I recently did a church installation that have been giving me headaches. I used good commercial equipment that performed well. My problem is getting enough gain before feedback occurs. I used 3 15' speakers with horns clustered close together to form an array. They were flown 20 feet up and approximately 8 feet in front of the pulpit. The coverage was very good, but I wound up using A Sabine Feedback exterminator as well as a Pitch shifter to help overcome this problem. The equipment handles the loud volumes that is generated by a high spirited service. As far as I know, I did things right. I have another installation coming up and I'm getting sceptical on where to mount the speakers relative to number of open mics.. I should comment that the discussed church was 65' X 35'shoebox shaped with a a vaulted wood ceiling. It also had carpeting on the floor. I would appreciate some help from the EAW guru's since I consider using your product. I know there is not a lot of specification here, but perhaps some rule of thumb ideas. Thanks in advance JJ
Been there, done that, did'nt like it! by Michael L. Jordan B.S.A.E, 3/14/96
As an audio systems designer/contractor who is also Media director for nearly 300 churches in the Southeast U.S. I frequently run into feedback problems and other "Demons". There was a very enlightening article in the January 96, I think, issue of Sound & Video Contractor magazine. This issue was dedicated to the total elimination of feedback (the "F" word).
In particular the article on the NOM curve has hepled immensely to eliminate alot of the problems, and this is a mathematical formula that I had'nt seen since college nearly 15 years ago. NOM is an acronym for Number of open mics. In high SPL environments, especially in environments that have moderate to high reverberation times such as many churches, feedback can become a constant foe, and an adversary that must be dealt with scientifically. I suggest that you have a thorough acoustical analysis done by an Acoustical engineer with experience in Church environments. Insist on them using the Techron TEF 20 or similar quality analyzer. Verify that there are no immediately obvious and harsh reflections that may be causing or contributing to your feedback problem.
Second verify that the systems EQ is properly set taking into account both direct sound and room noise, this will allow you to rule out overly compensated or under commpensated frequencies. In room acoustics a little touch can go a long way.
Thirdly and maybe most imprtantly, verify that the speech inteligibility, is good or at very least fair. Sometimes sound technicians attempt to overcompensate for low intelligibility by simply "carnkin' it up" as we say down South. There are a number of ways to deal with feedback and as many opinions as there are peanuts in Georgia. Again the best advice is to find someone qualified to analyze the situation and make recommendations as to the best way to approach the existing problems.
As for your future projects, there are three things you can do to make sure you get the most bang for God's buck. 1. plan 2.plan some more 3. Get a qaulified engineer to review your plan and make some recommendations. If all else fails I make myself available for advice to Churches only who need the advice from someone they can trust who is in Ministry in thia area.
Dan,
For a semi-affordable unit I really like the Yamaha Q2031A, 1/3 oct, 31 band, 2 channel equalizer. I would not however recommend the single channel version (I think it is the Q1031). To me (maybe it's all in my head) it does not seem to be of the same quality as it's 2 channel big brother.
Scott Dorsey
Other good options, if you have a bit more cash, would be the Ashley's eq's. Personally I am not crazy about Ashley, but most people DO NOT seem to share my opinion. Probably your best choice (and I doubt many would argue this) would be a Klark Technic, however be prepared to pay appropriately.
Finally, for a more reasonable price, you might check out the new DBX eq's. They're not bad FOR THE MONEY. I would definately stay away from the Alesis, DOD, Furman, Peavey, and other "low end" units that tend to create more problems rather than solve them.
Well, I can recommend the Ashly or the White parametrics, if you have the money. They have such a tight Q that you have to back off the filters for installed sound applications because the resonance points of the room change with atmospheric conditions! For a travelling PA system where you are dealing with many different rooms, they're feedback-eaters.
--scott
______________________________________________
In article 1996Sep29.182041.25765@jarvis.cs.toronto.edu,
Robert F. Enenkel
>I used an equaliser that covers 20Hz to 20kHz in 1/3 octave steps. The first
>thing I tried was to connect a sine-wave generator to the input, and put a
>microphone where my head is when I listen, feeding the mic into a tape deck
>so I could look at the dB meters. I set the generator to each center frequency
>of the equaliser in turn, and adjusted the EQ until the mic output was always
>the same. I did each channel separately. But I don't think much of the
>result, because if I move the mic a couple of inches, the levels (especially on
>the high frequencies) totally change. In fact, I could actually hear the peaks
>and nulls when I moved my head slightly. So much for that idea.
