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Considering the option of "DSP speakers"

“Go home or go big.”

I notice occasionally somebody would come in here and say something to the effect “go active” or “go DSP” or something as cryptic as “FIR”. I am more or less old fashion – big caps, big coils, and stuffs. But with all the DSP “come-ons” recently so I went lookup “what if I do DSP, what would it take, how much would it cost” … something like that. I also looked at some of the commercial self-powered speakers to see how they do it.

First of all, “DSP” is not bad as some would automatically think of it. It's the “cheap DSP” that is bad. Let's say it you were to do it, what are your options?


  1. The most simple way is to buy a “miniDSP 2x4” which is about $100 bucks. It has two analog RCA inputs, one for left and one for right channel, and four outputs, one for tweeter and one for woofer for each channel, and 2 channels as a 2way. You can download the software plug-in which is pretty simple to use for xover filter works. If you want 3way or 4way speaker, you just need to purchase two “miniDSP 2x4” modules – one module for left and one for the right speaker. The problem is you still need to purchase two 2-channel amps (or 3 or 4) that are small enough to fit inside the speaker cabinets. Most people would go with class-D amp since they are affordable and small enough to fit inside the speakers. Or if you already have an existing receiver with 7-channel amp for example, then you can use that for your amplification, albeit outside the speakers. The first problem is you still need an pre-amp and take the pre-amp output to connect to the “miniDSP” RCA input. But the REAL problem of the “miniDSP” is it would have to convert the input analog to digital by its built-in ADC to perform DSP then converting it back to analog using its built-in DAC, so it's kind of a double back and forth. For only $100, I guess the quality of the ADC/DAC probably have to be compromised. I am under no illusion to think a $100 unit will perform the same way as a $7K dCS unit. But for a low cost solution, I guess this is acceptable and if that's all you want. Personally, because of the serious compromise in sound quality, I would only use this setup for xover development.

  2. The next thing to improve is to eliminate the built-in ADC stage. Luckily, miniDSP has something called “miniDSP 2x4 HD” which is about $200, which can either take the Tos-Link digital output of your CDP or it can also take the USB output from your computer. For the Tos-Link input, I am not sure how you can control the volume control (unless your Tos-Link source has volume control), but at least you can use the USB input and control the volume from your PC or your streaming device. But even without the ADC stage, you you still have to deal with the DAC stage because the “miniDSP” still has to convert the digital back to analog, and again, being only $200 in cost, I have to assume the quality of the built-in DAC is somewhat compromised, but this time at least, you eliminate the build-in ADC stage. And same as in #1, you can either purchase a class D amp and install it inside the speaker cabinet, or using an existing 7-channel receiver. The problem with the “miniDSP 2.4 HD” is you can't buy another one to do 3way or 4way and also use the digital input, because you can't “split” the USB into two, and since the USB protocol can only allow one master, one slave at any one time, you can't “split” them. And if you use the “Tos-Link” input, I guess if your transport has two Tos-Link outputs (or your reciever has some pass-through capability” then you can purchase a second “miniDSP 2x4 HD”, then you can do 3way, or 4way speaker as long as you can provide enough amplifications.

    (BUT make sure the digital “Tos-Link” output does not have any difference in latency or delay within them. It is not something that can be sure of. If there is delay or latency between the two Tos-Link, it will definitely affect the sound since the timing of the left and right channel won't be the same.)

    Or you can just purchase a “miniDSP 2x10 HD” which has 10 outputs and has one digital input but the “2x10 HD” will cost about $500 but you only need one digital input so that would eliminate any possible latency issue. But still your digital source needs an ability for volume control to act as a pre-amp. I think if you have a PC connected to the “miniDSP” module all the time, then you can control the volume from the “miniDSP” software plug-in, but I think the point is not having any PC plug-in at all, so you can independently control the volume even without the PC.

  3. At this point, if for whatever reasons you hate the “miniDSP” corporation, you can simply purchase a set of Hypex modules, either 2way or 3way from Madisound and it has everything you need, including a built-in class-D amps and DSP processing. They are small enough that you can build them into the speaker cabinets. You still need to provide some kind of pre-amp control though, and just as with “miniDSP”, it takes analog RCA input and Tos-Link input and it also has a Tos-Link pass-through so you can connect that to the second speaker. This would eliminate the ADC stage, but you still have to deal with the built-in DAC. And of course you have to be mindful of the class D amp. Looking at the amp, it has a few tiny capacitors for power supply bypass compared to some huge caps on my separate amps and everything looks puny :-) And these things are not that cheap. A 2x4 module is about $500 and you need two so total is about $1000. That's definitely not CHEAP! (And that not including the speaker drivers).


So far, everything is simple enough. The only real reason to worry is with the built-in ADC and DAC, and therefore the sound quality could be compromised, but if you're looking at an affordable solution, then I guess all is good or good enough. Again, personally I wouldn't go with either 1, or 2, or 3 either unless I use them purely for my own speaker xover development works. Some commercial DSP speaker only use class A/B amp for the tweeter since the tweeter is sensitive to the noise of the digital amp.


