ATC speakers / Monitors

[tuga]

Because as I said before, even listening from very close distances, human auditory system combines incoming reflections with direct sound for about 30-40ms after receiving a direct sound. We hear a combination of sum of early reflections and direct sound even while sitting very close to speakers. As the distance between the listener and the speakers get wider, sound power becomes more and more determining factor in what we hear and that is way far field predictions of Amir is a combination of %44 direct sound, %44 early reflections and %12 sound power response.

From Toole's book:


You can read more about in Toole's book sound reproduction, check for the keyword "precedence effect". This is a topic which has multiple articles written about it. This isn't something new, first articles about the the topic is from around 1972 and most of the details of the precedence effect was revealed in an article in 1985.

Basically off-axis response matters and it matters a lot for nearfield listening. ATC designs speakers completely ignoring that fact.

Is it be possible that by moving the speakers closer to the listener and away from the boundaries the level of the relfected vs direct will be somewhat to a lot lower than when listening farfield in the same room?
Perhaps it would give and identical effect to adding absorption to early reflection areas?

 

[goat76]

direct sound is perceptually dominant however human auditory system starts a fusion interval for 40ms(after a single direct sound hits our eardrums), where all the incoming reflections combined/fused with the direct sound. Each incoming reflection adds timbral colorations to direct sound but simply early reflections are not heard as spatially separate events. This is called as precedence effect.

Read more about precedence effect in Toole's book for more information. Basically even while sitting very close to speakers, we hear a combination of direct sound and early reflections. That is why directivity of speakers matters a lot. Directivity is not just a silly measurement to calculate sweet spot width of speakers, directivity of speakers determine how the reflections are formed inside a room. This's why sum of early reflections are sound power response of speakers are very important to us and that is why most of us here find ATC designs outdated. ATC designs do not take precedence effect into full consideration. ATCs have awfully colored reflections and an measurements taken with omni microphones can't reveal that, only anechoic measurements can.

Sorry for the delayed reply, I had a long late workday today.


You are right about uneven directivity adding some coloration to the direct sound, but that's about all it will affect. But that got nothing to do with the precedence effect, which is a completely different thing.

The precedence effect is the localization of the first arrival sound and for this position to be altered in a significant way, the early reflective sound needs to be very strong. And opposite to what you seem to believe, the more similar this early reflective sound is to the direct sound, the more likely it will have a chance to alter the perceived localization of the sound. It's a similar effect to the phantom effect where two similar sounds arrive close together time-wise, but this time it will appear to be located somewhere outside of the physical position of the speaker, between the speaker and the closest side wall.

Working in a studio you simply don't want your own listening room/environment to alter the perceived localization of the sound at all, you need to be sure you know exactly where in the stereo field the different sound objects really are positioned, therefore it's better to reduce those early reflections from the side walls as much as possible. The studio engineer simply needs to know what he's doing and be sure he's not hearing too many effects that are specific to his own listening environment, those will of course not end up on the actual production.

But with all that said, I completely understand that some people in their own listening environments like the added effect of a perceived wider sound field, a sound that extends outside the speaker's position. And for that, a more even directivity is likely better if the goal is to create that type of enveloping sound. But that is of course a completely subjective preference.

 

[thewas]

Ok, so that's a home made anechoic chamber?

Which one are you referring to. the one of the Audio magazine or the Klippel NFS?

Those we did for some smaller manufacturers were small-ish and still valid down to a 100Hz or 80Hz.

Yes, the problem of small anechoic chambers that they are limited in the lower boundary, something which nowadays can be overcome with the NFS.

"Automatic gating" is a feature all apps have these days. It's not the same though. Data gathered is not comparable 1:1.

If their gating is wrong of course the data will not be comparable, that's why at serious measurements it usually adjusted by the user.

 

[thewas]

You missed 90% of the explanation.

As I didn't see any real convincing one, see

We don't share detailed design data with strangers (or competitors for that matter) so you would not have the information and knowledge to read that data since it is simply not publicly available. All of this is proprietary and the reason why we survive as a company in today's pro audio world. We do share 100% with clients though, that have full access to everything, at any point in time.

They use a psycho-acoustic principle called self-noises cues. It's a key aspect of the design

Could you provide some literature sources on this principle which is your key aspect as googling for it ("self-noises cues") gives only references to your company.

 

[thewas]

SCM11 is a DESKTOP speaker. Why do you bring off-axis issues here at all? This speaker is suppose to be 3-4 feet from your head and point straight at your ears. You cannot have the same arguments about near field desktop speakers and midfield or main monitors (or living room domestic speakers). Thus we either talk about ATC speakers below SCM40 or above that. They are made for very different use cases.