Yup, real bad idea. That's why NOBODY uses fixed-frequency sine waves to do what you are trying to do.
>The next thing I tried was to use a synthesizer keyboard as a signal source,
>setting it to do a frequency sweep over a small range. I repeated the above
>procedure, setting the synth for each center frequency on the equalizer in
>turn. This time, I didn't get the "peaks and nulls" effect with the high
>frequencies, and the result didn't vary much with small changes in the position
>of the mic. I went over the whole EQ several times since changing one slider
>also had some effect on the adjacent bands. Finally, I had the response flat.
>However, when I listened to some music, it sounded terrible. The bass sounded
>better, but high vocals and cymbals sounded awfully shrill and piercing. I
>wasn't able to listen very long at a high level without getting a headache.
Yup, your results are not in the least bit surprising, but, you continue...
>I realise my procedure depends on the mic having flat response, but all I
>could do was trust the manufacturer's graph showing 20-20,000 Hz +/- 3 dB.
>Is there a "right" way to do this, without super-duper measuring equipment?
>I also tried using an oscilloscope instead of the tape deck's dB meters, but
>that wasn't the problem either.
No, it's not necessarily the microphone's problem either.
The REAL problem is that the basic premise of what you are trying to do is, in many ways, fundamentally flawed. Many, if not most room problems are NOT problems that an EQ can correct.
Room problems occur, for the most part, because, for whatever architectural reasons, there is prolonged energy storage at some frequencies and not at others. With more sophisticated measuring equipment, you'd find, for example, that the reverberation time varies quite considerably with frequency. It might be very short at one frequency, and 1/3 octave away, it might be very long.
The problem being is that an equalizer does absolutely NOTHING to change the rooms reverberation time. The only thing it is doing is changing the amount of energy put into different bands: the energy storage and decay of your room modes remains unchanged. Now, the other problem is that while you may, on a steady-state measurement of the kind you are doing (that is, long term time avrages where the all-important time information is lost), you might well achieve a flat steady-state response (doubtful, but just maybe), but what your ears sense as frequency response is the first arrival signals. And using the equalizer to correct the steady state response of the room, you've royally screwed up the first arrival response from your speakers.
Sorry to say this, but, for MOST acoustical applications, equalizers are, for the very reasons you've demonstrated, worse than useless. The way to deal with the symptoms of acoustic problems is to deal directly with the acoustic problems. Screwing up the response of your speaker is NOT the way to do it.
There is also the possibility that the equalizer you're using is awful in and of itself, but the symptoms you describe are exactly what I would expect from an attempt to "flatten" a room using 1/3 octave steady state analysis and even the best of equalizers.
In a word, it doesn't work, despite the strident self-serving claims of manufacturers.
If you borrowed the equalizer, give back with a warm thank-you. If you bought it and own, well... It might be usefull, when used sparingly, for correcting bad recordings (something an equalizer is actually useful for in the right hands).
But for correcting bad acoustics in a room, the only thing an equalizer can do is make things much worse, as you have learned.
The $60 Radio Shack PZMs are NOT the same by a long shot as the real Crown PZM mikes. RS just licenses the basic design from Crown and has them made in Taiwan or Korea. I have two of the RS mikes and I also have two of the latest Crown PZM 30D mikes. There is no comparison. The actual mike elements in the RS are actually hearing aid elements.
The RS mikes have a lot more self noise, poor power system based on 1 AA battery and can't be powered from phantom power. They also are unbalanced high Z mikes with a 1/4" phone jack plug to consumer decks.
The Crowns have balanced 3 pin XLR connections that are low impedance and only accept real phantom power. They have a dead flat frequency response from 20-20k or via a built in switch a slightly rising top end for percussive based sounds such as piano, drums, guitar, etc. that have rapidly rising initial sounds. The self noise is a very conservatively rated 20dB slow A weighted. They have a SPL level in excess of 150 dB.
