if needed!

They also did time alignment with the old Urei studio monitors, and it really worked! And this was analog, and they were good sounding monitors!

At the present time, for what BSS dsp units cost, I would wait until dsp is actually perfected! The 366 is better than the 388, but....

Time aligment works, but it will not overcome the sonic inferiority of the dsp units to dedicated analog crossovers! If you dont like the sound of the 366 in the first place, nothing will change that! Have you actually listened to a DSP controlled system for any length of time? Let me also say that setting up a DSP loudspeaker processor takes alot of time to get it right as well!

My personal opinion is that while these units do all the things they claim, they also do too many things to be able to do ONE thing exceptionally well! I mean here we have units capable of 1st thru 4th order filtering, with selectable slopes; i.e. bessel, butterworth, linkwitz-riley, parametric eq, time alignment, limiting, phase alignment, and ability to store multiple system and room programs! While great analog crossovers, whether passive or active, are exactly that, Crossovers. The most you find on an analog crossover is frequency, output, and gain adjust. Maybe time delay.

Another thing to look at is consumer audio! Yes, while we do have digital room correction available such as TacT, no one is really manufacturing DSP active crossovers for the Home market! Most of the Top Guns in consumer speaker and electonics manufacturing are still analog, except for CD players! By this time, if DSP were perfected I tend to think you would see far more DSP units aimed at the consumer market! I also belive that companies that sell ultra expensive speaker systems, such as Wilson Audio would have already adopted DSP into some of their offered systems!

Again, you still have to listen to a DSP unit for an extended period of time and see IF you like how it sounds! When I say extended I mean at least 2 weeks to really get to know how the unit sounds!

Simple passive, well designed and built analog crossovers sound
great because they are simple, and do not do much to the audio signal except roll the signal off at a predetermined point.

I belive firmly in the Kis theory! Keep It Simple!

scott fitlin
10-08-2003, 04:02 PM
Oldmics points out that you are dealing with units that have many features that, in a home envirioment you will never use, at least not to the units fullest capabilty! At home I really dont see you ever using limiting or compression. Do you need to have the ability to store 60 individual room and system set ups? All these features add to the cost of the unit! The 366 is gonna cost you quite a pretty penny!

You can as Oldmics stated get an outboard delay unit and this is far more economical than Omnidrives! BSS also makes a good delay unit and its not that expensive!

Jus something to think about.

jim henderson
10-08-2003, 04:22 PM
My application is critical listening in my living room where we listen to music and watch movies.

My main speakers consist of 1400ND, 2123H, and a 2421A on a 2344A horn. My sub is 2235H-based. I am currently using a JBL active crossover at 300 Hz, and a passive network on the mid/high (1300 Hz) with CD horn compensation.

The subwoofer is currently connected to the LFE channel of my processor, but I think Id like to integrate it into the mains so that all signal sources utilize the sub. The EQ function of the BSS units would be quite nice to have.

My concern is the time-alignment of the 2421A, which sits about 4-1/2 behind the cone drivers, which is about 1/3 of a wave length at the crossover frequency. Is this significant?

I like the idea of having all processing being done in the digital domain where, presumably, no distortion or phase shifts are induced. However, my system is all analog, so the FDS-388/366 would have to convert to digital on input and back to analog on output.

Robh3606
10-08-2003, 05:40 PM
Hello Jim

I have had similar thoughts. I have 2122's and 2416's on the 2344's. I assume they are flush mounted to the baffle so you have the same spacing I do. One thing to consider is the 4430/4435 has a baffle step to move the horn forward to help make up the distance. Only problem is the 2122,2123 is not as deep as the 2235 so that won't really do it either. I am all analog too and I am hessitant to digitize just to get time alignment. Have you tried just moving the 2344 forward???

Rob:)

Oldmics
10-08-2003, 10:33 PM
Quote "My main speakers consist of 1400ND, 2123H, and a 2421A on a 2344A horn. My sub is 2235H-based. I am currently using a JBL active crossover at 300 Hz, and a passive network on the mid/high (1300 Hz) with CD horn compensation.

The subwoofer is currently connected to the LFE channel of my processor, but I think Id like to integrate it into the mains so that all signal sources utilize the sub. " So this would end up as a four way stereo system?If so two processers would be necessary if you choose to go with the 366.The 366 BSS units being discussed are are 2 in and 6 out.You need a 2 in 8 out processer to use in a four way stereo configuration.That brings you to the only choice of the 388 which is 2 in and 8 out.The 388 is an obsolete product although it can be found on eBay from time to time.Bringing this thread back to JBL products the DSC 280 processer by JBL could be the tool for your needs.These were made by BSS in England for JBL and are essentially the same as the BSS 388.Yes there are some minor differences but they are the same units.These also are obsoleted products but can be had on eBay(or PM me as I have a few to part with). All of the processers mentioned are analogue to digital and digital to analogue converters (with many more features than just conversion) in a box.This way all of the outputs are in the analogue domain for connection to your amplifiers.Digital amps are here but lets not rush these things yet.The 388 and DSC 280 both sound identical where as the 366 units sound different.I am going to limit my conversation to these two particular processers that you are questioning rather than going off on a tangent about the current realm of digital processers now being offered by many different manufactors.The hi freq componet offset that you have would be corrected with the alignment capacitys that these processers offer. A question to consider is "Do you have the tools to measure the time arrival differences from all of the componets? " This is another whole process that needs to be addressed.If you are only concerned about the hi freq time smear try to acquire an analogue delay and align the componets before delving into the world of digital.Digital is an interpetation of the original signal source.All of the units discussed are of good enough quality to overwhealm a listener who has not experienced the A to D and D to A trip previously.Upon hearing a digitised system you may not like the end result of converter interpetation to your favorite piece of music or you may love it.Only you can make that determination. Best regards,Oldmics

Ron K
10-09-2003, 07:37 AM
I have a system in my garage that uses a 2226H in a rear loading corner horn, 2123H mid, 2420/2421 on a 2342 horn.
The fronts of the frames and flange of horn are in the same plane.
this puts the bass driver and HF driver in the same acoustic plane. This is triamped and with a small amount of delay on the dids (AC23 crossover) the system is time aligned and sounds super. I used the 4435/4430 compensation on the horn but
C9 ended up as 3Mf with these drivers. I will post this system later with pictures. Ron

Robh3606
10-09-2003, 08:14 AM
Hello Ron

How do you like the 2342?? Do you get the expected high end response using that horn with the 4430/4435 compensation?? I have 2344's set-up with the 4435 schematic so C9 is 2.5Mf. Do you hear a significant difference without the delay?? How much delay are you using??

Rob:)

How are you attaching the horn??? The 2342 is a screw on. How did you rig an adaptor so it will work with a 2420/2421??

scott fitlin
10-09-2003, 11:14 AM
Originally posted by Ron K
I have a system in my garage that uses a 2226H in a rear loading corner horn, 2123H mid, 2420/2421 on a 2342 horn.
The fronts of the frames and flange of horn are in the same plane.
this puts the bass driver and HF driver in the same acoustic plane. This is triamped and with a small amount of delay on the dids (AC23 crossover) the system is time aligned and sounds super. I used the 4435/4430 compensation on the horn but
C9 ended up as 3Mf with these drivers. I will post this system later with pictures. Ron Its is NOT the front flanges and frames that have to match for time alignment! The voice coils of ALL your drivers must be lined up in the vertical plane to achieve time alignment!

Again, Oldmics says it the way it is! BSS did in fact manufacture the JBL line of DSP processors and essentiially they are the same as BSS!

Ill add another personal opinion here! In my listening experiences I have found that multi channel digital audio, properly set up is very well suited to movie sound and special effects that film soundtracks contain! But listening to music is a different story for me!

Also, Oldmics points out, and I have stated as well, these units are not the easiest to use in terms of correctly setting your parameters, such as time alignment! If you dont set it up correctly for your speaker/room setup, you do more damage than good! This is one feature of the 366 that comes in handy! The 366 can determine proper time settings! But, you would need two 366 units for 4 way flexibility, and this gets outta sight pricewise!

Another unit to possibly consider is Ashly! Yes, Ashly, because they make DSP processors, many swear they are warmer sounding than the BSS or JBL units, and are more affordable than BSS/JBL!

The BIG sound contractors and engineers that I know have all gone to XTA, and they say these are the best sounding of all the professional DSP processors available! Even more expensive than BSS!

Of course, theres always DBX Driverack! This is somewhat more affordable, and has the time alignment features and crossover functions!

Bottom line is YOU have to get your hands on one and really listen to your system with it. Digitizing your audio signal will sound different to you. I wont say better or worse, just different. And you will have to decide what you like and if its worth it!

:cool:

Ian Mackenzie
10-09-2003, 01:58 PM
Jim,

Stop worrying,

I would not trouble your self further, the pursuit of technical perfection may only reduce your listening pleasure

I had a very similar scenario to you until recently using the 2122 & 2344 and ended up leaving it as is.

It is very unlikey that your imaging will be effected by time delay effests. ie under 1 ms.

The effect of the driver dispacement only will be noticable only in the area of the crossover overlap where the dispacement angle of the acoustic centres will cause of lobe in the vertical polar amplitude response.

This can be minimised by either moving the 2344a forward on a sub baffle for shifting the 2344 further away from the 2123.

If the 2344a is near the 2123, be aware the 2123 will reflect off the 2344 cause more problems than it solves by moving it forward (I tried all of this stuff)

The time displacement is well under a millisecond, above 2 ms it might be a problem with some types of music.

On another matter, are you using the 3134/35 actual passive crossover values? Might be worth a try, its an allpass design with minimal group delay and appears as a very open sound stage.

(both drivers are in phase with this crossover, the LC values are tweeked to null the peak normally associated with 180 phase of 2nd order 2 pole designs)

macka

:smthsail:

Ron K
10-09-2003, 02:00 PM
Hope you know dids is mids, hit wrong key. The voice coils of the woofer and H.F. are in the same plane, the 2123 is 1.5 to 2" ahead so I use the electronic delay of the Rane AC23 crossover to align the mid with the others. The result is subtle but audible. I really like the sound of the EQed horns,the treble is adjusted slightly bright and still sound very smooth and accurate. I cut the threaded part off the horn and epoxied a home made flange on the horn because all my drivers are bolt on. I will post more on these speakers later and include pics. they are more functional and great listening than they are attractive. If others are interested in the bass horns I will give more details later. But they achieve an extra octave out of the 2226 that is not possible in other enclosures. I am getting much enjoyment out of this system. Ron

Ian Mackenzie
10-09-2003, 02:19 PM
This is the basic geometry of the lobe problem, actually JBL used a 15 degree lobe angle in the 4430 titling upward as they were often mounted upsidedown.

Sorry, my graphic skills are wanting.

Ian:o

Robh3606
10-09-2003, 02:43 PM
Hello Ian

Thats how you figure the lobe angle??? I thought you used the formula with the driver spacing and crossover frequency wave length. I posted a question about how to figure out axis tilt like offset in the 4430. So simply the tilt is the angle offset of the voice coils?? The driver acoustic centers don't contribute??

Help:hyp: :spin:

Rob:)

Ian Mackenzie
10-09-2003, 03:04 PM
O'kay Rob,

I have the Vance Dickason Cookbook in my lap.

He advises the exact acoustic centre is somewhat of a mystery.

When calculating the horizontal offset distance for crossover design purposes, the only important factor is the relative amount of offset and not the device acoustic centres.

A drivers acoustic centre varies with frequency and by definition is a function of the phase response of the driver.

At low frequencies where the group delay is the greatest, the acoustic centre of the speaker can be a substancial distance behind the driver (ZDP) zero delay plane.

He goes on to say for the purpose of radiation tilt offet, the radiating centre may be assumed to be the centre of the voice coil (midpoint on the driver front plate).

For further reading see pp 92 of the Loudspeaker Design Cookbook 4th edition.

Anyways, without impulse (moving from impulse power to warp driver says Capt Kirk) analysis measurement its a hit and miss affair.

Ian

:smthsail:

Earl K
10-09-2003, 04:03 PM
Hi Jim

My 2 cents ( okay there's more than that ) on this topic is;

I'd be more concerned with maintaining correct phase relationships than "time-aligning" those two components.. I realize that's an obscure / meaningless statement so I'll give you an example of a present dilemma I'm trying to work out a solution to . It all goes back to my playing around with DC blocking capacitors ( hence the title of my injoke - Ian should appreciate this ) & listening to the sonic signature of different dielectrics .

Some background; I use a 24 db/octave, LR slope type , active crossover ( 1st generation Behringer ). My only HF EQ is a simple RC network that depresses the mids and effectively adds a 6db per octave "boost" that fights against the 9 or 10 db/octave roll-off of my big 2441 drivers. This isn't a "flat" curve but the net result sure does sound musically correct to me. This setup on a small round-mouthed horn gives stellar results, like pin-point imaging and great depth of sound-stage. No synching up of the woofer & compression driver is involved or is a contributing factor in my musical enjoyment. The voice coils have about a 4" offset .

BUT - add in those damn DC blocking caps and my great soundstage collapses or compresses. For a couple of weeks I thought I was listening to a particular quality of the caps themselves until I twigged that this problem happens with all my types of DC blocking caps. The lurching-trail of logic has become that I am now reacting in a very negative fashion to an LR circuit that has been tragically mucked with giving the Hi-pass secton a 450 phase shift. This is easy to do with active components . For two years I've listened ( more or less ) to music with no DC blocking caps in place. So, the 90 phase-shift added onto that of the 360 hipass section has created something quite distracting to my ears. I can time-align the 2 sections but my ears tell me its far from the same. And I think the analog "bucket-brigade" delay section of the Behringer is a linear-phase type that begins correct in the crossover area and then just keeps adding increasing delay ( and phase offsets ) into the low pass section. In a couple of weeks , when work patterns permit, I'm going to turn my DC blocking caps into a 12 LR style network ( gloomed ontop of the choosen active frequency ) - then flip the drivers polarity and see if this does the trick for my ears by restoring the lost LR phase relationships . Going all passive may be a better option .

Obviously the domino effect of one change can be quite staggering .

Can I ask why you are concerned with time-aligning? It's mostly relevant ( to vertically alligned transducers ) within a single horizontal plane ( while these pesky phase anamolies will exist over a much broader listening area ).

If you go digital; I'd suggest a CD player with a digital out allowing you to keep it digital until you get to the best sounding DAs in your setup . This could also mean buying Apogee DACS and using them after the crossover if it has a couple of AES digital outs - which I can't remember if it does .

regards <. Earl K :)

Robh3606
10-09-2003, 04:30 PM
Hello Ian still:hyp: :spin:

Something seems weird there. If you look at the polars for the 4430 the 15 degree tilt looks to be way too much. You see the best at 0 and 10 up. At 20 you start to see some problems. Also if you apply that logic to the 4345 you have one axis from the midrange to the compression driver pointed at the ceiling. Much more tilt than the 4430 because of the much longer length of the horn. The compression driver to slot is pointed at a side wall. I must be missing something:confused: You have to include the crossover slopes in there too don't you?? 12db network Isn't the compression driver 90 degrees ahead and the woofer 90 degress behind so you invert phase. So you effectivly have just " physically moved" the driver 1/2 wavelength.


Rob:banghead:

Ian Mackenzie
10-10-2003, 05:30 AM
Robh3606,

(Going a bit off topic here...)

Yeah, sorting this out its like discerning French & English mustard, a mouthful of each and you can't work out which is which. (also both make you sneeze and blast hot wind)

My reference to the tilt was from the 4430 White paper in the AES.

Just looked at those curves myself, the 10+ degree up curve is quite smooth +-2 db and next one is +20 grees with 5 db variations. If you go down there are deeper troughs for sure, but only over a narrow band, I think this is JBL point and they would be eq'd out at the desk listening possy anyways.

I appreciate your comments regards the 4345, I'm not sure what the deal is as there are no known off axis curves that I am aware .

But looking at the 3145 network intuitively, the 4345 uses sharp high pass filter slopes in the upper mid and UHF regions of 18 db and this would mitigate the lobing effects in the crossover regions to a large degree. The slopes are also asymmetric 12db low pass/18db high pass and wired in phase.

The lobes are only an issue in the region of driver overlap, this is why the alignment is a big deal with 1st order systems like Duntech, Dynaudio etc.

It is normal practise to amend the LC values to minimise the lobing effects on axis or +- 30 degrees off axis. Having said that the 4345 is normally run with the slots on the outer left and right, could be the lobe is pushes inward in the crossover region

One can also reasonably assume the network in the case of the 4345 was smoothest on axis by design, and was not intended as a monitor with flat power response.

When I am in the test phase of the 4345 diy project I will certainly run some on/off axis curves and post them.

It would be interesting if Giskard can slooth from memories from GT on this topic.


Ian
:)

boputnam
10-10-2003, 09:00 AM
Originally posted by Ian Mackenzie
Yeah, sorting this out its like discerning French & English mustard, a mouthful of each and you can't work out which is which. (also both make you sneeze and blast hot wind) Then, of course, there's always VEGEMITE, the attributes of which completely evade me... :rotfl:

DamnYankee