I didn't bring up the SCM11 (I posted the plots of the SCM 40) and this issue is relevant to any speaker, also the SCM40 has the same tweeter and the same directivity issue.

 

[thewas]

No the resonances are at 9 and 13k. It doesn’t matter if they are Harmonic Distortion or not, the tweeter is distorting and resonating. Their correlation is not hard to pin in this particular case - they are very easily audible.

Wrong again, look at the waterfall which is totally clean at those frequencies, if it would be plotted up to 30 kHz you would see the break up mode clearly there:

 

[dfuller]

.



My career leads me to 99.99% agree with @Torbachkristensen here, but a couple times I had a band member who would say, "I want more *meaningless phrase* like *names competitor* on *names competitor's last album* here," ... and in an interesting loop-back to audiophilia itself, I would do absolutely nothing except re-name the file and send it back 48 hrs later, to his complete satisfaction. Expectation bias is everywhere.

LOL, 100% done this too.

 

[thewas]

I assume that report was created by frequency sweep. If resonance is at 27kHz, how was it triggered by a test signal below 20kHz? It should never happen.

The measurement sweep wasn't only up to 20 kHz as otherwise the 2nd harmonic measurement would go only up to 10 kHz and not 30 kHz, you can see on the right plot also that the reference goes up to 30 kHz:

Obviously on the left 86 dB measurement the sweep was done only up to 20 kHz and "magically" the 2 peaks disappear.

 

[Northward]

As I didn't see any real convincing one, see



Could you provide some literature sources on this principle which is your key aspect as googling for it ("self-noises cues") gives only references to your company.

Yes, because my company did the R&D, engineering, psychoacoustics study and came up with the design topology. We're protective of the data for that reason. It's proprietary. But the basic principle is easy to understand. How it's implemented not as easily.

It's a technology that allows us to fully guarantee the results to our clients on all our designs and have rooms that are extremely consistent in their performances, no matter the size (within certain minimum size boundaries though.)

A FTB room in NY performs in the same narrow technical envelope as one in Paris or Sydney. They're basically interchangeable, even if of different size.

 

[thewas]

All transducers have some distortion. In those cases the harmonics trigger a major breakup mode (as all metal drivers have) so it appears as much higher distortion than in reality. It's not really an audible problem unless you're driving the speakers really hard, in which case it could be high enough to create audible IMD (but this is unlikely).


Soft domes have much better controlled breakup behavior, but it's often broader Q and lower in frequency.

Correct, the same I wrote in #1,769

 

[thewas]

Briefly - the speakers' output fires into absorptive surfaces and doesn't bounce back.

This is not some new invention though and It is long known even how to place monitors to make the console reflection to be not directed to the mixing engineer:

Source: https://support.genelec.com/hc/en-u...ole-meter-bridge-Which-orientation-is-better-

 

[tuga]

The measurement sweep wasn't only up to 20 kHz as otherwise the 2nd harmonic measurement would go only up to 10 kHz and not 30 kHz, you can see on the right plot also that the reference goes up to 30 kHz:

Obviously on the left 86 dB measurement the sweep was done only up to 20 kHz and "magically" the 2 peaks disappear.

I wonder why the distortion traces only go up to 10kHz in the 86dB measurement and the FR curve stops at 20kHz...
This and other inconsistencies in the presentation of the data are disappointing.

 

[thewas]

Yes, because my company did the R&D, engineering, psychoacoustics study and came up with the design topology. We're protective of the data for that reason. It's proprietary. But the basic principle is easy to understand.

Sorry, but referring to a supposed psychoacoustic principle which isn't mentioned in any established literature publication is no science but reminds of typical audio snake oil marketing.

 

[thewas]

I wonder why the distortion traces only go up to 10kHz in the 86dB measurement and the FR curve stops at 20kHz...

Because obviously the left measurement was done only up to 20 kHz while the right one up to 30 kHz.

 

[Avp1]

The measurement sweep wasn't only up to 20 kHz as otherwise the 2nd harmonic measurement would go only up to 10 kHz and not 30 kHz, you can see on the right plot also that the reference goes up to 30 kHz:

Obviously on the left 86 dB measurement the sweep was done only up to 20 kHz and "magically" the 2 peaks disappear.