The RS mikes though are damn good for the money and can be modified simply using what is known as the Rastocny Mod. This upgrades the power system, makes them balanced and improves the floor noise. They still won't be the real thing but they will blow the doors off a ton of very high end mikes when it comes to performance.
By the way....if you have the very early RS PZM mikes (the ones that used to sell for about $45) you have a better mike than the current models.
I have several mikes including AKG, Shure, Audio Technica, and Sony and have to say that generally my favorite mike for DAT recording of groups, choirs, pipe organ, orchestras is the PZM 30D. If you can direct mike soloists or solo singers than I will go with the others but for groups or larger sound sources the sound of the PZM is a lot better. The ability to minimize the adjacency effect (sound in a drum due to out of phase reflections) makes the overall recording a much more faithful rendition of what I experienced in the audience.
By the way Crown does make cheaper mikes than the 30D (the 30D sells for a little under $300 from places like Full Compass) that eliminate things like the freq. switch and may use a slightly lower quality mike element...but I don't have any experience with them.
These mikes don't mount on a mike stand though. They must sit on a boundary...a floor, wall ceiling, or 2'x2' x1/4" sheet of plexiglass. Crown makes an excellent book called the application guide that is available from them for free that explains tons of unique setups for just about every possible situation.
Final comment....you really have to try hard to get a bad recording with the PZMs!
PZM Modifications, by Christopher Hicks (cmh@eng.cam.ac.uk)
=======================================
Last modified and copyright, 2nd October, 1995. Non-commercial
distribution encouraged; all other rights reserved by the author.
Firstly, You mess with your precious(?) RS PZM's entirely at your own
risk! Of course I disclaim all responsibility for your destruction of
your equipment - microphones, mixers, tape-recorders, lawn-mowers and
food-processors included.
I should point out that I have tried very few of these ideas; consider
this more as a starting point for your own experimentation. Most of
the ideas I present are equally applicable to other electret capsules,
such as those from Panasonic.
On with the interesting stuff...
Many people do not realise that the PZM as supplied by RS is actually
a balanced microphone. To convert it for connection to a balanced
mixer input is as simple as removing the moulded 1/4" plug, and
replacing it with a male XLR3. The shield goes to pin 1 of the XLR,
and the other two wires go to pins 2 & 3 - which is which doesn't
matter too much as long as you do them all the same if you are
converting more than one microphone.
Another simple modification is detailed in the manual. It is
suggested there that better dynamic range and higher maximum SPL can
be had by replacing the 1.5V AA cell with two 6V batteries which will
also fit in the battery compartment. The 6V battery is a bit
expensive and you can use a 9V PP3 if you bodge the connections and
tape it to the outside of the box.
Mechanical and acoustic modifications I have heard of include:
a) removing the bit of black fuzz from the front of the capsule.
b) enlarging the hole in the front of the capsule.
c) reinforcing/sealing the rear of the capsule with epoxy.
d) abandoning the metal plate in favour of a 6" square of perspex
(plexiglass for those in the US!)
Since the capsules are so variable in their manufacture it is
difficult to tell ahead of time whether any of these mods will lead to
an improvement in the sound.
Mods a and b aim to modifiy the HF response - as supplied most of the
capsules have a pronounced presence peak, and a fairly nasty phase
contortion around 4-6kHz due to the small size of the hole. This is a
deliberate resonance introduced to extend the hf response as far as
possible towards 20kHz. The felt pad is there to tame this resonance a
bit. Mods a and b in combination are an attempt to smooth out the
treble response by removing this resonance, at the expense of not
reaching quite as high into the stratosphere.
Mod c attempts to damp the phenolic backplate (which otherwise bends
in the acoustic breeze (allegedly!)) and sealing the back more
effectively should, in theory at least, extend the LF response a bit.
Note that this will not help compensate for the lack of bass
encountered when the baffle size for the PZM is insufficient.
Mod d replaces a highly resonant piece of metal which rings like a
bell with a piece of plastic which doesn't. This seems like a good
idea to me, but whether its effect is significant I can't tell without
trying.
