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Post by duke on Jun 15, 2017 13:55:02 GMT -6
Directional bass response would be good in that you could mitigate treatment options, or at least better manage bass monitoring/treatment solutions. I run a pair of HS-50's with no sub in my room, alongside a set of auralex foam wedges at the early reflection points. Speakers are about 2 feet off the back wall and with corner trapping on the front walls. I'm pretty happy with how it sounds, and I feel adding more bass would just add problems,currently. I'd doubt my next investment would be anything *but* rear wall trapping. A couple of notes is that typical nearfields are being used very near, very often in small-medium studios I see the listener position as being too close - either to deal with the interference of reflections on stereo image, or size or just ignorance. With this, I would assume the issue of driver-pinnae directionality becomes even more critical? I've owned several directional bass systems (dynamic dipoles, planar magnetic dipoles, electrostatic dipoles, passive cardioids). My observation is this: They do a very good job with pitch definition, but because they rely on cancellation to achieve their pattern control, they don't "pressurize" the room at bass frequencies, especially if it's a small room where the modal region gives way to the pressure region higher up than in a larger room. The "pressure zone" becomes the "no-pressure zone" with a dipole, for example. So they don't convey that tactile "THUMP" very well.
An active cardioid system could probably be programmed to transition from cardioid to monopole in a way that complements the transition from modal region to pressure region.
That being said, anecdotal feedback leans me in the direction of the distributed multisub system (like what I described to javamad) over an active cardioid system, but I can see arguments either way.
You mentioned that typical nearfields are often used with "the listener position being too close", perhaps "to deal with the interference of reflections on stereo image." I think that a more directional speaker could probably be positioned a bit further away, for the same degree of reduction in early reflections.
In response to your question, I would think that, the closer the speakers are, the more disruption small head movements impose on the angular relationships between drivers and pinnae. So I would think that head placement and angle relative to the speakers is more critical when the speakers are very close.
Do you think it would be a worthwhile advantage to be able to position the speakers a bit further away from the listener, assuming imaging isn't compromised?
* * * *
I can't help but notice that the two of you, jazznoise and javamad, are both from Ireland. My mother was an O'Niell. Note the incorrect spelling. Her great-great-grandfather started out as an O'Neill (the correct original spelling) and there was a member of Parliament with the same last name. Well turns out he hated the member of Parliament so much that he actually changed the spelling of his own last name! So the conclusion I draw is... a person hasn't really been hated until he's been hated by an Irishman!
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Post by duke on Jun 15, 2017 13:20:52 GMT -6
Hi Duke. I would fully support any developments that can help the long-suffering home studio in mixing. The biggest thing I believe they have to deal with apart from good stereo image is handling bass energy. Very often a home studio owner will have some very tight space constraints, as they get the use of a spare room or garage and there is no physical space (or budget) for the right amount of absorption required. So these domestic spaces are full of un-treated room modes. If by using a larger cabinet you are also lowering the total amount of bass energy the speaker puts out then I would say that you might be on to something. The physics associated with low frequency wave propagation make me doubt it can be done effectively at non-professional pricing levels though. Thank you very much for your encouragement! I'd love to do something at the creative end of the music chain. And not just because you guys have so much money to throw around.
I didn't ask about the bass region in my initial post because I'm starting out cautiously optimistic about being able to offer a worthwhile improvement at that end of the spectrum. Not that there won't be challenges of course.
Here's some quick psychoacoustic background information: The ear/brain system has poor time domain resolution at low frequencies (say below 100 Hz or so), to the extent that we cannot hear the woofer's output apart from the room's effects in small rooms (and just about anything we're talking about on this forum is a "small" room). The ear/brain system cannot detect the presence of bass energy from less than one wavelength, and must hear several wavelengths in order to detect pitch. Considering how long bass wavelengths are relative to our room dimensions, by the time we hear the bass, the room's effects are already all over it. So in the bass region at least, speaker + room = an inseparable system, from the standpoint of perception.
