a helmholtz resonator is a resonator as the name says. it can add a zero to kill an existing mode but add another mode deeper down.
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Paul Hales on room equalization on Soundstage! Network
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what is that experiment telling us? EQing something down for a limited period of time is a not a linear time invariant processing step, is distortion.
Sokel
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Testing about how EQ can affect impact is as simple as a pushing a button, even with a simple set-up using foobar and Mathaudio for example.
EQ the system properly for peaks (under 100Hz or so), adjust the amplitude of the before/after (also simple at this setup) and go.
I did it, 1000 times. And as my main room mode is at 30Hz or so, impact is the same as it resides far higher than that, at the 100Hz-250Hz area.
What's interesting about impact is that is closely tied with what some flowery terms would describe as "big sound" or "full-body" sound (similar to what a violin does up-close, which is near physical) and that even at far lower levels before it becomes physical and punches.
EQ the system properly for peaks (under 100Hz or so), adjust the amplitude of the before/after (also simple at this setup) and go.
I did it, 1000 times. And as my main room mode is at 30Hz or so, impact is the same as it resides far higher than that, at the 100Hz-250Hz area.
What's interesting about impact is that is closely tied with what some flowery terms would describe as "big sound" or "full-body" sound (similar to what a violin does up-close, which is near physical) and that even at far lower levels before it becomes physical and punches.
a zero that nulls a single frequency is a two tap fir filter. a zero or null is an interference and not energy storage. delayed sound from a wall reflection interferes with direct sound - even this is only a few msec.
Energy is stored in the standing waves. the sound power of the source gets trapped and accumulated in the standing wave and this energy is released with a slow exponential decay. it’s the energy storage that gives the time domain problems (modes ringing for seconds). in free space the sound power just travels out in the atmosphere and does not get trapped like in a room.
So dips/nulls/zeros is a completely different animal than poles/modes/standing waves which store energy and are global in the room. Zeros are extremely position dependent.
In a theoretical ideal, one front array (the entire front wall in an absolute ideal) radiates a plane traveling wave into an infinite tube, with no transverse modes being excited. Infinite rooms are impractical, even very long one, where I listened to few concerts (Vader's bass sounded unforgettable)), is now inaccessible. Therefore, this radiation must be somehow eliminated at the back wall. A passive absorber in a SBA or a second array in a DBA is needed for this. More details in the DBA threads.
Equalizers can be used to the compensate that reality differs from theory, including somewhat advanced ones. For a rear array, the practical goal of equalization may not be some kind of "flat frequency response," but rather matching the amplitude and phase of the front radiation as it reaches the rear wall (sic!).
He isn't the first speaker manufacturer to express this view. Later in the interview, he also addresses the multi-subwoofer setup. It makes me wonder if it’s really necessary to force the frequency response; for instance, if you have a 12 dB peak, perhaps creating a smoother response curve—maybe using Helmholtz resonators—would be better than relying on DSP. I certainly don't think
the interviewee or the
other
designers
are unfamiliar with the Fourier transform or how DSP works in general
audiofooled
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This is so complicated I am not sure what I am looking at and what is may meanCan you explain, thank you.
Well, this is just steady state (pink noise). If you look at L/R channel on each of the graphs, it shows the filtered impulse response at that frequency in case the right channel being inverted, so out of phase with respect to the left one, where the resulting pressure would be close to zero at this measurement position (3,5 meter distance). Yellow would be vector sum in case both of the channels are fully in phase and summing constructively (there's no difference in amplitude as the plots are normalized).
Now, if you look at particular wavelengths, it has to do with physical distance between the bass radiators and their position within the room, affecting impulse response behavior and what happens in room after the initial impulse. Frequency dependent summation with regards to the wavelengths shows timing differences at the peak of the impulse that are fractions of a millisecond, however, the response is nearly identical regardless of frequency. Whereas for each of the channels it is not, in comparison to the constructive summation.
It shows that each of the channels are contributing equally as far as energy in and energy out, at this measurement position, even though physical position is asymmetric to the room boundaries. Not easy to do. The practical benefits can be observed in case of complex signals, such as this:
Does 'envelopment' exist? Can it be measured?
"Envelopment" is a perceptual term meaning that it sounds like you are immersed in the soundfield, not "looking through a window" or some other sensation. This is "envelope" meaning "around the subject". The word "envelop", however, is also used in a technical sense as the Hilbert Envelop of a...
www.audiosciencereview.com
If you think about it, just about any music signal would have complex requirements and steady state measurements do not show the full picture.
BTW P. Hales produces reasonable (as far as can be judged from the photo) speakers:
I crop the excess part. And I would plug the holes in the subwoofers.
Watched Direct vs. reverberant sound and part of Can bad rooms benefit from room correction? I didn't find anything particularly interesting. Two comments: the 0.4-0.5 ms RT60 is a bit much for me, and instead of a single drum hit, one could listen to a matt test or, better yet, music. https://templumanimamorti.bandcamp.com/album/2019-remaster-2022 eg
I crop the excess part. And I would plug the holes in the subwoofers.
His https://www.proaudiotechnology.com/s2115sm above. Nothing wide, normal 90x60º.
Watched Direct vs. reverberant sound and part of Can bad rooms benefit from room correction? I didn't find anything particularly interesting. Two comments: the 0.4-0.5 ms RT60 is a bit much for me, and instead of a single drum hit, one could listen to a matt test or, better yet, music. https://templumanimamorti.bandcamp.com/album/2019-remaster-2022 eg
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Case with deep dip (s) refers to not well treated room.
Try to EQ that case is always question why.
Dips + Peaks are almost always series of knots in phase, so try to EQ just one peak + dip by 2 filters is wasting of filters.
