Article

Tuesday, August 02, 2011

RE/P Files: An Interview With George Martin At A.I.R. Studio London

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is an in depth look back at the career of a legendary engineer. This article dates back to the January/February 1971 issue.

George Martin at A.I.R. Studio London

bw: What do the letters “A. I. R.” stand for?

gm: Associated Independent Recordings.

bw: Has A.I.R. done any independent production locating the talent, etc. as yet?

gm: Yes, but not much. We left our respective companies just over five years ago—three of us left EMI and one left Decca—and we had to do a deal with EMI which lasted five years in fact; it ended about a month ago.

This was basically an independent deal but it also covered the servicing of artists that were contracted to the company anyway.

Obviously the Beatles came under that, and other artists that we handled—there were quite a few. So we had to maintain those artists and so our time for finding other artists was obviously limited.

But at the same time, as the years went by it became more and more difficult to get new artists not because they weren’t there but because the deal that we had with EMI was limited to an overall royalty which gradually became-well, in fact, very quickly became out of date.

So that by the time the contract ended we couldn’t possibly hope to secure any artists because we couldn’t offer them any money. We were bound by that and we couldn’t do anything about it.

Now that we’re free we can really look around—sniff the air—which is what we intend to do. But we decided, in fact, before we did that, to build a studio.

bw: Several of the studios I’ve visited in England are equipped, as is A.I.R., to handle visual material as well as audio. Do you feel that there is a potential in integrating the pop music field with visual technology ?

gm: Actually there aren’t all that many studios here that also do visuals. There are far more —fewer sound ones. But the tendency is, of course, to open up the visual side—mainly because, I think, this is inevitably the future. You’re bound to have video recordings they’re on our doorstep.

bw: What are your feelings about four channel sound?

gm: We haven’t built it into our boards mainly because it’s a very new development and most people in this country don’t know anything about it.

We know about it because we go to your country. I honestly don’t believe it’s a very important development. It’s quite nice; it’s pleasant; it’s a very nice gimmick, but I can not imagine the average person going to the elaboration of fixing up four speakers in their room so that they can hear the ambiance of the concert hall behind them . . .

You could have circular sound, of course, but when I was introduced to quadrasonic sound my comment was that if you’re using four speakers the ideal is not one in each corner of the room, but it is three in an equilateral triangle below you and one above you so that you’re in the center of a tetrahedron.

Then you’ve really got all around sound, in all manners—you’ve got up and down as well. But this is being idealistic and I really don’t think it’s for the average man. It’s very nice, but I can’t imagine Mrs. Jones of Wiggum or in your case Mrs. Bloomfield of Connecticut taking the trouble of fixing up her drawing room or . . . whatever you call it . . . the lounge with four speakers.

bw: Is there stereo radio transmission in England?

gm: Yes, there is, but it’s very limited. It’s third programme stuff; that is, you get classical concerts occasionally broadcast in stereo and occasionally you get stereo record broadcasts. I should think the number of people in England who listen to it is about .001 per cent.

And also, people don’t listen to radio much anyway. The average man in this country is glued to the television set.

bw: Would you describe what you feel the responsibilities of the producer are on a “rock” date?

gm: Yes. I’m glad you defined that because a producer’s responsibilities do vary an awful lot. For a rock date I think he’s got to get to know the group musically and obviously psychologically he’s got to know the people.

He’s got to get into their minds and he’s got to try to find out what they’re trying to express and if he can find out, it’s then his job to realise it in terms of sound.

So, his function is not to impose his will upon the group and produce his sound using the group as his puppet, but more to draw out from the group the best sound he can possibly get, and get them to play the best possible music.

bw: Then you feel that sound, as well as music, is a major responsibility of the producer?

gm: Yes. That’s the way I see it. It’s also psychological. I think you’ve got to learn how to get the best out of people find out when they’re going past it and so on.

bw: How would these responsibilities vary for a classical music session?

gm: Well yes, they vary enormously. To begin with in the classical session, unless it’s chamber music, you’ve only really got one person’s ideas to deal with, and that’s the conductor; and then, from the amount of classical recordings that seem to take place today, it’s more a question of the diplomatic handling of that conductor and trying to get the best out of him rather than the technical details of a good sound.

The classical producers of today, and I’m not calling myself a classical producer, seem to leave everything to the engineer and just act like a kind of . . . what shall I say . . . host to the conductor.

I don’t think they interfere too much musically, which I think is a pity. I think that classical music could be in fact improved by adapting certain pop techniques to it. I wouldn’t mind having a go at recording something classical in a different way.

bw: Would you, for example, use close miking?

gm: Yes. Most classical records are made like photographs of concerts, if you know what I mean—aurally speaking. The ultimate aim is to reproduce as naturally as possible the sounds of the orchestra as created in the concert hall.

Now I think this is terribly limiting. I mean it’s been done, and it continues to be done better and better because engineers and acoustics and recording techniques have advanced enormously. But I think we’re missing out on something.

I think that if Beethoven or Bach were alive today, they would call that a very timid approach, and I think they would go back to first base and say, “You’ve got tremendous tools here; let’s use them.”

