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Keith_W DSP system

BTW, I do a lot of experiments with DSP to satisfy my own curiosity, and sometimes this involves deliberately ignoring Toole and doing the opposite of what he says. I can tell you that the result isn't as disastrous as it's made out to be, but there is a niggling feeling of discomfort at the back of my head that comes from ignoring the experts. The reason I choose a FDW of 15/5 is to alleviate that discomfort.
That's the nice thing about having so much control over your system. You can try things just to see, (or hear) and then put it back later. My experience is similar to yours. Bad directivity doesn't necessarily sound so bad, can be intriguing at times. Better directivity seems more versatile, best overall effect with the most recordings.
 
- The Brueggemann target was felt to be "shouty" by one listener. The other was neutral about it. This is my preferred target, it has a nice midrange fullness and enough bass without sounding excessive.
I like a bold midrange. That whole range between 100 and 1000, (or maybe I should extend that to 3500) needs to be adequately prominent, and I think it won't sound shouty if the bottom end of it is not under represented. It'll sound bold and rich, solid, full, but not necessarily the clearest and most detailed. If there's anything I percieve to be a common error with most systems I've set up and others I've heard in people's homes, trade shows, high end shops, etc., it's that they're too thin in that range to sound realistic. The thing is, it can quickly get murky and chesty too if there's a little too much, or the room is causing problems. I think a lot of people inadvertently or on purpose cut in the lower part of that range because of room issues, and that resulting thinness in tone has become a practical hallmark of the sound of stereo to me. And I'll have to say, if I can't get that fullness without it sounding murky and chesty, then I'll go with the thinness. It's a practical compromise. Actually, I will tolerate a small amount of murkiness. I'm really liking what my system is doing right now, and it's because I boosted the range between 200 and 800 Hz by a couple dB over what measured to be a fairly flat room curve with about 0.7 dB/ octave slope. It's hard to tell in that range because the room response is rough through there.

I can back up that people cut in that range from conversations I've had with various clients who buy our products, working in mixing and mastering, and also talking to my co-workers about adjusting their systems. They frequently tell me that the range around 200-300 Hz is often problematic to their ears.
 
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I believe that preferred target curves can vary because of:

1. Different directivity characteristics of different speakers + room reflections. If your speaker has wonky directivity characteristics like mine, some target curves just don't work.
Exactly, that's why Toole also says that "correcting" a listening position response to such a predefined targets doesn't guarantee anything, as people prefer loudspeakers with flat direct sound and smooth directivity, the curves at the listening positions measured from such are results but not targets. That's why he also says that there is no such predefined "Toole target curve" and reacts allergic when people come up with such.
More reading material about it:
 
Also since direct sound dominates our perception above room transition frequency it might be interesting to see the corresponding direct sound curves with the LP correction curves you used and examine how well they correlate to the perceived tonalities and rankings.
 
Yes a result , the preferred speakers have roughly anechoic flat response on axis but in room due to their directivity properties the results are like these curves .

As your brain+ears listen primarily to direct sound above the transition region means that if the speaker does not naturally conform you have to botch the direct sound to get the reflected sound just right to conform to these curves ? you might like it but that not the point at all with these curves .

If you now the anechoic response you can maybe adjust for that above the transition region .

I also adjust to taste a tad warmer more bass, but its just me :)

Edit . Thanks for the test anyway it's still very interesting
 
Also since direct sound dominates our perception above room transition frequency it might be interesting to see the corresponding direct sound curves with the LP correction curves you used and examine how well they correlate to the perceived tonalities and rankings.

I am sorry, I am not sure what you are asking for. Do you want to see the uncorrected curves vs. the corrected curves?
 
I am sorry, I am not sure what you are asking for. Do you want to see the uncorrected curves vs. the corrected curves?
What I read him to say, and what I agree would be interesting to see, is the direct sound AFTER you made the room correction EQ. The direct sound dominates above the transition frequency, so it's interesting if altering it from flat ever consistently creates a preferred overall sound when the directivity is not smooth.
 
I am sorry, I am not sure what you are asking for. Do you want to see the uncorrected curves vs. the corrected curves?
The direct sound or listening window (so for example a gated response or a nearfield MMM) of your loudspeakers after the correction with all those LP targets you used. If you have it already without EQ of course no additional measurements are necessary as you could convolve them with your used filters.
 
The direct sound or listening window (so for example a gated response or a nearfield MMM) of your loudspeakers after the correction with all those LP targets you used. If you have it already without EQ of course no additional measurements are necessary as you could convolve them with your used filters.

Ah, I see. I am sorry, I don't have them. I generated all those filters in one sitting without doing any verification measurements. I only have sims, but I have found that the sims look very close to the actual verification measurement. I can show you the sims if you want. A nearfield MMM will look very different to a MLP verification measurement (or a sim of the MLP verification) but I am sure you knew this already. I am still not sure what you mean by "without EQ" and "after correction with targets" because it seems contradictory? By definition, after I correct it to a target, it has EQ applied? I think I am still misunderstanding you.

1715655413093.png


In any case, this is the "before" (red) vs. "after" (green) of the left channel only. The brown curve is the sim. I am continually amazed by how close the prediction matches reality. You can also see that I cut the peaks, left the dips alone, and applied "broad tone controls" to the upper frequencies.
 
These are measurements at the listening position so contain also some non direct sound parts, to get a mainly the direct sound you need to usually to get closer and use gating.
I am still not sure what you mean by "without EQ" and "after correction with targets" because it seems contradictory? By definition, after I correct it to a target, it has EQ applied? I think I am still misunderstanding you.
As written if you know or had measured the gated/anechoic response of your loudspeakers you can apply on it the filter functions that you generated to see how the direct sound of them after your EQ will look like.
 
These are measurements at the listening position so contain also some non direct sound parts, to get a mainly the direct sound you need to usually to get closer and use gating.

As written if you know or had measured the gated/anechoic response of your loudspeakers you can apply on it the filter functions that you generated to see how the direct sound of them after your EQ will look like.

The measurements I showed do have gating, AKA windowing. Both the before and after.
 
The measurements I showed do have gating, AKA windowing. Both the before and after.
I know, per Acourate measurements always do, but even for example with the typical values like 5-15 cycles they include a significant portion of reflected sound if they are taken at the LP and not close(r) to the loudspeakers.
 
I know, per Acourate measurements always do, but even for example with the typical values like 5-15 cycles they include a significant portion of reflected sound if they are taken at the LP and not close(r) to the loudspeakers.
Even with just 3 cycles the sound of a 50 Hz sinewave will travel already 20 m. So how do you exclude a reflected sound in a normal listening environment?
 
Even with just 3 cycles the sound of a 50 Hz sinewave will travel already 20 m. So how do you exclude a reflected sound in a normal listening environment?
Exactly, that's why closer measurements with a gate before the first reflection are more useful for the region where we care more about the direct sound, that is above room transition frequency.
 
I did some tests for long term loudspeaker compression today. This is the same test for dynamic compression, but noise is played for 1 min (IEC standard) so that voice coils, etc. heat up and any nonlinearities are exaggerated. Method:

1. Mic at 1m on axis to the speaker.
2. Output increased until SPL meter registered 76dB at mic position.
3. White noise played for 2 min.
4. Sweep done at 76dB.
5. Output electrically increased by 10dB (not by SPL meter). Sweep done at 86dB.
6. Output increased by 10dB. Sweep done at 96dB.

In Acourate:
- All measurements normalized to 0dB.
- 76dB measurement subtracted from 86dB and 96dB measurements respectively.
- 76dB measurement then represented as a flat line. The curve then shows the variance between 86dB/96dB to the flat line. We hope to see 3 flat lines stacked exactly on top of each other.

1719057216630.png


I was genuinely displeased to see this. What I see:

- Subwoofer: behaves extremely poorly. As volume gets louder, severe dynamic compression occurs.
- Woofer: behaves as expected at 86dB. By 96dB it is showing slight dynamic compression. I am unsure of the cause of the dip at 210Hz, could be some kind of cabinet resonance causing cancellation?
- Horn: exhibits the opposite of compression. In fact it exaggerates loudness. This might be why the horn sounds so lively.
- tweeter: behaves extremely poorly. Dynamic compression sets in at 86dB, but it "only" loses up to 2.3dB in loudness. By 96dB it is all over the place, that huge peak >15kHz is probably severe distortion.
 
I did some tests for long term loudspeaker compression today. This is the same test for dynamic compression, but noise is played for 1 min (IEC standard) so that voice coils, etc. heat up and any nonlinearities are exaggerated. Method:

1. Mic at 1m on axis to the speaker.
2. Output increased until SPL meter registered 76dB at mic position.
3. White noise played for 2 min.
4. Sweep done at 76dB.
5. Output electrically increased by 10dB (not by SPL meter). Sweep done at 86dB.
6. Output increased by 10dB. Sweep done at 96dB.

In Acourate:
- All measurements normalized to 0dB.
- 76dB measurement subtracted from 86dB and 96dB measurements respectively.
- 76dB measurement then represented as a flat line. The curve then shows the variance between 86dB/96dB to the flat line. We hope to see 3 flat lines stacked exactly on top of each other.

