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Everything you always wanted to know about cardioids* (*but were afraid to ask)

The F205 has radiation control with acoustic ports and several internal chambers, with ports inside the cabinet. There is a rendering (picture below "Technology" section) which shows some of the internal construction, without damping materials.

The purpose of those internal chambers and ports and damping material is to shape the frequency response and delay of the sound emitted from the acoustic ports. This all works together in a rather complex way, that is only possible to see properly using simulation of the acoustic system.

The cabinet with the ports and internals is one part of the design - the choice of radiation pattern is also a part of the design process. A desired pattern is defined, and then the cabinet is optimized to create this pattern, by placement and size of acoustic ports, and tuning of the internal chambers.
I read it. Cool stuff.

However, I wouldn’t call your design a horn. An acoustic horn converts large pressure changes within a small area into a low pressure changes within a large displacement area. As far as I can see that’s not how your subwoofer works. Am I correct?
 
I read it. Cool stuff.

However, I wouldn’t call your design a horn. An acoustic horn converts large pressure changes within a small area into a low pressure changes within a large displacement area. As far as I can see that’s not how your subwoofer works. Am I correct?
They generally have too small mouth radiation area for pattern control, so in that context, they are certainly not horns. But where do we set the limit - where does it stop being a horn - not many bass-horn designs that can be called a real horn, if the mouth must be large compared to the wavelength. But they do actually convert a large force/pressure into higher velocity/lower pressure - they act as an acoustic transformer - and that is what a horn is. Even the very small T6 does that, and compared to a typical ported design, the acoustic loading works across a much wider bandwidth - from 30Hz up to around 100Hz in the T6.

The acoustic ports for radiation control in what we call passive cardioid does not work like horn loading or like a typical reflex port - they do not provide acoustic loading, and they do not contribute to increase low frequency extension. In a typical bass port, the sound emitted from the port has a 180 degrees phase shift, so that the port output then sums in-phase with the sound radiated from the front of the cone. The ports in a cardioid emit sound that is in-phase with the sound emitted from the back of the driver - the sound inside the cabinet, so the port output is out-of-phase with the sound from the front of the cone. The sound is not totally cancelled, because the port output is at a different location - which creates a delay, and typically at a lower level. Off-axis and backwards, the timing between the port output and the cone output changes, so that the cancellation effect increases.
 
where does it stop being a horn - not many bass-horn designs that can be called a real horn, if the mouth must be large compared to the wavelength. But they do actually convert a large force/pressure into higher velocity/lower pressure - they act as an acoustic transformer - and that is what a horn is. Even the very small T6 does that, and compared to a typical ported design, the acoustic loading works across a much wider bandwidth - from 30Hz up to around 100Hz in the T6.
Your design certainly is an acoustic transformer but horn is just one type of acoustic transformer. As far as I can see yours does not achieve that with a horn. In a horn there is a tapered waveguide as the sound guide. As far as I can see (not much) you do not have that. I am also not sure if you actually increase efficiency as much as a horn will do.

Do please realise that I am talking about semantics here. You have a unique design. I congratulate you. I do have a question though: why are you so worried about the (low) loudness capacity of your speakers? What is stopping you to build larger versions?
 
Your design certainly is an acoustic transformer but horn is just one type of acoustic transformer. As far as I can see yours does not achieve that with a horn. In a horn there is a tapered waveguide as the sound guide. As far as I can see (not much) you do not have that. I am also not sure if you actually increase efficiency as much as a horn will do.

Do please realise that I am talking about semantics here. You have a unique design. I congratulate you. I do have a question though: why are you so worried about the (low) loudness capacity of your speakers? What is stopping you to build larger versions?
They follow the same principles of physics as all speakers do - which means size, low frequency extension and efficiency follow a fixed relationship. For any given size, there is a limit for low freq and efficiency.

The narrow horn channel increases slightly in area, and it is long compared to wavelength, this is different from a typical ported box where the port is very short compared to wavelength, and acts like a pure acoustic mass.

It is capacity at the lowest frequencies that sets the limit for a bass-system, especially for movies. The acoustic loading makes it possible to achieve good low frequency output utilizing drivers with more powerful motor system and less moving mass, thus increasing output capacity at higher frequencies.

Once the required extension is in place, a larger bass-system can be created by stacking more units. A very large subwoofer is impractical - it is hopeless to handle, ship and place.
 
They follow the same principles of physics as all speakers do - which means size, low frequency extension and efficiency follow a fixed relationship. For any given size, there is a limit for low freq and efficiency.
On a pure horn, the standard efficiency relationship of Vb * f3^3, which is the fundamental formula of all electromagnetic drivers in an enclosure, doesn’t apply. I do however agree that the overall size and f3 still affect efficiency.
 
