I have a pair of Kef SP1022 B200 drivers that I am trying to build into a Fried Model H "Coffin" which will be used to provide the low frequencies in support of a pair of smaller nominal 8Ω speakers each of which will contain a B110 and a T27 with a suitable crossover (e.g. LS3/5a). If it makes any difference, I could perhaps even use this with a pair of Quad ESL-57s for example.
The Fried Model H design includes a three-way crossover in a separate box. Since I already have a crossover in my "Satellites" I do not want to do this. My plan is to drive the "Coffin" from a single Amp and have the signal split into low frequencies (probably below 100Hz) driving the B200s and higher frequencies (e.g. above 100 Hz) which will be taken to the "Satellite" speakers. The coffin would therefore end up with four pairs of speaker binding posts - two in from the Amp and two out to the "Satellites".
I have spent a fair amount of time on the AudioHolics and DIyaudioandvideo websites and am probably more confused than I was before I started. Therefore I wonder if anyone would be willing to clear up a couple of points for me.
1) I don't really understand the benefits of and logic behind the various "Butterworth" designs. Based on two drivers which effectively approximates to what I want to have, I have worked out that a 1st order Butterworth uses just an inductor to stop high frequencies reaching the bass driver and a capacitor to stop low frequencies reaching the mid/tweeter.
A 2nd order Butterworth uses just an inductor in series with the bass driver and a capacitor in parallel with it and in order to stop low frequencies reaching the mid/tweeter it uses a capacitoir in series and an inductor in parallel.
I have also looked at 3rd and 4th order Butterworths and they use greater numbers of capacitors and inductors configured in much the same way. I have included below some sample designs taken from the DIyaudioandvideo website.
I would be really grateful if someone could explain in Noobie's terms how the various Butterworth filters work and what the effective benefits are of the different "orders"?
2) taking the 2nd order Butterworth design as a basis I get the following values for the Capacitors and Inductors:
100Hz - C1 & C2 = 140.63 uF and L1 & L2 = 18.01 mH
150Hz - C1 & C2 = 93.75 uF and L1 & L2 = 12.01 mH
200Hz - C1 & C2 = 70.31 uF and L1 & L2 = 9 mH
300Hz - C1 & C2 = 46.88 uF and L1 & L2 = 6 mH
Do the fractional parts make any significant difference? In the first instance would 140 uF and 18 mH be satisfactory?
Are Inductors of these values actually available? I believe that on can achieve the correct value by combining components but I am not sure whether it should be in series or in parallel?
Finally, will my plan actually work? Does it make sense?
I know that this is all pretty basic stuff but I would be VERY grateful for any relevant help and guidance 🙂
The Fried Model H design includes a three-way crossover in a separate box. Since I already have a crossover in my "Satellites" I do not want to do this. My plan is to drive the "Coffin" from a single Amp and have the signal split into low frequencies (probably below 100Hz) driving the B200s and higher frequencies (e.g. above 100 Hz) which will be taken to the "Satellite" speakers. The coffin would therefore end up with four pairs of speaker binding posts - two in from the Amp and two out to the "Satellites".
I have spent a fair amount of time on the AudioHolics and DIyaudioandvideo websites and am probably more confused than I was before I started. Therefore I wonder if anyone would be willing to clear up a couple of points for me.
1) I don't really understand the benefits of and logic behind the various "Butterworth" designs. Based on two drivers which effectively approximates to what I want to have, I have worked out that a 1st order Butterworth uses just an inductor to stop high frequencies reaching the bass driver and a capacitor to stop low frequencies reaching the mid/tweeter.
A 2nd order Butterworth uses just an inductor in series with the bass driver and a capacitor in parallel with it and in order to stop low frequencies reaching the mid/tweeter it uses a capacitoir in series and an inductor in parallel.
I have also looked at 3rd and 4th order Butterworths and they use greater numbers of capacitors and inductors configured in much the same way. I have included below some sample designs taken from the DIyaudioandvideo website.
I would be really grateful if someone could explain in Noobie's terms how the various Butterworth filters work and what the effective benefits are of the different "orders"?
2) taking the 2nd order Butterworth design as a basis I get the following values for the Capacitors and Inductors:
100Hz - C1 & C2 = 140.63 uF and L1 & L2 = 18.01 mH
150Hz - C1 & C2 = 93.75 uF and L1 & L2 = 12.01 mH
200Hz - C1 & C2 = 70.31 uF and L1 & L2 = 9 mH
300Hz - C1 & C2 = 46.88 uF and L1 & L2 = 6 mH
Do the fractional parts make any significant difference? In the first instance would 140 uF and 18 mH be satisfactory?
