• Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!
Where did this assumption come from? And how do you know it is even achievable?
Hi again, you spotted my poor phrasing, implying intensity should be equal at any distance. I meant the spectral content should ideally be equal at any distance.
And Constant Directivity is not the ONLY thing.

Still, wherever i find myself (not only in a fixed listening position, like Siggy), i hope the direct sound outlevels the reflected sound, the late reflections WAY outnumber and/or outlevel the early ones, and said reflections tend to uphold the spectral content of the direct sound. Until i find a better definition, these three will be among the things i strive for.

Tx for keeping me on the straight n narrow..
 
You will find no evidence that reverberation can be higher in level than direct sound without detriment to localizability and intelligibility.

For the Elbphilharmonie, here are the acoustical parameters: https://www.akutek.info/Mitt Bibliotek/IOA Auditorium Acoustics Hamburg 2018/Additional/papers/p11.pdf

View attachment 504918

None of the acoustics parameters show that direct sound is lower than reverberant sound: C80 is positive across all frequencies, center time is low, as is early decay time, particularly in high frequencies where it is important.

This is a modern hall designed with electronic reinforcement and seats positioned to receive direct sound first, with deliberately low early reflections and highly-scattered late reflections. As such, clarity and consistency across frequencies were prioritized for as many seats as possible. This is in contrast to traditional rectangular halls where the above acoustical parameters as well as the strength of early reflections significantly changed depending on the position of the seat.

Again, this has little to do with small rooms.
Ha! But it does have to do with my now better phrased (not yet shifted) paradigm:

Wherever i find myself (not only in a fixed listening position, like Siggy), i hope the direct sound outlevels the reflected sound (by a lot, preferably), the late reflections WAY outnumber and/or outlevel the early ones, and said reflections tend to uphold the spectral content of the direct sound. Until i find a better definition, these three will be among the things i strive for.
 
Ha! But it does have to do with my now better phrased (not yet shifted) paradigm:

Wherever i find myself (not only in a fixed listening position, like Siggy), i hope the direct sound outlevels the reflected sound (by a lot, preferably), the late reflections WAY outnumber and/or outlevel the early ones, and said reflections tend to uphold the spectral content of the direct sound. Until i find a better definition, these three will be among the things i strive for.
Your goals are a mix of already-achieved and impossible, in one case.

In a normal listening situation, direct sound will always be louder than early reflections, and late reflections will always outnumber, but never be louder than, early reflections.

The spectral content of late reflections depends on the design of the speaker foremost and the room second, although most home spaces tend to act the same way. I agree constant directivity or similar designs make the most sense, particularly if they maintain pattern control down to around 100Hz. Although, frankly, many speakers sound very good even if they deviate as long as the overall radiation pattern is relatively even and there are no resonances.

You can manipulate the strength of early reflections with absorption panels and do something similar for late reflections by using secondary highpassed speakers firing away from you and into the walls and ceiling, preferably at an angle.
 
Hi again, you spotted my poor phrasing, implying intensity should be equal at any distance. I meant the spectral content should ideally be equal at any distance.
And Constant Directivity is not the ONLY thing.

Still, wherever i find myself (not only in a fixed listening position, like Siggy), i hope the direct sound outlevels the reflected sound, the late reflections WAY outnumber and/or outlevel the early ones, and said reflections tend to uphold the spectral content of the direct sound. Until i find a better definition, these three will be among the things i strive for.

Tx for keeping me on the straight n narrow..
This seems like a good / reasonable goal to me, but I think you'll also need to manage room acoustics pretty rigorously to achieve the late / early balance?

As for the overall goal of achieving a realistic illusion of presence, I feel like the parameters of the recording need to be controlled also. I don't have data for this but my gut tells me the same speaker and room won't make a 2-mic stereo live orchestra recording and a multi-tracked studio production sound real. One needs preservation of cues in the recording and the other needs cues to take shape de novo.
 
Your goals are a mix of already-achieved and impossible, in one case.

In a normal listening situation, direct sound will always be louder than early reflections, and late reflections will always outnumber, but never be louder than, early reflections.

