That SB34 suffers from one fairly big flaw IMO, which is a small spider. Large excursions aren't limited so much by the surround, but moreso by the spider itself. They should have given it a larger spider, which would have lowered THD considerably at higher power levels further down low. That big 3" VC is sort of pointless if the driver can't remain linear to take advantage of the electrical power capacity.
The NRXL version of the SB34 is a bit better. The spider has slightly taller waves. allowing more linear suspension compliance at higher cone excursion levels. Many larger pro sound drivers have this issue as well.
Thats why I like the Eminence KL3012LF so much, as it can do more linear cone excursion as a 12" pro sound driver with 37 hz Fs. That's exceptional IMO for a higher sensitivity driver if it's type.
The NRXL version of the SB34 is a bit better. The spider has slightly taller waves. allowing more linear suspension compliance at higher cone excursion levels. Many larger pro sound drivers have this issue as well.
Thats why I like the Eminence KL3012LF so much, as it can do more linear cone excursion as a 12" pro sound driver with 37 hz Fs. That's exceptional IMO for a higher sensitivity driver if it's type.
Yep. I guess that says it all, for explaining the sims.Since the simulation has no input for the suspension limits, it considers it limitless
Oh me too, i would still be a stat die hard if I could get the dynamic bass I want out of them.
If using IIR, I strongly agree with your take on filter slopes.
Happily though, I've found with complimentary linear phase xovers, I can use any order i like with zero sonic penalty...in fact, i think with sonic gain.
I will always be an analog lover, but DSP with linear phase filters are the only way to EQ the low end properly without damaging the time domain. EQing and crossing the low end with analog filters never sounds that good and is incredibly hard to get right. The narrow banded filters required to do that would wreak havoc on the low end integrity having such abrupt phase swings.
Once you get into the 300 hz + range, EQ of any kind is verboten in my systems. That range has to be unmolested and pure. While its not easy, having pristine audio without any processing is the only way to obtaining the highest resolution possible from a system, especially at the macro-dynamic level. Some DSP tends to pre ring and suffer from unexplainable artifacts. It depends on the type of algorithm the DSP uses and how much processing it needs to accomplish on the fly.
Once you get into the 300 hz + range, EQ of any kind is verboten in my systems. That range has to be unmolested and pure. While its not easy, having pristine audio without any processing is the only way to obtaining the highest resolution possible from a system, especially at the macro-dynamic level. Some DSP tends to pre ring and suffer from unexplainable artifacts. It depends on the type of algorithm the DSP uses and how much processing it needs to accomplish on the fly.
Sorry for the excessive technical details - the room dereverberation artifacts are very explainable. the RIR is not invertable, so you end up playing tricks. It is much easier to do dereverberation with delay, but that adds pre-echo. According to the neuroacoustics it shall not be audible - but it is because everything depends on the level of your attention. There is an additional problem with the mic type - you shall not use an omni mic as the reference because your ears are not omni, and dereverberating the omni RIR does not help cardio-like ears. All in all, the idea of DSP preprocessing only sounds good before you try it. I killed a few years on blind dereverberation ... in most cases, it's a scam.
Hi,
I think that problem with eq. is not related to phase in most cases at all. Lets say, with e.q it's attempted to drop a peak in the room because of some resonance using a steady signal. Amplitude is corrected, but sound power is still distributed across longer time domain now with lesser initial value, so another problem is created.
Also many musical instruments are not theoretical point sources either, so phase distortions are in the recording already, maybe including reflections from recording space. Not to mention different distances to drivers in multi way systems, and reflections from walls etc. where phase can be reversed many, many times.
Regards
Those are great points. Regarding inaccurate simulation of large excursions, I'm sure Hornresp (which I've never learned to use) isn't the only guilty party. Most sim software will calculate excursion using TS parameters, which as I understand it are really only intended as models of small-scale behaviour. As you point out, BL will in reality be a curve, not a single number, and suspension compliance the same. Basta, which I play with a lot, undoubtedly is going to suffer the same inaccuracy as Hornresp, because it's calculating from TS parameters.
I'd love to think that this is right. But I used to have Linkwitz Orions, with Peerless/Tymphany XLS 10" woofers in an H baffle. With bass-heavy programme they would sometimes produce that horrible 'click', which I assumed was the coil hitting the backplate. I always backed off the volume quickly, and fortunately they never suffered any damage. Was it maybe not the coil hitting the backplate?