A number of electrical modifications are also possible, and their
effects are a bit more predictable than for the mechano-accoustic
ones! First, the FET inside the capsule can be replaced with a
quieter one, though I have never managed to do this without destroying
the capsule. This is more due to my own heavy-handedness than
anything else. Definitely not an easy mod, but some folks have
reported success.
The electronics of the stock microphone are really let down by the
transformer and electrolytic capacitor lovingly installed by RS. Both
are really ghastly examples of their type, but what else can you
expect for the money.
To drive an unbalanced input from a battery-powered mic, but removing
those two components is quite simple. I make no claims regarding the
benefits or otherwise of doing this!
+---------------------------- battery +ve (3 to 12 Volts)
|
2k2
|
o---------- 10u -------o----- output
|+ |
CAPSULE 10k
|- |
+----------------------o----- GND, and battery -ve
The 10u capacitor can either be a high-quality plastic-film type
(expensive and large), or an electrolytic in parallel with a small (eg
100nF) plastic-film capacitor (cheap and small). In the latter case
the positive of the capacitor should go to the capsule, and the
negative to the 10k resistor. See note 5 below for the capsule
polarity.
I have also designed two schemes for completely replacing the innards
of the microphone with better, phantom-powered electronics. The first
method is the simpler; the second is more complex, but provides a
lower output impedance, thereby allowing longer lines to be driven.
Both remove the horrible transformer, and both remove the equally
horrible electrolytic capacitor from the original RS circuit. Neither
is intended as a complete cookbook method, but both can be made to
work well with a little electronic skill.
Method 1
========
10u
+---------o-------------||------o----------------- HOT (2)
| | |
| |- |
| CAPSULE 22k
| |+ |
| | 10u |
| o-------------||------|------o---------- COLD (3)
2k2 | | |
| 2k2 | 22k
| | | |
| o--330R---o----o------o------+
| +| +| |+
| 10u 12V 10u
| -| -| |-
--o---------o---------o----o------------------------ GROUND (1)
Notes:
1) The component "12V" is a 12 Volt zener diode
2) The 10u capacitors in the HOT and COLD signal leads should be
high-quality plastic film types. The values of these may be
reduced to 2u2 if the preamp input impedance is 10k or greater.
3) The 10u capacitor in parallel with the zener should be a tantalum
type, and can have a 10n plastic film cap in parallel if you wish.
4) The cable to the capsule should be twin+shield. The shield should
be connected to ground near the zener diode, and left unconnected
at the capsule.
5) The polarity of the capsule is important. The + side is the one
connected to the casing. (Odd but true, at least in the case of
the RS PZM.)
6) The pinout given is the standard for XLR3 mic connectors.
7) If you want to use the existing RS box you will find that the 10u
capacitors do not fit. If you *must* then use electrolytics for
these (>50V working) and bypass them with 100n plastic film caps.
Method 2
========
+-----o---------------------330R--------+
| | |
| 2k2 +---10k----+ |
| | | | |
| | | E--o---|--------------- COLD
| o---------||----o--o-------B |
| | 1u0 | C |
| |+ +-100k-+ | |
| CAPSULE |---o------o
| |- +-100k-+ | |
| | | C |
|+ o---------||----o--o-------B |
10u | 1u0 | E--o---|--------------- HOT
|- | | | |
| | +---10k----+ |
| | o----+
| 2k2 +| |+
| | 12V 10u
| | -| |-
--o-----o---------------------------------o----o---------- GROUND
Notes:
1) Notes 1, 3, 4, 5, 6 from above apply here too.
2) Component EBC is a PNP bipolar transistor, eg BC479
Ideally these should be hand-picked for low noise and
matched gain. Bear in mind that VCE can be up to about 36V.
3) The 1u0 capacitors should be high quality plastic film types
4) This circuit will fit in the existing RS box, but a metal one
is recommended for the additional screening it affords.
5) The circuit may benefit from the addition of 22pF capacitors in
parallel with the two 100k resistors.
6) For minimum hum pickup the two 2k2 capsule bias resistors should
be accurately matched.
-----------------------------------------------------------------------
I built the FET circuit. Works great.
David
____________________________________________________________
"Risk your life on every note" C. Haden
............................................................
David L. King | ktone@chitown.com
Chicago Psyberview-Music and Arts | http://www.chitown.com
............................................................