The spectral content of late reflections depends on the design of the speaker foremost and the room second, although most home spaces tend to act the same way. I agree constant directivity or similar designs make the most sense, particularly if they maintain pattern control down to around 100Hz. Although, frankly, many speakers sound very good even if they deviate as long as the overall radiation pattern is relatively even and there are no resonances.

You can manipulate the strength of early reflections with absorption panels and do something similar for late reflections by using secondary highpassed speakers firing away from you and into the walls and ceiling, preferably at an angle.
Hi again. Doesn't managing the room help, by having less reflected energy in the first place? Such as dipole nulls, which are in large part located where early reflections would come from? And below 100Hz, the most problematic area for rooms, where the reflected bass energy is down 4.8dB (depending) compared to closed box? How about living room Line Arrays avoiding floor and ceiling reflections? And implemented carefully, CAN have a radiation pattern similar to what dipole does in omni range?

Since rooms are not always within our control, i would generally prefer a speaker to be dependant on room properties as little as possible.

Tx for your input, appreciated!!
 
This seems like a good / reasonable goal to me, but I think you'll also need to manage room acoustics pretty rigorously to achieve the late / early balance?

As for the overall goal of achieving a realistic illusion of presence, I feel like the parameters of the recording need to be controlled also. I don't have data for this but my gut tells me the same speaker and room won't make a 2-mic stereo live orchestra recording and a multi-tracked studio production sound real. One needs preservation of cues in the recording and the other needs cues to take shape de novo.
Btw, i'm on my phone, without glasses, so perhaps be back later. But recordings also beyond our control, except when selecting what to buy... :-(
 
You will find no evidence that reverberation can be higher in level than direct sound without detriment to localizability and intelligibility.

For the Elbphilharmonie, here are the acoustical parameters: https://www.akutek.info/Mitt Bibliotek/IOA Auditorium Acoustics Hamburg 2018/Additional/papers/p11.pdf

View attachment 504918

None of the acoustics parameters show that direct sound is lower than reverberant sound: C80 is positive across all frequencies, center time is low, as is early decay time, particularly in high frequencies where it is important.

This is a modern hall designed with electronic reinforcement and seats positioned to receive direct sound first, with deliberately low early reflections and highly-scattered late reflections. As such, clarity and consistency across frequencies were prioritized for as many seats as possible. This is in contrast to traditional rectangular halls where the above acoustical parameters as well as the strength of early reflections significantly changed depending on the position of the seat.

Again, this has little to do with small rooms.
Although not directly relevant to home audio (Repro of recorded AOT original live; Huge distances AOT small, often too small) that is still an inspiring article. Fascinating they built a suspended room-in-room at that size, incredible. Makes me wanna go there and listen to a concert. Five hrs by train or car from where i live.

Your statement "You will find no evidence that reverberation can be higher in level than direct sound without detriment to localizability and intelligibility" agrees with all information i have come across in my life up until now. It makes sense, too.
There are a few circumstances where it might not make it too much worse, like steady state organ notes in a large cathedral, but better? I would need some serious explanation on the physics of that.

So, tx for your involvement!
 
Your goals are a mix of already-achieved and impossible, in one case.

In a normal listening situation, direct sound will always be louder than early reflections, and late reflections will always outnumber, but never be louder than, early reflections.

The spectral content of late reflections depends on the design of the speaker foremost and the room second, although most home spaces tend to act the same way. I agree constant directivity or similar designs make the most sense, particularly if they maintain pattern control down to around 100Hz. Although, frankly, many speakers sound very good even if they deviate as long as the overall radiation pattern is relatively even and there are no resonances.

You can manipulate the strength of early reflections with absorption panels and do something similar for late reflections by using secondary highpassed speakers firing away from you and into the walls and ceiling, preferably at an angle.
So what is a normal listening position to you? If the critical distance in my room is 10ft/3m, that means i will spend 80% of my time in the far field, were reflections are louder than direct sound for the given CrDi. In the kitchen 20ft behind the 'normal' listening position, at the diner table, 15 to the side of that, etc. I don't sit, i hover about the house and like to have a band playing while i do. So i am aiming to add most of the room to the nearfield, needing a CrDi. of about 10m/33ft. Am i on the wrong track?