And I think if you go back to the actual music and adopt, really, very modern recording techniques and produce a work of art which is different from what you hear in the concert hall, and not necessarily inferior which most people might think.

bw: Then the rock producer presently has more room for creativity?

gm: Unquestionably. That’s what appeals to me.

bw: (Before AIR Studios were built) Your responsibilities also include selection of the studio and engineer?

gm: Yes.

bw: In recording a rock group, will you attempt to capture a “live” studio performance, or will you construct a recording using, for example, overdubbing.

gm: I’m afraid the latter is true. One doesn’t go for a performance as such in the studio because you know darn well that if you do that there are going to be shortcomings in various other departments.

You might get a great vocal performance, and the bass line may not be so great.

So, there are various things that you can do-you can go and overdub the bass line if you’ve got good enough separation.

You’ve seen us working recently .. . what I was trying to do yesterday, in fact, with Peter, with the whole group, was to try to concentrate on Peter’s performance tying to get something out of him, and then worrying about the rest of it.

But in fact we’ve reversed the process today because we’ve decided that Peter will probably do as good a performance by overdubbing anyway. So we’re going back to first base and concentrating on the actual sound.

It doesn’t seem to impair the total result. Most rock recording is done that way today. You obviously get a much better sound on everything; you are able to pay much more attention to detail.

bw: You mentioned before the importance of psychologic ally understanding the group. Could you be more specific?

gm: It’s just instinct really a kind of sixth sense you build up. You’ve got to get to know people and sense what’s happening.

bw: Would you say that a sense of humor is important?

gm: Oh yes, a sense of humor is terribly important. Absolutely. If you didn’t have a sense of humor on rock dates, then everybody would go sour. I can’t bear people who take themselves loo seriously, including rock musicians.

bw: Do you find that you do a lot of producing during the mix down stage as well as during the recording stage?

gm: It depends on the artist and the record you’re making, and what techniques you’re using.

If you’re making a record like Sgt. Pepper, for example, the mixdown is just as complicated, in fact more so, than the original recording because you’re painting a picture in sound and you’re using extra things: you’re bringing in sound effects, you’re distorting sounds, you’re playing with them, you’re soil of shaping them-sculpting them, if you like—and mixing them down at the same time.

So that kind of production is probably more complicated and more important in the mixing stage than at any other time. But if you did that all your life, you’d be spending all your time mixing and none of it recording.

bw: Then it varies greatly from group to group?

gm: Very greatly, yes.

bw: When mixing down, do you physically operate the console, or do you direct an engineer?

gm: Like most producers I like to get my hands on the controls, and it’s wrong. Sometimes I do—sometimes you have to—because sometimes the mixes are so complicated that one pair of hands won’t work.

In fact, on many Beatles mixes, we would have the engineer sitting in the middle, me sitting on the right, and one of the guys on the left.

It depends whose song it was—it might be Paul or John or George. And we would all be playing with the faders, the three of us; we would actually be playing a sort of triple concerto.

But the snag with that is that you still need someone else to listen because when I’m controlling the controls on a mix, I’m listening for certain things that I’m controlling and I don’t have that essential requirement of being able to listen to the whole thing with absolute impartiality.

So nowadays I tend to get out of that scene and say, “This is wrong. You shouldn’t be handling the controls. You should be standing back and telling people what to do, and listening to the whole thing.” It’s only by being free that you can really see the whole picture.

bw: What qualities do you look for when selecting an engineer?

gm: Oh, that’s a big question. First of all, he’s got to be an enthusiastic engineer. I’m very fortunate with Bill (Price); he really is a dedicated engineer.

He must be keen on his job, keen on sound, and preferably— and there will be many people who will quarrel with this—preferably without the ambition to be a record producer, because I think that gets in the way of good engineering.

bw: Why is that?

gm: Well, there are an awful lot of engineers who become record producers, which is fine; I’ve got no gripes against that. But I don’t think you can do two jobs at the same time.

And there’s always the transition period when the engineer tries to do a bit of production, or goes back to doing a bit of engineering after he’s been a producer. And I think that they lose out because of that. They are two separate jobs and they need detached minds.

bw: Anything else?

gm: He’s got to be good at his job; he’s got to know a lot about recording—that goes without saying. He’s got to know the board, and he’s got to have a good ear. He’s got to have a personality where, without being servile, he makes it plain that he is there, in fact, to serve the group.

He doesn’t have to be a humble person. On the contrary, he must be a person of some authority and some spirit; but he must always give that impression, that he is there to get the best sounds out of people, just as the producer should give that effect.

bw: So you don’t care if the engineer has a musical background?

gm: No, not really; not personally because that should be the job of the producer.

bw: What kind of language do you use to communicate with your engineer? You mentioned to me before that you were non-technical, therefore / assume that you do not communicate in technical terms.

gm: Well, in fact, I do. I’m non-technical, but I still say to him, “I think we need a bit of top at 4,000 (Hz) on that, or try it a little lower down.” When I say I’m not technical, I mean I haven’t any technical training.

