Well yes - of course I missed something:
8.) due to the nature of time delay being created by 3D, 100% correction is limited to a single point in that 3D space
there simply is NO method to 100% correct for CMP distortion at more than a single point in space - which may be of no big concern anyway - just for the record
Michael
8.) due to the nature of time delay being created by 3D, 100% correction is limited to a single point in that 3D space
there simply is NO method to 100% correct for CMP distortion at more than a single point in space - which may be of no big concern anyway - just for the record
Michael
Interesting Gizmo
Nice that it can optimize in time and frequency domains, and independently correct for loudspeaker FR and phase distortions while treating room modes separately.
Very different than the auto-equalize gizmos seen in HT receivers and midgrade prosound gear, which are often little more than digital implementations of parametric equalizers, with no FIR (time domain) correction at all.
Trinnov Multichannel Optimizer
and
Trinnov Audiophile 4-channel Optimizer
Disclaimer: No connection to Trinnov, but it looks like an interesting application of auto-EQ technology. Thought you guys might be interested in what the commercial sector is up to.
Nice that it can optimize in time and frequency domains, and independently correct for loudspeaker FR and phase distortions while treating room modes separately.
Very different than the auto-equalize gizmos seen in HT receivers and midgrade prosound gear, which are often little more than digital implementations of parametric equalizers, with no FIR (time domain) correction at all.
Trinnov Multichannel Optimizer
and
Trinnov Audiophile 4-channel Optimizer
Disclaimer: No connection to Trinnov, but it looks like an interesting application of auto-EQ technology. Thought you guys might be interested in what the commercial sector is up to.
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Hello Lynn
It's not only in the Pro domain. A version is available is some of the newer surround pre-processors. Many of the older ones use Audyssey which is similar.
Rob
Stereophile: Music in the Round #42
It's not only in the Pro domain. A version is available is some of the newer surround pre-processors. Many of the older ones use Audyssey which is similar.
Rob
Stereophile: Music in the Round #42
I think we need to review your conclusions.
1.) We are facing CMP behavior anytime when we have to deal with constructive and destructive interference caused by virtual sources – be it by reflections, diffraction, looped echos or whatever
The constructive and destructive interference of the "whatevers" have nothing to do with MP.
2.) CMP behavior itself is not dependent on frequency
What does this even mean?
3.) the most basic characteristic of a CMP system is that there is time delay involved
While not technically required I will agree that in most cases a time delay is involved.
4.) as a direct consequence of the delay time involved, we are facing a frequency response that depends on the time we look at
Again, not true. In fact, all your models are linear and time invariant. "Time invariant" should be a clue. The frequency response of a linear time invariant system is the FFT of the system impulse. You are confusing the transient behavior and the effects of stored energy with frequency response. As I said in my previous post, the impulse response defines the response to any input. And as I have also said before, any two system which have the same impulse response will have the same frequency and transient response. It doesn't matter how that impulse is generated. It could just be a system with funny impulse or it could be the result of a summation of multiple impulses. And it doesn't matter if it is MP or not.
5.) as a direct consequence of the delay time involved, we are facing a time span - at the beginning and at the end of any sound reproduced - that causes CMP distortion if not corrected
Same deal, any transient distortion arises from the system transfer function regardless of how that transfer function is generated.
6.) there is a method to 100% correct for CMP distortion
No. If a true null (100% cancellation) exists it can not be corrected. 1-1 = zero. A x (1-1) still = 0. regardless of the magnitude of A.
1.) We are facing CMP behavior anytime when we have to deal with constructive and destructive interference caused by virtual sources – be it by reflections, diffraction, looped echos or whatever
The constructive and destructive interference of the "whatevers" have nothing to do with MP.
2.) CMP behavior itself is not dependent on frequency
What does this even mean?
3.) the most basic characteristic of a CMP system is that there is time delay involved
While not technically required I will agree that in most cases a time delay is involved.
