The design of active crossovers- Douglas Self wants your opinions

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Time Coherency

Doug,

I looked over your book's table of contents and your work appears to be EXTRAORDINARYLY well done. I will order it shortly.

Have you considered a chapter on acheiving time coherency in the composite system? I'm working with Roy Johnson of Green Mountain Audio on this topic right now for a future collective product. Would you consider working with us and perhaps have the opportunity to add a chapter in a future release?

-Jeff Jaska
Enliten Audio

:)
 
I have bought it as well.

What I like most about it are the noise and distortion examinations which are made from the theoretical and practical point-of view. I.e. for many gneral purpose blocks he made actual measurements of practical implementations.

Second good point is that there is a lot of information in just one book that one would otherwise have to search around for.

What I don't like about it is the fact that it doesn't really cover the topic about how a driver response is included into the crossover function. There are some EQ circuits described which one normally uses to straighten out driver irregularities. But there is no description how one would take the rollloff at the upper and lower end of a driver's response into consideration. Unfortunately also the carefully worked-out design example at the and of the book is just what one would call a "textbook crossover".

Transient-perfect (i.e. time-coherent) crossovers are treated within the book but not at a very deep level. There is some info on subtractive-delay crossovers and the classic constant-voltage crossovers are covered as well - but the latter only in their very basic form where the derived branch is 1st order. Higher-order or symmetrical ones aren't covered at all.

Regards

Chalres
 
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I bought it when it came out, and it is a very useful work.
Like you, I was disappointed that Doug seems (like most people) to be unconvinced by 'transient perfect' crossovers. Of course, such a crossover is very demanding on the out of band characteristics of the drive units, but my own listening tests have shown that the stereo image from a pair of speakers is noticably better when using such a crossover. I can only hear the improvement with stereo speakers. Listening to a single loudspeaker, I cannot hear any difference between a 'transient perfect' crossover and a conventional active one.

I guess everyone knows the excellent 2-way TP crossover design with a 2nd order roll-off on both the HF and LF units that first appeared (to my knowledge) in the National Semiconductor app-note for the LM833?
 
Hi,

I bought it when it came out, and it is a very useful work.

I don't know.

I only looked at the TOC and I cannot see anything that is not the ninth rehash of 1980's circuitry.

A much better treatment of generic active crossovers that included how design the driver response into crossovers was (is?) on Siegfried Linkwitz's site, sans obolus.

Active Filters

A number of excellent articles on electronic time alignment, transient perfect subtractive crossovers et al appeared at one time or another in Wireless World and Electro and many now also floats around on the World Wide Web, gratis. Here is one:

Three-way active speakers using chip amps.

I suspect for the basics of active crossovers Rod Elliots Website, accessible pro bono, may be a good starting point.

DIY Audio Articles Look under beginners luck...

Ciao T
 
I guess everyone knows the excellent 2-way TP crossover design with a 2nd order roll-off on both the HF and LF units that first appeared (to my knowledge) in the National Semiconductor app-note for the LM833?

I know it but have never used this one. I usually employ circuits that filter each branch seperately but give a response like this one.
The big advantage when using this kind of filters (so-called state variable) is that the driver response can be taken into consideration using the properly weighted sum of multiple outputs. And this should even work for both - allpas crossovers (like LR) and constant voltage.

Regards

Charles
 
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Hi,

Is Linkwitz = Sig

Mr. Linkwitz is originally German.

His first name is Siegfried (Victorypeace if literally translated to english), in german we often abbreviate the full name to "Siggie", to be pronounced loosely like Ziggie (as in Ziggie Stardust), if this abreviated further you end up with Sig...

Ciao T
 
I am a believer in using the nicknames we each adopt at signing up to be Members.
If Mr. Linkwitz were a Member here, I would used his nickname when referring to him.

I most certainly not use the names applied by his friends and family.
I would report anyone who revealed his real name.
 
Yes Siegfried Linlwitz. I met him at RMAF and discussed time domain crossovers for quite awile there. He's quite passionate about that subject at least for Sig. For what I got from the discussion, he feels you don't have a correct crossover unless you handle the time domain and I agree. From what I see, most crossover designs comercially (like $100K speakers!!!) ignore this and do not have time coherency.

It's time for that to change.
 
