Doing this with a phased array is an extremely complex undertaking. I think I'll lead with "are you sure?" having built such arrays (yes, successfully).
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Constant Beamwidth Transducer (CBT) Speakers
- Thread starter Rick Sykora
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dziemian
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No, I am not sure at all. Thats why I am asking. I just cant figure out how it would work. I used to use Dirac Live with many speakers in the past but all of them were typical 2-way boxes including constant directivity Geddes Abbey mentioned here by Bjorn. I loved the results.
MrKlinky
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My thoughts exactly, and for years I have been asking the same question regarding time-smearing wherever I think I might receive a worthwhile reply. Nothing to date...
To me, transient response is key to exciting and accurate sound, which appears on the face of it to be impossible given the multitudinous different path length to the listener's ear from an array of sources.
Yet, according to Keele's published data, his CBTs can reproduce pretty good square waves from the upper bass through the lower treble.
Quick grab:
Don's square wave measurements were conducted in a fairly small room with very close proximity to surfaces/furnitures. Even though it's a nearfield measurement, it's likely to look better with anechoic measurements.
What you do not understand is that all the impulses line up in time in the direction that the array is pointed.
Outside of that space, they CANCEL each other. This is why the array is directional.
So, I'm sorry, but you don't understand how an array is steered.
May ask then how this speaker has a 140 degree degree beam width? Was JBL’s application of CBT different?
You do understand that an array can simulate a point source, yes? Well, no, maybe you don't. But that's the answer. There's no mystery to this at all.
Consider, inside the limits of the array's spatial sampling, you can create any shape of wavefront you want, given enough control over each individual driver input. For instance if you take a flat array and you set it up so that the time delays are such that the wavefront emerges in a circular shape, you have a point source (i.e. omni, most likely over only the front half of the array) pattern.
If you drive them all such that the impulses line up at 50 degrees, you have a far field plane wave that points to 50 degrees.
It's much like FIR filter design in that the delays and weights for each driver (including frequency weighting for constant beam widths across frequency) allow you to design a pattern you want.
For near field (under 5 times the height of the array) you can also do near-field focusing, in which the level is maximized at one point (or line) in space, and is less energetic both nearer and farther. I've done this myself by making a focus at 1.8 meters, and putting a lower level diffuse (i.e. reverb plus other stuff) into this driver, so you hear the "far away" effect until you're at about 2.2 meters away, then the source moves to RIGHT THERE ON YOUR NOSE (quickly over about 30-40 cm) and then back AWAY as you get closer.
This is, note, only true for an array with control over delay, gain, and frequency weighting per driver, and careful design. You can get something approximating (for straight in front of the array) a properly sampled array when the drivers are below 1/2 wavelength. Over 1/2 wavelength lobes happen. If you're shooting end-on in an array it's 1/4 wavelength. (any delay effectively moves a driver (or mike, works both ways) directly away from the point being observed, regardless of the angle of the array. This complicates aiming the array to some extent. Just more math.
I'm not trying to be mysterious here, there's actually a crapload of math involved, and I do mean a crapload.
A lot of this is rather counterintuitive, which is why I hesitate to try to explain thoroughly. Because of the differences in wavelength between bass and treble, the effects are very, very different than at radar frequencies, for instance.
Thank you for the explanation. It seems I took your short sentence: "this is why the array is directional" literally and failed to understand that you were talking about arrays in general and not necessarily about CBT arrays. As the thread is about CBT arrays introduced to the Hi-Fi world by Keele I was thinking about the CBT Curved‐Line Array with shading (Legendre). As far as I know such a line source propagates sound as a cylinder as opposed to a sphere, i.e. a non-dimensional point source.Consider, inside the limits of the array's spatial sampling, you can create any shape of wavefront you want, given enough control over each individual driver input. For instance if you take a flat array and you set it up so that the time delays are such that the wavefront emerges in a circular shape, you have a point source (i.e. omni, most likely over only the front half of the array) pattern.
If you drive them all such that the impulses line up at 50 degrees, you have a far field plane wave that points to 50 degrees.
It's much like FIR filter design in that the delays and weights for each driver (including frequency weighting for constant beam widths across frequency) allow you to design a pattern you want.
For near field (under 5 times the height of the array) you can also do near-field focusing, in which the level is maximized at one point (or line) in space, and is less energetic both nearer and farther. I've done this myself by making a focus at 1.8 meters, and putting a lower level diffuse (i.e. reverb plus other stuff) into this driver, so you hear the "far away" effect until you're at about 2.2 meters away, then the source moves to RIGHT THERE ON YOUR NOSE (quickly over about 30-40 cm) and then back AWAY as you get closer.
This is, note, only true for an array with control over delay, gain, and frequency weighting per driver, and careful design. You can get something approximating (for straight in front of the array) a properly sampled array when the drivers are below 1/2 wavelength. Over 1/2 wavelength lobes happen. If you're shooting end-on in an array it's 1/4 wavelength. (any delay effectively moves a driver (or mike, works both ways) directly away from the point being observed, regardless of the angle of the array. This complicates aiming the array to some extent. Just more math.
I'm not trying to be mysterious here, there's actually a crapload of math involved, and I do mean a crapload.
A lot of this is rather counterintuitive, which is why I hesitate to try to explain thoroughly. Because of the differences in wavelength between bass and treble, the effects are very, very different than at radar frequencies, for instance.
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There is a reason why we talk about radio waves and audio waves
For a line array, the result is a cylinder. It takes a square array to do a point source.
Done that. Works fine. Costs a lot. Not clear it's worth the effort.
First generation.
HOLOPLOT X1
HOLOPLOT X1 blends line array and electronic beamsteering technology for complete sound control. Empowered creative freedom for your boldest audio concepts.
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Maybe the first publicly announced generation, perhaps.First generation.
HOLOPLOT X1
HOLOPLOT X1 blends line array and electronic beamsteering technology for complete sound control. Empowered creative freedom for your boldest audio concepts.