But you can’t grow up in the recording industry, and go from mono recording through stereo and multi-track, working all the time on boards, without picking up a little knowledge.

bw: Then you feel that the producer should be able to operate the console himself-at least in his head?

gm: I think it helps—anything that gives a greater understanding between people. I think that if my engineer knows that I know what’s going on, then he will respect me more and he’ll work more closely with me. If I don’t know what I’m talking about and I ask him for something that is patently impossible, I’ll lose his respect, and he won’t work so well with me.

bw: Do you prefer to work straight through with one engineer?

gm: I prefer to work with one engineer for a particular job, but I don’t want to work with that engineer all my life.

bw: Many Beatles recordings employ techniques or tricks such as phasing very tastefully. Did the ideas for these techniques come from engineers? Or, to put it another way, do you encourage your engineers to make suggestions?

gm: I certainly would encourage engineers to make suggestions. But in fact, all the techniques we used that you’ve described have come about not because the engineers made suggestions, but because we actually asked for particular sounds.

Phasing came about as a result of experimenting with the automatic double tracking, ADT, which was, in fact, suggested by an engineer, who strangely enough wasn’t a balancing engineer. He was a backroom boy who came forward with this idea.

He was an EMI bloke; he’s now in fact running EMI studios, which is nice. And so phasing came about as a result of that—playing with ADT. In most other cases they’ve been a result of personal experimentation in the studio. My experience with spoken word recordings—building up sound pictures without music was invaluable in that respect.

bw: Are there any special considerations that you keep in mind when producing a 45 rpm single release?

gm: Obviously it’s got to be a little more concise than an album track. There are a lot of things which you put on an album, which stand up on an album because they are part of a long scene, which obviously wouldn’t mean anything on a single.

In any case, you are making records to a certain extent for a particular market. One is well aware of the nature of the music that is played on the top 100 in the “states”, so you’re obviously thinking of that when you select your single.

bw: Is there any instrument, or instruments, that you consider particularly important, especially with regard to singles?

gm: No, I don’t honestly consider any one thing to be particularly important—I think they’re all important. When I’m doing a recording of a rock group, I do actually, mentally, go through every sound that I’m hearing, saying, “Is that the right sound?” I apply the same devotion to each one. If you miss out on one, you’re not doing your job.

bw: Is it true that the early Beatles records were remixed by Capitol for release in the states?

gm: They weren’t remixed by Capitol; they might have been re-equalized by Capitol. Yes, in fact, I’m sure they were. The story was in those days that American record players were different from English record players, and therefore they had to cut their own masters to suit their own tastes. And they did that; and I didn’t like the results, but I couldn’t do anything about it.

bw: Could you describe the differences in sound between the American and British releases?

gm: I didn’t think they (U.S. releases) were as good. It’s difficult to get a good answer to that one because I was hearing their records on my machine and I don’t know what they would have sounded like if I had heard them on their machines.

They may have been alright, but they generally sounded much thinner and harsher than our sound, and less bass certainly.

bw: Early Beatles records were characterized by a particular vocal sound which has been very influential on pop music in general. How did this come about?

gm: Because we had particular kinds of vocalists, really.

bw: You mentioned ADT.

gm: That was a particular sound we put on. You know, once we got over the first hurdle of being a success, they were always looking for something new. They were continually coming to me and saying, “Do something different.”

They were always prodding and trying to push some things out a bit further. John hated the sound of his own voice, which I personally thought was a great voice, and quite often he would come to me and say, “Can’t you do something with my voice; it sounds terrible.”

He’d say “I know it is terrible, but let’s do something about it. Don’t make it sound like me,” which was worrying in a way because he expected magic.

I don’t know quite what he was expecting to hear, but it wasn’t what he was producing and consequently we did play about with the voices quite a bit. Sometimes, I think the results weren’t very good, but in a lot of cases they were.

bw: Is it true that Sgt. Pepper was recorded on four-track machines?

gm: Yes, absolutely true. It was done four to four.

bw: Who did the engineering on Sgt. Pepper?

gm: Geoff Emerick, I think he did all of it.

bw: What other Beatles records has he worked on?

gm: I couldn’t give you a catalog—there are quite a few. When we started out, the engineer we had was a guy by the name of Norman Smith. I can’t give you which record he stopped on, but we could find that out easily the facts arc there.

But he came to me one day and said he wanted to be a producer… he was an EMI engineer. . . and did I mind. And I said, “No, fine. Off you go.” He said, “The only thing is, I want to go on engineering the Beatles.” And I said, “Well, now, I don’t think you can do that.”

I was very firm, but quite polite, and I said, “If you want to be a producer, that’s one thing and that’s fine. Go and make some good records. I’m sure you can, but I don’t think you can go on engineering at the same time,” which comes back to your previous question.

So he made the plunge and he left and became a producer, and he’s done some extremely good stuff. He made all of the Pink Floyd’s early records. He’s now a staff producer for EMI. But then I had to find another engineer.

Now there were lots of engineers senior to him at EMI, but I decided at that time that I wanted someone very new and young. I’d been looking around—looking for talent, so to speak, and I decided to give the chance to Geoff Emerick, who in fact had done very little recording before.