Rhythm City - Dance/R&B | http://www.mcs.net/~ktone/rc.html
____________________________________________________________
Roger Andersson speaks:
>I will now say a few things about speaker cables.
Of this, we are sure.
>Everything said in this article can be scientifically proved and is
>uncontestable. The sound technical society of Sweden can prove that
>everything in this articel is the one and only and final truth.
This sort of attitude is, regrettably, is one of the worst imaginable examples of "science." Nothing scientific is "uncontestable," and it the height of arrogance to claim otherwise. The value of the scientific approach is that everything IS contestable, and must stand on it's own merits.
Nor is science, in any way, interested in any such thing as a "one and only and final truth." One and only truths are the province of religions, not of science.
Over and above that, these claims of "uncontestable" and "one and final truths" are made even more ridiculous because of fundamental technical mistakes in Mr. Andersson's assertions. Let's explore each one in turn.
>If the amount of signal proportionally missing at the end of the cable is
>both independent of the frequency as well as the level then the cable is
>a perfect damping device. This cable would theoretically be a perfect cable.
What you are describing, Mr. Andersson, is simple resistance, nothing more, nothing less. Let's call it what it is.
What, though, is a "perfect damping device?" This is a term unheard of in the technical lexicon of cables, loudspeakers, and such. And, further, why should a cable with resistance (which ALL cables do) by this undefined "perfect damping device?"
Regarding the damping of, say, the fundamental mechanical resonance of loudspeakers, the damping of the system is dependent upon the total effective electrical, mechanical and acoustical losses in the system. Of these losses, the greatest is, in the vast majority of cases, electrical. Those electrical losses are, with the exception of rare pathological examples, dominated completely by the DC resistance of the voice coil.
Unless you are willing and (needless to say) able to show that the resistance provided by the cable is a significant part of that electrical loss: more specifically, that the resistance of the cable is significant compared to that of the DC resistance of the voice coil, then the cable provides NO significant contribution to the damping of the system at those points where it is most important: at the high reflected motional impedance due to the resonance of the system.
Typically, the DC resistance is on the order of 4-8 ohms, and unless you can show cables resistance greater than 5-10% of THAT, your claim of a cable being influential in the damping of the loudspeaker is so much nonsense. That it makes a "perfect damping device" is utter hogwash.
If you are going to talk about the technical aspects regarding cables and such, you are obligated to use the language and the principles suited to the purpose. Inventing new untofore heard-of technospeak does nothing to advance your cause, whatever it may be.
>If the missing signal is proportionally the same at all levels but is
>dependent on the frequency, then the cable will cause an error in the
>frequency response. This can be compensated for by some kind of tone
>control. This is not a perfect cable.
And what is cable?
>If the proportionally missing signal at the end of the cable is both
>depending on the frequency as well as the level, then the cable causes
>severe distortion. The cable's behavior is totally unpredictable. This
>is the worst kind of cable and it is absolutely useless. Unfortunately
>most expensive audiophile, high end cables have this type of problem.
Sorry, Mr. Andersson, you're quite incorrect and several of your assertions here.
First, by what mechanism does a cable have the level-dependent behaviour you describe? I doubt that you or your friends at the magazine have, in fact EVER observed such an effect.
Second, IF the cable DID have the non-linear behavior you describe AND if had a fequency dependent behavior, simply by characterising the non-linearity in both domains, it's behavior is COMPLETELY prodictable. You may not know how to predict the behavior, your firends at your magazine may not know how, but the techniques for doing so have existed for nigh on a century.
>The worst problem with cables is the cross-impedance-modulation. This
>distortion is caused by electrons jumping from one strand to another
>in a cable with multiple strands. This happens in all cables with
>multiple strands no matter what sorts of materials are used. Because of
>this all cables with multiple strands are useless for hifi-applications.
Sorry, Mr. Andersson, but this entire paragraph is absolute techno-drivel. Where on earth did you or your magazine friends ever come up with the term "cross-impedance-modulation?" This is complete nonsense.