I would say unwrapping the in-room phase is always helpful.
Try to EQ that case is always question why.
Dips + Peaks are almost always series of knots in phase, so try to EQ just one peak + dip by 2 filters is wasting of filters.
I would say unwrapping the in-room phase is always helpful.
I know one guy who can afford whatever is expensive.
He lives in functionalist villa with glass walls and large rooms, where is almost nothing except glass table, glass showcases, all on glossy concrete floor.
He asked me a question what hi-fi high-end system I would recommend.
I said no one, it is wasting of money, and even cardioidic would not work here.
He said he has got an offer of complex system hidden in plaster walls by buzzing those, but by geometry it was impossible to gain uncompromised result of stereo.
I know other guy who sold expensive system to similar space, and he said...It is not bad for what it is...and big smile.
I do not understand expensive houses and or spaces that are built without any passive acoustic technology included in the project.
He lives in functionalist villa with glass walls and large rooms, where is almost nothing except glass table, glass showcases, all on glossy concrete floor.
He asked me a question what hi-fi high-end system I would recommend.
I said no one, it is wasting of money, and even cardioidic would not work here.
He said he has got an offer of complex system hidden in plaster walls by buzzing those, but by geometry it was impossible to gain uncompromised result of stereo.
I know other guy who sold expensive system to similar space, and he said...It is not bad for what it is...and big smile.
I do not understand expensive houses and or spaces that are built without any passive acoustic technology included in the project.
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The claim is that notching out room modes decreases transient impact because the notch reduces the bass in the transient direct sound. Im saying that notch dosnt make much audible difference to the transient, I believe the claim was the first 6ms. So remove the bass from the transient and see what it sounds like.
So punching in Eq (de-essing .eg) is distortion? And so is compression/limiting (try recording/mixing with out it) and even ridding a fader and is making distortion. I know it is, its amplitude modulation, but why dont we hear it as "distortion"?
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Why not?))) Let's listen to the recording without a compressor\limiter. Where can we find it?
Tom Danley mic recordings are excellent, but it's not music.
I highly recommend everyone turn the volume up to maximum and listen\enjoy to Danley's recordings. Disclaimer: if your LS50s blow up, it's your and their fault, not ours.
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that’s not possible mathematically - bass takes time…
it’s all nonlinear or not LTI. in my opinion all these things drain the life out of the recordings. weapons of mass destruction
Curvature
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Must be different physics for pianos below Schroeder, then.
?? Take a kick drum sample and EQ the bass out. Take the first 6ms of that and quick crossfade (prevent clicks) it with the orignal sample. No bass in transient.
Thats my point. Bass takes time to peak but the LF wave starts at zero and EQ can remove the little bit thats there. And I believe that this listening test still reveals something about the claim that EQing bass out of direct sound to lower room modes ruins transients.
it’s all nonlinear or not LTI. in my opinion all these things drain the life out of the recordings. weapons of mass destruction
Than why do many recordings (that use a lot of processing) still sound undistorted. Im not talking about dynamics. Have you done much recording/mixing/mastering?
audiofooled
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Attached you may find a recorded kick drum transient (duration roughly 200 ms).
This is a REW scope capture at MLP (3,5 meter distance) of my system reproducing this signal:
For the sake of simplification, this is the impulse of the signal downmixed to mono:
And this is system's impulse response from the scope data:
Overlay of the impulse responses:
Phase around 44Hz fundamental:
Overlay of ETC plots:
Wavelet 1/1 of the signal:
Wavelet 1/1 of the system response:
This is a rather crude method, measured at a single point. However it can give you a rough idea on the potential signal demand and some idea about in room reproduction of it.
This is a REW scope capture at MLP (3,5 meter distance) of my system reproducing this signal:
For the sake of simplification, this is the impulse of the signal downmixed to mono:
And this is system's impulse response from the scope data:
Overlay of the impulse responses:
Phase around 44Hz fundamental:
Overlay of ETC plots:
Wavelet 1/1 of the signal:
Wavelet 1/1 of the system response:
This is a rather crude method, measured at a single point. However it can give you a rough idea on the potential signal demand and some idea about in room reproduction of it.
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HighImpactAV
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I visited the Theory Audio Design demo room at CEDIA 2026. Paul Hales did a 60 minute demo of his speakers and subwoofers in a 30' x 30' x 15' room. I was at the show from Wednesday-Friday and visited all the theater demo rooms. No EQ was used in the room and all volume leveling/matching was done by ear. This was easily the best sounding room at the show. The bass was visceral and detailed.
Here is an interview with Paul Hales at CEDIA:
Here is an interview with Paul Hales at CEDIA:
Three of my rooms will fit into this room if you put them on the side wall.
Is the correct interpretation of this that your system spreads out the time of the kick drum transient by a factor of 4 so the perceived "impact" dP/dt, is 1/4th of what the original signal would have created if it was reproduced accurately? Can you tell what factors contribute to these delays and by how much i.e. speaker drivers, crossover filters, DSP corrections, or room interactions?
audiofooled
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The signal requires very steep rise time and many harmonics arriving on a very tight schedule. No one knows how it was recorded but there's no propagation delay in the signal itself, whereas at 3.5 meter distance, there is.
Still the wave front shape is preserved from the initial start of the impulse. Rise time, dP/dt and perceived atonal impact would depend on acceleration but from the last graph you may get the idea that the time of delivery would depend not only on measurement distance, but also the initial wave front would affect the time of delivery of all the harmonics riding that wave. Energy down to DC at each of the ribs on the timeline would have taken round trip from front to the back wall.
This is not the right way to measure this as it shows only the pressure at a single point and I can't even speculate how CNS handles this as far as body sensation.
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