He’d been balancing for six or nine months before I gave him the job with the Beatles. He jumped at that and it was really tossing him over the deep end; but he was marvelous—he came out with colors flying. And after Geoff we used other people as well, but in fact, we brought Geoff back for Abbey Road.

bw: He didn’t, then, work on the Beatles white album?

gm: No, he didn’t.

bw: Would you describe some of the techniques used on Sgt. Pepper, for example on “For The Benefit of Mister Kite”?

gm: That’s really quite simple when you know about it. John wanted a calliope kind of sound. He wanted to get the impression of a fair ground and he played me this song that he’d written, and asked what I could think up to give it that kind of fair ground atmosphere.

And I thought a lot about it, and I decided the best way to do it was to use some of the techniques I’d done with spoken word records.

I decided that to get the kind of swooping, steam organ noise he wanted, I got him on one Hammond organ and me on another; actually I think he was on a Lowry and I was on a Hammond.

And we recorded some half speed organ, and I did some chromatic runs with the tremelo on fairly fast over two octaves and then sped them up to double speed. That was one of the things—the swooping noises.

But for the background mush, I got lots of steam organ tapes, genuine fair ground organ recordings of all sorts of pieces of music—“Stars and Stripes Forever” and those kinds of things—and cut them into short lengths (of tape) and threw them up in the air, literally, and just told the engineer to pick them up again and join them all together. He thought I was mad.

We played it and of course the result was very cacaphonic. We used that as just a general background, mingling mush, which gave the required effect ... all kinds of funny jumping—some of it was backwards—but it worked.

bw: Beatles records are also characterized by constructive use of echo effects. Do you pay particular attention to echo on your recordings?

gm: The right kind of echo, yes. There’s a tendency these days to use plates an awful lot, in fact exclusively. We have plates here but we also have an echo chamber, which I must confess I haven’t used a great deal yet.

But I believe that a good chamber can beat a plate any day. I used chamber mainly on Beatles records.

Actually, we used a combination of chamber and tape, which we called “steed”—I don’t know why we called it “steed”—but it was basically sending the delayed signal by means of tape into the chamber.

bw: Why weren’t any of the engineering teams credited until Abbey Road?

gm: EMI policy, and they didn’t like it even then. (Abbey Road)

bw: Beatles records, especially since Sgt. Pepper, have caused a rekindling of interest in the electric bass. Was bass a particular problem in recording the Beatles?

gm: Paul was always worrying me to get more bass on the records, certainly, and it was my job to try and get that bass on, true. Probably it was the single most worrying factor, of any sound that we produced, because Paul is a perfectionist and even when we got a great bass sound he didn’t think it was very good.

Now, you say that we got some great bass sounds, which is nice to know. I’d like you to relay that information to Paul.

bw: I’d be glad to.

bw: Could you describe a technique you used on the bass on Abbey Road, say, for example, “Come Together”?

gm: I think on that particular one we used a combination of direct injection and live sound.

bw: And limiting/compression?

gm: Yes, of course, and also a little bit of echo too. But each sound is treated on its own merits. That’s why we, in fact, got lots of varied sounds, some of which were not so good as others.

bw: The instruments and voices on Abbey Road have a particular clarity and presence that seem to be derived from close-miking or similar techniques. Was directed it in the studio. Everything else was mine. this your aim?

gm: I was aiming for clarity, but oddly enough, it isn’t very necessarily close-mike techniques that provide this.

This essence of that clarity that you talk about is the ability to differentiate one sound that is interfering with your bass, for example, then you do something about it. You change it. And I think the clarity comes from having distinguishable sounds anyway.

bw: Then from a production standpoint, if you’re going to have two sounds in the same frequency range, they should be playing approximately the same part, or else they will muddle each other?

gm: That’s right.

bw: Did you do all the horn and string arrangements for the Beatles?

gm: Yes, with one exception. Oh, I certainly didn’t do the Let It Be one, which Phil Specter did. I was quick to disown that. There was one exception; it was one of the string ones, which an English arrange did.

He gave us the score because I wasn’t around at the time and Paul wanted it done very quickly. Mike Leander it was on one title. He gave us the score and I directed it in the studio. Everything else was mine.

bw: Do you think that you’ll work with the Beatles again, or any of he Beatles?

gm: In the answer to the first question, I think it’s possible if the Beatles ever work together again. As to the individual Beatles, I don’t know. Each one of them is very talented, two of them in particular, in fact George, John, and Paul are obviously more talented than Ringo.

All four of them are very talented anyway, but none of them is as strong as the four of them together. The four individual parts were not as great as the entire whole. The Beatles, four people together, did something that nobody else had ever done before, and the fact that they’re not together I think is a very sad thing.

 
Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay.

After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Monday, July 25, 2011

Featured Audio Cartoons: The Faders

For several years, Live Sound International magazine has presented several ongoing series of cartoons with a professional audio slant - “The Faders” by Frank Frombach.