4.) as a direct consequence of the delay time involved, we are facing a frequency response that depends on the time we look at
Again, not true. In fact, all your models are linear and time invariant. "Time invariant" should be a clue. The frequency response of a linear time invariant system is the FFT of the system impulse. You are confusing the transient behavior and the effects of stored energy with frequency response. As I said in my previous post, the impulse response defines the response to any input. And as I have also said before, any two system which have the same impulse response will have the same frequency and transient response. It doesn't matter how that impulse is generated. It could just be a system with funny impulse or it could be the result of a summation of multiple impulses. And it doesn't matter if it is MP or not.
5.) as a direct consequence of the delay time involved, we are facing a time span - at the beginning and at the end of any sound reproduced - that causes CMP distortion if not corrected
Same deal, any transient distortion arises from the system transfer function regardless of how that transfer function is generated.
6.) there is a method to 100% correct for CMP distortion
No. If a true null (100% cancellation) exists it can not be corrected. 1-1 = zero. A x (1-1) still = 0. regardless of the magnitude of A.
Reply to your review, John
Ad 1)
*How* the virtual source is created is not subject of the discussion here – the results concerning CMP are simply not affected – so the „whatever“ has only be seen in this context
Ad 2)
That – for example in case of OB where you seem to have the most difficulty to agree with me – my conclusions regarding CMP behaviour apply one octave below dipole peak just as well as anywhere else in the frequency band.
Ad 3)
I'm happy we partly agree
But furthermore CMP *always* is connected to delay – its in the naming itself : *consecutively* min phase behaviour.
This implies and focus on the fact that a min phase source - like an ideal speaker – changes its properties dramatically when virtual sources of that speaker enter the picture.
The most dramatic change : correctability is out of reach forever, though *theoretically* possible.
Which means: we have to live with CMP distortion, no matter what...
Ad 4)
I hold my position until you show that my measurements presented (the sine bursts) are flawed.
Here most clearly is demonstrated that until the virtual source enters the picture – of course – there is difference in amplitude. As the difference in amplitude is not the same throughout the frequency band (due to the nature of comb filtering) we have to put it just like I did.
To put it simple: comb filter effects do not happen immediately and in the time interval before comb filtering happens (the delay time between original source and virtual source) we are facing the speakers “natural” FR
Ad 5)
Agree – with changed signs.
Also – we are not discussing non-MP systems – at least not non-MP sources, so to speak.
Ad 6)
A paradox situation at a first glance, but I hold to my position as clearly shown in my simus. The point to focus at is, that the “summation to null” is a process *after some time* and not an immediately process (hence your math is incomplete).
Meaning we can add any signal we like at any time we like – the overlay towards “summation to null” will happen later - so “the signal of desire” can *always* be created until overlay with the virtual source happens.
As the overlay always happens later on, we can push and push and push just as may times we like.
That we will rapidly run out of steam is quite another thing (as said: ”practically” out of reach forever, though *theoretically* possible).
Michael
Ad 1)
*How* the virtual source is created is not subject of the discussion here – the results concerning CMP are simply not affected – so the „whatever“ has only be seen in this context
Ad 2)
That – for example in case of OB where you seem to have the most difficulty to agree with me – my conclusions regarding CMP behaviour apply one octave below dipole peak just as well as anywhere else in the frequency band.
Ad 3)
I'm happy we partly agree
But furthermore CMP *always* is connected to delay – its in the naming itself : *consecutively* min phase behaviour.
This implies and focus on the fact that a min phase source - like an ideal speaker – changes its properties dramatically when virtual sources of that speaker enter the picture.
The most dramatic change : correctability is out of reach forever, though *theoretically* possible.
Which means: we have to live with CMP distortion, no matter what...
Ad 4)
I hold my position until you show that my measurements presented (the sine bursts) are flawed.
Here most clearly is demonstrated that until the virtual source enters the picture – of course – there is difference in amplitude. As the difference in amplitude is not the same throughout the frequency band (due to the nature of comb filtering) we have to put it just like I did.
To put it simple: comb filter effects do not happen immediately and in the time interval before comb filtering happens (the delay time between original source and virtual source) we are facing the speakers “natural” FR
Ad 5)
Agree – with changed signs.
Also – we are not discussing non-MP systems – at least not non-MP sources, so to speak.