Also, the ear is particularly sensitive to SPL differences in the bass region (peaks and dips, especially the peaks). If you eyeball a set of Fletcher-Munsen curves, you'll see that they bunch up south of 100 Hz or so. This means that a small change in SPL makes a disproportionately large change in perceived loudness. In fact, a 6 dB change at 50 Hz makes just about as much of a perceptual difference in loudness as does a 12 dB change at 1 kHz! Thus a 6 dB peak in the bass region really sticks out like a sore thumb. Given that small rooms inherently have the worst peaks and dips in the bass region, they also have the most room for improvement.
The approach I embrace in pursuit of that improvement is somewhat counter-intuitive: Four bass sources (small subwoofers) distributed asymmetrically around the room. Each will inevitably produce a nasty peak-and-dip pattern at the listening position... but each of these nasty peak-and-dip patterns will be different. The sum of all these different peak-and-dip patterns will be significantly smoother than any one alone.
At low frequencies, speaker + room = a minimum phase system. What this means is, the frequency response and time-domain response are so closely tied together that what affects one, affects the other. And when we fix one, we have simultaneously fixed the other! Absorption works on the decay times which therefore fixes the frequency response. And a distributed multisub system works on the frequency response which therefore fixes the decay times. But the thing is, in most cases we can make a greater improvement (and with far less space occupied) with distributed multisubs. And we can use both techniques.
One nice side effect of this approach is, the benefits hold up throughout the room... in contrast to relying in EQ, which usually worsens the bass response elsewhere in the room in exchange for improvements at the mix position.
Credit to Earl Geddes for the asymmetrically distributed multisub concept. I'm using his idea with his permission.
Just to make sure my point isn't misunderstood, the argument for using multiple subs has to do entirely with quality, not quantity. As for quantity, too much bass is far worse than not enough. But if the quality is there, then the quantity (loudness and bottom-end extension) can be adjusted until it's right for the application.
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Post by duke on Jun 14, 2017 20:51:41 GMT -6
Duke. The room I work in is a total mess by its dimension and build materials form around 1920. I used tons of broadband base absorption and the speakers ended up very much left corner of the room because we had the best base response there. I use Genelecs 8040 in extreme Nearfield position because that was where we got the best measurements. The Mointors are about a 2,62 feet away form the ear distance. Great sweet spot very stable. Behind the monitors is the DEAD END. In my back I have two Schroeder diffusers. I can make a layout plan tomorrow. Its all against theory but everybody who used to work with me agreed that it sounds good. Its a typical home recording compromise situation and I think we hear to 80% direct signal. Thank you very much for providing this information!
My understanding is that small rooms are the hardest to get right, and what ends up working well may be very different from what theory predicts, as in your case. I remember George Augspurger saying that room modeling programs do a good job with big rooms but that he's never seen one that worked well for small rooms. His observation was that "slight details make a much bigger difference in a small room."
Those Genelecs have a waveguide around the tweeter so imo that is a choice that makes a lot of sense for your space. And since the waveguide is elliptical, the vertical spacing between woofer and tweeter is minimized, which is good for close listening distances.
Thank you for the offer, but no need to make a layout plan.
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Post by duke on Jun 14, 2017 18:30:17 GMT -6
The room I work in is a LEDE thing and very dry ø 200 ms I like dry rooms and its yet the best way to do my mixes. If I understand correctly, the idea behind an LEDE room is, to minimize the relatively detrimental early reflections while preserving, as much as reasonably possible, the relatively beneficial later-arriving reflections.
I think this is hard to pull off in a small room because, in most cases, enough absorption to suppress the early reflections ends up absorbing more than just the early ones. Because the reflection paths are so short in a small room, within a given time interval there will be many more reflections that bounce into an absorptive panel.
So I'm thinking that monitors with fairly narrow radiation patterns would need less absorptive treatment to adequately suppress early reflections, allowing us to still get the clarity of a good LEDE room, but we'd have a bit more sense of spaciousness and envelopment because our T60 would be a bit longer due to the reduced amount of absorption needed.