It’s proven quite popular with readers throughout its run, highlighting conversations between three mixing console faders - Darth, Bump and Slam.

Here we present several installments of “The Faders,” and more will be coming soon.

By the way, Frank’s a long-time audio professional in addition to being a talented cartoonist and all-around great guy. He works with SurgeX, serving as both southern regional sales manager for SurgeX USA and international sales manager for SurgeX International.

- Keith Clark (.(JavaScript must be enabled to view this email address))

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Wednesday, July 20, 2011

Old Soundman: Stage Volume & Singer Freaking

A guy who really needs to learn how to mix drums and whining from the talent (what a surprise). The Old Soundman is on the case.

Dear Old Soundman,
I occassionally run sound for a band that tends to play at local hole-in-the-wall venues.

O.K., we feel sorry for you, now move on!

The “stage” for the band is always in one of two places: a nice boomy corner, or better yet, right in front of a brick or paneled wall.

One of many problems I run into (including the lead guitarist who insists he hears better with his knees)…

I know that guy! And I think half of our readers at home do, too. He must have cloned himself a dozen times in each and every state of the union!

… is cymbal bleed-thru on the vocal microphones. If I try to spare the audience the shrill ring of these upper frequencies by pulling back the highs on the board, I seem to lose clarity in the vocal.

That is not an illusion. It is indeed what is happening, you are perceiving it correctly.

This problem gets worse when the guys are playing at a loud stage volume, and I need to crank a little more vocal, which of course starts to feed back when the ring of the cymbals hit the mics, then comes through the monitors and hits the mics again. You know the sad, sad story.
Help!!
Stip

I do indeed know the sad story, and even sadder is the fact that the list of remedies is a very short one. I’m a straight shooter, Stip. Move back the drum riser.

But you can’t - you’re stuck in this little club with a stage the size of a saltine. Now that you mention it, some cheese and crackers would really hit the spot right about now!

The drummer can be asked to use lighter cymbals with a shorter decay time. But since he’s a club guy, getting paid very little beyond the endless chain of longnecks he consumes, he probably only has his local music store’s finest, thickest bang-a-langa models. Don’t tell me he wears those warm-up things on his wrists? You do have it rough, Stip.

You want a hypercardioid mic for your singer, and he needs to stay right on top of it. I’m not gonna lie, everything I’ve said boils down to band-aids. It’s hellish there where you are. Would it make you feel better to hear how Jacquie gets treated?

Dear OSM:
Just had an outdoor gig. The singer was freaking out, saying “the sound sucks” when in actuality it didn’t suck at all. I tried to tell him (from my limited experience) that running sound outdoors is quite a bit different from running sound indoors. And, we are using quality equipment.

Since I’m a rank amateur at this, is there anything specific I can tell him to shut him up? He’s a great singer, but like most musicians, he has high-end hearing loss.

Thanks mucho. You crack me up.
Jacquie

Thank you, Jacquie! My, what excellent taste you have in humor.

Most of the self-righteous hornblowers over on the Live Audio Board (LAB) would be real quick to say that you should proudly tell this character off and then march off into the sunset, with your pride intact and your wallet quite empty.

Well, I guess some of the more sensible ones who read a lot of self-help books would advise you to talk to the guy when he is calmer (since right after a gig is a notorious time for musicians to make ludicrous remarks, usually due to lack of confidence in their own abilities).

He may continue to say “Wull, I dunno, Jacquie, it just sucked, y’know?” Most of us would shake your hand if you just hauled off and slugged him. But we live in a very litigious society, so it’s best not to.

What you are digging for is him to say something like “there was too much low end” or “it was too trebly.” Precise technical terms like that.

Luv –
The Old Soundman

There’s simply no denying the love from The Old Soundman.

Check out more from the OSM:
OSM: Lawyers & Open Mics
OSM: Deep Questions & Intriguing Quests
OSM: A Youngster’s Revolt

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What’s Under The Hood? Power Amplifier Sections, Connectors & Classes

A power amplifier is not just a black box that makes signals stronger; rather, this complex device has a number of functional sections, connectors and circuit classes that differentiate one model from another.

As audio professionals, the more we understand what’s under the hood of modern power amplifiers, the better we can make a wise buying decision.

What are the main sections or parts of a power amp?

Every power amplifier includes a power supply, an input stage, and an output stage. Most amps also have protection mechanisms; some have DSP, and a few have networking capability.

Let’s explain each feature…

Power Supply
Basically, a power amplifier uses the input signal to modulate DC from its power supply. This supply receives 120 volts AC from the mains outlet and converts it to DC to operate the transistors, FETs and MOSFETs and so on in the amp circuitry.

Two types of power supply are analog and switching. A typical analog power supply rectifies the incoming 50 or 60 Hz AC and low-pass filters it to create DC for the power amplifier circuitry.

A switching power supply converts the incoming AC to DC, switches it on and off at an ultrasonic rate, runs those pulses through a small, lightweight transformer, then rectifies and filters the waveform to produce DC. The switch-mode supply can be smaller and lighter than the analog supply, but is more complex.