Ad 6)
A paradox situation at a first glance, but I hold to my position as clearly shown in my simus. The point to focus at is, that the “summation to null” is a process *after some time* and not an immediately process (hence your math is incomplete).
Meaning we can add any signal we like at any time we like – the overlay towards “summation to null” will happen later - so “the signal of desire” can *always* be created until overlay with the virtual source happens.
As the overlay always happens later on, we can push and push and push just as may times we like.
That we will rapidly run out of steam is quite another thing (as said: ”practically” out of reach forever, though *theoretically* possible).
Michael
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Agreed that the amplitude of the dipole and monopole radiation must be equal in the far field and that this would require an amplitude variation of the front and rear drivers as well as a phase variation because of the changing phase of the two components in the far field. This is shown in my book. And I agree that achieving a pure cardiod behavior would be complex (this too is shown). But I do not agree with your last diagram because it relies on point sources and is not true for a radiating sphere (all sources are at the same radius - there is no d/2 speration of the dipole components form the monopole). I think that our differences come directly from our models. You think in term of point sources seperated in space and I did my models as a sphere with sources on either side. These are NOT the same models and may not lead to the same conclusions. In that case, one has to look at which model is closer to reality. I think that this should be readily apparent.
That there is some transfer function F(w) that when multiplied by the rear source will yield an ideal cardiod polar response should be apparent. It's easy to show. It is also obvious that achieving this polar response will also require additional EQ to the "system" to make the axial response flat, so lets call this transfer function G(w). Hence, if we source the front driver with G(w) and the rear driver with F(w) * G(w) we will simultaneously achieve both a flat axial response AND a polar cardiod response - the ideal. Now how close can we get to this ideal if we 1) don't want cardiod down to low F's and 2) we want to do it passively. That I do not know, but its an interesting exercise. But I don't think that your comments shed any light on the answer.
Hello Lynn
It's not only in the Pro domain. A version is available is some of the newer surround pre-processors. Many of the older ones use Audyssey which is similar.
Rob
Stereophile: Music in the Round #42
I've been upgrading my HT setup - not easy trying to find equipment that is sonically compatible with the Ariels and the kind of DHT-triode sound I enjoy.
Just finding a Center speaker turned out to be a major project; I was dumb enough to set up two basic criteria: good dialog intelligibility, and decent enough at music that they would sound good in mono. I just wanted to buy the stupid thing, not run off with ANOTHER project. I mean, who cares about Center speakers?
The two criteria ought to simple enough, and simple enough to test: turn off all EQ in the pre/pro, set to Dolby Pro Logic II on Cinema mode so the processor crams most of the soundstage into the Center speaker, and listen to my favorite CD's to see if the speaker is at least as musical as an iPod or Karna's little $129 Sharp compact stereo. What could go wrong?
Bowers & Wilkins, not happening. The mass-market HT favorites with titanium-dome tweeters - no way. Sonus Faber Cremona - almost, but dynamics not as good as the Ariels. I was starting to get a little discouraged, actually - the competition for the Ariels seemed to be in the $10,000 to $15,000 bracket, not a happy discovery when I'm trying to find a commercial loudspeaker that more or less sounds like an Ariel.
I started looking at the $1000 price point, but nothing there was even remotely suitable - dialog was OK, but music in mostly mono? No. Most center speakers were really colored. My existing Tannoy M1's were really horrible, and I didn't see any point to spending a lot more and having something that sounded just as bad.
The only ones I heard that were similar in overall character to the Ariels - to my surprise - turned out to be the Dynaudios, which are not exactly cheap. The one I took home was the Contour SC X. Yes, I know, way more expensive than I originally intended. But I really did look, and nearly all of the market was aimed in a really different direction than what I was looking for. It's loudness, loudness, and more loudness for most of these folks - Competition Car Stereo in the home.
Maybe I was stupid in expecting music from a Center speaker - but, doggone it, I actually like surround music. I got my start in the industry with the Shadow Vector decoder, and I expect surround to actually sound good with music, not just dinosaur thumps, car explosions, and machine-gun fire. If I want that, I can set off a string of firecrackers in my living-room and get way more realism and SPL than a movie soundtrack.
It's turned out the same way for replacing the wretched-sounding Denon 2805. I've had that thing for nearly five years, and never liked it. The iPod Touch sounds better - granted, I use Sennheiser HD580 phones and uncompressed AIFF files, but still, an iPod is a pretty low standard. And much of the HT gear manages to rumba under that low bar with ease. Most HT receivers are not as good as mid-fi Pioneer, Sansui, and Kenwood receivers from the Seventies. No ability to drive a 4-ohm speaker. Low-rent opamps (like the JRC 4558, worst of them all). Feeble power supplies.
The middle line of Yamaha receivers actually power the analog op-amps from the digital +5V supply, so the "pre out" can only deliver 1.7 Vrms before entering gross clipping. By comparison, any opamp that is driven from the industry-standard +15V/-15V supply can deliver just short of 9 Vrms before clipping. Even worse, aside from the huge noise penalty of using the noisiest supply in the receiver, is the requirement for an electrolytic cap at every input and output of the opamp, since the supply is single-ended (like an iPod or car stereo). But no worries: many receivers below the $800 price point no longer have "pre outs" anyway.
But I'm glad to see the S/PDIF connection finally go away. Although the rapid changes in the HDMI spec has created chaos with the specialist audiophile manufacturers, a connection that finally supports lossless multichannel 96/24 is a big, overdue step forward. The lack of an industry-standard connection is part of what did in SACD/DSD and DVD-A's, and I'm glad to see that the better receivers and pre/pros support DSD and multichannel PCM over HDMI 1.3 and higher. Good. But - another downside - support for DSD and DVD-A is dwindling. It's starting to go away from many Blu-Ray players and receivers. You have to dig deep into the manual to see if support is there, and trust the vendor is telling the truth about the feature set.
I really wish the Audyssey had a "below-300 Hz only" feature. I don't like what I hear above 300 Hz at all - it sounds very much like a parametric EQ that's EQ-ing both direct-arrival and overall room sound at the same time, instead of a FIR corrector for the crossover time distortion. That's much more subtle, and that's not what I'm hearing. I'm hearing what sounds like frequency-based EQ changes, in the theater THX style. So my preference is "OFF".
Some people might see my enthusiasm for 300B direct-heated triodes and large-format horn speakers as incompatible with surround sound. That's a modern artifact of the split between the weirdos in the high-end and the mass-market HT market. My first exposure to Stereophonic Sound was in 1958, at a 70mm wide-screen theater in Osaka, Japan, at the international premiere of "Ben Hur". That theater used three Altec A4's behind the screen, six all-analog magnetic soundtracks, and all-tube electronics from the microphones to the mix desk to the theater amplifiers. When I saw 2001 in the Cinerama theater in Washington, DC, in 1969, that was also a theater with three A4's behind the screen, Altec tube amplifiers, and six-channel mag-track sound.
You could make a good case that movies from the widescreen era - the Fifties through the Sixties - really needs to reproduced through tube amplifiers in order to be heard as they were intended to be heard. The Crown DC300 gradually replaced the vacuum-tube Altec and RCA amplifiers throughout the Seventies, but by then 70mm and mag-track sound has disappeared. Movies reverted to low-fi mono optical tracks and much smaller screens in the new multiplex theaters. When Dolby Digital finally replaced mono (and stereo) optical sound in the Eighties, Altec Voice of the Theater speakers had been replaced with THX-approved JBLs, and theaters were entering the kilowatt realm of amplifier power. This was a completely different sound than the earlier era. Much, much louder, and a radically different tonal quality.
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Ahh, now I have some idea what the Ariels might sound like. I've listened to their 30th aniversary model (can't remember the exact name), and also the some of the Evidence series.
Rene says that all "in a box" HT receivers are crap, from many standpoints. His only suggestion was to look carefully at Outlaw for a controller and develop your amplifiers to taste.
In addition, I am experimenting with the M Audio 610 mixer, with full duplex digital at 24 192 and fire wire connection to a computer. Hands down the best Red Book DAC I have ever auditioned, right in the same category as Rene's Berkley for this resolution. I will know more about it's capabilities at higher rez in a month or so.
In addition, I am experimenting with the M Audio 610 mixer, with full duplex digital at 24 192 and fire wire connection to a computer. Hands down the best Red Book DAC I have ever auditioned, right in the same category as Rene's Berkley for this resolution. I will know more about it's capabilities at higher rez in a month or so.
Outlaw has replaced their own house-brand pre/pro with the Marantz SR5004 receiver. Based on that, and subtle recommendations on the HT-review sites - one-line allusions by the writer that they actually enjoyed listening to music on the Marantz - I tracked down a Denver-area Marantz dealer.
Almost against my will, I was quite impressed with the Marantz AV8003 & MM8003 combination. For some reason, they didn't sound like the Onkyo, Pioneer Elite, or Denon high-end products, despite similar appearance and build quality. As mentioned earlier, I didn't care for the Anthem products at all - way too much Class AB transistor sound for my tastes.
Dunno why these things sound the way they do. You certainly can't tell from build quality, or the useless power ratings. My guess is most are using very poor quality power supplies, the opamps are substandard, RFI interference from the TV and digital stuff is making a hash of the analog circuitry (ground-borne noise and radiative emission into adjacent circuits), and the power amps are very poor quality, akin to car stereo stuff.
The DACs are probably pretty decent, but then, you really have to go out of your way to find bad DACs these days, when 96/24 delta-sigmas for less than a dollar are the norm. I don't realistically expect to find a Burr-Brown 1704 ladder DAC in any of the equipment at this level.
I wonder if anyone is taking, say, a Marantz receiver and simply removing all the wires between the main PS capacitors and the power-amplifier sections. At a stroke, noise emission in the amplifier would drop, and all the rest of the remaining circuitry would be in Class A in terms of current draw on the power supply. You'd certainly want to have the schematic in front of you - and Marantz schematics are not easy to find, I've been trying to find some myself - and verify that voltages wouldn't swing too high to overpower and overheat the regulators for the low-voltage stuff.
A really crude fix would be a big fat power resistor to drag down the PS rails if necessary. Although an ugly solution technically, at least the resistor wouldn't radiate huge Class AB switching pulses into the rest of the circuitry. This is real problem, by the way - when I was Audionics, I could change the entire distortion spectra by just moving the power-supply wires around. The big induced current pulses are from Class AB switching and harmonics of 100/120 Hz switch noise radiated from the power trans + rectifier bridge + PS caps circuit loop. Shield and reduce the loop area of these two EMI offenders, and the amp gets much quieter. It doesn't measure all that different, but the sound quality changes quite a lot.
It would probably make the receiver a much better pre/pro just from a noise and PS perspective alone. Solid-state devices do not like EMI emission into the input nodes - they are pretty easily slewed with any signal that's going fast enough, and feedback doesn't work very well at frequencies beyond the audio passband. I suspect a big part of ultrafast solid-state is its improved immunity to all the RFI trash floating around the chassis of modern electronics.
That's the main reason swapping power cords cam make a difference to the sound - what you're really doing is changing the emission characteristics of a big antenna that's radiating a 100/120 Hz comb spectra into everything around it. Same story for those overpriced "power conditioners" - all they do is slightly improve the emission of RFI and EMI into adjacent electronics.
Almost against my will, I was quite impressed with the Marantz AV8003 & MM8003 combination. For some reason, they didn't sound like the Onkyo, Pioneer Elite, or Denon high-end products, despite similar appearance and build quality. As mentioned earlier, I didn't care for the Anthem products at all - way too much Class AB transistor sound for my tastes.
Dunno why these things sound the way they do. You certainly can't tell from build quality, or the useless power ratings. My guess is most are using very poor quality power supplies, the opamps are substandard, RFI interference from the TV and digital stuff is making a hash of the analog circuitry (ground-borne noise and radiative emission into adjacent circuits), and the power amps are very poor quality, akin to car stereo stuff.