And when early reflections are just outnumbered by the late, regardless of frequency range, do they not then carry more energy and over longer distance, thereby overwhelming the earlies? As in dipoles that have a null up to about 1.6kHz, and starting a cylindrical dipole wavefield above that up to approx. 10kHz, creating a continuing null, just connected to dfferent princple? Thereby scarcely illuminating the side walls and ceilings close to them, and liberally the front and back walls plus side walls, floors and ceilings surroundiing those? Over most of their frequency range? So these reflections will all be recognized as separate secundary events by our acoustic brain?

Not a wise guy but genuinely asking you these questions, because i have arrived at a solution that would at least approach this description, but it is all in theory so far. The theory seems to hold.
But, as a Dutch saying goes: The difference between theory and practice is that in theory there is no difference, but in practice there is!

Not that you have to supply me with answers, but, as you brought it up, after i brought it up, etc.
If it starts to annoy you, don't worry at all about dropping it!

Gtz
 
None of the acoustics parameters show that direct sound is lower than reverberant sound: C80 is positive across all frequencies, center time is low, as is early decay time, particularly in high frequencies where it is important.

C80 is a clarity index for large venues and music, particularly complex static sounds (such as an orchestra or organ). It has nothing to do with ratio direct sound to indirect sound. In contrary, the calculation is based on the assumption that all ´early´ reflections contribute to clarity and perceived loudness, while later ones are defined as deteriorating clarity. The threshold is, as the name c80 is hinting, 80ms. Which is is indeed a good indicator for static (over time) sounds, but have nothing to do with direct sound, localization, and absolutely nothing with intelligibility. In contrary, intelligibility is depending on direct sound and very early reflections dominating, in the early litarature one might find the C50 index for speech (and singing with declamation). Which is still a very long window and no guarantee to really understand every word.

Key thing to understanding why this concert hall sounds different, is the lack of a support wall behind the musicians, like it is installed in so many other, similarly-shaped concert halls. This contributes to a pretty diffuse reflection pattern even within the early time window.

You will find no evidence that reverberation can be higher in level than direct sound without detriment to localizability and intelligibility.

The ratio of direct sound to indirect sound is depending on the directivity of the source and the distance between source and listener/microphone, and for most of venues and seats the indirect sound will be dominant. That does not mean localization stability and intelligibility are deteriorated. If that is the case, is vastly dependent on how consistent the reflectogram pattern of the room is and from which time window within in the reverb pattern we can assume something close to perfectly diffuse reverb (which includes directional information).

this has little to do with small rooms.

Certainly true, I just wanted to give examples of critical listening distance and intelligibility being independent from each other.
 
This seems like a good / reasonable goal to me, but I think you'll also need to manage room acoustics pretty rigorously to achieve the late / early balance?

As for the overall goal of achieving a realistic illusion of presence, I feel like the parameters of the recording need to be controlled also. I don't have data for this but my gut tells me the same speaker and room won't make a 2-mic stereo live orchestra recording and a multi-tracked studio production sound real. One needs preservation of cues in the recording and the other needs cues to take shape de novo.
Hey Kemmler, to start with the latter, you are obviously right about recording practices. But the only way to control that is to not buy bad recordings, no? Little else i can do about it..

About controlling the room acoustics, also obviously right, but there's various approaches. I always like to start out with a speaker that does not excite the room more than necessary to begin with. Therefore, open dipoles.

But these are flawed also, at least in that dipole nulls and peaks have to be managed (which can be done by driver and baffle size and shape, and choice of XO frequency) and in that dipole behavior stops when reaching comb filtering wavelengths. Which would in my plan be at about 1.6kHz given baffle shape and size of the driver array operating from ~700Hz to 18kHz. Above that 1.6kHz i'll try to maintain a null, this one caused by figure of eight radiation from a dipole line source.