Some amps have a separate power supply for each channel so that high demands on one channel don’t affect the other. A few also have a separate power supply for the input stage, which is the part of the amplifier that does not drive the loudspeakers.

It’s important that the power supply have enough power reserve to supply power for transients or signal peaks. That happens when the supply uses big filter capacitors that store energy, and releases it when needed.

If the amp is heavily loaded down (that is, it is driving a low output impedance), the power supply voltage may drop or “sag,” causing distortion. Using a separate power supply for the input stage prevents distortion in the input stage caused by the output stage’s supply voltage sagging.

Amplifiers of very high power draw lots of current through the power cable from the AC outlet. To avoid limiting the current that can be drawn, the AC power cable has to be heavy gauge and short.

And the circuit breakers feeding the amplifier’s AC outlet need to be 20 A or higher rather than 15 A. Low-current AC outlets can prevent the amplifier from reaching its maximum power output.

Input Stage
The input stage or “front end” accepts the input signals and feeds them to the output stage to be amplified. Here you’ll find connectors that mate with the input cables.

Level controls and any plug-in modules are part of the input stage as well. The level controls do not affect the gain of the amp; rather, they affect the input sensitivity – the input voltage required to drive the amp to full power.

Turning down an amp’s level controls does not make it less powerful or reduce its wattage rating. Instead, this requires the amp to have higher input signal to drive it to full power.

Put another way, turning down the level controls reduces the level to the output stage of the power amplifier. If you send the amp a high enough signal level, you can drive the amp to its full rated power even with the level controls turned down from maximum.

In fact, it’s standard practice to set the amp’s level controls for proper gain staging. Set up the sound system’s mixer so that signals peak around 0, then gradually turn up the power amp’s level controls until the sound is as loud as you want it. This results in the best system signal-to-noise ratio and headroom.

If you turn up the amp’s level controls to maximum, you’ll often hear mixer noise through the system loudspeakers because the mixer will have to be run at levels well below 0 on its meters.

Let’s look at other parts of the input stage. LED’s on the front panel indicate signal level, clipping, and overheating, so they can be used for diagnostics if you hear no sound or distorted sound.

Connectors in the input stage are on the back panel of the amp. You’ll see these types of connectors:

• 1/4-inch phone jacks: These are most often seen in portable PA or small band PA systems. TS (tip-sleeve) is unbalanced; TRS (tip-ring-sleeve) is balanced and is preferred for its rejection of hum and noise.
• Female XLR: This three-socket locking connector mates with a male XLR and provides a balanced connection. It’s used in portable PA and touring sound applications.
• Terminal block (terminal strip, screw terminals). This type is intended for permanent installations. It lets you eliminate connectors and their cost because the input cable is hard-wired to the terminal block.
• RCA or phono connectors: These are used for background music systems and home stereos.

All XLR inputs, terminal blocks and most phone jacks are wired balanced which rejects hum and noise on the input cable.

Output Stage
This stage amplifies, or increases the power of, the input signal up to a level sufficient to drive the loudspeakers.

In this stage are the power transistors (output devices), which tend to generate a lot of heat. Also in this stage are the output connectors which are wired to the loudspeakers.

Four types of output or loudspeaker connectors are phone jacks, five-way binding posts (banana jacks), Speakon connectors and terminal blocks (screw terminals).

• Phone jacks are inexpensive connectors for low-power applications. They are often seen in portable PA systems.
• Five-way binding posts provide a temporary or permanent, high-power connection to banana plugs, spade lugs or stripped wires. You’ll find these connectors in amps for touring sound.
• Speakons are a high-power, locking, cylindrical connectors used in touring sound.
• Terminal blocks are mainly used in installed sound applications to eliminate connectors and their cost.

Figure 1 is the back panel of a Crown I-Tech HD power amplifier, showing XLR, Speakon and five-way binding post connectors.

The best power connectors have low contact resistance. As contact pressure and contact area increase, contact resistance goes down. High-pressure contacts increase current flow by helping the current to penetrate through the surface films. They also increase contact area by flattening out the contact surfaces.

So when you use banana plugs, it helps to “stretch out” the ribs in each pin to increase contact pressure. Use a small screwdriver to bend the ribs.

Protection
The better power amps include circuitry that protects the loudspeakers and the amp itself from overheating and burning out. Some include a limiter to prevent the output power from getting too high and causing clipping, which can destroy tweeters.

Others prevent DC and ultrasonic signals from reaching the loudspeakers in the event of amp failure. Low-end units just blow a fuse or trip a circuit breaker if the current draw is too high, while high-end amps limit the output power so that the music doesn’t stop.

Cooling
The main cause of amplifier failure is overheating, so most amps include heatsinks and fans to keep the amp cool. In some units the fans come on only when needed. Some Class D amps tend not to get hot, so they don’t need fans.

DSP
A few models include built-in digital signal processing: compression, limiting, EQ, filtering, and so on. The advantages are:

• There is less gear to lug around in racks.
• EQ and limiting presets can be set up in DSP to work with specific loudspeaker models. Just select a preset that works with your chosen speakers.
• Some DSP includes diagnostics such as load monitoring to check for blown speakers, error logging, and so on.