4.  So what if you want to go DSP, but still want some really good quality sound? First off, at least you have to eliminate the built-in ADC/DAC all together. Use external DAC. Use external quality amplification. Luckily, miniDSP does offer a nice little “miniDSP NanoDiGi 2x8”, which does everything in digital domain. It has a Tos-Link input and has 4 digital SPDIF channel outputs (or 8 channel total). I believe miniDSP also offers a balance version so all inputs and outputs are balance, but being digital it's not that big deal. There is still a concern of the jitter of the SPDIF outputs and that can affect the sound quality on the downstream equipment. MiniDSP does offer more expensive “all-digital DSP” processing solutions but it costs quite a bit more (about $500 I think).

But still you have to provide some type of external DAC and amplification. At this point, things start to look a bit complicated (which ironically DSP promises the opposite). For a 3way speakers, you need 6DAC's, a 6-channel amplifications. There are off-the-shelf purely DAC modules that can provide multiple channels, and of course the good one will cost quite a bit of money. Or if you have purely digital amp, that is class-D amps that take the SPDIF input directly so you can actually eliminate the DAC stage all together. Just plug the SPDIF outputs of the “miniDSP” outputs directly to the amp. Again, it's the cost again. And I wouldn't be surprise if you want a better SPDIF amp, the more it will cost. Another practical thing to consider is how will these amps will fit inside your speakers or if you decide to have all the amp outside? Finding a plate SPDIF powered amp that will fit inside the cabinents can be difficult – a practical consideration. I can already imagine a bunch of cables criss-crossing!

At this point, it's no longer a straight forward “plug-and-play” but you probably have to do some research, but still it's probably not that bad.  As for sound quality, as for the performance vs cost trade off, if you just go with some run-of-the-mill external DAC, class D amp, then it why just go with option 1, or 2, or 3 and save your money.  Sure you can get the best external DAC or amplifications but then again it all comes down to cost, and if you go all the way with these, I assume it will cost a lot of money and gets complicated indeed, which goes back to the beginning, that is , is it worth it? Then why not just go with good old analog? I think you can only go so far with “plug-and-play” off-the-shelf solutions though (unless you want to turn your living room into an equipment rack). Which leads to #5.


5.  I think truly to have good DSP system is if you go “Meridian” way, that is to go “BIG” and I mean it as both figuratively and literally. Figuratively, you really need to have some real R&D, a real lab, and hiring real engineers to develop the hardware from the bottom up and everything is custom made – from the digital stage, DAC and amplification and electronic xover works with the drivers at hand. Literally, “Meridian” was referring to some of the stuffs from Meridian and I think those are the only true high-end DSP speakers. And if you can develop your own hardware in-house, then you can scale up your design and in the process, save cost but having optimal performance. But this would exclude any chance of DIY.

I've also looked at some sub-$1000 DSP speakers, and Harman Kardon “Citation Tower” for about $3000 and Elac Navis floor stander for around $2500. Some of them actually uses only class A/B amp for the tweeter, and the class D for the woofer so I guess they realize the difference in amplification quality. Anyway, the only truly high end DSP stuffs that I've have seen only come from Meridian. The others I would characterize them as “life style”, and I am sure they provide good quality sound but I wouldn't call them high end – at least not at the same level as Vandersteen or Sonus Faber, B&W, or Magico.

Most high end speakers are still analog, And I think the main reasons why there are not many out there because it costs a lot of engineering money to develop a good system and only a few big companies can afford it. You can't just plug-and-play and call it high-end. Of the off-the-shelf stuffs I've seen, I don't think they are up-to-snuff. But I also think more and more will have some type of DSP in the future and that is just inevitable.








andy2
andy2 OP
1,522 posts
 
Colonel Kurtz "You don't like my methods?"
Captain Willard "I don't see any method."

Since we're on the topic of "DSP", with regards to some of the methods, it seems like some of them try to correct the entire frequency spectrum, which maybe a little overdone.  I've listened to a few and although it is a bit interesting and it does make the sound a little "clearer" with "tighter bass", but something about the treble that doesn't sound right.  You could see right away the sound is somewhat processed not not quite "real" - makes for some nice party music I guess..

I think room correction should be done in the context of the bass frequencies.  Not the mid range. And certain not the treble.