There is a difference between sweep range and measurement range. Sweep can be limited to 20kHz, while measurements done all way to 60kHz to get 3rd harmonic. But anyway sweep limited to 20kHz should not excite resonance at 27kHz. This is of cause when measurement is done correct and results displayed properly. I understand that breakup mode at 27kHz can create SUB-harmonics when excited. But you won't see it in individual harmonic vs frequency graph. But it will be seen in THD+N graph vs frequency as a major N component raise at around breakup point. Sub-harmonics will be measured as noise in that context.

 

[Pearljam5000]

They really need to make this one in the raw aluminum finish

 

[617]

 

[Northward]

Sorry, but referring to a supposed psychoacoustic principle which isn't mentioned in any established literature publication is no science but reminds of typical audio snake oil marketing.

My company does not owe you anything, certainly not years of expensive R&D, important prototyping investments and over work burn-outs. You are far out of the reality we work in.

The name "Self-noises cues" itself literally describes what it is (*points at voice path bouncing off a nearby wall back to ears*) and anyone with even a limited psychoacoustics background understands exactly the path described by the name alone. I've described on numerous occasions the principals of it, it's not quantum physics. It's only Audio. How it is implemented and within what parameters is the stuff we don't share.

We'll be ok without your Anonymous arm chair peer review.

 

[Northward]

Fairly informal interview.

I can understand the need for confidentiality. But manufacturers have used this principle to their advantage to sell junk, false promises or mediocrity.

Maybe you aren't willing to divulge trade secrets. Fair enough. But beyond the work you do for individual clients, no one will get the education they need.

Here we do the best we can discuss the subjects, look up papers etc. Maybe you could share the papers you see as most fundamental to your approach? We may be able to understand it from the ground up, or your reasons for preferring ATCs.

I'll try and find time to gather a few papers yes (I'm traveling now, hence the unusual free time to be on a forum). I remember one about canonical spaces being very interesting.

 

[Blockader]

Sorry for the delayed reply, I had a long late workday today.


You are right about uneven directivity adding some coloration to the direct sound, but that's about all it will affect. But that got nothing to do with the precedence effect, which is a completely different thing.

The precedence effect is the localization of the first arrival sound and for this position to be altered in a significant way, the early reflective sound needs to be very strong. And opposite to what you seem to believe, the more similar this early reflective sound is to the direct sound, the more likely it will have a chance to alter the perceived localization of the sound. It's a similar effect to the phantom effect where two similar sounds arrive close together time-wise, but this time it will appear to be located somewhere outside of the physical position of the speaker, between the speaker and the closest side wall.

Working in a studio you simply don't want your own listening room/environment to alter the perceived localization of the sound at all, you need to be sure you know exactly where in the stereo field the different sound objects really are positioned, therefore it's better to reduce those early reflections from the side walls as much as possible. The studio engineer simply needs to know what he's doing and be sure he's not hearing too many effects that are specific to his own listening environment, those will of course not end up on the actual production.

But with all that said, I completely understand that some people in their own listening environments like the added effect of a perceived wider sound field, a sound that extends outside the speaker's position. And for that, a more even directivity is likely better if the goal is to create that type of enveloping sound. But that is of course a completely subjective preference.

I explained before that this is simply not true, you can read my full post here. https://audiosciencereview.com/forum/index.php?threads/atc-speakers-monitors.17448/post-1313629

Haas called this first as echo suppression effect,

Haas described this as an “echo suppression effect.” Some people have taken this to mean that the delayed sound is masked, but it isn’t. Within the precedence effect fusion interval, there is no masking—all of the reflected (delayed) sounds are audible, making their contributions to timbre and loudness, but the early reflections simply are not heard as spatially separate events. They are perceived as coming from the direction of the first sound; this, and only this, is the essence of the “fusion.”

And this phenomenon was included in Benade's summary in "Generalized Precedence Effect"

Benade (1985) contributed a thoughtful summary under the title “Generalized Precedence Effect,” in which he stated the following:
1. The human auditory system combines the information contained in a set of reduplicated sound sequences and hears them as though they were a single entity, provided (a) that these sequences are reasonably similar in their spectral and temporal patterns and (b) that most of them arrive within a time interval of about 40 ms following the arrival of the first member of the set.
2. The singly perceived composite entity represents the accumulated information about the acoustical features shared by the set of signals (tone color, articulation, etc.). It is heard as though all the later arrivals were piled upon the first one without any delay—that is, the perceived time of arrival of the entire set is the physical instant at which the earliest member arrived.
3. The loudness of the perceived sound is augmented above that of the first arrival by the accumulated contributions from the later arrivals. This is true even in the case when one or more of the later signals is stronger than the first

Quotes are from Floyd Toole - Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms.