Is it possible to cover the whole spectrum, high SPL, low distortion with a 2-way?

Pop Quiz, topic; Room Acoustics.... Why does the Room Mode that is ~60hz - and 30hz not have a linear relationship with input into the mode.
Microphone is 2.5" from the mouth, which is my sub on the floor in a corner.
The lower level traces have what appear to be low frequency noise similar to what I've seen from heat pumps at several frequencies marked with red arrows.
Noise.png


Are those bumps what you are calling the "Room Mode"?
 
Well that was something, I am really interested in knocking out this Horn situation. I have an idea for a hanger for the horn. A square or possibly hexagon base that sits on top of the woofer enclosure. From that base, a "hangman" that erects from the rear of the base, some 28" or so. Possibly 2 hangmans spaced about 6" or so. Rope used to suspend the horn. I think I would need the suspension line to be fixed that the front the rest I imagine just wrapping around the horn.
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I Was trying to suspend my ejmlc300 when I stumbled on a big bag of washed sand in my garage. I just put that on top of my bass bin and rested the horn on it, easy to adjust, stable and it damps both the horn and the box, bingo! That is two years ago and it will stay....
//Anders
 
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Omg talk about a deal. I was looking at ~250usd or more shopping at Menards/HomeDepot...
Now I just cut cost to $30 Ty God 😩


For the hexagon base:

  • Hexagon Base Total Length: (6 \times 18") = 108 inches
For two "hangman" supports:

  • Two Upward Sections: (2 \times 27") = 54 inches
  • Two Forward Sections: (2 \times 32") = 64 inches
Adding these together: [108\text{"} (\text{hexagon}) + 54\text{"} (\text{upward sections}) + 64\text{"} (\text{forward sections}) = \boxed{226\text{"}}]

If you are building two identical structures, each requiring a hexagon base and "hangman" supports, then you would need to double the total length of material calculated for one structure.

For one structure:

  • Hexagon Base Total Length: (6 \times 18") = 108 inches
  • Two Upward Sections: (2 \times 27") = 54 inches
  • Two Forward Sections: (2 \times 32") = 64 inches
Total for one structure: [108\text{"} + 54\text{"} + 64\text{"} = 226\text{"}]

For two structures: [226\text{"} (\text{for one}) \times 2 = \boxed{452\text{"}}]

So, the grand total length of material needed to construct both structures is 452 inches.
 
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Yeah I need help... I don't want it... I need it lol
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Just when I thought I did good I found these boys for 20 bucks a pop. 3.15" x 2"x19ft seems more promising than 1x1. I feel the need to change the horn stand design once again but I think I am getting closer to being confident enough to start cutting and wielding. Im trying to keep the aspect of Rope Suspension alive in the design, and I think I might be able to hide most of the frame work behind the Horn which is a huge plus for me cause If I had it my way, you'd never see it at all except for maybe something under the horn. I need to have the ability of pitch and yaw adjustment as well as elevation.

Funny play on perspective; Perception is everything, I am calling this a mastering monitor, and the big dispute about my choice in waveguide is the lack of constant directivity. In critical listening, a singular point between the 2 loudspeakers is the highest resolved listening position one could have....As one moves off axis with my waveguide, there is an acoustical reminder that you are leaving the target listening position. I've seen some pretty good Bsing... I was the one to present Active Radiator Technology, Mbat with his Low Frequency extended waveguides, Docali with his Acoustic loading optimized Horns... Sounds good right? What could I call the rising DI of my horns in relation to the need to stay in the center of the loudspeakers.... I has be ridiculous as possible... Surely AI will make short work of this :ROFLMAO:

Auditory Bullseye Beamforming
Sonic Centrality Vortex
Acoustic Anchoring Aura

Possibly my Favorite so far - Auditory Alignment Alert System

Thats not bad actually and sounds almost as ridiculous as I want it to. You thought it was just a mechanical low pass filter... Wrong!...My horns have an Auditorial Alignment Alert System that reminds the listener to return to the optimized acoustical center designed for pinnacle Critical Listening.

If you needed to laugh today, please, do so at my expense :ROFLMAO:

ps - Acoustic Apex Advisory System:ROFLMAO: or maybe Acoustic Clarity Compass (ACC): Like a compass pointing true north, this feature guides listeners back to the true center of audio clarity.
 
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Early reflections tend to enhance spaciousness (the sense of a spaces acoustic ambiance) but they degrade imaging (the clarity of the sound image/location of the musicians in the recording.)

@IamJF I found it!
That's more true at LFs than HFs. At HFs we hear more of the direct sound since the ear processes these faster and the reverb field just adds spaciousness. At LFs the room almost completely dominates the situation. Hence, the anechoic response is very important at HFs, but less so at LFs.
"At HFs we hear more of the direct sound since the ear processes these faster" - Masking of low level details in the mix, is still a possibility, I think. masking only requires a masker. Indirect sound level can be the masker.
 
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I already agree with this. At some point I told myself, that as a studio engineer one should have something that represents an "everybody situation".
I do agree that mastering is meant to be perceived well on the average consumer product....for that same reason I have a pair of 4" two ways + 1" dome.....5-4" being the most popular bookshelf speaker design on Amazon.
Theres too many aspects of change between the mastering monitor and the average listener to rely solely on them. If we can find the loudspeaker aspects that are the most popular/common through out America, that would be a great secondary.
I would be looking for the most common Xmax, Driver Size, Driver Count, and maybe even diaphragm material. You might throw in baffle as well. One could go as far as looking to investigate and find the mean Room size and RT60 of your target audience.
 
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If the Spl level at 2 meters was 80db, the direct portion might be 70db and the indirect portion might be 79.54db... If this was the case, along with exaggerated decay of the room... Masking has to be a thing. Things happen fast at HF, correct? If we are listening at 80db... and in the mix, there is details at -10db of that... isn't this a cut and dry discussion?

What first comes to mind is that maybe this is a bad room... but my point is that in an average room, the high directivity is winning already, in clarity. That suggest only in clinical/studio conditions can we begin to start to compare a dynamic BE tweeter to a high Directivity AL/Ti tweeter... still at least on paper... The higher directivity will potentially measure better, given that it is a AL/Ti tweeter of high quality. The more directivity the less the room matters.

In a well-treated studio environment, the direct sound at 2 meters from the monitors is typically intended to be more prominent than indirect sound. Assuming an SPL of 80 dB at the listening position:

  • Direct SPL: This could be quite high—perhaps around 75-78 dB or so—since studios are designed for critical listening and aim to maximize direct sound.
  • Indirect SPL: The level of indirect sound would ideally be significantly lower than that of direct sound. It might range from about 65-70 dB, depending on the effectiveness of acoustic treatment and room geometry.

Even here, doesn't it seem, that masking is possible? how far down are we expecting these low level details?
 
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Low-level details in a mix, such as subtle reverbs, quiet background vocals, or soft instrumental textures, can often be 20 to 30 dB below the average level of the mix. These elements are critical for depth and space but may not be as prominent as lead vocals or main instruments.

In a studio environment:

  • Mix Engineers strive to balance these details so they're audible yet not overpowering.
  • Mastering Engineers ensure that during playback on various systems, these details remain perceptible within the dynamic range of the recording.
The ability to hear these subtleties is essential for high-quality mixing and mastering which is why studio monitors and room acoustics are designed to minimize masking from indirect sound reflections.

If I can trust this data.... Then I would confidently guess that even in the studio, a high directivity waveguide will provide better results over a dynamic tweeter. So much so that a well designed Al/Ti tweeter with high directivity could possibly perform a Dynamic Be Tweeter. I know must ask, Could it be possible that results could be perceivable all the way until the room is down 30db from the Direct sound.

I got onto this topic discussing potency of increasing accuracy, Diaphragm material vs Directivity.

In terms of potency:

  • Beryllium affects the inherent damping properties of the tweeter itself—how quickly it ceases vibration after excitation.
  • High directivity influences how much of those vibrations reach the listening position without interacting with room surfaces.
While both reduce perceived decay in their ways—a beryllium tweeter directly through its physical properties and high directivity indirectly by managing room interactions—it is difficult to quantify which is more potent without specific measurements. In practice, combining both could yield optimal results; however, if one had to choose between them based on effectiveness against decay specifically within a studio environment where acoustics are controlled, then high directivity might have a more significant impact since it addresses multiple paths by which sound reaches our ears beyond what is produced at source level alone.
it is my belief that Increasing Directivity is one of, if not the most potent aspect, of increase Accuracy. This is of course, given that quality designs are being compared. Accuracy to me, does not exclude Dynamics. To report Dynamics incorrectly is just another form of inaccuracy, that is, it is not enough to have a smooth FR and low decay.
 