I'm curious... can you let me know what monitors you use, and your approximate room dimensions? That information (combined with what you posted above) would be valuable data to me. Danke!
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Jun 14, 2017 11:34:16 GMT -6
Post by duke on Jun 14, 2017 11:34:16 GMT -6
Sorry I made you sad/disappointed/I'm not sure...
Here's a link. Hope that helps!
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Post by duke on Jun 13, 2017 19:00:35 GMT -6
Use good solder, makes a big difference.
So... basically what you're saying is... "Don't be a solder miser"... ?
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Post by duke on Jun 13, 2017 15:17:07 GMT -6
I've DIYed hundreds of kits and here's what I use. A 15 watt soldering Iron Soldier that contains at least or > 2% silver. Hakko 808 desoldering gun ( the only way to fix mistakes ) Wen 100 watt soldering iron ( for the big wires ) Harbor Freight headband magnifier ( It moves with your head ) Good luck Very nice. I like your advice better than mine.
Not long ago I bought another roll of 2% silver solder from my usual supplier and had a really hard time getting good solder joints. Changed tips on my iron but that didn't help. Then I read the description more carefully and discovered that it was now "lead free" solder, in packaging virtually indistinguishable from what I'd been using before. So... my fifth-grade teacher was right, reading comprehension does matter after all. Anyway I'm definitely not a fan of lead-free solder.
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Post by duke on Jun 13, 2017 14:39:38 GMT -6
Considering getting a decent soldering iron and attempting to learn how to replace failing caps and possibly upgrading opamps in my own stuff. Im sure I will destroy a few or many things during the process. Anyone care to recommend a newbie 101 setup? Is Weller WES51 a good choice? I do not have the best vision so I would need a magnifier and also a stand.. I know just enough about this to probably buy the wrong tools and waste $$. Any suggestions? Thanks in advance. I can't say the Weller WES-51 is THE "best" choice, but it's what I've been using for the past dozen years or so. You might want to get a long sharp tip for precision work; I use mine mostly for loudspeaker crossovers where a fatter tip for better heat flow makes more sense. Also, I wear a Coast brand headlight (the battery pack in the back keeps it balanced so it doesn't walk down my face as I work) along with strong reading glasses when I solder. That being said, take the advice of anyone experienced with precision soldering over mine.
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Post by duke on Jun 11, 2017 13:24:55 GMT -6
The NS 10 was a small high-end consumer speaker and not designed for monitoring. For a while the Yamaha NS4 was the biggest selling consumer speaker and was used for mixing as the JBL L-100 and KLH6 had previously been. The optimal setup is BOTH reference speakers and full range speakers. I did not know that about the NS-10! Very interesting! Lucky design!
Makes sense that both types of speakers are optimal.
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Post by duke on Jun 11, 2017 13:11:16 GMT -6
La Scala will definitely tell you when there's too little dynamic range, everything sounds overly-present and fairly harsh. Thank you very much for correcting my misconception!
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Post by duke on Jun 11, 2017 13:02:51 GMT -6
A better speaker will handle larger dynamic swings which may be truly out of balance, yet sound fine because the brain adjusts and accepts and there's no technical indication of a problem. A lesser speaker may obviously fart with the same material, alerting you to a mix problem. I was doing some late night listening on Klipsch La Scala at low volume. They sound so huge and loud at any volume, I usually have an urge to turn them down on the assumption i must be disturbing someone. Pulled the SPL meter out, they were running at roughly 48dB, which was roughly 10dB louder than ambient background. Yet they seemed loud, clear, and full. I could detect the bottom resonating the walls and floor. Good mixing monitors are considerably more complicated and specialized little beasts than I had realized! I would not have thought of dynamic compression as a virtue in a monitor, but what you say makes a lot of sense. And obviously something like the La Scala would be the wrong tool for judging how much compression to use in the mix.