Ignoring the bizzarre, inaccurate and completely non-accepted terminology for the moment, Your assertion about electrons "jumping" about between strands is implicitly based on the requirement that for an electron to, indeed, move from strand to strand requires there exist a potential difference between strands. Can you, in fact, show that such a potential difference exists as an a priori condition of conduction in multi-strand cables?
And, despite your hokum nonsense about "cross-impedance modulation", so what if the occasional electron jumped between a strand. So what? You went ahead an invented a non-existant term to describe a non-existant phenomenon, what difference could it make and why?
>The only good cable is the solid-core-cable. The solid-core-cable has
>only one strand and this makes it a perfect cable.
Well, given the completely unstable foundation for upir multi-strand assertion, this carries little weight.
>The skin-effect is just something the high end cable manufactorors have
>made up. In practice it doesn't exist.
Well, Mr. Andersson, in this particular respect, you and your magazine friends are completely, provably wrong.
SKin effect is NOT something the "high-end cable manufacturers have made up." It is not something that "in practice, doesn;t exist." It is a very real phenomenon that is trivially verifiable in measurements. It was well understood, described, measured and accounted for LONG before there was a high-end notion of audio.
Whether the skin effect is SIGNIFICANT at audio frequencies or not is one issue, and it can be argued that . But your declaration that it simply doesn't exist flies completely in the face of a half century of well documented work. You and your friends are quite ill informed in this respect.
>Everything said in this article can be scientifically proved
Sorry, Mr. Andersson, nearly everything you have said in this article of yours can be easily shown to be technically incorrect, regardless of whther or not your particular selection of cable types has any audible advantage.
> and is uncontestable.
No, because it is so inaccurate, because of nonsense terminolgy, because of your outright dismissal of well-understood electrical phenomenon such as skin effect, it is not only quite contestable, it is provably bunk.
>The sound technical society of Sweden can prove that everything in this
>articel is the one and only and final truth.
The technical assertions of the "sound technical society" are technic ally quite unsound. Indeed, many of them are simply wrong.
> In The Molt, the journal of the sound technical society, it has been
>proved that all expensive high end cabels with multiple strands is
>unusable for all audio applications.
If the above is an example of their "proofs", it proves nothing more than a very poor grasp of the technical concepts surrounding the electrical properties of cables. I would grant that the above may or may not accurately reflect the proceedins of the "sound technical society," but the above is all we have at our disposal on which to make an assesment. And from this evidence. we are forced to conclude that the assertion presented therein suffer from serious technical flaws. They are not "a final truth", they are not "uncontestable", they are wrong.
>I would personally not pay $1 for a pair of these cabels.
A $1 cable, even of the worst kind, is a better deal than the kind of techno-babble claptrap presented here.
This kind of "science" is little more than an annoying embarrasment to the name of science.
--| Dick Pierce | | Loudspeaker and Software Consulting | | 17 Sartelle Street Pepperell, MA 01463 | | (508) 433-9183 (Voice and FAX) |----------------------------------------
I've found that most RFI problems are the result of RF currents, picked up by cables acting as antennas, and feeding these currents into the electronics. Note that it's not necessarily RF voltage ACROSS the input or output connection. It's likely to be RF current flowing into the grounded lead of the audio unit from the cable. (A simple antenna). Yes the grounded lead may actually be sensitive to RF currents, demodulate it in an audio stage, and there's your music.
Try putting some RF chokes (maybe a few turns of wire around a ferrite core) in series with the connections. Do this as near to the connector as you can. Try using a large enough core, (toroidal is probably good) to wrap the entire cable (shield, hot lead(s), and all) around the core. You may want to splice the cable to a thinner, smaller gauge, cable for the coil.
Here's a technique that I've had success with. Connect the output of the mixer to a battery powered headphone amp, or battery powered speaker. If you hear a radio station, start disconnecting inputs, AND OUTPUTS, to the mixer. You may be able to narrow it down to one connection that is the culprit.