With a 103mm baffle width, shaped like a very shallow waveguide in both front and back, with rounded outer edges (radius 15mm) over its 800 or 1200mm height (to be determined), the line array will be prepared to deal with baffle step and diffraction issues. But i don't expect any, because of the wide dispersion angle caused by all wavelengths being pressurized through 6mm horizontal width openings. Not the full 180 degrees front and back, but not too far from it. Above 1.6kHz it stops being omni, hence no dipole null, but it starts being a true line source.

If it works as engineered, proposed sizes and radiation patterns avoid any baffle step.

See anything wrong? Shoot! Shooters rock.
 
Last edited:
The concept of Ripol i have to research,

Compact folded-baffle dipol subwoofer, looking like this:

Ri.jpg


but it seems reasonable we would want to minimize room reflections to keep the acoustics of the recording dominant in near field. If the room is anechoic however, i'm not sure it would be a good extreme of that proposition.

For two-channel stereo, a compromise is mandatory. While under anechoic conditions, a surprisingly consistent imaging including limited angle of reverb and depth-of-field can be achieved with recordings containing a significant level of reverb from the concert venue, this is not the case with ´dryer´ recordings or such with mostly decorrelated reverb (like pop, rock productions), they appear with annoying proximity. The second problem is our auditory sense being used to a certain level of reverb coming in from the sides, the rear and from above, which we can distinct thanks to HRTF. If you attenuate this to the level of being inaudible, you create a strange effect of ´looking through a window from a black-out room´.

If reverb in the listening room is kept sufficiently diffuse, not overly dominant in level, and of similar tonal balance compared to the direct sound and reverb in the recording, a pretty good compromise is possible.

The question is how to achieve this. Diffusion in the room, suppressing early discrete reflections, while keeping all indirect sound tonally balanced, are certainly a good starting point.

the way i foresee my project is by minimizing room reflections by a combination of several physically implemented parameters controlling directivity, including driver- and baffle- -sizes and -shapes, carefully choosing frequency ranges, OB, LA, etc. I may share some sketches and more detail later, after i see where the somewhat hostile athmosphere on this forum is going.

Sounds very very promising, but controlling directivity over vast frequency bands, is really very difficult.

The tweeters i'm referring to are the PT6816-8, which are dipole, emitting through two rows of vertically arranged ovular holes in the rear and four rows in the front. The holes are 6mm wide for a 15mm wide pair of rows in rear and a 35mm foursome in front.

If I recall it correctly, I know such planar transducer, companies like PS Audio and Radian using units with surprisingly similar geometry yet own diaphragm/drive structure.

The structure of the holes is almost irrelevant, as the geometry of the diaphragm and array of holes is creating a homogeneous wavefront anyways. That becomes a problem due to the sheer size of the diaphragm, which is causing a planar wavefront with usable listening window narrowing towards higher frequencies, particularly vertically, as expected from a truncated linesource. As the diaphragm is also pretty broad at the same time, you have a threshold frequency above which the directivity index is skyrocketing and horizontal listening window narrowing down as well.

If you want to design anything resembling a constant directivity concept with such driver, turning it into a midrange and keeping the passband pretty narrow seems to be a solution. For example PS Audio are using such transducers from approx. 500 to 2,500Hz which are surprisingly low crossover frequencies ( @Chris Brunhaver seemingly is a member here, maybe he might want to share his thoughts behind it). The problem how to combine it seamlessly with a tweeter, stays unsolved.

The other question is if you want to operate such planar magnetostat as a true dipole, or put it in an enclosure. From practical experience, I do not see much point to use a dipole here, as the sound emanating to the rear, under usual living conditions is hitting a wall quickly, contributing to a lot of early reflections hence significantly reducing critical listening distance.
 