Are Inductors of these values actually available? I believe that on can achieve the correct value by combining components but I am not sure whether it should be in series or in parallel?
Finally, will my plan actually work? Does it make sense?
I know that this is all pretty basic stuff but I would be VERY grateful for any relevant help and guidance 🙂
The first order is the only "sum to unity" type of crossover, which gives better waveforms, but requires drivers
with wider bandwidths, because of the slower slopes. The fourth order is considered the best type overall by most.
You want reasonably close tolerances on the parts, as well as good channel to channel matching. Examples of parts:
Compare Inductors: The Madisound Speaker Store
Compare Capacitors: The Madisound Speaker Store
I wish we could explain it in layman's terms, but unfortunately, I have some bad news: crossover design is about the hardest thing involved in loudspeaker design. If the drive units are the heart of a speaker, the XO is its brain and nervous system. It is not an easy thing to master, there are a myriad of complex factors involved, and ultimately, it is the XO in a multiway loudspeaker that determines the bulk of the end sound / 'voicing'.
Firstly, I think you need to read this: http://www.diyaudio.com/forums/mult...designing-crossovers-without-measurement.html
A very valuable thread.
OK, next up. Those on-line calculators? They're basically junk, except as theoretical exercises. The problem is that they assume the frequency response of the drive units is flat, and the impedance is also flat and stable. Unfortunately, loudspeaker drive units are anything but flat under 99.999% of practical conditions, nor is their impedance, so if you base your XO on the assumption that either / both of these are (when in fact they are not): you're in trouble.
Some basic theory for you on filters.
-A high pass, as the name suggests, allows high frequencies above point x to pass through and rolls the response off below that. A low pass does the opposite: it allows low frequencies below point x to pass through, and rolls the response off above that. OK so far?
-Different orders of filters basically roll the response off more or less quickly.
-A 1st order filter rolls the response off at 6dB per octave, i.e. 6dB for each doubling of frequency, e.g. 1KHz, 2KHz, 4KHz, 8KHz, 16KHz and so on.
-A 2nd order filter rolls the response off at 12dB per octave
-A 3rd order filter rolls the response off at 18dB per octave
-A 4th order filter rolls the response off at 24dB per octave
And so on, up to infinity. Each time you go one order higher, you get another 6dB of attenuation. Filters are not 'brick walls' (not this type anyway): it's not a sudden stop, it's a more or less gradual attenuation.
-Filters with slow rolloffs have the least phase shift (good). However, they do not protect delicate tweeters very well because of that very shallow / slow attenuation, and if a bass driver has a big peak in its response just above the crossover, it won't help attenuate that very much either. Higher order filters roll the driver off faster, give better protection to the tweeter and shunt unwanted resonances above / below the crossover frequency (remember they're not brick-walls with a sudden stop) to a lower level where they'll be less audible. They do, however, have more phase-shift (not so good, but not automatically a problem -this depends on implementation).
-All first order filters are Butterworth types. The can't physically be anything else. Butterworth filters are named after a physicist called Stephen Butterworth.
-Higher order filters (2nd, 3rd, 4th etc.) can have different sub-types, e.g. Butterworth, Linkwitz-Riley, Bessel etc. They all ultimately have the same order of rolloff, but the details of how they do so is different. Butterworth filters are what's known as maximally flat. That means their amplitude stays flat to the lowest possible frequency, and then they roll off consistently, without deviation. That sounds ideal, and in some ways it is, but it's rarely very practical for a variety of reasons. For example, even order Butterworth high pass and low pass set for the same frequency will cross at -3dB, which results in a +3dB peak at that point as the outputs sum. Not so good. Other even-order types, like Linkwitz, sum flat, and are generally more useable. There are exceptions, but you really need to know what you're doing, so forget that.
That's all theory. In practice, drive units do not have a flat frequency and impedance response -not usually sufficient for pure theory to work anyway. Let's take an example. Say you have a tweeter which has a flat response from 2KHz upward, and below that, it rolls off 6dB per octave, so it is naturally 6dB down at 1KHz. OK so far? You've no doubt seen similar in the frequency response plots of different tweeters. Now, if you put a 1st order electrical filter on that tweeter at 2KHz, those things become additive. You get a 2nd order acoustic rolloff, because the tweeter was already rolling off with a 1st order slope all by itself, and you've given it a 1st order electrical filter in addition to that (this is an over-simplification, but basically accurate enough to give you an idea). Got a headache yet? 😉 No? You will do, because as soon as you put a driver into a cabinet, you get a bunch of other factors to consider, such as baffle-step losses, diffraction effects, floor-bounce etc., all of which play merry havoc with the frequency response and will need some degree of correction.
I'm not trying to discourage you, but I do strongly advise not going in at the deep end, because it will be a recipe for failure, since you are using drivers that are not known for their ease of use (they're a swine) and you'll probably a/ get frustrated, and b/ become discouraged. Which we don't want. So: read that thread I linked to above. Carefully. Take your time. The maths looks a bit daunting, but it's not all that bad if you go through it carefully with a calculator. Alan used a simple 2-way speaker as an example, but it will give you a starting point to work from, and one that is much better than any of those on-line calculators, because you will get an understanding of what you are doing and why. Generally speaking, the best method of designing a crossover is to measure the frequency and impedance response of the drive units in the cabinets, enter this data into modelling software, and design the filter that way. That is a big new skillset in itself, and not for the faint-hearted. Once you have familiarised yourself with the thread I linked to above & feel a bit more confident about the basics, you may then want to move on to read these pages, which will give you an idea of how you can use some simulation software (good) without necessarily investing a large amount of time & potentially money into measurement equipment.
https://sites.google.com/site/undefinition/diy-faqs-provendesigns
https://sites.google.com/site/undefinition/simulated-measurements
A lot there I know, but I hope it helps a little.
Firstly, I think you need to read this: http://www.diyaudio.com/forums/mult...designing-crossovers-without-measurement.html
A very valuable thread.
OK, next up. Those on-line calculators? They're basically junk, except as theoretical exercises. The problem is that they assume the frequency response of the drive units is flat, and the impedance is also flat and stable. Unfortunately, loudspeaker drive units are anything but flat under 99.999% of practical conditions, nor is their impedance, so if you base your XO on the assumption that either / both of these are (when in fact they are not): you're in trouble.
Some basic theory for you on filters.
-A high pass, as the name suggests, allows high frequencies above point x to pass through and rolls the response off below that. A low pass does the opposite: it allows low frequencies below point x to pass through, and rolls the response off above that. OK so far?
-Different orders of filters basically roll the response off more or less quickly.
-A 1st order filter rolls the response off at 6dB per octave, i.e. 6dB for each doubling of frequency, e.g. 1KHz, 2KHz, 4KHz, 8KHz, 16KHz and so on.
-A 2nd order filter rolls the response off at 12dB per octave
-A 3rd order filter rolls the response off at 18dB per octave
-A 4th order filter rolls the response off at 24dB per octave
And so on, up to infinity. Each time you go one order higher, you get another 6dB of attenuation. Filters are not 'brick walls' (not this type anyway): it's not a sudden stop, it's a more or less gradual attenuation.
-Filters with slow rolloffs have the least phase shift (good). However, they do not protect delicate tweeters very well because of that very shallow / slow attenuation, and if a bass driver has a big peak in its response just above the crossover, it won't help attenuate that very much either. Higher order filters roll the driver off faster, give better protection to the tweeter and shunt unwanted resonances above / below the crossover frequency (remember they're not brick-walls with a sudden stop) to a lower level where they'll be less audible. They do, however, have more phase-shift (not so good, but not automatically a problem -this depends on implementation).
-All first order filters are Butterworth types. The can't physically be anything else. Butterworth filters are named after a physicist called Stephen Butterworth.
-Higher order filters (2nd, 3rd, 4th etc.) can have different sub-types, e.g. Butterworth, Linkwitz-Riley, Bessel etc. They all ultimately have the same order of rolloff, but the details of how they do so is different. Butterworth filters are what's known as maximally flat. That means their amplitude stays flat to the lowest possible frequency, and then they roll off consistently, without deviation. That sounds ideal, and in some ways it is, but it's rarely very practical for a variety of reasons. For example, even order Butterworth high pass and low pass set for the same frequency will cross at -3dB, which results in a +3dB peak at that point as the outputs sum. Not so good. Other even-order types, like Linkwitz, sum flat, and are generally more useable. There are exceptions, but you really need to know what you're doing, so forget that.
That's all theory. In practice, drive units do not have a flat frequency and impedance response -not usually sufficient for pure theory to work anyway. Let's take an example. Say you have a tweeter which has a flat response from 2KHz upward, and below that, it rolls off 6dB per octave, so it is naturally 6dB down at 1KHz. OK so far? You've no doubt seen similar in the frequency response plots of different tweeters. Now, if you put a 1st order electrical filter on that tweeter at 2KHz, those things become additive. You get a 2nd order acoustic rolloff, because the tweeter was already rolling off with a 1st order slope all by itself, and you've given it a 1st order electrical filter in addition to that (this is an over-simplification, but basically accurate enough to give you an idea). Got a headache yet? 😉 No? You will do, because as soon as you put a driver into a cabinet, you get a bunch of other factors to consider, such as baffle-step losses, diffraction effects, floor-bounce etc., all of which play merry havoc with the frequency response and will need some degree of correction.
I'm not trying to discourage you, but I do strongly advise not going in at the deep end, because it will be a recipe for failure, since you are using drivers that are not known for their ease of use (they're a swine) and you'll probably a/ get frustrated, and b/ become discouraged. Which we don't want. So: read that thread I linked to above. Carefully. Take your time. The maths looks a bit daunting, but it's not all that bad if you go through it carefully with a calculator. Alan used a simple 2-way speaker as an example, but it will give you a starting point to work from, and one that is much better than any of those on-line calculators, because you will get an understanding of what you are doing and why. Generally speaking, the best method of designing a crossover is to measure the frequency and impedance response of the drive units in the cabinets, enter this data into modelling software, and design the filter that way. That is a big new skillset in itself, and not for the faint-hearted. Once you have familiarised yourself with the thread I linked to above & feel a bit more confident about the basics, you may then want to move on to read these pages, which will give you an idea of how you can use some simulation software (good) without necessarily investing a large amount of time & potentially money into measurement equipment.
https://sites.google.com/site/undefinition/diy-faqs-provendesigns
https://sites.google.com/site/undefinition/simulated-measurements
A lot there I know, but I hope it helps a little.
Hi,
Just copy the Fried Model H, i.e. Inductor and capacitor
on the B200 and a series capacitor to the speakers.
However a word of warning : a passive sub x/o
requires the whole thing to be designed as a whole.
The model H x/o probably won't work well with
a LS3/5A type speaker, the sub being too loud.
It may or may not work well with Quad electrostatics,
certainly the Quads input impedance is very different.
There is no such thing as as a universal passive subwoofer.
It must match the speakers properly or won't work well.
Do you have the correct version of the B200 for the coffins ?
TL Links
The alternative is making an active subwoofer and high
passing the speakers at line level, active or passive.
If you want flexibility on levels and x/o point, go active.
rgds, sreten.
Just copy the Fried Model H, i.e. Inductor and capacitor
on the B200 and a series capacitor to the speakers.
However a word of warning : a passive sub x/o
requires the whole thing to be designed as a whole.
The model H x/o probably won't work well with
a LS3/5A type speaker, the sub being too loud.
It may or may not work well with Quad electrostatics,
certainly the Quads input impedance is very different.
There is no such thing as as a universal passive subwoofer.
It must match the speakers properly or won't work well.
Do you have the correct version of the B200 for the coffins ?
TL Links
The alternative is making an active subwoofer and high
passing the speakers at line level, active or passive.
If you want flexibility on levels and x/o point, go active.
rgds, sreten.
Due to the radically changing impedances as you go this low, and the size of the parts needed, getting a passive XO to work well is extremely hard if not "impossible". The cost of parts will very likely exceed a reasonable plate amp.
I cannot advice strongly enuff that this is bi-amp territory.
dave
I cannot advice strongly enuff that this is bi-amp territory.
dave
If you are a newbie to XO, getting a miniDSP and a pair of amps will let you play with and experiment will all of these XO types and slopes on the fly. It will not cost much more than passive components for a stereo Linkwitz 4th order XO. It's like having an infinite set of perfect XO's to experiment with. You will at least have some hope of making a XO that sounds decent on your first try.
Many thanks to all for their tips and guidance - particularly to Scottmoose, I have started reading the thread you linked and I am learning a lot. The detail in your post has also clarified a fair amount, in particular I think I now understand the significance of the different "orders".
Following sreten's suggestion, I am now minded to experiment with the bass "part" of the Model H crossover since I get the impression that I could simply swap over the Inductor and Capacitor on the leg exiting this section to feed the satellites. I guess that I could use resistors or a variable resistor to attenuate the woofer level? I will have to think about this, does it make any sort of sense at all?
planet10 - you mentioned Bi-Ampling in my other thread but as I said there, I don't understand how this would be set up or why?
xrk971 - I have thought about active crossovers before. They seem to be incredibly expensive and whilst I don't doubt that they have tremendous potential to modify the crossover points they do just seem OTT tome - sorry.
