Why 2nd Order Is best or not

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You are correct about those little foam things. No, real base traps. Remember, this is DIY! I am talking about large ( 2 x 2 x 16') cavities built into my ceiling filled with rolls of insulation and a rear wall made of various different sizes of this sheetrock glued by the edges over tapered damped cavities.

We agree, that's real bass trapping. 🙂
(Note my "unless" clause above.)

Most of the time when people talk about "bass traps" they mean stupid little pieces of foam in the corner or whatever, that really don't do anything.

I use DSP for testing and tuning. Then I implement it in higher quality analog. Someday, there will be a DSP unit I can afford that is of sufficient quality, and I will gladly use it. My Bheringer is not.

Let's assume arguendo that the Berry box is not sonically transparent. Given that the AD/DA loop is not the answer (again, see Meyer and Moran), it may be useful to look at why it's not sonically transparent. There are two main options, inadvertent user error, or innate shortcomings of the analog circuitry.

The first place to double-check is your gain structure.

There was a good thread somewhere (I don't honestly remember where, may have been here on DIYA) where Bob Brines noted that the 2x4 miniDSP didn't sound sonically transparent, and proved it with distortion measurements. Long story short, it turned out he had too much gain prior to the A/D conversion, which was overdriving the chip. When he flipped the input sensitivity jumper on his little box, the distortion he heard and then measured went away.

With the Berry DCX2496, there's also the potential for noise getting in during balance/unbalanced interface changes, which may be audible as treble grain. If your whole system is balanced, that's probably not the case, though. Or the output stages might not be up to par. I can't speak to that unit, as I've never used one. I can only say that, unless the gain structure is messed up, nobody would be able to distinguish a flatlined miniDSP in the signal chain compared to the circuit with the miniDSP bypassed.

But the one thing we do know is that a single AD/DA loop is sonically transparent. And unless somebody can point to a similarly rigorous study as Meyer and Moran that comes to the opposite conclusion, any reasonable and educated person simply has to accept their findings.

For one thing, it by default filters at about 22K. Useless for investigating breakup modes of metal tweeters. Handy for not exciting them.

Given the paucity of program material with 22kHz signal, who cares?

A copy of UE, with a stack of decent D2A's is an idea, but streaming audio through Windows, a non-real time OS, just has too many issues.

I've been a Mac guy since the TiBook of 2001, so my knee jerk reaction is that any use of windows just has too many issues. 😉
 
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Quite true Steve, the Le will limit the top end. So does mass, rigidity, and physical dimension.

Visaton Visaton - Lautsprecher und Zubehör, Loudspeakers and Accessories does seem to have some interesting drivers. They don't seem to have wide distribution over here, but they make a very wide product line. E-Speakers carries a few of them. Go figure. The little FRS may be a decent wide range mid like I was mentioning, crossed over to a real tweeter. I am going to look into them further. B80 looks to have possibilities for that 600 to 6K mid I am looking for. ( if anyone knows a full line international distributor, please speak up)

When you say "almost perfect", I fall back on my core argument. Almost perfect for what? A specific set of requirements they meet very well for you. I may have very different requirements that the referenced system will not be suitable for at all. I am not at all disparaging the design, just pointing out that speakers are a set of trade offs we select to meet the needs of a specific project. Every project is different. " It Depends"

If there was anything close to "perfect", we would all know it and have only one or two Asian rip-off companies making every speaker in the world. As there are thousands, clearly we don't have anything close to perfect anything.
 
There are several DCX threads here with a wealth of info and different approaches to modifying it. Let's just say the digital section is NOT the problem.

You might also cast a glance at my "What is Gain Structure?" article in the articles section.

Please don't get me wrong. I am not anti-digital. I don't dislike my DCX. It is very good for it's intended use, and quite useful for quick prototype. I just don't consider it quite "hi fidelity" to my ears.

Many issues, but this thread is about why a second order crossover does not fit all applications. I would rather get back to why slope is more dependent on the actual drivers that some easy rule of thumb that says just use second. There are plenty if issues involving phase and distance to discuss. We might actually comment on how easy it is to switch between slopes with the DCX for prototyping, Q vs center point asymmetry to deal with bumps, dips, BSC and how to translate that to passive crossovers.
 
" It Depends"

If there was anything close to "perfect", we would all know it and have only one or two Asian rip-off companies making every speaker in the world. As there are thousands, clearly we don't have anything close to perfect anything.

I don't quite agree. There are designs where changing anything makes it worse. That means it's perfect, doesn't it? 😀
http://www.diyaudio.com/forums/multi-way/212926-what-makes-sound-stick-speakers-8.html#post3032880

Nothing wrong with that tweeter. Off-axis it does fine and these are bookshelves so you're not going to toe them in. The slight peak at 14kHz is just a beaming thing from the horn-like cone.