Are there tasks in the recording and producing process where speakers that reveal the full dynamic range of the recording would be an advantage?
I've seen speakers that look like midfields at first glance advertised as being suitable for both mixing and mastering. Please forgive the naive newbie question - is that reasonably feasible, or mostly marketing?
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Post by duke on Jun 11, 2017 12:57:02 GMT -6
You can hear more deeply into the sound of transparent speakers. What sounds like apples vs. oranges on the transparent speakers can sound like right vs. wrong on less transparent speakers. With my Duntechs I learned that when diffraction is properly handled, a speaker that measures flat will sound flat. When I remove the felt diffraction treatment, the speaker measures the same but sounds bright as all get-out. Where NS-10s excel is sitting on a large console meter bridge. The low end goes away anywhere else. I would not have thought of apples & oranges vs right & wrong. Well that makes figuring out where the goal posts are for a really good mixing monitor considerably more complicated!
What you say about the Duntechs makes total sense. Diffraction has very little effect on the frequency response, but can be a significant source of audible coloration because it is not covered up by the ear/brain system's "masking" mechanism. Masking doesn't work in the time domain, and diffraction arrives a little bit later due to the additional path length, which in effect it makes those sounds last longer. And while an SPL meter wouldn't pick up on this difference, the ear/brain system interprets "lasts longer" as "louder". The frequency region where the ear is most sensitive to diffraction is right around 4 kHz. This just happens to be at the bottom end of most tweeters' ranges, where their radiation patterns are extremely wide, and therefore where they are sending a lot of energy towards the enclosure edges where it will be diffracted.
Any small speaker's low end will be increased by boundary reinforcement from placement on a large console meter bridge, but the NS-10 seems to have been specifically designed with just such reinforcement in mind.
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Post by duke on Jun 11, 2017 9:09:00 GMT -6
The only difference is that many audiophile speakers are too flattering or have dips in the response which can lead one into a rude awakening when listening elsewhere. Transparency and full range rules for mastering. For mixing less transparent "reference speakers" are required in addition for setting balances. This is because balance isn't nearly as critical with transparent speakers. Some "reference speakers" have been found with experience to "work" while others haven't. The usual problem is too many dips in the response. For tracking it is important that the speakers musicians judge performances with are not so revealing that left brain audio problems that can easily be fixed later are not distracting from right brain performance. Yes!! Some variation of the "smiley-faced" curve shows up in a lot of audiophile speakers. And a dip in the 3-4 kHz region can help make a speaker "more forgiving". Imo that region is the most important to get right, and a dip there is "cheating", but sometimes it's the lesser of two evils.
I often shoot for a gently downward-sloping trend, rather than "flat", for high-end audio. However ime the closer the system comes to full bottom end extension, like all the way down to 20 Hz, the closer to "flat" the general trend can be without the tonal balance sounding bright and therefore tending to be fatiguing long-term.
Could you elaborate on this statement?: "For mixing less transparent "reference speakers" are required in addition for setting balances. This is because balance isn't nearly as critical with transparent speakers."
Why is balance less critical with transparent speakers, apparently making them less desirable for mixing?
Thanks!
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Post by duke on Jun 10, 2017 23:41:28 GMT -6
My understanding is that the goal posts are in slightly a different place for recording engineers than for high-end home audio, at least when it comes to loudspeakers. You want to hear what's wrong so you can fix it, and that's usually not the top priority in home audio.
And recording engineers have so many things they need to spend money on that most hard-core audiophiles are going to out-spend them many times over on the playback chain, source -> preamp/amp -> speakers -> room... EXCEPT for that last one. You guys usually have a far greater appreciation for the role that the room itself plays, and so you spend money on professional-level room treatment. Relatively few audiophiles do so.
Also, at least in the playback chain, you guys are much more inclined to figure out the best way to plow with the horses you have, before you start spending money on a new toy that promises to deliver better sound. You approach setup professionally, giving priority to results, whereas many audiophiles give top priority to something else, perhaps aesthetics. Not only do you make measurements whenever it makes sense to do so, but your ears are really well calibrated.
So I would say that many if not most audiophiles are in a position assemble a "better sounding" (which may or may not include "higher resolution") system than most recording engineers, but relatively few of them capitalize on that. It seems to me that you guys are getting 110% of the potential out of what most audiophiles would consider to be a budget system.
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Post by duke on Jun 9, 2017 14:06:49 GMT -6
Thank you for elaborating, that makes sense. I don't normally listen nearfield so I really appreciate your raising my awareness.
The ear's vertical sound source localization is poor in comparison with its ability to localize in the horizontal plane, and localization ability peaks in both planes in the 4 kHz region. In the vertical plane, the ear tends to mis-judge tones higher than about 6 kHz as coming from a location physically higher than they actually do, and likewise tends to misjudge tones lower than about 1.5 kHz as coming from lower than they actually do. This might be an argument for using two-way monitors upside-down, or at least giving it a try.
Because of the ear's greater angular resolution in the horizontal plane, I would think that the minimum distance is greater for a speaker on its side, unless the speaker is axi-symmetric (MTM or coaxial).
Obviously we'd want the drivers as physically close to one another as possible, but I would think that arrival time also plays a role in coherence at close range, so small timing differences arising from relative phase and relative driver depth can come into play. Lots of stuff to take into account.
If you don't mind, could you tell me which 3-way monitor you use? That information, combined with your 6 foot listening distance, will give me a good data point.
Thanks!
I am using KRK 10-3's set up horizontal with the mid/hi rotated vertical. Not super hi-end speakers, but they translate really well for me. Thank you, that is quite helpful!
Looks like that KRK is a lot of speaker for a very reasonable price, and if they "translate really well", that's a great!
Aside from going deeper and louder (if needed), a 3-way often has this advantage over a smaller 2-way: The off-axis energy's spectral balance is closer to that of the on-axis energy. This is because the "steps" in going from one driver diameter to the next are inherently smaller, which reduces the magnitude of any off-axis glitches in the crossover regions as we transition from a driver that's starting to beam to a smaller driver that's not. (This is a generalization - there are of course exceptions.) Ime significant spectral discrepancies between the first-arrival sound and the reverberant sound can result in listening fatigue. This is not the only potential source of listening fatigue of course, but it's one of the more sneaky ones.
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Post by duke on Jun 9, 2017 11:10:38 GMT -6
Every speaker and room are different and interact differently. Increasing the baffle size is going to have a minimal effect, so I wouldn't pay to much attention. Imo we can probably make some assumptions about the environment that a monitor will encounter in a project studio, and hopefully go on to use those assumptions to our advantage. For instance, there will usually be a reflective surface below the monitor and in between the monitor and the listener, so the narrower the pattern is in the vertical plane, the less energy in the (undesirable) early reflection off that surface.
True that baffle size itself makes little acoustic difference except down in the "baffle step" region, but things like cone diameter do make a difference, and those things usually set the minimums for baffle size.
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Post by duke on Jun 9, 2017 10:31:36 GMT -6
Thank you mulmany...
Can you tell me your ideas about optimal listening distance? I really haven't been thinking in those terms, so anything you have to say will be very helpful.
In my experience, speakers have a critical distance where you cannot distinguish the crossover region. IMHO near fields are used too close to the listener. I have a preference for 3-ways and have found that I need at least a 6ft equilateral triangle so that the speakers have blended correct ly at the crossover. Hope that makes sense. I guess you could say the distance that the waveforms have combined to form the acoustic center of the speaker. Thank you for elaborating, that makes sense. I don't normally listen nearfield so I really appreciate your raising my awareness.
The ear's vertical sound source localization is poor in comparison with its ability to localize in the horizontal plane, and localization ability peaks in both planes in the 4 kHz region. In the vertical plane, the ear tends to mis-judge tones higher than about 6 kHz as coming from a location physically higher than they actually do, and likewise tends to misjudge tones lower than about 1.5 kHz as coming from lower than they actually do. This might be an argument for using two-way monitors upside-down, or at least giving it a try.
Because of the ear's greater angular resolution in the horizontal plane, I would think that the minimum distance is greater for a speaker on its side, unless the speaker is axi-symmetric (MTM or coaxial).
Obviously we'd want the drivers as physically close to one another as possible, but I would think that arrival time also plays a role in coherence at close range, so small timing differences arising from relative phase and relative driver depth can come into play. Lots of stuff to take into account.
If you don't mind, could you tell me which 3-way monitor you use? That information, combined with your 6 foot listening distance, will give me a good data point.
Thanks!
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Post by duke on Jun 8, 2017 23:53:04 GMT -6
Within reason, it makes no difference to me if the speakers are larger, smaller.. mount horizontal or vertical... When I get them, I spend time putting them in the "right spot" and there they stay, for a decade or more... till I change them for the next set. The treatment, room, placement and acoustic treatment are all part of the process of getting it right... so to speak.. in my environment. hope that is of some help cheers Wiz Yes that does help - thank you! I was hoping that speaker size would be low in the list of priorities, and it sounds like it is for you.
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Post by duke on Jun 8, 2017 23:51:46 GMT -6
I did not consider size in my last speaker purchase, but I did look and price and performance. If you can build a better speaker for less and the trade off is size... Go big or go home! I think a better metric is optimal listening distance. Thank you mulmany...
Can you tell me your ideas about optimal listening distance? I really haven't been thinking in those terms, so anything you have to say will be very helpful.
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Post by duke on Jun 8, 2017 23:27:05 GMT -6
Just so you guys know, I'm not a recording engineer, just a speaker designer. So apologies in advance for whatever misconceptions I still have – please feel free to correct me.
One of the things I've noticed in reading threads on this site is how much attention you guys pay to the acoustic treatment of your rooms. That makes total sense to me, especially given the restricted room size of most project studios. Small rooms have the shortest reflection paths, and in general early reflections are the ones most likely to be detrimental.
Acoustic treatment can affect the intensity, spectral balance, diffusiveness, and decay times of reflections. Ideally we want the direct or first-arrival sound to be fairly loud relative to the reflections, and we'd like to minimize early reflections without totally killing the later ones (which would make the room too dead). We'd also like the reverberant field to be spectrally similar to the first-arrival sound, to be diffuse, and for the reverberant energy to decay smoothly as far down the spectrum as is reasonably feasible. Please feel free to add to or correct any of this!
Studio monitor design by itself can't do much about the diffusiveness of reflections nor their decay times (those being the domain of room treatments), but monitor design can at least affect the amount of energy in early reflections, the relative loudness of the direct vs reverberant sound, and the spectral balance of the reverberant energy.
Are improvements in these areas worth pursuing? I think so, but welcome input from any of you on this topic.
Unfortunately few improvements in loudspeaker design come without tradeoffs. One of the tradeoffs we'd have to make in order to get better radiation pattern control is, size. In order to get good pattern control down to a given frequency, the theoretical minimum enclosure height and width dimensions usually need to be about one-half wavelength at that frequency (it is possible to reduce the size penalty somewhat but it's expensive to do so and other penalties arise). So if we want to make improvements in radiation pattern control, the net result is going to be, wider monitors... and in practice, probably taller as well.
So here is my hypothetical question:
Could you live with monitors that are several inches wider and taller than what is normally used in a project studio, if the result would be an improvement in room interaction, at least in the mids and highs? The answer is probably “it depends”, but for the sake of this hypothetical, let's assume enclosure width and height are the only things that change.
I realize it's highly counter-intuitive to talk about bigger monitors for smaller rooms, but small rooms are the biggest challenges from an acoustics standpoint, and thus arguably would benefit the most from improvements in radiation pattern control.
Thanks in advance for any and all critiques, suggestions, comments, and insights.
