Phase only EQ is almost never a good idea. The only time it's known to work is linearizing the phase rotations in IIR crossovers...and even then with mixed results.
All-pass is a fine tuning stage. The difference in phase between your green and red traces you were asking about, is still in the find the ballpark stage.
Sim phase should be closer to real life than not, I think, when we know what we're doing both with the sims and with real measurements.
And yep, measurements, optimizing and verifying, is work. But it's how you move past sims and get onto the real playing field.
You have been going on and on about a 1 meter listening distance, is your basement only a bit over a meter wide?
Keeping excursion low to prevent intermodulation distortion of a woofer covering 20 Hz to over 300Hz makes sense. If you cross the 4x18" ~120Hz, IMD won't be a problem for them at excursions near Xmax.
The pair of front load 15" mids would not have any IMD to worry about running from ~120Hz to >600Hz, and your Axi2050's IMD will not be a problem once it's crossed over an octave or so above the horn's Fc.
The 4 x18" alone (~4800 cm2) with no corner gain could do that level at only 3mm excursion, with corner gain probably greater than 116dB at 20Hz with excursion under 7mm.
No reason to cripple the LF excursion of woofers in a 3-way system.
No, my vote is for a three-way system, using your single 15" per side, and as many 18" as you want to use.
Same amount of crossovers and amps as 2.5 way, but much better system headroom.
The pair of 2x18" slot load you built should be plenty if you are only looking for 30 Hz 110dB/1m.
Art
A 70 year old reminder of 'the basics'.
The following list includes all the types of distortion which are at present known to have an important effect on the quality of the reproduced sound:
Nelson Pass' take on IMD (in amplifiers):
"An important thing about distortion – when you run a signal through a device which is even slightly non-linear, you have changed the signal forever. You can use various techniques to reduce distortion after the fact, but you can’t go back.
In Figure 8 we see a waveform consisting of 7 non-harmonically related tones of equal amplitude from 100 Hz to 2800 Hz.
My point? IM distortion is the elephant on the dance floor.
Much of the time IM distortion simply forms a complex “noise floor” which masks musical detail. At lower levels, it takes the life out of the music and makes it uninteresting, even irritating. It isn’t as noticeable with very simple music, but it stands out with orchestral material as if the instruments were covered by a veil.
At high distortion levels, the sound simply turns to mud, and we turn it down. Or off."
The following list includes all the types of distortion which are at present known to have an important effect on the quality of the reproduced sound:
- Frequency-amplitude distortion.
- Reproduction noise.
- Harmonic distortion.
- Intermodulation distortion.
- Transient distortion.
- Phase distortion.
- Frequency-modulation distortion.
These distortions occur within the reproducing system, and are capable of measurement and analysis. However, there are certain other fundamental limitations which are present in any attempt to reproduce sound. These might properly be called acoustic distortions:
- Differences in acoustics between the room in which the sound originates and the room in which it is reproduced.
- Spatial distribution effects.
- Limited dynamic range in reproduction.
Nelson Pass' take on IMD (in amplifiers):
"An important thing about distortion – when you run a signal through a device which is even slightly non-linear, you have changed the signal forever. You can use various techniques to reduce distortion after the fact, but you can’t go back.
In Figure 8 we see a waveform consisting of 7 non-harmonically related tones of equal amplitude from 100 Hz to 2800 Hz.
My point? IM distortion is the elephant on the dance floor.
Much of the time IM distortion simply forms a complex “noise floor” which masks musical detail. At lower levels, it takes the life out of the music and makes it uninteresting, even irritating. It isn’t as noticeable with very simple music, but it stands out with orchestral material as if the instruments were covered by a veil.
At high distortion levels, the sound simply turns to mud, and we turn it down. Or off."
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65" apart where I am sitting right now, so no equilateral triangle, horizontally, but its just a matter of position when the time comes. Sitting at the computer desk, I am 1m away from them. The mouth of the lower ppsl is ~32" from the side wall and 42" from the front wall. Rear of cab is about 9" from the wall, and the side cab is almost touching the side wall, cab yawed towards center.
I think I may have stumbled on something. Equilateral. Why is this complimenting? I've been experimenting with LR crossover in Vituixcad and have found that when listening distance and ctc spacing of the TM are the same, it forms another equilateral triangle and the vertical polars clear up. The "sound power" sky rockets too. I don't know what it is.
Could you please elaborate?
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@Ro808 I don't know if I am more or less confused after reading that lol... Comb filtering is not desirable.... "If the subwoofers are on the floor, then propagation is parallel to the floor and there is no ground bounce. Hence, there is no comb effect." @phase_accurate suggested one woofer in the air as a plus. It seems, that, is only complimenting when the two sources are coupled. This sim above is my mock up of a TMM with WW on the back. In contrast to the MTW type, shown previously. Both polars look good. The difference is the efficiency, placing the WW in the back of the cabinet buries it further in the corner, and the gain goes way up.
@GM I have simulated all sources as omni I presume, as I've not entered any type of directivity information. I mentioned comb filtering because it appears to me that with low enough crossovers, the comb filtering is mitigated. In these sims I am using a higher XO point than I have in the past, in real life. Lower the XO, the better the XO performs.
In audio you can strive for and even approach perfection, never to achieve it.
So it all comes down to the good 'ol selection and optimization of a number of compromises.
A former BBC engineer rightly said that (analogue) audio is inherently chaotic.
Wesayso's arrays are the epitome of lobing, but there are certain advantages that (may) outweigh this compromise.
This comment from AVSforum contains a healthy dose of common sense, derived from science:
"Psychoacoustics matters a lot. All speakers in a room with flat surfaces will have an element of comb filtering, they may not be audible to our deaf human ears because of the time involved in the cycle relative to how our brain interprets it, and it matters what the frequency range is, as we are more sensitive to some rangers than others per neuroscience. So psychoacoutics plays a role of course, because we do not hear the way a microphone does; the brain is a prediction engine with poor reflex, so we hear what we hear not because its reality (objective) but rather what our brain interprets after the fact, or predicted and biased towards it (subjective).
Lots of metrics don't matter in the audible realm, making them more academic, and are basically too minute to physically hear the difference for most people's ears. Does that mean they're not important? No. There's just a distinct difference between scales. Our hearing's scale is course and low resolution. A microphone's scale is fine and high resolution over the same period of time. This said, today's culture has access to high resolution data compared to zero data or low resolution limited data of our audio-past. So despite this, I think metrics are still incredibly important to know about from an end-user experience, to understand the what and why of your hearing's information that your brain will bias.
It's not like people talking in a room sounds weird or artificial as we hear speech in a room. Our brain handles the subtle differences in SPL to result in similar experience for the most part. A more interesting question would be... where on average does our brain's interpretation of audio (psycchoacoutics) become noticeable by an average listener (ie, over 3db or more? Less?). Comb filtering or wave front cancellation (we're more interested in the loss, not the standing wave here) when heard by our ear would be loudness (audible) versus not (inaudible, insignificant in terms of end user experience)."
So it all comes down to the good 'ol selection and optimization of a number of compromises.
A former BBC engineer rightly said that (analogue) audio is inherently chaotic.
Wesayso's arrays are the epitome of lobing, but there are certain advantages that (may) outweigh this compromise.
This comment from AVSforum contains a healthy dose of common sense, derived from science:
"Psychoacoustics matters a lot. All speakers in a room with flat surfaces will have an element of comb filtering, they may not be audible to our deaf human ears because of the time involved in the cycle relative to how our brain interprets it, and it matters what the frequency range is, as we are more sensitive to some rangers than others per neuroscience. So psychoacoutics plays a role of course, because we do not hear the way a microphone does; the brain is a prediction engine with poor reflex, so we hear what we hear not because its reality (objective) but rather what our brain interprets after the fact, or predicted and biased towards it (subjective).
Lots of metrics don't matter in the audible realm, making them more academic, and are basically too minute to physically hear the difference for most people's ears. Does that mean they're not important? No. There's just a distinct difference between scales. Our hearing's scale is course and low resolution. A microphone's scale is fine and high resolution over the same period of time. This said, today's culture has access to high resolution data compared to zero data or low resolution limited data of our audio-past. So despite this, I think metrics are still incredibly important to know about from an end-user experience, to understand the what and why of your hearing's information that your brain will bias.
It's not like people talking in a room sounds weird or artificial as we hear speech in a room. Our brain handles the subtle differences in SPL to result in similar experience for the most part. A more interesting question would be... where on average does our brain's interpretation of audio (psycchoacoutics) become noticeable by an average listener (ie, over 3db or more? Less?). Comb filtering or wave front cancellation (we're more interested in the loss, not the standing wave here) when heard by our ear would be loudness (audible) versus not (inaudible, insignificant in terms of end user experience)."
MTW 2.5way... there is no highpass filter on the bottom of any woofers passband, the lowest woofers handle most of the bass, The upper woofers roll off sooner. The lower woofer plays about 5db louder than the upper, but what I think is important is that both sections are able to have very similar FR. I think that whatever models in a way that seems conducive to your goals, let it be. The configuration was easier to tweak with good vertical polar results. The TMM with rear WW, had more phase issues summing the bass sections. I think with more advanced dsp, things would be better but I'd rather be able to have a decent performance with least tweaking. After adjusting the spl to represent about where the drivers should achieve 2mm at 30hz the sim suggested in room response, in the 115db range For the MTM2.5. Its sorta like The 2way I already have with a complimentary woofer above the tweeter, slightly closer ctc and lower Qts. The low xovers is what makes it work, possibly allowing the best of the tweeter in the middle configuration. If used as MTW the low crossovers are still a benefit. As a MTW2.5way like a described above. I see two sources that can cover their pass bands with similar FR, upper with a 36db LR LP, lower with 12db LR LP and the tweeter with a 24db LR HP. The lower plays about 5db louder than the upper. So its like a two way but the image will be brought slightly up towards center. Using the same low crossings as I've already demonstrated to myself, to work. In my opinion, the imaging was great, as a 2way, with the addition of another woofer above the tweeter, I think I would like a XO point closer to if not, 300hz, which really takes care of any doubt of headroom issues for the tweeter. Even near 300hz the polar performance should allow for a clean response, near 30 degree wide, on the vertical.
Stuffed vented cad tuned low, or just sealed. At domestic levels basically 95db rms at the most, probably closer to 82db rms, at 1 meter lol. I figure if the 2way version worked, then this should work as well.
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OK, then no clue what this measurement is showing us in the ~150 - 1 kHz BW or whether it's an issue and/or even audible.
Its my mock up of the MTW as it would be in basement, in the corner. Horn is at ear height ~44inches, 18's are 32" below horn and the 15's are above the horn about 17". Driver #1 is 18"ppmsl and Driver #2 is the 15" ppsml. Driver 3 is the tweeter. Everything is modelled as point sources. This should be the worse case scenario.
There is some cancellation but there is a good clear space close to 30 degrees wide, on axis. A very gentle squeeze on the vertical axis, no more so then on a really good two way, just on top and bottom of the tweeter. Low Crossovers, sound better.... Less comb filtering.
There is some cancellation but there is a good clear space close to 30 degrees wide, on axis. A very gentle squeeze on the vertical axis, no more so then on a really good two way, just on top and bottom of the tweeter. Low Crossovers, sound better.... Less comb filtering.