Change to reflex? Why? It'll have slower bass.

Change box dimensions from 0.618:1:1.618 aka 5:8:13? That is optimum to minimise resonance. Anything else is worse.

Change the crossover? No, it uses voicecoil inductance and Zobel capacitance to near perfection. And to get away with second-order is about as good as it gets on group delay. Phase and imaging is near perfect.

Free-standing? No, you lose efficiency and run into bafflestep and diffraction issues and the wife is going to hate it.

Now maybe you could find better low inductance drivers, maybe vented voicecoils too. Perhaps the optimum shape for a cabinet is an egg or a doughnut for all we know, but some designs just have a RIGHTNESS about them. All I did was crib the lovely bookshelf BBC LS3/5A really. 🙂
 
There are several DCX threads here with a wealth of info and different approaches to modifying it. Let's just say the digital section is NOT the problem.

You might also cast a glance at my "What is Gain Structure?" article in the articles section.

You might add some comments about the differences in adjusting level at different points within the system between analog and digital methods. It could be eye opening to some how the end to end process from mic to your ears can result in far less than 16 bits of resolution.
 
I don't quite agree. There are designs where changing anything makes it worse. That means it's perfect, doesn't it? 😀
http://www.diyaudio.com/forums/multi-way/212926-what-makes-sound-stick-speakers-8.html#post3032880

Nothing wrong with that tweeter. Off-axis it does fine and these are bookshelves so you're not going to toe them in. The slight peak at 14kHz is just a beaming thing from the horn-like cone.

Change to reflex? Why? It'll have slower bass.

Change box dimensions from 0.618:1:1.618 aka 5:8:13? That is optimum to minimise resonance. Anything else is worse.

Change the crossover? No, it uses voicecoil inductance and Zobel capacitance to near perfection. And to get away with second-order is about as good as it gets on group delay. Phase and imaging is near perfect.

Free-standing? No, you lose efficiency and run into bafflestep and diffraction issues and the wife is going to hate it.

Now maybe you could find better low inductance drivers, maybe vented voicecoils too. Perhaps the optimum shape for a cabinet is an egg or a doughnut for all we know, but some designs just have a RIGHTNESS about them. All I did was crib the lovely bookshelf BBC LS3/5A really. 🙂


We are actually in agreement. For your intended use and the set of drivers selected, quite possibly a good design. You have used the set of tradeoffs to advantage for you. What is perfect for you just may not be perfect for me. My house has 6 stereos in it. All with different speakers. All have different criteria. In one room, the sub is further away than another. Different issues. Two of them do not have subs. Different choices. One of them is mostly only background music. Different issues. One of these days I need to build a very high efficiency bookshelf so I can see how much progress I have made on my 6W tube amp. Totally different problems from where I have a bank of Rotels, Parasounds, or my Rotel and Denon AVRs.

I find Anchor Porter to be the "perfect" winter brew. Others prefer a Miller Lite, a choice I almost can't comprehend.
 
Many issues, but this thread is about why a second order crossover does not fit all applications. I would rather get back to why slope is more dependent on the actual drivers that some easy rule of thumb that says just use second.
Yep. The actual acoustic slopes, the phase, the out of band stuff is what's important (not to mention harmonics). Those are easy to tweak and adjust on the DCX. It can make a world of difference. Forget these cheesy "1 simple rule!" "Try this one stupid trick" adverts. There is no "1 trick" for crossovers.
Sure, there are some simple recipes that can work, but careful attention to the crossover works much better.

It could be eye opening to some how the end to end process from mic to your ears can result in far less than 16 bits of resolution.
Sure. I used to worry and fret about digital volume control. But it was eye and ear opening to me when I started doing blind listening tests and rigorous measurements. All I could find was a loss of S/N ratio, nothing else. No harmonic or inharmonic distortion - nothing. Of course I don't do more than about 10dB of digital attenuation. I just don't worry about it any more. Sorry for the OT.
 
I suspect I'm just extremely retro at heart. I'm trying to recreate those nice little bookshelf speakers like the Wharfedale Linton and Melton that gave me so much pleasure in cramped accommodation when I was at College. 😀

Now since we are talking about second-order filters here, has anyone had any luck at all lately with series crossovers?

An externally hosted image should be here but it was not working when we last tested it.


I decided that second-order was the most interesting one, but damned if I can get them to behave even with some "top-secret" Fried designs at my disposal. Every time I iron out a wrinkle somewhere, another one pops up somewhere else. 😕
 
Come on Steve, you can uncover the secret and put it into todays terms.

I was always trying to recreate the sound of pre Thiele/Small oversized closed boxes driven by valves, and how did I find it? By starting with what 'most' people would consider to be 'right', and adding my own 11 secret herbs and spices 😉
 
Please don't get me wrong. I am not anti-digital. I don't dislike my DCX. It is very good for it's intended use, and quite useful for quick prototype. I just don't consider it quite "hi fidelity" to my ears.

Many issues, but this thread is about why a second order crossover does not fit all applications. I would rather get back to why slope is more dependent on the actual drivers that some easy rule of thumb that says just use second. There are plenty if issues involving phase and distance to discuss. We might actually comment on how easy it is to switch between slopes with the DCX for prototyping, Q vs center point asymmetry to deal with bumps, dips, BSC and how to translate that to passive crossovers.

Most likely tweeter will have second order for improved HF, a robust midrange/bass driver gives you more flexibility.


I don't quite agree. There are designs where changing anything makes it worse. That means it's perfect, doesn't it? 😀
http://www.diyaudio.com/forums/multi-way/212926-what-makes-sound-stick-speakers-8.html#post3032880



Change to reflex? Why? It'll have slower bass.

Sorry cannot agree with this , i have found it to be the opposite ....
 
Change to reflex? Why? It'll have slower bass.

Sorry cannot agree with this , i have found it to be the opposite ....

It's simple physics, my friend. All things being equal a closed box 2nd order butterworth bass rolloff has less group delay than the sharper rolloff of a 4th order reflex.

When you pull the speaker away from the wall, you then need to apply bafflestep correction aka bass boost. The coil that does that also adds group delay, as illustrated for this 3kHz 2nd order filter.

Bottom line is the modern freestanding bass reflex has slow bass. You'll hear this on drums particularly, which lack punch! Put a sock in your reflex and try it. 🙂
 

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When you pull the speaker away from the wall, you then need to apply bafflestep correction aka bass boost. The coil that does that also adds group delay, as illustrated for this 3kHz 2nd order filter.
Not so.

It's not valid to look at the group delay caused by the baffle step correction circuit by itself.

The loss of bass due to the baffle step effect is a minimum phase phenomenon, and the BSC applied in a crossover which accurately corrects this loss of bass is also minimum phase. Fix the amplitude response, and the phase response and therefore the group delay will also be fixed.

The group delay of the speaker with accurate baffle step correction will be better than the one without correction, not worse...
Bottom line is the modern freestanding bass reflex has slow bass. You'll hear this on drums particularly, which lack punch! Put a sock in your reflex and try it. 🙂
Well first you have to define what that nebulous audiophile term "slow bass" actually means. Contrary to popular opinion it has nothing to do with whether it's bass reflex or closed, it's more about the in room tonal balance between low bass and mid/upper bass.

Too much low end bass and not enough mid/upper bass sounds "slow", not enough low end bass and too much mid/upper bass sounds artificially "fast". It's a phenomen that can be fully described by looking at the actual in room amplitude response at the listening position.

Those that believe that the extra resonant pole of a bass reflex cabinet tuned to an adequately low frequency (<~40Hz) is to blame for "slow" bass need to read a study that was done to determine our sensitivity to high Q resonances in the bass, which found that ability to detect resonances drops rapidly below 100Hz and is almost non existant below about 50Hz, and that the only thing we detect below that is a peak in the amplitude response, not any ringing from the Q at roll off. (This article was linked in another thread, I'll see if I can dig it up if anyone is interested)

In other words a bass reflex design whose tuning frequency is significantly below 50Hz (below 40Hz to be conservative) is perceptually no worse than a closed box design provided that the low end response is flat and without peaking before roll off.

Of course a speaker with a flat extended bass response will typically have an in room response that rises towards the bottom end often with some peaking in the amplitude response at the lowest room mode due to room effects, and this can lead to a slow sounding bass response. In this case the closed box with its earlier more gradual roll off can sometimes neatly compensate this room gain, if you're lucky, so that the in room response is flatter with less peaking at the bottom.

But you can also fix this with EQ without throwing away the dynamic range benefits of a bass reflex design...or if you like, choose an early gradual roll off bass reflex alignment instead of a maximally flat one.

There is nothing magical about a closed box just because it is a second order system...get the in room frequency response balance right at the listening position and you won't have "slow" bass with either a closed box or a bass reflex.
 
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I really don't like wasting time on the sort of nit-picking that goes on here, Simon. Far rather do new and innovative stuff. And SO SHOULD YOU! 🙂

I think a picture is worth a thousand words some times, which is why I post so many:

An externally hosted image should be here but it was not working when we last tested it.


Anyone who knows filters knows which of these two curves has more group delay. The sharper one. The reflex. N'est ce pas? 😀
 
Easy System7, balshy behavior, isnt a good idea.The greater question is HOW MUCH group delay is audible. I know fron experience in making.fixing playing with guitar FX pedals that, in a chorus pedal, a delayed version of the signal is mixed with the original. A range of delays is used. The point where the doubling effect is audible, to me, is around 8 to 10 ms. Its likely that the lower in frequency the less sensitive to its effects.
 
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