On a pure horn, the standard efficiency relationship of Vb * f3^3, which is the fundamental formula of all electromagnetic drivers in an enclosure, doesn’t apply. I do however agree that the overall size and f3 still affect efficiency.
Yes, for horn the efficiency also depends on the design, still there will be a similar trade-off between size and low cut-off and efficiency.

But for a modern design, it is all about max output capacity - once, when a 20W amplifier was considered powerful, we have much more cheap and good power available, and drivers that can handle this power. Relevant for cardioid - or acoustic ports for pattern control - is that this principle also trades efficiency for better radiation pattern, but this does not matter in the end, because you just add more power and end up with sufficient capacity still.

But it is better with a larger radiating mouth area - it couples better. There is a difference in perceived sound from the smaller versions of my subwoofers to the larger ones - larger sounds more powerful.
 
Just the thread I was looking for.
It seams to me like there is a trend of making Cardioid speakers latly. Kvasvoll, Sigberg audio (both from Norway), D&D are all active on the front. This might be a stupid queastion for you technically enabled guys. But still I wonder. What are the main differences between cadiod and dipols? expet the obvious designs.

I'm asking because I have followed Forsman and Likwittz speaker designs the last 20 years and am a happy owner of the D2-10s for the last 4 years. Rarly do I hear speakers comparable when it comes to ease of room integration, imaging, stage and depth. The only pitfall I encounted is the bass. Less then 1m from the backwall and its gone. I use Sonarworks for room correction and get good measurements from 27hz. The placement issue can be easily fixed with separate subs thought.

 
Hey guys, I was just curious - been trying to understand loudspeakers and room interactions a little better lately

As an example of how dipoles can give a smoother response due to their dispersion pattern interacting less (or in a more unique way?) compared to more traditional designs - which got me thinking...could it be fair to describe a cardioid design as basically being similar to a dipole, but that you're catching the rear wave?

Essentially, you're "catching" and dealing with the rear-circle of a dipole's figure-8 pattern, such that it's really only the forward part of it that's being sent out into the room?

Does that make sense to you guys?
(Not sure if I'm wording the pieces correctly here, but I feel like y'all catch what I'm trying to describe)
 
How far from side walls should a cardioid speaker need to be placed to function correctly?
 
How far from side walls should a cardioid speaker need to be placed to function correctly?

Cardioid speakers cancel sound out as they leave the speakers (as opposed to being canceled out when they return as reflections from the wall). So the effect itself is placement independent.

That being said, you will find there are places in your room the speakers will work more or less optimally just as with any other speaker.
 
Hi,
I'll augment a little to help everyone think it through:

My first thought about this stuff some years back was that cancellation of sound would happen in a finger snap at certain location, as if sound would disappear and wouldn't exists anymore. This is not true though, because sound cancellation is not destructive event but a superposition.

Both the primary sound (front of speaker) and the secondary sound, that is utilized to make cancellation and a pattern, must both exists and expand together to same direction so that the cancellation is effective at a distance. Attenuation behind a cardioid box is not there because there is no sound toward that direction, but because there is two sounds in superposition, with different phase. Both of them must exist in order to have destructive interference, cancellation/attenuation of sound happen. Best cancellation is at a frequency where both primary and secondary sound have exactly same amplitude and exactly 180deg phase difference, but since the primary and secondary sound sources cannot be at same physical location both amplitude and phase varies with observation angle, and thus the pattern changes per observation angle and frequency, which relates to wavelength, which all depends on what kind of construct you have, what are path lengths around the construct to various directions.

It is very hard to make cancellation at high frequency, because short wavelengths interact more strongly with physical structure of the speaker, which makes both primary and secondary sound vary to all kinds of directions, due to edge diffraction for example, sound outputting any side of the box is going around the whole thing in 3D and so on. You would see this in a BEM simulation: Put a tweeter horn above open baffle mid woofer, and the horn makes the OB mid pattern break in vertical direction, because the horn makes different shaped obstruction to both front and back sound of the OB woofer, which affects both sounds differently, and now there is no proper dipole cancellation above the speaker as there would without the horn. Initial idea might have been that the horn is fine there and shouldn't affect mids sound as the mid should have null toward the horn, right?:) Null is there only without the horn, and the horn breaks the null affecting the superposition.

So, expect that in the near field of speaker both (all) sound sources on the speaker box have local effects, which makes the pattern vary some from what it is in the far field, especially on higher frequencies.So, whats far field then, a simple rule of thumb that floats around is roughly three times the speaker biggest dimension. So if you want to put speaker 10cm away from wall, you should make it smaller than 3cm for everything to behave like assumed/expected. Or, just put it there and not care too much, listen and tweak and so on, compare :) You could simulate with BEM and try to figure out what works better and what doesn't work that well within the distance you must use, what's the shape of the speaker, where the sound sources locate, and so on.