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Moved
Jun 8, 2017 23:17:13 GMT -6
Post by duke on Jun 8, 2017 23:17:13 GMT -6
I meant to post in the "Pro Audio" forum, so I moved the post there.
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Post by duke on Jun 8, 2017 21:27:55 GMT -6
Hi Dave,
Thank you very much for taking the time to educate me about the effects of filters (electronic and mechanical) and even humidity on microphone performance! Extrapolating from what you said, I'm guessing that humidity probably affects speaker high-frequency performance as well. Next time I do an audio show in some dry place like Las Vegas, I'm going to bring something for misting!
Yes the JBL 4430 was imo a revolutionary design. To the best of my knowledge, among other innovations, it was the first time anyone had combined a constant-directivity horn with pattern-matching in both the horizontal and vertical planes in the crossover region. While I believe there have been improvements in horn design since then, I do not believe that basic concept has been improved upon.
Duke The effects on humidity on speakers varies with material, I do know Electrostatics love constant humidity! I had been thinking in terms of the transmission of high frequency energy through the air, but you are right, with electrostatics humidity is a significant factor. For many years I lived with electrostats that had a user-adjustable bias voltage control, and the optimum bias setting definitely changed with the humidity.
Paper-based cones can be hygroscopic (tend to absorb moisture from the air), which can change the cone's mass and probably its rigidity and, *ahem*, internal damping as well. I'm under the impression that most paper cones are sufficiently sealed that this is not normally an issue, even if they're not treated to the point of being "waterproof".
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Post by duke on Jun 8, 2017 11:32:57 GMT -6
Hi Duke, thanks for your observations. I think we are both right!!! What I mean by perception of sound is how the ear perceives sound. Which relates to the Fletcher Munson Curve. For, example there is a 10db difference in the perceived low frequency energy between monitoring at 60db than 80db. Music generally sounds better when its louder. This often fools folks. If you take the inner mesh away from a condenser microphone the level will go up because you have increased the sound pressure reaching the capsule's diaphragm. So, if you compare the before and after at the same preamp setting it will usually sound better with the inner mesh removed but you have most likely reduced the headroom several db. Notice, the Beatle's engineers added a 3rd metal "pop" filter to front of the U47's used at Abbey Road. This would have reduced the overall level reaching the microphone's diaphragm and softened the attack. It would reduce the popping also but effect the HF at 15khz by less than 1/2 db. Now, the Fletcher Munson Curve is the frequency response of the listeners ears at different monitoring levels. They took a group of folks and did listening test and averaged out the results to get the Fletcher Munson curve. So, we all hear slightly differently from each other at different volumes levels. So, each of us will have our own personal Fletcher Munson curve. It is important to pick a consistent montor level when you are referencing gear that average listener can agree too. Probably not your friend "Animal" the Heavy Metal Harley driving, guitar player with a double stack of Marshall 100 watt amps that he likes to turn to 11. At the Kim Novak mixing theatre on the Sony Picture's lot, they have a SPL meter and the average level must never exceed 85db SPL at the mixing desk. Psycho-acoustics also suggests that we hear differently on different days depending on the humidity. Today, in a studio control room with good HVAC we can usually keep the humidity consistent. In Howard Tremaines, The Audio Encyclopedia he alludes to Sound Stages in the LA during the 30's, misting the air before takes to increase the HF response when they had very dry days with low humidity. Sound travels faster as the air becomes more humid. However, this increase in speed, is quite small. Sound travels about 0.35 percent faster in 100 percent humidity than it does in zero percent humidity. Someone must have heard a difference or just got back from science class?? I know that humidity seemed to made a huge difference to the sound of our Reverb Plate. Sound travels on average 343M/Sec in air but in stainless steel it travels 5790 m/sec. I had two 40 watt light bulbs in series inside the plate reverb to keep it at a consistent temperature and humity. A properly working Neve 1073 or API 512 will often make a mediocre microphones sound "better" than the average transformerless preamp to my ears. I see you are using Bi-Radial horns in some of your speaker designs. We had a pair of JBL 4430 speaker in Control Room "A" at Ocean (circa 1980) that we custom flush mounted in the soffit. The had very even dispersion horizontally compared to the Altec 604E which were in vogue at that time. The didn't sound as harsh as the Altec's when the volumes go rocking. You could also listen to them longer than the Altec 604E speakers without as much ear fatique. The also had a time aligned passive crossovers. There was a great article on the design in AES magazine. We drove them from a Sansui BA-5000 that could deliver 300 watts per side RMS into 8 ohms. Cheers, Dave Hi Dave,
Thank you very much for taking the time to educate me about the effects of filters (electronic and mechanical) and even humidity on microphone performance! Extrapolating from what you said, I'm guessing that humidity probably affects speaker high-frequency performance as well. Next time I do an audio show in some dry place like Las Vegas, I'm going to bring something for misting!