Many thanks to everybody, I really do appreciate your time and patience.
Following sreten's suggestion, I am now minded to experiment with the bass "part" of the Model H crossover since I get the impression that I could simply swap over the Inductor and Capacitor on the leg exiting this section to feed the satellites. I guess that I could use resistors or a variable resistor to attenuate the woofer level? I will have to think about this, does it make any sort of sense at all?
planet10 - you mentioned Bi-Ampling in my other thread but as I said there, I don't understand how this would be set up or why?
xrk971 - I have thought about active crossovers before. They seem to be incredibly expensive and whilst I don't doubt that they have tremendous potential to modify the crossover points they do just seem OTT tome - sorry.
Many thanks to everybody, I really do appreciate your time and patience.
Here's the thing with subs. At these frequencies the room is in control, so you don't cross them like mids and tweeters. Sound bounces around and creates patterns. A low frequency driver can only be in one place and so the room 'modes' it will link into and cancellations or peaks it creates elsewhere should be 'crossed' by another driver covering some of the same frequencies and balancing these spatial abberations. In short, you may want the satellite
to cover some of the same frequencies as the sub or subs. You measure your system while moving the sub about in space, level and/or phase.
Active can be simple. Instead of using a capacitor in series with your tweeter you put it at the input to the amp. The capacitor and amp can be smaller giving you some options for the treble, and the speaker impedance isn't a factor. This is usually called a passive line level crossover and requires a separate bass amp.
to cover some of the same frequencies as the sub or subs. You measure your system while moving the sub about in space, level and/or phase.
Active can be simple. Instead of using a capacitor in series with your tweeter you put it at the input to the amp. The capacitor and amp can be smaller giving you some options for the treble, and the speaker impedance isn't a factor. This is usually called a passive line level crossover and requires a separate bass amp.
On the contrary, they are less expensive if you ever make more than one XO. I have just priced a passive XO for a two way second order Linkwitz Riley for a FAST with XO at 350Hz and parts (trying to use budget caps and coils) cost $87 from Parts Express. If I had upgraded to the next mid level caps and coils and resistors, price would have doubled to about $170. If using boutique caps and coils - price goes through the roof. Upwards of $500 easily.
A miniDSP with 2x4 XO plugin and shipping is $125. If you have an amp already then get an additional class D amp for $30 and you are set. Even buying two dedicated class D amps is not too bad.
Now the moment you want to change a XO point and new parts are needed the cost of the active stays fixed while passive increases with each new XO design.
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TLS.org | Passive Line-Level Crossover
This is really cheap -- the cost of connectors & a box are more than the parts inside usually (if you build a separate box)
You would approach bi-amping in one of 2 ways:
1/ pre-amp > XO > Amp1 (driving sats) +Amp2 (driving woofers)
2/ drive sats as normal, use a plate amp to drive woofers. Take off for the woofers can be hi-level or from the speakers.
1 is usually preferred since it has a high pass on the sats. But REL recommends the 2nd is simplier and it can be used in situations where 1 is not possible.
dave
Hi,
There is a 3rd option, high pass the main amplifier to the satellites.
Then from the speaker output build an inverse filter to drive the
line level sub input with a flat frequency response for the bass.
Works well with low powered integrated (valve) amplifiers.
rgds, sreten.
There is a 3rd option, high pass the main amplifier to the satellites.
Then from the speaker output build an inverse filter to drive the
line level sub input with a flat frequency response for the bass.
Works well with low powered integrated (valve) amplifiers.
rgds, sreten.
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Can you give more details - sounds interesting but I don't know how to do what you say.
Hi,
Say you decrease the amplifiers input coupling cap so the main speakers
are rolled off @ 100Hz, the speaker level input into the sub is useless.
With a passive @ 100Hz inverse filter you can drive the subwoofer.
The gain comes from the difference in speaker level and line level.
rgds, sreten.
Calculations don't work for imperfect (all) drivers so there is no point in choosing the type of crossover for the final design at this time. It does gives a hint of what will sound ok when you listen to the calculated filters.
After you find a type of arrangement that sounds ok you start playing with the values until you reach a certain type of sound. After you measure with equipments and judge for yourself if the deviations from a straight line response need to be corrected or if it sounds best with certain defects left unchecked.