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Hello camplo

WRT the room the only thing you are changing is the differences in power response between the 2 systems. The room is the room.

More about if there were any improvements made by using a "better" material would be easier to hear on which system.

CD vs. Non CD
I invite you to read the above and then see what you think. The idea of Exponential vs Constant Directivity waveguides isn't really the issue. The idea is more of a certain amount of directivity versus a higher amount of Directivity. I think the above gives evidence that there is are possible difference in clarity all the way until the point that the room is down 30db from direct sound, with masking of low level details being the culprit....

Still havent went into the idea of multiple arrivals and its effect on transient perception... obviously the more arrivals the worse transient accuracy.
 
I was moving fast when I put together some of the above data... I need to go back through, and weed out some errors. In particular I don't think I asked AI detailed enough question about Direct vs Indirect spl levels. Generally speaking indirect spl level will be lower than Direct but I would think it is frequency dependent. My focus was on tweeter performance which is not a mode dominated part of spectrum for normal rooms. Back to the drawing board.
 
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If the Spl level at 2 meters was 80db, the direct portion might be 70db and the indirect portion might be 79.54db... If this was the case, along with exaggerated decay of the room... Masking has to be a thing. Things happen fast at HF, correct? If we are listening at 80db... and in the mix, there is details at -10db of that... isn't this a cut and dry discussion?

That's why critical distance is such an important parameter in a pro situation, you can have at most 50/50.

But... from my experience the real advantage of pro rooms is in what is called 'isophonie' in french ( sound isoltaion from/to the outside). It's crazy how much sound pollution there can be especially in low end. It makes for a lot more details 'reading' in the outcome.

-20db for reflected sound is a good compromise i found ( and guess what... it's the 'recommended' DR target for 'acoustic' music( classical, jazz,...)). Anything with lower DR ( 90% of music style) is then 'dry'... ;)
 
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At HFs we hear more of the direct sound since the ear processes these faster and the reverb field just adds spaciousness. At LFs the room almost completely dominates the situation. Hence, the anechoic response is very important at HFs, but less so at LFs.

I definitely need to start here, I am trying to get you guys to respond, here, as I will be going on with this until I understand what it is a want to know, in the way I want to know it, that might take some discussion, and because I talk a lot, I don't want to dominate that thread with my post about psychoacoustics.

Hello camplo

WRT the room the only thing you are changing is the differences in power response between the 2 systems. The room is the room.

More about if there were any improvements made by using a "better" material would be easier to hear on which system.

CD vs. Non CD

Directivity changes the ration of direct sound to diffuse sound - as it would with decreasing the distance to a less direct speaker. But it does not change the decay or sound of the room itself.
I would for sure not go so far and say controlling the room and directivity will sound the same - it will not!


There are a few mechanisms happen all at once, you have to learn them one by one and then put it together to what our ear and brain receives.
The big difference between a microphone and our ear is that our hearing is time dependend. We hear a reflection differently when it arrives 5ms or 15ms after the main signal.
Just roughly - in the first 10ms the ear adds all the impulses to one. 10ms are about 3m path - so your side reflections or ceiling reflections would need to travel 3m longer as your direct signal. That never the case in normal size living rooms. So these reflections add to the main signal and when off axis frequency response is not linear - speaker response is not linear for our ear.
When building a studio room these first 10ms are crucial and the goal is to keep them free of strong reflections - absorption is one of the methods to do so.

There are different types of masking - Wikipedia should be helpful here.
There is "level masking" (a loud sound masks a silent sound) which depeds on the distance between the frequencies (as you describe). There is also time masking - a loud sound AFTER a silent sound can mask the silent sound! (That's how the cochlea and our nerves work - a loud signal get's processed faster, could be danger) All these masking effects are well researched and the reason why compression like mp3 is possible.
The "basic dynamic range" of the ear is about 60dB btw. The auditory ossicle helps to switch "gears" when signals get louder. Therefore it takes a while after a loud impact to hear more silent signals again.

Psychoacoustics ... it's such a great science!


There is a concept named "non environment room" which I built in my listening room. "It exposes the truth" is a good description ;)
https://www.soundonsound.com/techniques/sos-guide-control-room-design


edit: here an interesting graph about time and power of reflections and how we receive them:

View attachment 1275914
"Directivity changes the ration of direct sound to diffuse sound" that is identical to what room treatment does.
 
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