C80 is a clarity index for large venues and music, particularly complex static sounds (such as an orchestra or organ). It has nothing to do with ratio direct sound to indirect sound. In contrary, the calculation is based on the assumption that all ´early´ reflections contribute to clarity and perceived loudness, while later ones are defined as deteriorating clarity. The threshold is, as the name c80 is hinting, 80ms. Which is is indeed a good indicator for static (over time) sounds, but have nothing to do with direct sound, localization, and absolutely nothing with intelligibility. In contrary, intelligibility is depending on direct sound and very early reflections dominating, in the early litarature one might find the C50 index for speech (and singing with declamation). Which is still a very long window and no guarantee to really understand every word.

Key thing to understanding why this concert hall sounds different, is the lack of a support wall behind the musicians, like it is installed in so many other, similarly-shaped concert halls. This contributes to a pretty diffuse reflection pattern even within the early time window.



The ratio of direct sound to indirect sound is depending on the directivity of the source and the distance between source and listener/microphone, and for most of venues and seats the indirect sound will be dominant. That does not mean localization stability and intelligibility are deteriorated. If that is the case, is vastly dependent on how consistent the reflectogram pattern of the room is and from which time window within in the reverb pattern we can assume something close to perfectly diffuse reverb (which includes directional information).



Certainly true, I just wanted to give examples of critical listening distance and intelligibility being independent from each other.
It seems to me, assuming you and others that seem informed speak truth (all the greats i read on one side, the acoustics engineers for Grosser Saal in Hamburg on the other), that the desired room properties of a 2100 hundred-person vinyard shaped concert hall for original music production on the one hand, and a large-ish home listening room with a disproportionally large near field for recording-repro on the other hand, are diametrically opposed.

In the home, early reflections are detrimental, because they blur the image. We can't separate them from the direct sound when under 6-10 milliseconds, resulting in them being perceived as part of the direct sound, aka time smearing.

Late reflections less so, because we can separate them, recognizing them as a separate event (echo), which adds space. Not always good, but better than smearing.

Dipoles and Line Arrays (and various other speaker concepts) address both. Less total room reflections on account of their radiaton patterns, but much less early reflections and only somewhat less late. While at it, also improving spectral balance of said reflections.

Again, shoot me if i'm wrong!
And thanks for being here at all..
 
Last edited:
Compact folded-baffle dipol subwoofer, looking like this:

View attachment 505028



For two-channel stereo, a compromise is mandatory. While under anechoic conditions, a surprisingly consistent imaging including limited angle of reverb and depth-of-field can be achieved with recordings containing a significant level of reverb from the concert venue, this is not the case with ´dryer´ recordings or such with mostly decorrelated reverb (like pop, rock productions), they appear with annoying proximity. The second problem is our auditory sense being used to a certain level of reverb coming in from the sides, the rear and from above, which we can distinct thanks to HRTF. If you attenuate this to the level of being inaudible, you create a strange effect of ´looking through a window from a black-out room´.

If reverb in the listening room is kept sufficiently diffuse, not overly dominant in level, and of similar tonal balance compared to the direct sound and reverb in the recording, a pretty good compromise is possible.

The question is how to achieve this. Diffusion in the room, suppressing early discrete reflections, while keeping all indirect sound tonally balanced, are certainly a good starting point.



Sounds very very promising, but controlling directivity over vast frequency bands, is really very difficult.



If I recall it correctly, I know such planar transducer, companies like PS Audio and Radian using units with surprisingly similar geometry yet own diaphragm/drive structure.

The structure of the holes is almost irrelevant, as the geometry of the diaphragm and array of holes is creating a homogeneous wavefront anyways. That becomes a problem due to the sheer size of the diaphragm, which is causing a planar wavefront with usable listening window narrowing towards higher frequencies, particularly vertically, as expected from a truncated linesource. As the diaphragm is also pretty broad at the same time, you have a threshold frequency above which the directivity index is skyrocketing and horizontal listening window narrowing down as well.

If you want to design anything resembling a constant directivity concept with such driver, turning it into a midrange and keeping the passband pretty narrow seems to be a solution. For example PS Audio are using such transducers from approx. 500 to 2,500Hz which are surprisingly low crossover frequencies ( @Chris Brunhaver seemingly is a member here, maybe he might want to share his thoughts behind it). The problem how to combine it seamlessly with a tweeter, stays unsolved.