_ I just « finished » to read the Douglas’s book !
In 2012, like Wim, I agree that a home DIY digital crossover filter is out of the reach of most people, as DSP are in big SMD or PGA packages… plus the software development time & tools price !
And even DIYed in Digital, to reach the quality of an good Analog stuff … its not with a common DSP nor Software found ! IMO, “standard” digital in 2012 is ok for THX, MultiChannel AV, Stage Events, but surely not for an HiTech HiFi home system.

_ I think it was intended to be Crossover “all in one” book design, so all overall separate “building blocks” are inside, precise but concise ! (Each in Theory +Simu +DIY): so this is why you can find some parts here in other books…
_ Impressive work …. As i am an electronics engineer, but more focused in video processing, I found very interesting topics about loudspeaker’s enclosure & boxType configurations and location.
Also, I found a lot of valuable informations about THD in OAmp, Transformer’ s 0zOut & Capacitors.
(nothing about inductors excepted page12… it is an Active crossover book !).
Many different Filters & HiOrder AllPass types are reviewed, and the last remainers are well argumented.
Crossover summing results & Eq Filter Variable Notch are helpful.

_ A remark:
I did not find the Time GroupDelay of Summed 2sd Order LRiley, nor the Step Response of summed LR of 2 & 4th Orders, which could be interesting !
Also, nothing about AudioPro “Quasi Floating” output… (I know, it is a Home app book !)

Finally, it is a very instructive book, and I will keep definitively. :)
 
Loudspeaker Time Alignment Compensation

First, many thanks to Mr. Self for writing this excellent text. The book is currently the reference guide for the design of active crossovers. I recommend it highly.

I have a few questions/concerns in regard to Chapter 10: Time Domain Filters. Mr. Self recommends the introduction of time delays into the crossover outputs to correct for the physical alignment of drive units.

Siegfried Linkwitz also makes use of all pass, time delay filters in his analog crossover designs (ASP), and has gone so far as to claim that "Multiple sections may delay the tweeter output and compensate for the driver being mounted forward of the midrange. Active crossover circuits that do not include phase correction circuitry are only marginally useable."

However, it's well known in science that the human ear is very insensitive to phase. Given the complexities of normal listening environments, different sound frequencies will reach your ear through different paths. Per the "The Scientist and Engineer's Guide to Digital Signal Processing," from a physics standpoint, the phase of an audio signal becomes randomized as it propagates through a complex environment.

As expressed from Lenar Audio: "Small shifts caused by vertical crossover alignments are difficult to hear and measure. A trained ear can hear subtle differences of alignment shifts, during real time switching comparison only but not when stationary. That is, the ear is unable to pick which speaker is in alignment, forward or back"

Given Mr. Self's well rationed, objectivist, engineering, and science based approach to audio design in his previous works, he gives little attention to the theory behind the supposed necessity to introduce time alignment compensation. In addition, Mr. Self makes several references to Wikipedia through the text. I find this lacking for a serious engineering reference. At a minimum I would like to see original source references.

Can someone knowledgeable please give a science based answer as to why delay compensation does or does not make the slightest bit of difference for normal listening environments? I'm NOT interested in an audiophile, golden ears based subjective answer...... because I've tried altering the delay in my digital crossovers and I can't tell the least bit of difference in my living room.

Another design factor to consider when implementing analog time delay compensation / equalization: a slight change in listening position will cause a variation in frequency response. Also, any object in the room will alter the standing wave pattern, thus changing the frequency response at any particular location. Time delay compensation must account for listening distance and height, and also room temperature. All these factors are variable across listening environments and people, which makes me question why time delay is useful for any real world domestic situation.
 
Actually, the ear is very sensitive to phase in harmonics.

We can train ourselves to hear through the messes that we do..... and thus we actually are quite phase sensitive, in the final analysis. we can hear and differentiate in the nano and pico seconds of clocking jitter, so we are definitely phase sensitivity capable.

The problem is 99% of the people out there are programmed, wiring and experience wise, to hear through noise, to decode through noise.

Thus few people understand phase noise and work to correct it, as that is literally against the design of their internal wiring.
 
I find real time switching not so good for listening tests in general. If a certain part of a passage has been identified to reveal differences more obviously, then real time switching is a good way.

As I have mentioned in many occasions, the delayed release of stored energy in speakers have more influence unless it drops below some 15 20 db withing the first 0.3ms as shown in a CSD plot. Once you get to that ball park, other things start to show up.
 
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I fear you chaps have the wrong end of the stick. Time delays are used in crossovers to make sure that the sound produced by different drivers is in phase where the two drivers are producing the same signal, (in the crossover band). If not, you get amplitude anomalies via cancellation.
 
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