Someone mentioned Vandersteen.  I think that may be preferable.  They only try to "room correct" the bass which usually is most of what people want.  

golfmd2
27 posts
 
There are other high end fully acive speakers out there.  Dynaudio, Kii among others.  I have the Avantgarde zero 1 xd pros.  Fully active, time and phase aligned with room correction options.  Awesome speakers.  Very High end. 
andy2 OP
1,522 posts
 
An interesting excerpt from an interesting article regarding to "room correction:

An example will show this point more clearly. Consider a loudspeaker standing in a room. Mr A measures impulse responses in a certain listening volume and finds to his dismay that the magnitude response has a substantial broad dip at some rather low frequency, say 300 Hz, in all positions. He calibrates a peak filter and fills up the hole in the magnitude response, which is then confirmed by measurements. Enter Mr B. Mr B is a musician and he listens to the equalized system. “It sounds horrible! What have you done to the system!? It sounds all swollen and strange!” Mr A becomes nervous, as Mr B is an important customer, and calls his trusted friend Mr C. Mr C answers: “Ah, yes of course. The dip was really due to reflections. You should never
Dirac Research AB 5
boost any dip, because they are typically due to reflections.” So Mr A removes his equalizer filter and lets Mr B listen again. Mr B, however, is still not happy. “It is better, but it’s not good. There is something hollow about the sound.” At this time Mrs D enters the conversation. She’s been listening, sitting quietly in a corner of the room, and says: “Mr A was wrong because he forgot about the time domain. Looking only at the magnitude of the Fourier transform and interpreting it as strongly related to our concept of frequencies, he thought that he could boost that region and obtain better sound. The problem is that he uses minimum-phase filters and consequently adds energy at that frequency early in time. But if we only look at the direct wave there is no hole to be filled in the frequency response. The hole never exists if we look at a short window at any time.” Mr B frowns: “So Mr C was right to say that we cannot do anything about it. But if that’s the case, why do I still hear a strange sounding oboe on my recording?” Mrs D looks sternly at him: ”Mr C was wrong too. The problem is due to the time domain properties; the reflection causes the problem and it can only be corrected for by a time-domain approach. If we design a filter that reduces the reflection, you will end up with the interesting result that the hole will be gone and the oboe will sound more natural.” “But,” Mrs D adds, “don’t take this example as evidence that you can always correct dips this way! In this case it was possible, because all positions experienced the same problem.”

steve59's avatar
steve59
1,640 posts
 
Has it been determined that 24/96 is about as good as we can hear? If you get a dsp system that unfolds 24/196 mqa would it be considered mediocre. Also why wouldn’t dsp speakers be upgradeable? Meridian is doing it so why couldn’t other manufacturers keep up with the sota? Since the dsp system is optimized the need to keep searching for the perfect combination is done short of playing with power cords.

 I like changing my gear occasionally because it refreshes my music collection, but having a complete system that minimizes mistakes in the playback actually broadens the amount of quality recordings I can play and there’s still always the ability to move things around to tweak the sound. Idk if my dsp kit is my final system, but I’m really noticing the advantages.
andy2 OP
1,522 posts
 
One of the main motivation is that one can implement any forms of filtering - your imagination is the limit.  And of course, if you want to implement "time coherent design", using passive filter may be very difficult, but with "DSP", it's much easier.  That is your speaker can have 0 phase shift, proper step response and so on.  

As it happens that if you go with miniDSP solution, there is a program called "rePhase" that is run in Phython, that can be used in conjunction with miniDSP software, that will implement the xover that will give you "perfect 0 phase shift, time coherent design".  The only constraint is you speaker has to use "LR" filter such as LR2 or LR4.

Or if you're pretty good with DSP, you can implement yourself using the same strategy as "Bang and Olufsen" uni-phase approach which doesn't seem that complicated, that is as long as you have a background in filter theory and DSP.
Here is a link to the article the "B&O" approach.
https://www.tonmeister.ca/wordpress/2015/10/29/bo-tech-uni-phase-loudspeakers/

Assuming you have a three way speaker - woofer, mid, and tweeter.  Each will have it's own filter - and therefore will have its own filter transfer function. 

Tweeter transfer function: FS1
Mid transfer function: FS2
Woofer transfer function: FS3

So you have the total speaker transfer as : FS1 + FS2 + FS3 = FStotal.

Now most speaker with inverted polarity and so on, the total speaker response, FStotal, will have some phase shift - be it 180 degree or 360 degree and so on.  So in order for a speaker to have 0 phase shift, time-coherent response, FS1 + FS2 + FS3 will have to be equal to "1" that is:

FS1 + FS2 + FS3 = 1

Here is how "Bang and Olufsen " approached it:
First they implement the speaker as a two way - with just the woofer and tweeter - in this case, initially you have only FS1 + FS3 = FStotal.  But FStotal is not yet "time coherent" and FStotal is not yet equal to one "1". 

At this point, what "B&O" did was using FS2 (the midrange) as a sort of "tuning", to turn FStotal to "1".

In term of equation, here how it goes.

FS1 + FS2  + FS3 = 1   (This equation is made to be "1".  We will solve for FS2)

FS2 = 1 - FS1 - FS3.  (Here is what FS2 has to be, to get FStotal to "1")

So as long as you implement FS2 according to the equation above, that is:  FS2 = 1 - FS3 - FS1, your speake will have perfect time-cohernet response.

Of course, you probably don't have a three way design, but similar approach will get you time-coherent, if a little more complicated.


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