View attachment 376728

I was genuinely displeased to see this. What I see:

- Subwoofer: behaves extremely poorly. As volume gets louder, severe dynamic compression occurs.
- Woofer: behaves as expected at 86dB. By 96dB it is showing slight dynamic compression. I am unsure of the cause of the dip at 210Hz, could be some kind of cabinet resonance causing cancellation?
- Horn: exhibits the opposite of compression. In fact it exaggerates loudness. This might be why the horn sounds so lively.
- tweeter: behaves extremely poorly. Dynamic compression sets in at 86dB, but it "only" loses up to 2.3dB in loudness. By 96dB it is all over the place, that huge peak >15kHz is probably severe distortion.
That can't be right.
By the looks alone of your speaker,96dB should be a walk at the park.
Are you sure there nothing going on electrically?Or some severe EQing messing with gain structure or asking the drivers going beyond their linear range?
What weighting did you use at the SPL calibration?
 
That can't be right.
By the looks alone of your speaker,96dB should be a walk at the park.
Are you sure there nothing going on electrically?Or some severe EQing messing with gain structure or asking the drivers going beyond their linear range?
What weighting did you use at the SPL calibration?

The performance of the subwoofer, I believe. I can hear it making a "flappy" sound when it was pushed to 96dB. The performance of the tweeter, I also believe. Plasma tweeters don't go very loud, that's why they need horns. Push them too loud, and the flame becomes unstable.

I used C-weighting on my SPL meter.

Thank you for trying to make me feel a bit better, but I am a realist :) I don't mind showing all the warts on my system for all to see and criticize.
 
I did some tests for long term loudspeaker compression today. This is the same test for dynamic compression, but noise is played for 1 min (IEC standard) so that voice coils, etc. heat up and any nonlinearities are exaggerated. Method:

1. Mic at 1m on axis to the speaker.
2. Output increased until SPL meter registered 76dB at mic position.
3. White noise played for 2 min.
4. Sweep done at 76dB.
5. Output electrically increased by 10dB (not by SPL meter). Sweep done at 86dB.
6. Output increased by 10dB. Sweep done at 96dB.

In Acourate:
- All measurements normalized to 0dB.
- 76dB measurement subtracted from 86dB and 96dB measurements respectively.
- 76dB measurement then represented as a flat line. The curve then shows the variance between 86dB/96dB to the flat line. We hope to see 3 flat lines stacked exactly on top of each other.

View attachment 376728

I was genuinely displeased to see this. What I see:

- Subwoofer: behaves extremely poorly. As volume gets louder, severe dynamic compression occurs.
- Woofer: behaves as expected at 86dB. By 96dB it is showing slight dynamic compression. I am unsure of the cause of the dip at 210Hz, could be some kind of cabinet resonance causing cancellation?
- Horn: exhibits the opposite of compression. In fact it exaggerates loudness. This might be why the horn sounds so lively.
- tweeter: behaves extremely poorly. Dynamic compression sets in at 86dB, but it "only" loses up to 2.3dB in loudness. By 96dB it is all over the place, that huge peak >15kHz is probably severe distortion.
Remember that none of this is anechoic.

This means: issues below 100Hz may be because the microphone position picks up nulls, issues around a few hundred Hz show interference, although the latter may also be some mechanical issue.

Agreed the tweeter can't handle those SPLs, but then it doesn't need to most of the time given the spectral distribution of music signals. The expansion you see with your midrange horn could be, and is likely a combination of, distortion from the tube amp and distortion from the horn, which is common at high levels.

Those are my best guesses.
 
4*12" drivers that have 18mm xmax really should not collapse so drastically at just 86dB
 
Remember that none of this is anechoic.

This means: issues below 100Hz may be because the microphone position picks up nulls, issues around a few hundred Hz show interference, although the latter may also be some mechanical issue.

That is a very good point. Also, the subwoofers are located about 1.5m further away from the speakers. So to make "86dB" at the mic, they probably have to make closer to 90dB.

Agreed the tweeter can't handle those SPLs, but then it doesn't need to most of the time given the spectral distribution of music signals. The expansion you see with your midrange horn could be, and is likely a combination of, distortion from the tube amp and distortion from the horn, which is common at high levels.

I am very skeptical that the tube amp is distorting at all. The horn has a quoted sensitivity of 98dB/W/m, and the tube amp puts out a nominal 110W. The amp would be barely chugging along at that volume.

BTW, I was surprised by how loud 76dB was. 86dB was intolerable, and 96dB had me plugging my ears. I normally listen at about 60-70dB. It is really quiet where I live. I find that when the noise floor is so low, music does not need to be turned up so loud. For example, I typically listen louder in the day and softer at night. I wonder if you guys have the same experience.
 
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