The other question is if you want to operate such planar magnetostat as a true dipole, or put it in an enclosure. From practical experience, I do not see much point to use a dipole here, as the sound emanating to the rear, under usual living conditions is hitting a wall quickly, contributing to a lot of early reflections hence significantly reducing critical listening distance.
That Ripol driver duo seems interesting. The openings are to the sides, so a dipole null will be present there? Then it's a practical way to sum two woofer outputs in minimum real estate. I may apply it if true.

The structure of the holes in this case seems paramount. The (18mm wide) active surface of the membrane being easily over twice the surface of the hole array, the holes will be loaded, especially on the rear (which i may use as the front). The loading makes the holes into physically independent point sources much smaller then even the highest frequency produced. Of course i have to test this. But they sure work well as tweeters crossed at 700Hz up to some 18kHz without issue. Some of the best tweeters he ever tested, said Chua (ampslab-spk.com). Few artefacts, great HD, he only did not publish of axis SPL, so i will mail him if he has any data on that.

Btw, not designing speakers for a little room with a fixed small sweetspot at 10ft CrDi. That's easy, there's thousands of those, and i seldom sit just sitting. Instead looking for speakers that envelop most of a largish room into the near field, with somewhat diminished late reflections and much diminished early reflections, and close to constant spectral content in a mostly uniform radiation pattern up to 8~10kHz.

Tx for the warning! Shoot me again whenever you like!
 
Last edited:
I have encountered several setups, particularly broad-baffle speakers in smaller rooms leading to dominant early reflections from the sides

I can see a broad baffle lowering the baffle-step frequency, but it's not obvious to me how it would contribute to "dominant early reflections from the side." Can you elaborate, or give an example?

I'm NOT disputing you; I'm trying to learn... imo broad baffles and avoiding dominant early reflections can both be desirable.

How is the broad-baffle GGNTKT M3, which you mentioned earlier, in this respect? Seems to me its 140-degrees-wide horizontal pattern could be a bit too wide in some situations.
 
HI EVERYONE, MAJOR EMERGENCY EDITING ALERT!

I posted an opening statement at the beginnig of this thread that was hastily and sloppily phrased.
This caused it to contain inaccuracies, falsehoods and phrasing prone to misinterpretation.
I am like that. Jumping in. Would usually do better when i wait a bit and let it settle.
Not too smart to begin with (in certain respects anyway), jumping in with sloppy posts makes me look even dumber than i already am!

SO HERE'S MY REVISED AND EDITED OPENING STATEMENT FOR THE THREAD ('FORUM'). HOPE I DID A LITTLE BETTER THIS TIME.

2CD is key in the search for reproducing a credible illusion of live recordings (from concert hall or music studio) in our home listening environment.

CD simultaneously happens to be the acronym of two parameters highly relevant to attaining this goal:
-Constant Directivity (ideally closely approached)
-Critical Distance (ideally far, to create a wide listening area)

Both very important and difficult to manage well in small(ish) reproduction listening environments, as well as critical and, on top of that, the second can improve greatly by good implementation of the first.

Both Open Baffle configurations and specific "Living Room Line Arrays" (among other speaker concepts) have shown to be a step, or several, in the right direction when implemented well. Since a while actually, as a very famous and specific type of line array (Don Keele's CBTs of the 1970's) and very famous dipoles (Quad ESL63s and Siggy's {RIP} Orions/LX521s of the 1960's and 1980/1990's) remind us.

Quoted years may not be entirely correct, but the concepts have been around for at least half a century. Yet (almost) no manufacturer has made serious attempts at continued and sustained mass production for home audio, for obvious marketing and other practical reasons.

That means little funding has gone into the research also, leaving us in the hands of some great contributors in the area between Pro and DIY audio, among which the aforementioned greats. There's some on ASR, some on DIYaudio, many AES or BAS members, some have their own websites, some are professionally involved but share their knowledge unreservedly and their white papers are freely accessible. .