------------------------------------------------------------The FCC closed 12 field monitoring stations this summer and has little interest or ability to track down RFI reports. The FCC's position for a number of years has been that RFI problems are the fault of poorly designed or shielded consumer and semi-pro audio gear, not the RF generator. If the voice is clear(AM) or sounds like Donald Duck(SSB) then the mic/preamp/cable combination is receiving and detecting the RF envelope. No indication of a faultly transmitter is likely to be found. A transmitter with spurious emmission or other defect does not produce more interference to audio systems. There are few transmitter problems that can result in radiation at audio frequencies. The problem needs to be fix at it's source - the poorly shielded or filtered audio gear with a nonlinear stage that is acting as an envelope detector. --------------------------
--------------------------------------------------It has also been shown that some mixers that use quazie balanced inputs ( that is when pin 3 is shorted to ground through a resistor or other component ) can pick up RF without anything connected to the mix because the chaise of the mixer is physical in phase with a radio station transmitter hundreds of miles away. Moving the mix a few few feet will show whether or not if this is the case.
Another source of RF is also from low cost balanced mic inputs. Pin 2 and 3 carry the signal. As the signal arrives in the mixer, the signals are reversed and any RF above or below the frequency range of 20hertz to 20K or 30K are canceled. But! of the input components of pin 2 or 3 - if either side is out by .05% in tolerance, RF and signal loss occurs.
Mixer with transformer balanced inputs or high quality electronically balanced inputs/outputs have a long service life. Usually 20 years or longer. Mixers of lesser quality have a service live of 10 years or less.
Question
>Is there a text, similar to those available on the basic principles of ray-tracing, which I could use to get an overview of the physics involved in room acoustics?
Reply
There are a number of texts on room acoustics (and many more on hall acoustics
Asking
>The reason I ask is that my company is investigating a development project involving the use of simulation software to assist architects in acoustical design. To my limited knowledge of the subject, the principles - although complex - must be quantifiable, and to that end, we could use a 'bible' to work to.
Reply
If you are planning to change the world with a piece of software, you are in for a rude awakening. You and your company should be aware of the following points:
1) Software simulation of room and hall acoustics is an active university research topic, especially in Europe. A number of people have gotten PhDs working on these problems, before moving on to do work that they could actually get paid for.
2) There are already a number of commercial software packages that attempt to do acoustical simulation. I can think of five, just off hand.
3) Most of these packages produce results which are thoroughly unreliable. Even in the rather limited arena of PA cluster simulation which is often their target application, they produce results which are woefully inadequate, as pointed out in the latest issue of the Journal of the Audio Engineering Society [Seidel and Staffeldt, JAES, Vol. 44, No. 7/8 (July/August 1996)]. New high-resolution polar data for loudspeakers may improve the situation in the future, although the modeling of the room itself will still be wrong.
4) Even if the algorithms in the software package were reliable, the real limitation is the absence of a good data base describing the acoustical performance of various surface treatments. All the packages I've seen have used tabular data that are twenty or thirty years old. These data don't reflect current commercial products, and they don't account for performance differences arising from variability in installation. I'm sure the top two or three acoustical consulting firms have better data bases than what you find in the back of textbooks; I'm equally sure that they won't divulge these proprietary data to outside parties.
5) Sound system designers use computer modeling packages anyway because they are better than nothing, at least if you understand their limits. It is pretty common to ignore the program's predictions for reverberation time and enter measured field data instead. Of course, if the building in question doesn't exist yet, that's not possible.
6) One program that has a pretty good reputation for accuracy is B. Dalenback's CATT-Acoustic package. [http://www.netg.se/~catt/] The last time I saw it, it didn't have much of a user interface -- using it was more like programming than running a CAD package. This may have changed in later releases. I think it would be a useful tool for an acoustical consultant, but I can't imagine an architect using it. There's also a package called ODEON that appears to be pretty complete, and seems to have a more polished interface. [http://www.dat.dtu.dk/~odeon/]
7) The market leaders in the US are EASE, CADP2, and Modeler. I don't think any of them are very accurate, and they all take a long time to learn properly. Most people who buy them end up going to class for a week, although I learned EASE on my own.
8) An architect has no business using any of these packages to avoid hiring an acoustical consultant. It would be like buying Microsoft Excel in hopes of avoiding hiring a registered professional engineer to approve the foundation design.
There, I've finished venting my spleen. I've also just saved your employer a lot of money.
David L. Rick
drick@hach.com
Copyright (c) 1996 JdB Sound, Acoustic Lab.
Last updated, May, 1997