Networking
Another feature in many modern amps is networking. Network-capable amps can be part of an interconnected audio network, so they can be controlled and monitored from a central computer. This beats walking around on stage trying to figure out which amp has shut down.

Amplifier Class
Let’s turn now to another aspect of power amplifier design. Amplifier class refers to the circuit design of the output stage, such as Class A, Class AB, Class D, and so on.

As a background for this section, remember than an audio signal has a positive half of the cycle and a negative half of the cycle (Figure 2).

Transistors are basically rectifiers; they can conduct (pass current) only during the positive or negative half unless they are biased by a certain amount. The bias can create a DC offset in the signal.

Here are the features of the most common classes:

Class A
• Has enough bias (DC offset) to shift all of the audio signal into the positive region in the output devices (Figure 3). As a result, positive/negative signal halves become more positive/less positive changes.

• Because the output transistors are always on, current flows at all times. This design generates a lot of heat. Some power is dissipated even when there is no music playing.
• Lowest distortion.
• Least efficient (typically 20 percent); wastes a lot of energy.
• Typically used in audiophile applications up to 300 W per channel.

Class B
• The output devices are in push-pull pairs: one device amplifies the positive half of the sine wave signal, and the other amplifies the negative half. Each device of the pair is on for half of the signal cycle (positive or negative voltage) and off for the other half of the cycle. Each device conducts for a half cycle.

• Much more efficient than Class A (typically 60-70 percent).
• Less heat.
• There is discontinuity at the transition point between transistor signals near 0 volts (Figure 4). This results in high “crossover distortion” or “switching distortion” – low sound quality.
• Typically used for pocket radios or clock radios.

Class AB
• Both output devices in each pair are biased slightly on which reduces crossover distortion. Each transistor operates slightly more than half the cycle but is off for a fairly long time, which reduces heat dissipation. (Figure 5).

• About 50 percent efficiency.
• Low distortion.
• Typically used for home stereos and pro audio amps up to 600 W per channel.

Class G or H
• Both of these classes include two DC rails (DC supplies) of low voltage and high voltage. The high-voltage rail is switched on only when the input signal demands it, which reduces the amount of heat in the output devices. The power supply is signal-controlled.

• In Class G, one output stage is fed by the low voltage rails and another stage is fed by the high-voltage rails. The low-voltage stage is always on, and the high-voltage stage turns on only when the signal exceeds a threshold level. Class H uses only one output device stage which is fed variable supply voltages depending on the amplitude of the signal.
• Fewer output devices and less heat sinking are required, which reduces the weight and size of the amp.
• Tends to have elevated distortion at high frequencies due to the switching.
• Typically used for LF or MF applications from 400 to 3000 W per channel.

.Class D
While the other classes operate in a linear (analog amplifier) mode, Class D amplifiers run the output devices as switches (on or off) at an ultrasonic frequency to create a continuous series of square waves or pulses.

The input signal is made to modulate the pulse width, an operating principle called pulse width modulation. The pulses’ high frequencies are filtered out, and this filtered output (an analog signal) drives the loudspeaker.

• Very efficient (90-95 percent). That’s because the transistors are either on (with high current and no voltage) or off (with high voltage but no current). So there is very low power dissipation compared to linear amplifiers.
• Tends to be small and light because of less cooling and fewer output devices.
• Typically used for high-power car stereos and pro applications where heat, weight and size are considerations.

Class I
• Class I or BCA (Balanced Current Amplifier) design developed by Crown is an efficient system based on Class D switching amplifier technology. The amp needs only a little AC power to operate.
• According to the manufacturer, a BCA amp generates one-tenth the heat of conventional amplifiers so it can work with much less air movement. This reduces fan noise, heatsink size, filter maintenance, and failure due to heat, so the amp can be small and light.
• A BCA amp re-uses the energy returned from the speaker rather than dissipating it as heat or forcing the amp into premature current-limiting. This helps BCA models handle 2-ohm loads without shutting down.

I hope we’ve clarified some of the differences among power amplifiers. They are another component in the chain of audio equipment that we need to understand well.

Take the PSW Real World Gear Tour of the latest power amplifiers.

AES and Syn Aud Con member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.

 

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Thursday, July 14, 2011

RE/P Files: Carole King, Lou Adler, And Hank Cicalo In Session At A&M Records

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is an interesting look at the studio approach for a legendary artist. This article dates back to the September/October 1971 issue.

The door of Studio B at A&M in Hollywood sports a sing which reads, “CLOSED SESSION - NO ADMITTANCE PLEASE.”

Inside, Carole King, looking much more like a friend that the superstar she is these days, is recording her third album.

At this writing here last album, “Tapestry,” has been #1 on the charts for twenty weeks.

The sign on the door is indicative of a refreshing professionalism going on in the studio. The people in there are working. Doing what they enjoy, but working nonetheless.

There is little of the temperament that often acts as an excuse for lack of skill. In the first two days of these sessions, eight tracks were cut for the album.