Simplified takeaway would be that, the closer you are at a wall there might be some break in the pattern and bit unexpected response behavior, unless the speaker was optimized to be positioned that way, at a particular distance from wall for desired effect.
 
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Forgot that ripple tank is fantastic for quick experiments and to get intuition on this stuff.

See, here I've setup ripple tank to show random ordinary speaker box at various distances of a wall, it's a top down view. Wavelength and all settings what ever they happened to be, just to demonstrate how the pattern changes with various distances to wall. On first image a speaker positioned back corner against a wall and sound doesn't get around the box that side but reflects back towards "forward direction", a listener would be somewhere bottom right corner.
close.pngbit-further.pngfurther.pngfar.png

Notice, when there is some space between wall and the speaker sound now goes around both sides of the box and different kinds of patterns emerge to various directions, great difference behind the speaker.

Here illustration that somewhat cardioidish pattern emerges with any boxed normal speaker due to sound going around the speaker on both sides, and observed from and angle, say at 130deg so somewhere behind there is cancellation, bacause sound all the way around the speaker took longer to get you than that which came right around the closer corner, they interfere and on some frequency there is some destructive interference and at some frequency there could be some constructive interference. Directly behind a speaker there is mostly constructive difference, because path length around both sides of the speaker are the same. I say mostly because real world is not 2D like the sims here but sound diffracts all around a box/construct, from all sides, and all/most pathlengths differ, depending on the size and shape of the structure.

Here simple experiment to illustrate. If I remove the wall there are some nulls forming behind the speaker, easily visible in the image:
freefield.pngnulls.png
direction and existence of these nulls is function of wavelength and box dimensions, both width and depth and all.

If I add an obstacle on another side to prevent sound from going around the box from that side, the nulls don't exists anymore. There is attenuation behind the speaker now, especially to the side where the sound had to wrap around a lot and lose energy to diffraction.
no null.png

One could do cardioidish system in the ripple tank as well, or a dipole, using two sound sources and manipulating their phase.

From this and previous post one can continue the thought process and reason features for a cardioid system :)
Fun stuff, starting point from "simple" physics of sound, happy experimenting!:)
 
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We can understand the concepts a bit without getting too deep. Like much in acoustics, it's fundamentally about distance versus wavelength, connected by the speed of sound.

Because the 2 sources are in fixed locations, their phase relationship is different at every wavelength/frequency AND also across the polar pattern. This develops exactly the same sort of complex interference pattern as monopoles or dipoles; it's only the exact details that change.

I think I've set this up correctly in VituixCAD.
  • A pair of imaginary drivers with perfectly flat frequency response & omnidirectional dispersion, i.e., ideal point sources.
  • 200mm spacing between them. This is good to experiment with, but 200mm covers a useful span of midrange.
  • Because the 2nd driver is 200mm farther from the imaginary microphone, its amplitude is increased 0.9dB to match the main driver's. (In the real world, you need to EQ to match FR as well.)
  • We'll keep all of those things constant while altering the 2nd driver's delay & polarity.
Monopole - no delay, positive polarity. This shows essentially perfect omnidirectional behavior up to 500Hz. Higher, we see massive comb filtering on the main axis & serious lobing elsewhere.
View attachment 42383

Dipole - no delay, inverted polarity. Massive cancellation below ~250Hz, an octave below the monopole's transition. A nice dipole pattern covers the 2 octaves from 300Hz to 1.2kHz. The same combing & lobing above, but nulls & peaks switch. No surprise that this matches the monopole's results only reversed.
View attachment 42384

Cardioid - 580us delay (200mm spacing, or 1 wavelength at 1.7kHz), inverted polarity. The gradient behavior again drops by an octave or so, giving a good pattern from roughly 150Hz to 700Hz. Combing & lobing yet again, though a very narrow null directly behind the speaker remains clean.
View attachment 42385

The driver spacing causes LF reinforcement when both have the same polarity, while cancelling in designs with one inverted. Just as in a dipole, the cardioid's secondary driver nulls the main output at all angles, once the wavelength becomes long enough to overwhelm the geometry. (Note that both Kii & Dutch^2 use monopole subs below 100Hz or so.)

Regardless of the configuration, 2 drivers with this spacing will exhibit major lobing at higher frequencies. The waves simply cannot combine properly when the sources are a significant fraction of a wavelength apart.

So, covering 2 octaves isn't too hard, 2.5 appears reasonable. More than 3? You're gonna need 2 cardioids or another way to control directivity. (Note that this bandwidth fits a 4-way, which is pretty much where all of the serious DIY - read "doesn't need to be marketable" - builders end up. If you're beginning to suspect that physics & our hearing have conspired just to annoy loudspeaker designers, we're on the same page.)
 
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