The DACs are probably pretty decent, but then, you really have to go out of your way to find bad DACs these days, when 96/24 delta-sigmas for less than a dollar are the norm. I don't realistically expect to find a Burr-Brown 1704 ladder DAC in any of the equipment at this level.
I wonder if anyone is taking, say, a Marantz receiver and simply removing all the wires between the main PS capacitors and the power-amplifier sections. At a stroke, noise emission in the amplifier would drop, and all the rest of the remaining circuitry would be in Class A in terms of current draw on the power supply. You'd certainly want to have the schematic in front of you - and Marantz schematics are not easy to find, I've been trying to find some myself - and verify that voltages wouldn't swing too high to overpower and overheat the regulators for the low-voltage stuff.
A really crude fix would be a big fat power resistor to drag down the PS rails if necessary. Although an ugly solution technically, at least the resistor wouldn't radiate huge Class AB switching pulses into the rest of the circuitry. This is real problem, by the way - when I was Audionics, I could change the entire distortion spectra by just moving the power-supply wires around. The big induced current pulses are from Class AB switching and harmonics of 100/120 Hz switch noise radiated from the power trans + rectifier bridge + PS caps circuit loop. Shield and reduce the loop area of these two EMI offenders, and the amp gets much quieter. It doesn't measure all that different, but the sound quality changes quite a lot.
It would probably make the receiver a much better pre/pro just from a noise and PS perspective alone. Solid-state devices do not like EMI emission into the input nodes - they are pretty easily slewed with any signal that's going fast enough, and feedback doesn't work very well at frequencies beyond the audio passband. I suspect a big part of ultrafast solid-state is its improved immunity to all the RFI trash floating around the chassis of modern electronics.
That's the main reason swapping power cords cam make a difference to the sound - what you're really doing is changing the emission characteristics of a big antenna that's radiating a 100/120 Hz comb spectra into everything around it. Same story for those overpriced "power conditioners" - all they do is slightly improve the emission of RFI and EMI into adjacent electronics.
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They aren't exactly the same. The Dynaudios have stupendous dynamics for a low-efficiency speaker. I'm actually at a loss how the SC X manages to be as inefficient as it is: they claim 86 dB/metre/2.83V, with a 4-ohm impedance no less, which implies that each 7-inch driver is an astounding 83 dB/metre/2.83V. The Esotar tweeter has to be around 90 dB, if it's anything like the classic Esotar, so the low efficiency rests with the pair of 7-inch drivers. Maybe the voice coil is very long and mostly out of the gap; I dunno. The Ariel is a genuine 92 dB/metre/2.83V, so there's a big difference there.
The Ariels do not sound good on most transistor amps - flat and 2-dimensional. Even so, they are better than the Sonus Faber Cremonas, which had much more limited dynamics than I expected, based on the glowing reviews. Nice balance, no obvious and gross flaws, and attuned to acoustic music, but dynamically less than the Ariels by maybe 5 dB or so. Higher efficiency always helps with dynamics, both micro and macro.
The Dynaudios might have a slightly cooler balance, but I'm not sure about that. In terms of transparency and immediacy, they are quite similar - this was the big difference between the Dynaudios & Ariels compared to B&W, which I found very, very disappointing. The Kevlar midrange coloration was very obvious, and the crossover evidently doesn't bother to filter it out. The metal-dome tweeter had the usual generic metal-dome coloration in the upper treble. It just didn't sound all that good on the kind of acoustic music I like - maybe for recordings of heavy-metal rock concerts?
There are lots of high-end speakers where I just don't understand the appeal. Anything by Wilson, for example. Never liked the Watt/Puppy when it first came out, and I don't like them today. Wretched tweeter with obvious and impossible-to-ignore resonances, a colored and coarse-sounding midrange driver, grossly over-emphasized bass, and a crossover with poor driver integration. I guess the cabinets are good if you ignore diffraction. This is a speaker I just don't get, never mind the price. I don't see the appeal at $100 each, much less $20,000 or more.
Don't get the Magico Minis, either. A tipped-up minimonitor with constrained dynamics is what I hear, with some nasty upper-mid and HF resonances. No thanks.
My only conclusion is that audio reviewers have weird tastes, or at any rate, tastes that have little or nothing in common with mine. The kind of records they mention in the reviews is music I despise and wouldn't listen to if you paid me. Sonically, they're on a different planet than I am - most of the reviewer-speak is just gibberish to me. I've heard some of their own systems in their own homes, and I just didn't understand why they liked what they were listening to.
It gets embarrassing when I'm asked point-blank "how I like it". This has happened more than once; the dealer or reviewer just wouldn't give up, and kept pestering me to tell them what I really thought - well, I told them and didn't get asked back. Ever. This has happened several times, so I try my best to just keep quiet and not say anything at all.
In a way, it simplifies show-going. Most of the rooms drive me out within seconds. A few other rooms, the exhibitors throw me out when they hear my demo material. A few rooms might sound interesting, so I rush around and try to find them before the show ends. Maybe one, two, or three might sound really good. I try and give these a writeup when I find them. It doesn't mean that I'd necessarily like them in my own home, it just means it was a fun discovery at the show. A few that come to mind are Dr. Geddes' Summa, anything by AudioKinesis, the Orions, the Harbeth and Spendors, the Oswald's Mill Audio system, and a few others. All fun and enjoyable. The others, not so much, so I keep on walking.
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Twice a week I do volunteer work at a resale shop, where some little old ladies sort out the donations and put aside all the electronic stuff , and anything else that gets “plugged in“, for me to test.
Tons of Sony, Denon, Onkyo, Yamaha, & Pioneer HT receivers come through. Many of them are DOA ( and at a very young age), while the rest just sound dismal at best.
Even worse than the mediocre sound are the poor designs that are so counter-intuitive to good function. I‘m always wondering when I come across this crap, “whatever happened to intelligent industrial design?” I used to be able to figure out how to operate a receiver by just looking at it from half way across the room (like with a Marantz, for example). Now I need glasses and a flashlight just to find the basic functions, which, even then, might require several steps to get at.
Every once in a while something decent will come in. Last week it was an early 90’s Yamaha receiver, and one with a big power supply. (the model number is….hold on, let me get my glasses and a flashlight…..okay, it’s rxv870) The display light had gone out, so, as usual around here, someone just donated it. It’s in my office now and sounds super…the only decent sounding Yamaha that I’ve heard…out of hundreds. Thanks, in part, to a decent power supply.
Tons of Sony, Denon, Onkyo, Yamaha, & Pioneer HT receivers come through. Many of them are DOA ( and at a very young age), while the rest just sound dismal at best.
Even worse than the mediocre sound are the poor designs that are so counter-intuitive to good function. I‘m always wondering when I come across this crap, “whatever happened to intelligent industrial design?” I used to be able to figure out how to operate a receiver by just looking at it from half way across the room (like with a Marantz, for example). Now I need glasses and a flashlight just to find the basic functions, which, even then, might require several steps to get at.
Every once in a while something decent will come in. Last week it was an early 90’s Yamaha receiver, and one with a big power supply. (the model number is….hold on, let me get my glasses and a flashlight…..okay, it’s rxv870) The display light had gone out, so, as usual around here, someone just donated it. It’s in my office now and sounds super…the only decent sounding Yamaha that I’ve heard…out of hundreds. Thanks, in part, to a decent power supply.
I don't have to refute any of this. What I have said covers it all. 1) The output of a system is the input signal convolved with the system impulse. 2) Any MP response can, (in theory), be equalized to any other desired MP target response using MP equalization and the impulse of the equalized system with be that of the target.
If your results don't agree with those two statements then they are incorrect. You have pontificated for several pages on this subject and all you have done is to obfuscate what was my correct response to Lynn initial statement about equalization in the frequency domain making things in the time domain worst.
Backman showed that it is possible to use an all pass filter so that at low frequency the front and rear source sum in phase and as the frequency rises the all pass delay combines with the source separation to yield a cardioid, but as the frequency continues to rise the polar pattern continues to morph since the all pass delay is not constant. I have experimentd with Backman's approach and never found it very satisfactory. I would like to see what what your transfer function isas I seem to have missed it.