Yes the JBL 4430 was imo a revolutionary design. To the best of my knowledge, among other innovations, it was the first time anyone had combined a constant-directivity horn with pattern-matching in both the horizontal and vertical planes in the crossover region. While I believe there have been improvements in horn design since then, I do not believe that basic concept has been improved upon.
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Post by duke on Jun 6, 2017 13:20:13 GMT -6
HI Dave, great paper! I don't remember seeing it before. Great exploration into why tube amps can be driven further into overload than transistor amps and op-amps before the sound gets too bad. Very interesting observations the author makes about the objectivity of musicians' ears. I've found musicians to be amazingly discerning.
Quoting Dave/aamicrophones: "I am looking for some methods of measurement that confirm what we hear minus the psycho acoustics." (Sorry my quoting skills often prove to be inadequate on this site.)
The term "psychoacoustics" has unfortunately acquired a negative connotation, as if it's marketing snake oil. Correct me if I'm wrong, but in your sentence that I quoted just above, it sounds to me like that's how you're using the term. The "book" definition of the term is more like, "the scientific study of the perception of sound", and I believe that is how the term is used in the paper you cited.
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Post by duke on Jun 5, 2017 13:16:07 GMT -6
Audibility is a very very complex subject because of the way our hearing works. We can hear amazingly low levels provided nothing happens to be masking the frequency range of the artifact or we have hearing damage preventing nearby frequencies from masking it. I'm hardly an expert but I know enough to know that all sweeping generalizations about what is or is not audible are pure, utterly ignorant B.S. The more I've learned about the subject since the psychology of music class I took in 1968, the more amazed I am that anything sounds good to more just than the person turning the knobs. Yes!! Whether or not the ear's "masking" characteristic comes into play has a huge effect on the audibility of distortion components. Masking does not work well in the time domain, by which I mean, if the distortion occurs later in time, even just a tiny bit later, then it is far more likely to be audible and objectionable. The higher-order distortion components in an amplifier which are the product of large amounts of global negative feedback occur after a slight time delay, relative to the original signal, and this works against masking.
Or if distortion occurs during a quiet time, like crossover distortion in an amplifier, or a noise floor that is present no matter what, it is far more audible than if it occurred at some other time when it might be masked.
This general principle extends to the domain of loudspeakers as well: For example, diffraction causes only a small disturbance of the frequency response, but in effect it is a spectrally-distorted reflection that arrives just a tiny bit later in time (because of the path length distances to the ear), which means that it is not masked, and therefore is more likely to be audible and objectionable. Diffraction horns, which intentionally use discontinuities (kinks or sharp bends) within the horn itself to diffract the energy and thereby widen the high-frequency coverage pattern, tend to sound more and more harsh as the level goes up. This is a prime example of the ear having a non-linear perception of what is actually a linear distortion. Diffraction is not a characteristic of all horns - a horn designed to minimize diffraction can be a very low coloration device even at high sound pressure levels - but unfortunately diffraction horns have been responsible for many people assuming that all horns are harsh, because the ones they heard sure were.
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