In the past I used stacked quad57 electrostats with Decca ribbon tweeter and subs. When "right" the midrange was extraordinary but the subs never integrated well enough with them. It masked the midrange with much program material. Ultimatley this was even with a pair of motional feedback subs with triple chamber dual 8" drivers per side (12 x 8" drivers in all) and even though the distortion could easily be heard dropping on sine waves in the 20 to 30 Hertz region (drastic reduction in audible doubling). The subs also contained a pair of parametric equalizers per side to drop out a couple of the primary room resonances. Ultimately these subs went to my brother in law for about 20 years until one failed, being used with satellites sitting on top. These subs integrating much better in a less resolved system.
Later I went to single 12" motional feedback per side in larger boxes and with some porting. These turned out much better, though required a much larger room than I had, just to begin to work. In a large loft, playing E Power Biggs classic Toccata and Fugue (the record) the vibrato in the lower registers was extraordinary that in a small room (and my equipment at the time) couldn't at all match. I do remember he was using Threshold gear designed by some guy named Nelson I believe.
Nevertheless I am currently in possession of six Electro-voice 15" drivers. A couple of "Force" 15" being planned for frequencies above 200 Hertz or so (with higher frequency drivers as well), and four Electro-voice DL15X's for lower frequencies. The plan is to use the DL15X's in a dipole configuration on each side for the low frequencies (more side to side) with a front facing Force driver, using DSP A to D, and D to A conversion and filtering on the low frequencies only. IMO analog networks for the lowest frequencies are highly problematic in being too inflexible to fine tune and adjust for variant room positions and it isn't clear if digital systems are at all problematic in this frequency range.
Later I went to single 12" motional feedback per side in larger boxes and with some porting. These turned out much better, though required a much larger room than I had, just to begin to work. In a large loft, playing E Power Biggs classic Toccata and Fugue (the record) the vibrato in the lower registers was extraordinary that in a small room (and my equipment at the time) couldn't at all match. I do remember he was using Threshold gear designed by some guy named Nelson I believe.
Nevertheless I am currently in possession of six Electro-voice 15" drivers. A couple of "Force" 15" being planned for frequencies above 200 Hertz or so (with higher frequency drivers as well), and four Electro-voice DL15X's for lower frequencies. The plan is to use the DL15X's in a dipole configuration on each side for the low frequencies (more side to side) with a front facing Force driver, using DSP A to D, and D to A conversion and filtering on the low frequencies only. IMO analog networks for the lowest frequencies are highly problematic in being too inflexible to fine tune and adjust for variant room positions and it isn't clear if digital systems are at all problematic in this frequency range.
Right, the TS parameters can't predict what the actual limits of a driver are, and Xmax is calculated differently by many manufacturers. Even if Xlim/Xmech is given, there is seldom an explanation of which part of the speaker will limit first- surround, spider(s), or voice coil former contact with the back plate.
As the Orion construction plans state:
"A separate amplifier channel is allocated to each of the 10" woofer drivers to obtain output capability that is commensurate with the midrange, yet minimizes bottoming and the risk of mechanical damage to the woofers, or having to reduce the speaker's low frequency extension. At very high sound levels the amplifiers clip first, which gives a clearly audible warning to turn down the volume."
Amplifier clipping may sound like a horrible click.
Peerless' GBS woofers state they are designed to allow full excursion without "bottoming", but I haven't seen that claim for the P830452 XLS-10.
From it's 12.5mm Xmax rating, the large 1/2 roll surround, and 35mm magnet depth, my guess would be the XLS-10's spider may hit the front plate before the coil former would hit the back plate:
A spider glue joint hitting the front plate wouldn't sound much different than a coil former striking the back plate, but would be less likely to cause damage.
That said, I prefer to push the cone of a woofer by hand to see and hear what and where the mechanical limits are to guessing
Art
Attachments
I didn't use the ATI amps that Siegfried was talking about - my amps were a bit more powerful, so were definitely capable of driving those XLS woofers beyond the quoted linear xmax. Which was silly, probably.. But I think I was far from being the only Orion ownerr who did this.
That makes so much sense. I've always been hesitant about getting hands on with drivers, but you're right, guessing is a lot more hazardous.Peerless' GBS woofers state they are designed to allow full excursion without "bottoming", but I haven't seen that claim for the P830452 XLS-10.
From it's 12.5mm Xmax rating, the large 1/2 roll surround, and 35mm magnet depth, my guess would be the XLS-10's spider may hit the front plate before the coil former would hit the back plate:
A spider glue joint hitting the front plate wouldn't sound much different than a coil former striking the back plate, but would be less likely to cause damage.
That said, I prefer to push the cone of a woofer by hand to see and hear what and where the mechanical limits are to guessing
Art
It's interesting to see how many people repeat the same concept of what "a good speaker/good sound reproduction" is. They mainly look at the RF, and assume that a "flat" answer says it all. Distortion doesn't make much sense, since all speakers distort, a lot.
Real life instruments too. Even if you are an excellent performer, no note will be "pure", because they emit a fundamental and its harmonics, it is not an audio generator.
An acoustic system - snare, OB, horn, etc. - from one or more speakers can have a non-linear response throughout the frequency spectrum, but sound very pleasant and real. The reason is that it is reproducing "tones" similar to real instruments, and they are attributable to the quality of the transducers, the design, etc. And one of the very important parameters is CSD analysis. And very few designers talk about it. Just my two cents.
https://audiojudgement.com/cumulative-spectral-decay-csd-plot/
Real life instruments too. Even if you are an excellent performer, no note will be "pure", because they emit a fundamental and its harmonics, it is not an audio generator.
An acoustic system - snare, OB, horn, etc. - from one or more speakers can have a non-linear response throughout the frequency spectrum, but sound very pleasant and real. The reason is that it is reproducing "tones" similar to real instruments, and they are attributable to the quality of the transducers, the design, etc. And one of the very important parameters is CSD analysis. And very few designers talk about it. Just my two cents.
https://audiojudgement.com/cumulative-spectral-decay-csd-plot/
Which in this context is of course totally irrelevant - one case is production, the other is reproduction.
//
CSD= Impulse=Frequency response. It is not a stand alone indicator: perfect SPL= perfect CSD. And CSD analysis certainly not a parameter...
You lacked mastery of time, the third variable. The amount of energy retained by the diaphragm when the signal ceases. Simple.
Ok, it's not a parameter. And what would you call it, spectral accumulation analysis? Find me a suitable synonym other than that one.
@Boden I am not sure I agree with this as a universal statement. I believe it is possible to apply EQ to a driver resonance, and achieve a flat frequency response. But this is not equivalent to a driver with no resonance. The FR of the two drivers may be the same, but the impulse response and CSD plot will be different. Perhaps you meant your statement in a certain context, and I have misunderstood your point. ?
j.
In my book as per Fourier this is near impossible: same FR - same impulse. Mathematically they are linked.
But: resonances will always show in the SPL as well. Look at what e.g. the Bodzio Ultimate Equalizer does, although some higher order modes in alu cones are near impossible to equalize.
Hi Jim, meant to reply to this a while back and forgot....
Thanks for the follow thru, checking excursion with VCad. Would appear it uses a limitless suspension too, huh?
And yep, you make good common sense with re operator usage, Pmax, and LT transforms.
If I may now switch gears to current topic in next post...
I'm in the same camp as Boden, i think it's just mathematical equivalencies we are dealing with, including resonances.
But we need to make sure when thinking about this, that frequency response = equals frequency magnitude response and phase.
I've found when doing super duper corrections, using pure impulse response inversions, that waterfalls / CSDs clean up just like mag and phase do.
Examples: here two sections of a coax CD on a horn, measured and corrected in a clothes closet.
Lower CD section Measurement; Impulse inversion with xover added; and Processed result
Upper CD section Measurement; Impulse inversion with xover added; and Processed result
Here's the two processed sections summed together. It was a little mind blowing to see such a clean acoustic output.
Sure, Conditions aren't real world: a. in a closet; b. pure impulse inversion w no smoothing, no FDW, no nothing; and c. to a single mic spot
But the fact these are acoustic measurements showing this kind of cleanliness, .....well it says to me the Fourier math is spot-on bingo!
And handles spectral decays too, I think..