In the case of real drivers, the rear (180 degree) response from the front driver is either equal to that on axis (wave length large compared to source and box dimensions, below the baffle step) or has some frequency dependence (above the baffle step).
If I consider the figure to the left below,
representing a speaker box with front and rear mounted drivers. The circle represents the polar response of the front source at frequencies where the wave length is much greater than the box dimensions, so the response is omni-directional. If I look only at the response at 180 degrees, then the question is what would the rear driver have to radiate to null the response at 180 degrees? Obviously the amplitude would have to be equal to the amplitude of the front source at 180 degrees, which is just the amplitude at the front source since it is omni at this frequency. The rear source's phase would have to be 180 degrees relative to the front source plus a delay defined by the additional propagation distance the front wave must travel around the box. This is on the order of d/c. This is the classic two source cardioid generated using two omni source (conventional drivers at low frequency).
Now looking at the figure to the right. Here we are looking at the polar response of the front source at higher frequency where the box shadows the response. So the polar pattern is no longer omni. It is weaker to the rear. At 180 degrees, to null the response, the rear source would again have to match the back side amplitude from the front source with inverted phase (now less than the front source on axis) with the addition of the same delay due to propagation distance differences. Of course, the polar response of the front source will change as the frequency changes once the box shadows the front response, much like a disk at the end of a long pipe. In any event, I would agree that it would be possible to high pass a rear driver so that at low frequency only the front, omni polar response would be obtained and at higher frequency, where the box shadows the front response the rear source could have its amplitude tailored to null the 180 degree response. But a delay would still be required due to propagation distance differences and while the 180 degree response could be nulled, the radiation pattern of the rear source would have to have very unique characteristics to create a classic cardioid, or so it seems. And the rear source would potentially still have some effect on the on axis response of the system, thus requiring additional eq as you note.
But over all this seem very little different than just choosing a baffle size so that the transition form 4Pi to 2Pi radiation occurs over the frequency range of 100 to 200 Hz, no?
I don't know, I'm not sure that I understand the comment. In my model this transition occurs naturally so its not something that can be gleaned by looking at the curves.
What I did was to model a piston in a sphere - complete polar and frequency response at all far field points. Now take this model and add to it (superimpose) the same model rotated by 180 degrees, this represents the forward and rear facing sources (equal in area for simplicity, but this is not required). By finding a transfer function for the rear source that exactly causes the rear radition to be zero, everything is accounted for, including diffraction (2Pi to 4Pi transition) etc. The exact solution does have delay requirements, but I simulated doing this without delay as well. You cannot get a pure cardiod without delay, but you can get some directivity.
Whats the advantage over an OB - clearly efficiency - it can be a monopole at LF so that there isn't the low efficiency dipole problem which requires seperate amps and EQ. Having now tested the Orion, the EQ and all the amps required are a real killer of value in that design. Is there a way to get the Orion directivity at LFs without the need for that extreme level of complexity and cost?
If I get some time I'll post the curves that I derived.
I guess the question I would ask is , over what frequency range? Clearly the piston in a sphere is omni directional at low frequency. So it seems to me, since you are interested in omni at low frequency, you are using the high passed rear driver to cancel the rear radiation form the front driver in the baffle step and above range. I.e. where the front piston does not radiate omni directionally. Be it a sphere or box, it seems you are looking at frequencies where the front driver directionality and size of the spherical enclosure (or box) are limiting the magnitude of the rear radiation of the front piston. No?
Certainly if I place a mic at 180 degrees and measure the response I can generate the inverse transfer function and HP filter it and correct the delay to null the rear radiation as a function of frequency. I'm just not sure what that would actually do to the on axis and polar response????
Certainly if I place a mic at 180 degrees and measure the response I can generate the inverse transfer function and HP filter it and correct the delay to null the rear radiation as a function of frequency. I'm just not sure what that would actually do to the on axis and polar response????
Thats in my book, but, as I said, if I get the chance I will post the figures.
Hello Lynn,
I have never found the 2 to be mutually exclusive, having high efficiency doesn't mean having bigger dynamics, interesting you should say so...