I'd like to participate in this research, accompanied by folks some of whom are better educated and more experienced than myself, hopefully.
My own research focuses on other's research. Meta- or comparative research so to speak, so mostly theoretical so far. I invite anyone to shoot me down at any time. If you have valid arguments, knowledge or experience: put me in my place. But i found comparative research can be illuminating and very satisfying.

I lack a formal electronics or acoustics education, but i am an engineer, tend to grasp logics (and a tiny bit of logic) and am generally able to adequately absorb basic concepts discussed in white papers slightly outside my field (by academics, engineers or otherwise). Of which there is still an abundant treasure to be found about Open Baffles and Line Arrays on www and in print.

Standing on the shoulders of giants i hope to take some small steps i could never accomplish on my own, and then do a test build and give back measured results.

If any of the above resonates with you, take your shot. Bring a machine gun!


For those not familiar with the concepts:

1.
Constant Directivity.

The most used term for Equal Power Response, Uniform Radiation Pattern and some similar terms, all meaning the same:

For our auditory senses (two ears + psycho-acoustic brain centre + what else involved?) to recognise an auditory event to be genuine, as in "i am hearing musicians, not a loudspeaker-cone in a coffin", the intensity of the sound should radiate more or less evenly in all directions, even maintaining its spectral balance when reflected. This incidentally means off-axis listening will get better: although the SPL will lower, the spectral content will remain more or less unchanged (best case scenario).

Given the physical limitations of acoustic transducers and their applications, as well as the acoustics of most rooms, only very few loudspeaker-listening room combinations attain this goal.

ALL well designed audio devices sound decidedly identical, given the enormous differences between ANY two makes of loudspeaker in any two rooms, OF ANY QUALITY LEVEL, AT ANY PRICE.

That should tell us something.


2.
Critical Distance.

The distance away from the sound source (speaker) past which reflected SPL is higher than direct SPL.
Also known as the "Near Field/Far Field" transition. They signify the same:

When reflected sound overwhelmes direct sound, articulation becomes blurred (specifically on transients/impulses), intelligibility of speech is compromised and eventually the whole sound stage drowns in a sea of undefined acoustic rubble.

In a very reflective room this distance and the near field are small, in a 100% anechoic room this distance and the near field could (theoretically) cover the entire room.

Whether that makes an anechoic room the ideal listening venue, however, is under debate IMO and will be addressed later (in this thread or elsewhere).
Cheers!
 
Where did this assumption come from? And how do you know it is even achievable?
Hi Amir M, i rephrased my opening statement, because it was sloppy and full o' ****.
While my basic starting points remain the same, re-reading i saw falsehoods, sloppy phrasing and inaccuracies.
I can imagine it didn't make much sense to you in the first attempt.
Thanks for not blocking me forthwith..

Cheers
 
No thank you.
I actually phrased the opening post to this thread horribly sloppily, is what i found upon re-readng it today.
Happens sometimes.

So although, if you're not interested, posting was totally unnecessary, i can totally imagine the level of irritation that would make me comment as you did.
That was still quite merciful and gracious actually!

Tried to make more sense of the post as to factual accuracy and more careful description of topics of interest, tx to you.

Gtz
 
Hi again. Doesn't managing the room help, by having less reflected energy in the first place? Such as dipole nulls, which are in large part located where early reflections would come from? And below 100Hz, the most problematic area for rooms, where the reflected bass energy is down 4.8dB (depending) compared to closed box? How about living room Line Arrays avoiding floor and ceiling reflections? And implemented carefully, CAN have a radiation pattern similar to what dipole does in omni range?

Since rooms are not always within our control, i would generally prefer a speaker to be dependant on room properties as little as possible.

Tx for your input, appreciated!!
The full null for dipoles is at 90 degrees. At every other angle there is some output. Remember that there is a timing, angle and spectral difference to reflections. Those originating from nearby boundaries, second in pathlength to direct sound, are what we call early reflections and they will have a perceptual effect.

