Erin,
It seems like an extremely well balanced thread with lots of contributors giving well thought out balanced views on the benefits and shortcomings of both approaches.
I don't use digital actives just analogue, I do not recognise the issue with a supposed lack of "grain free" sound, quite the contrary!
Hi Thoriated, I won't go into too much detail here as it will be pulling the thread off topic, It sounds like perhaps I had already through experimenting in the sim found some of the "tricks" that could result in me removing my 2khz dip. The problem was that they resulted in impedance dips to a bit under 2 ohms. I'll post a pic of the various impedances. Black is the simulated impeadance of the current crossover (I haven't measured it since I added in the notch at 1.4Khz) Dark blue is the simulation of the new crossover. The lower overall impedance is due to the new low DCR coils I've just made. The 4k notch currently has a coil with ~300m ohm dcr. The other two are the raw (actual) impeadances of the individual drivers. Current crossover is 3Khz 2nd order bessel. New one will be 2.8Khz 4th order bessel.
Abrax, yes the dip is in the simulation. I've not yet done the new crossover. I wound the last two coils today, and still have two to dip in varnish. I've found speaker-workshops simulation to be very close to actual results (at least for spl) I've not compared simmed vs actual impedance.
1st pic impeadances, 2nd pic simmed crossover (black) vs actual measured crossover blue. I guess this could come under the trials and tribulations of passive crossover design (a + for active) but It is getting off topic, so if you want to discuss more I should open another thread
edit: you can see from the spl curve why I added in the notch at 1.4K also I know the impedance curve looks like a roller coaster, but thats because of the two notches on the midbass'
Tony.
Abrax, yes the dip is in the simulation. I've not yet done the new crossover. I wound the last two coils today, and still have two to dip in varnish. I've found speaker-workshops simulation to be very close to actual results (at least for spl) I've not compared simmed vs actual impedance.
1st pic impeadances, 2nd pic simmed crossover (black) vs actual measured crossover blue. I guess this could come under the trials and tribulations of passive crossover design (a + for active) but It is getting off topic, so if you want to discuss more I should open another thread
edit: you can see from the spl curve why I added in the notch at 1.4K also I know the impedance curve looks like a roller coaster, but thats because of the two notches on the midbass' Tony.
Attachments
Last edited:
Well, I did read all 18 pages, and it seemed that most contributors were in favour of active crossovers. I was mostly addressing digital active crossovers in my last post. Analog active crossovers eliminate the digital disadvantages.
Analog active crossovers will address the losses in the passive crossover, but not likely to address phase issues and also wont address frequency peaks and dips. Most analog active crossovers use opamps which not all people are happy with. All types of crossovers appear to have advantages and disadvantages.
I'm quite content with passive crossovers for the moment. A passive crossover means that you only need one good amplifier. Active requires more amps as well as the active crossover in the signal path.
There are many ways to get excellent sound. Passive is IMO a more "purist" way to achieve it.
I think it comes down to which school of thought you come from. And also your own experience with active vs passive crossovers.
Like I wrote earlier, I have no doubt that digital active crossovers can result in a flatter in room frequency response, but whether this is all that matters is very much open to debate.
I am not criticizing anyone who uses an active crossover, just airing an alternative viewpoint.
Last edited:
That's true but you can also do hybrid crossovers where you have the actual crossover point in the active and then use passives for any response tailoring. An example of this would be using a compression driver with a CD type horn and doing the compensation with a passive network. When you biamped the JBL 4430 using the the JBL 5235 crossover with 4430/35 cards that's what you ended up doing. I have my compression drivers in my HT done this way.
Rob
Last edited:
While I do not favor of one approach over the other as I see advantages to each, the idea that analog active crossover won't address peaks and dips is certainly incorrect. If you are limiting the term "active analog" to off the shelf type active crossovers using text book filters, then yes, I agree. But active analog to me means designing an active crossover which does what it needs to using gyrators to deal with peaks and dips, shelving filters to deal with baffle step or dipole eq, pole shifters, LP and HP stages with poles staggered as required, etc.
Even in the digital domain, if IIR filters are used there is very little digital can do that active analog can not other than a pure time delay. Digital is a little easier because all that is done is programing the filter to emulate the required transfer function. With analog, once the transfer function is defined it becomes a circuit design process, active or passive.
No mystery AFAIK. I recently saw an electronics industry article that enjoined designers to use X7R dielectric capacitors for commercial audio instead of the normal Y5V, Z5U run typically specified for audio coupling & decoupling 'to minimize audible distortion'. The capacitors under discussion were basically all barium titanate types.
Last edited:
So I have a pair of balls on my forehead, and I can tell you I am quite happy with them.
Please don't make the mistake to equate active xovers with DSP. Analog active is the way to go for high end quality.
vac
how do you implement passive BSC in a 2way, with active xo
Coil bridged by resistor in series with the driver, but it would seem a bit clumsy to do so given that you can realize a low shelf filter with 1 opamp and 3 passives.
vac
Well, it's *a* way to go to get high end quality. The sonic degradation the active analog circuitry imposes has to be weighed in the context of the overall application. For instance, if I were to, say, use an op amp based active EQ, I might as well go SS all the way for the amplification as well.
That is a bit of a cliff hanger, but I would still prefer the xover analog for at least three reasons:
1) most likely not all sources will be digital, at least, in my case there still is vinyl and the occasional airwave.
2) gain structure. I like to have my amplification as close to the loudspeakers as possible (no wire beats cryogenic treated silver anytime), as well as the xover. DSP is best if fed with a well managed signal, so you can't put a volume control in front of it.
3) latency. DSP works by remembering part of the signal, analyzing it, and calculating the desired transformation. That works great for 1 channel. However, the latency in doing all this is not similar for 2 DSP's. They will have ever so slightly different latencies, typically increasing with the dissimilarity of the two channels. Since interaural phase differences are a main cue to the perception of stereo image, this is not what you want. Analog does not suffer from this. This is not just theory, but also something I verified in listening tests.
vac
Erin,
You most certainly can address peaks and dips just as John K says in his reply. The mistake Diy-ers could make is slapping in off the shelf crossovers, this is equally problematic for passive speaker design too, know thy drivers.
Certainly excellent speakers can be made with passive crossovers, this is self evident, otherwise passive would have died off completely years ago, other than at the cheap end of the market. The vast majority of the best speakers are passive. This is mature technology, just as the best CRT TV's can still produce superb images.
I do not believe passive is going to be the best way to design speakers forever, but it may be the best way now for many, such as for the more simple diy projects and affordable retail speakers, it may also be the best way for the current generation of speaker designers whose many years of valuable expertise lies in passive design.
For DIY experimenters active is becoming a very welcome tool, of course it does not have to be the only way. It will be interesting to see how the balance changes over the years.
Passive is ultimately less flexible (when cost is less of an issue) also active is actually a more efficient system (when properly designed, if that is a desired feature) despite the suggestions of those who moan about multiple amp's etc. However perhaps the real world gets in the way for many diy-ers who will use multiple traditional metal box amp's (their old hifi plus extra boxes!), hence potentially less efficient systems.
People are continously looking for better or at least equal results in ever more compact packages, when high quality and compact is the requirements for your design active operation is going to be the better solution.
I know the purist worries about non discrete components but basically they are obliterated when you hear a speaker with a properly equalized crossover as good as Pluto (and it needs it with the driver in a tube!), I have no doubt this applies to the heavily equalized Orion with it's filthy, common op amp's too.
No need to feel threatened, passive design is completely dominant in retail Hi-fi and simple solutions suit diy better so there is a good few decades in passive design yet, who knows maybe it will always remain so.
If I heard a speaker that I felt was superior and would suit my needs I would go with a passive again, of course why not, all my other speakers have been passive but my sacred cows of "discrete design required" or for that matter "SE valve amp's required for great sound" bubbles have been well and truly burst.
Which brand of DSP did you listen to which did this? I think we should all be told so we can steer well clear
Which brand of DSP did you listen to which did this? I think we should all be told so we can steer well clear
It is inherent to the technology, so it would be very unfair to mention a brand name.
vac
This I think is wrong. Can you back it up with measurements?
My own DSP outputs the bits for all channels at exactly the same time, since they all use the same clock.
Latency itself is of course an issue, but not if you dont have to sync with video. I'm happy with the 1 second latency of my own DSP.
Once I have a speaker design optimized, I would just as soon finalize component values and call it a wrap. That way, much of my work is not potentially wasted if I transfer the result to a new environment, on top of the fun things passive xovers can do that actives can't such as voltage boost. Plus, I'm thinking of my 8 speaker HT setup, for instance which are decent standalones w/o any external xover. 16-24 channel amplification, anybody? Yow! And as far as taking advantage of digital EQ, why settle for half a loaf? I would want a full digital environmental EQ, not just an 'active xover'.
So, I resolutely have a foot in each camp.
Last edited:
This is really not true at all.
Yes, DSP systems introduce latency, typically on the order of 1-2 ms however any decent stereo / multi-channel digital DSP system will have identical and deterministic propagation delay down to a single sample for each channel regardless of equalization adjustments. The designers of DSP processors are not morons.
True, if you are mixing and matching different channels on separate pieces of DSP equipment which are different brands/models (which will probably have different latencies) or have some speakers going direct or via analog EQ and some via digital DSP you'll run into problems with differences in latency, something which will catch out the unaware.
Most decent DSP systems will also allow you to add additional adjustable latency down to a very fine degree in case you do have some need to use two different pieces of DSP equipment in the same system on different channels. (I would still only ever use a stereo DSP for left and right channels though)
Double blind tests ?
Did you also measure the phase shift and time delay of two channels on a stereo DSP to see if there was a discrepancy, or are you only going by a sighted listening test ?Besides, it's important to realise that inter-aural phase differences are only used up to about 800Hz by the brain to determine direction, above this frequency its amplitude differences and HRTF together that provide left-right localization.
Last edited:
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.