Omnidirectional speakers. The future?


I have been interested in hi-fi for about 25 years. I usually get the hankering to buy something if it knocks my socks off. Like most I started with a pair of box speakers. Then I heard a pair of Magnepans and was instantly hooked on planars. The next sock knocker was a pair of Soundlabs. I saved until I could afford a pair of Millenium 2's. Sock knocker number 3 was a pair of Shahinian Diapasons (Omnidirectional radiators utilizing multiple conventional drivers pointed in four directions). These sounded as much like real music as anything I had ever heard.
Duke from Audiokinesis seems to be onto the importance of loudspeaker radiation patterns. I don't see alot of other posts about the subject.
Sock knocker number four was a pair of Quad 988's. But wait, I'm back to planars. Or am I? It seems the Quads emmulate a point source by utilizing time delay in concentric rings in the diaphragms. At low volumes, the Quads might be better than my Shahinians. Unfortunately they lack deep bass and extreme dynamics so the Shahinians are still my # 1 choice. And what about the highly acclaimed (and rightly so) Soundlabs. These planars are actually constructed on a radius.
I agree with Richard Shahinian. Sound waves in nature propagate in a polyradial trajectory from their point of source. So then doesn't it seem logical that a loudspeaker should try to emmulate nature?

holzhauer
Dear Newbee - I wasn't really serious about the DSP. But I listen only to vinyl and the DBX (even a 3bx) is IMHO essential. I really have pretty good ears and have tried to find a fault with this thing and can't. It's the only add-on I have. The sound is soooo much improved. I haven't had anyone listen who wasn't floored by it. It's takes compressed analog (like almost every record ever made), and restores the dynamic range and punch that it had - there's no going back once you hear it. There's nothing magic about compression that can't be reversed with a proper algortihm.
I know many vinyl/audio snobs would have a kneejerk negative reaction to such a device, but then they've just bought a $5,000 tonearm to listen to highly compressed source material with numerous clicks and pops? It ain't that accurate to begin with.

Damned if you don't and damned if you do I guess....

Any Audiogoners in the SF Bay area who'd like to stop by hear it are welcome. You'll be on Ebay buying one within hours.

OK - I really have to get out of here this time...
Opalchip: A microphone picks up whatever is fed into it, both direct and reflected energy. It can't discern if the primary or reflected signal should should dominate as it can't differentiate between arrival times and their individual intensities. In effect, it becomes a recorder of acoustic activity at that specific point in time and space based on the specific pick-up patterns of the mic being used.

The Walsh driver simply re-radiates the energy that was captured at the mic as a point source and re-radiates it into the listening environment as a point source. The fact that the original ambient sounds heard during the recording could be heard at every point in the room, and are pretty much preserved and reproduced due to the pseudo-omni radiation characteristics of the Walsh design, is one of the most endearing properties of these speakers. The fact that there is only one driver acting as a point source for each channel reduces time / phase distortions to a minimum, hence the preservation of natural harmonic structure. This too is a very endearing quality of this speaker design. The effects of binaural recordings as heard on these speakers is pretty amazing.

Other than that, each musical note has a primary frequency and multiple harmonic frequencies. These harmonics vary in spectrum and intensity. Any device that tries to separate the audio spectrum into different segments will introduce distortions into each note reproduced. That's because the time, phase & amplitude of the primary note vs that of the harmonics will not remain cohesive in presentation.

As a case in point, the specific device that you mention is capable of expanding multiple different frequency regions at different rates. When doing this, it means that a harmonically rich instrument ( like a Cello ) that is centered in one specific frequency section may be expanded at a different rate than the harmonics, which might fall into one or two different frequency spectrums. As such, each spectrum is / can be expanded at different rates, which in turn varies the amplitude of the harmonics in respect to the amplitude of the primary notes.

The reverse of that is also true. That is, an instrument that covers a very wide range of the audio spectrum ( like a piano ) can have different levels of expansion applied to it across the entire band due to the spectrum segmentation that the device does as part normal processing. This would take place on both the primary note and the harmonics.

As such, expanding a compressed recording could only be done optimally if the algorythms used during recording and playback were exactly the same. Given that this is next to impossible given the differences in recording, mixing and processing techniques, the end results of attempting to expand a compressed recording can be very "interesting" to say the least. I will agree that "expanded" music sounds noticeably more dynamic and "punchy", but at the same time, it also has a certain "artificiallity" to it. On top of that, quite a bit of electronically generated music IS compressed, even when played live. Electric guitars, bass guitars, electronic keyboards, etc... are often processed in a certain manner with the musician specifically desiring certain sonic attributes that compression / clipping bring along with them. Trying to "undo" what was meant to be, both live and on the recording is nothing more than a distortion. These distortions may be pleasant on certain recordings, but it all boils down to a matter of personal preference vs articulate preservation of what is on the recording. Sean
>
Opalchip...The sound generated by a point source speaker becomes, within a few feet, a planar wavefront, just as you have described. By generating a planar wavefront to begin with the planar speaker is (neglecting for the moment any reflections) simulating a point source at a greater distance. As a result of this the SPL falls off much less rapidly as distance to the listener's position increases, producing (IMHO) a very stable and uniform soundstage throughout most of the listening room. This characteristic of planar speakers, more than their inherent bidirectional nature, accounts for the sound that some people like.
Opalchip, I gotta tip my hat to you for consistently holding to your convictions, even if they're very different from mine. I have a feeling your ideas are more the norm than what you're finding on this thread - I think you've stumbled into a hotbed of believers in planars and/or poly-directional loudspeakers (to borrow Dick Shahinian's term).

A comment about one of your arguments, if I may: While it is true that the microphone picks up hall ambience cues, microphones are usually placed much closer to the performers than listeners would normally be. So, relatively speaking, they pick up a much higher proportion of direct to reverberant sound than what a listener would hear in the same venue. This isn't always the case, but usually is.

Also, the direction from which reflections arrive make a difference in how they are percieved, and in most venues the reflections arrive from all around rather than from the exact same direction as the first-arrival sound. Reflections that arrive from the sides, and well delayed in time, are particularly beneficial in conveying a sense of ambience and acoustic space.

My father has done research in anechoic chambers, and he reports that music live or reproduced in an anechoic chamber has incredible clarity but also sounds dead and boring. I have not listened in an anechoic chamber, but having severely overdamped my listening room as an experiment let's just say I'm sure it wouldn't be my cup of tea.

My conclusion from fairly extensive research in the Journal of the Audio Engineering Society and other publications, and from my own crude experiments, is that the ideal would have the direct sound to arrive completely free from early reflections, then for the reflected energy to begin to arrive perhaps 10 or more milliseconds later, and then that reverberant energy would build up and decay over about 50 or so milliseconds.

However, if I understand your position correctly, you hold that all reflections are colorations - even those inevitably part of a live performance. So there is little point in me arguing that there's a right way and a wrong way for a loudspeaker to interact with the room if you see all room interactions as inherently detrimental. I doubt you and I will find much common ground here other than our passion for audio well reproduced, whatever that may mean. Hey, that's enough for me. I'd love to hear your system some day, and if you're ever in New Orleans give me a holler and come hear mine.

Cheers,

Duke
Opalchip,
Not once in your discourse, albeit cohesive, have I perceived any experience on your behalf of omni or pseudo omni loudspeakers. Might you find it in your realm of acceptance, the off chance that the culmination of your scientific understanding could be eclipsed by real experience?
Many with respected opinions (not me) have chimed in on your post. I suspect because they find you intelligent enough to consider their opinions. Listen to some real live music, then listen to some MBL's, German Physiks, Ohms, Shahinians, Quads etc. If you don't like them, fine we'll agree to disagree. Until then, please remain agnostic.