The most problematic frequency area is 100-500Hz, where you have nonminimum phase cancelations that can't be EQed, the domain of speaker boundary interference response (SBIR). It occurs when reflections are similar in strength to direct sound, and the wavelengths are long enough to cause local peaks and nulls. This is where pressure-based (like panels) absorptive treatment is the most useful.

Below 100Hz, the 4.8dB figure you mention is not clearly related to anything I can think of. Anyway, there it's the room dimensions which dictate frequency response.

For line arrays, I've personally never heard an implementation which was truly floor-to-ceiling or one that was mounted in a corner. Depending on the exact design they would easily produce the same issues as regular speakers.

This topic seems a little abstract. I'd like to say that there is no way to erase a room. You have to think in terms of wavelengths and expanding spheres (whose direction of travel can be simplified into straight lines you can follow). Constant directivity or any other design which controls radiation patterns and reflection patterns do not eliminate reflections but rather dictate the strength of that expanding wave relative to a particular angle.

I think we have strong baggage as audiophiles which translates into how we set goals. One version of that is: because I want fidelity to the source, I cannot allow the medium of transmission to impart any character. There has been a long history of acoustical engineers which have understood that to mean that the room itself should be eliminated as much as possible. I think this approach culminates in Tom Hidley's nonenvironment studio room designs.

Recent work has made it very clear that speakers and rooms work together to make pleasant, realistic sound. Psychoacoustics shows that there are thresholds to everything, that a wide array of physical circumstances can lead to a similar percept.

Achieving a good balance of all the various elements is getting pretty easy with tools like mixed-phase correction (e.g., Dirac) and a decent set of speakers (decent meaning competently designed using high-resolution measurements). We don't have to set such hard goals or struggle so much to achieve them. Which I think is a nice, comforting conclusion to reach, and is supported by the most realistic, nuanced and hardline scientific approach informed by the latest research. You would think that the science would demand the most precise and labor-intensive approach, but it's not that way at all.
 
Hi there Curvy,

Many tx for your elaborate post. You mention a lot of relevant issues. And raise questions as much as you answer them, which is good (research is in part about asking the right questions).

1.
About the digital measures, i'm probably conservative for now, still prefering to only apply any after having exploited every last physical option at managing speaker/room acoustics and wavefield behavior. I may change my mind with progressing insight, but that is like innocent until proven guilty for me.

2.
So the gradually diminishing shadow on both sides of the 90° will still remove a large part of the earliest reflections (in the omni- up to comb filtering range), wouldn't you agree?

My 4.8dB figure comes from Siggy in the below quote from his chapter "Loudspeaker Directivity and Room Response".
(Not sure, but reason has it this applies to bass and lower voice mostly, on account of comb filtering starting to interfere with fig-8 radiation as frequency rises? Since those are also the hardest to manage in acoustically small rooms, this does not seem to make the quote less relevant.):

1 -An open baffle, dipole speaker has a figure-of-eight radiation pattern and therefore excites fewer room modes.
2 - Its total radiated power is 4.8 dB less than that of a monopole for the same on-axis SPL. Thus the strength of the excited modes is less.
3 - A 4.8 dB difference in SPL at low frequencies is quite significant, due to the bunching of the equal loudness contours at low frequencies, and corresponds to a 10 dB difference in loudness at 1 kHz.
Thus, bass reproduced by a dipole would be less masked by the room, since a dipole excites fewer modes, and to a lesser degree, and since the perceived difference between direct sound and room contribution is magnified by a psychoacoustic effect.

(end of quote)

Siggy does not reference every statement individually to his sources, but he does list his sources meticulously and is generally solid in his approach.
Though not infallible, his website has proven a valuable source for many general principles.

Dipole design and execution does have to follow many specific details to meet the attributed acoustic advantages to their full extent, but if done right i see very few drawbacks, especially compared to the inherent distortion and serious non-constant directivity issues of boxed speakers.

So you gave me a lot to consider, tx for that. Points 1 and 2 are probably the ones i'll abandon last, but, everything is up for debate.

Gtz
 
Last edited:
Back
Top Bottom