I find that without a reference speaker/system it is nearly impossible to build a good sounding crossover. All the maths, listening, log book notes and response curves lead me to a dead ends.
After you find a type of arrangement that sounds ok you start playing with the values until you reach a certain type of sound. After you measure with equipments and judge for yourself if the deviations from a straight line response need to be corrected or if it sounds best with certain defects left unchecked.
I find that without a reference speaker/system it is nearly impossible to build a good sounding crossover. All the maths, listening, log book notes and response curves lead me to a dead ends.
I take your point about the potential benefit of the miniDSP route. However, I suspect that experimenting with / utilising one may not be significantly less challenging than assembling a passive crossover for drivers with known parameters? I haven't done either yet so I may well be wrong.
I have had a look at the availability of the miniDSP in the UK. It is available for £95 from a company called Chromapure. To this you would need to add a PSU, probably about £15 ($25).
I'm not sure about Amps but I have seen a "TDA7498 100W+100W Class D Amplifier Board DC20V to DC36V"on eBay for about £18, once again one has to add a PSU (32V) - say £15 again.
The cost therefore of a miniDSP, a pair of Class D Amps and all associated PSUs comes to about £176 (say $270).
However, this is not the end of the cost, realistically one would need to add a UMIK-1 at £80.
I accept that this configuration would probably offer a great "learning opportunity" and eventually, if everything works properly, a fine active crossover. However, one would still be out of pocket to the tune of some £250 ($400).
The only bit of this that (I think) I can claim to understand is the first option where I take it that one would be placing a crossover (handling low voltages) after the Pre-Amp and before a pair of Power Amps - effectively equating to powered speakers? Can you explain how and why this would make sense?
As ever, thanks to all for the education 🙂
passive first order crossovers are very tolerant of small errors in component values (tolerances).
As you increase the order of the filters from 2nd to 4th, the errors multiply and the resultant filter+driver is very intolerant of typical component errors.
High quality manufacturers will measure every driver and every critical component in their inventory. Then use a computer to select the components to the drivers to get good "pairs" that match each other AND get an overall performance that meets their tolerances.
This difficulty with using normal components limits us to first and second order passive filters, unless we are able to measure and adjust each drivers output to match "OUR" specification.
Forget passive 3rd & 4th order filters.
A filter has a few quite onerous conditions to work as predicted/expected.
The Source impedance should be near zero ohms, quite easy to meet with SS amplifiers.
The load impedance should be infinitely large. This is impossible to meet with normal 4ohms and 8ohms drivers.
Instead we adjust the filter to match the actual load impedance and that brings in a further condition. The driver impedance must be constant over a wide range of frequency around the filter turn over frequency.
This requires a lot of skill on the part of the filter designer to meet. There are tricks that help get the first or second order filter to work as expected/required. It is not easy for the high pass filter, it is a little bit easier for the low pass filter.
Using an active filter for the bass only driver's low pass and another active filter for the satellite's high pass filter is a much easier solution.
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Thanks AndrewT, that has helped further my understanding in relation to the "order" of filters, I probably would have aimed to go for a 2nd order anyhow.
I get the feeling that like xrk971 and perhaps also planet10, you believe that "Active" is the way to go? As it happens, I had a further thought about complications associated with this: I have a fairly powerful Desktop PC in the study and a laptop which might be almost anywhere in the house. I doubt that either of them include a decent sound card which I presume would be essential to use the UMK-1 & REW?
I get the feeling that like xrk971 and perhaps also planet10, you believe that "Active" is the way to go? As it happens, I had a further thought about complications associated with this: I have a fairly powerful Desktop PC in the study and a laptop which might be almost anywhere in the house. I doubt that either of them include a decent sound card which I presume would be essential to use the UMK-1 & REW?
The miniDSP can be powered from USB connection while being programmed or any cheap $5 USB power supply like for phone charger. It also handles up to 24v so can tap a amp power supply. The TPA3116D2 is $15ea and laptop 19v 4.6amp smps is $6ea. I have built a whole 40watt/ch active system for $150. You can order miniDSP direct from miniDSP website and shipping is $25, so total is $80+$25+$10(plugin)+$2(volume pot)=$117. Amps $30 total shipping included. You may have laptop supplies already if not $12 more.
The miniDSP software makes it very easy to program with GUI.
If you want to hear how good they sound listen to my sound clips on FAST projects or even Trynergy horn.
The miniDSP software makes it very easy to program with GUI.
If you want to hear how good they sound listen to my sound clips on FAST projects or even Trynergy horn.
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