The pace obviously is very quick. It is quick not because the people involved are rushing, but because they are not fooling around. They don’t need ten takes to get a vocal part or a guitar lick right. They know what they’re after, and they get it.

There are no secret techniques being used here. The success of the albums is based on a combination of experience and openness.

“You have to be open to new ideas. I’ve been around this business for seventeen years, and I could be set in my ways. But that’s wrong. I’ll try anything. I learn something every day.” - Hank Cicalo.

Hank is the engineer on this session, as he was on “Tapestry.”

His words are confirmed by Lou Adler, the producer: “I’ve only worked with two complete engineers. A lot of engineers are complete electronically but more important, there’s a disposition, a compatibility, and a knowledge and feeling for the kind of music we’re doing that’s necessary. Bones Howe and Hank are the only two complete engineers I know.”

What distinguishes Hank? His use of microphones, although not terribly strange, is certainly creative. He mikes the piano with a Sennheiser 421 D, inside and with the lid closed.

The piano is then enveloped with two covers. This is done for isolation’s sake, as Carol sings for the band during recording.

The bottom end of the piano is rolled off slightly to compensate for the boominess, caused by the piano being closed and covered.

On Carole’s vocal is an AKG 202 E. Although most of her vocals are done in their final version after the instrument tracks are down, some of the cuts on her last album were actually vocals she sang to the band while doing the instrument tracks.

Two AKG C 12s are used on the drums, one for both the toms and ride cymbal, and one overhead. An Electro-Voice RE-15 picks up the hi-hat, and another the snare.

The bass drum also uses an RE-15, placed deep inside near the head. The head is deadened by two heavy sandbags placed against it, giving a “tight” sound to the drum.

A lot of instruments and overdubs are used in this session, robbing the engineer of the luxury of several tracks for drums. But Hank is not that enamored with multi-track drum sounds anyway: “A lot of times I’m against that sort of thing. I’ve seen guys mix drums across 5 tracks of a 16-track, and the stereo effect was horrible.

“The guy got so wrapped up on the effect that it sound like an 18-foot set of drums. Who has an 18-foot set of drums? I would rather work to a tighter sound.”

That tighter sound happens on two tracks, one for a complete drum mix, and the other for bass drum alone.

This allows the bass guitar and the bass drum to be mixed against each other, independent of the total drum mix.

When the drums are limited, it is often just the tom. A touch of echo is sometimes added, especially if the part requires a slow rolling sound.

Bass guitar is taken both direct and with a microphone (Neumann U 87), with a ratio between the two of about 85/15. The direct feed is limited 2 to 5 dB. Electric guitar also employs a U 87, and acoustic guitar a Sony C-22.

Percussion and conga come though a U 87 feeding an Allison Gain Brain, resulting in a tight sound with a great deal of presence, as well as an even ratio between the high and low conga. Such a ratio is often quite difficult to obtain.

A Fender Rhodes utilizes an RE-15, and a Wurlitzer Electric Piano is taken direct.

The Wurlitzer seems to have some electrical noise problems though its own amplifier, but when taken direct and EQ’d properly, it produces a very warm sound.

The Rhodes sounds quite good through its own amplifier, resulting in its being miked and not taken direct.

Inside the control room, the producer has at his disposal a “playback panel” that allows him to mix independent of the engineer, and without affecting the recording.

Thus the producer can begin getting a perspective on a final mix while the recording is still in progress.

Lou, as producer, takes full advantage of this, a fact which certainly contributes to the success of his work. In his words: “From the time I start an album, I’m mixing. Every day and every night I’m always thinking about a mix. Sometimes in my sleep I’ll hear the machines rewinding.

But I’m always sure what I’m after. I’m always mixing for myself, but taking into consideration the likes and dislikes of the artist, which I’ve picked up during the session.

“If Carole says, ‘Can you turn the bongos down?’ while she’s listening to a playback, I remember it when I get to the mixdown. All those things are programmed in my head.

“Recording is important. I do that more than anything else in my life. I work more than I sleep. I work more than I socialize. But it’s a complete enjoyment when I do it.

“I like to get the best sound out of an artist. I don’t have my own sound. I think it’s entirely possible that a person could play all of my albums and not identify them as mine.”

Lou is in control of the session from the time it starts. He feels that as long as his is open-minded, and the artist knows he can be communicated with, his control is both accepted and appreciated.

The sessions are closed for several reasons. The fewer people there are around, the more work gets down. And the fewer people are around, the less confusion there is for the artist.

Lou doesn’t not like anyone standing behind the console: “An artist should always have one person to look to when they have a question. If they say, ‘What do you think’‘, and there are four different expressions, they have no idea where they are.

“They should look to me… but if there’s a person in the booth, and he’s happy just to be there, and the artist comes into the room, sees the person beaming, and I say, ‘We’d like to do it again,’ it’s confusing.”

The music ranges from ballads to rollicking rock and roll. The musicians and the atmosphere are cheerful. The musicians are not sidemen; they come with Carole. They have to be interested and involved in the music. Otherwise, they are not on the next date.

The arrangements are written by Carole, as well as being made up as the session rolls along.

Every number seems to “cook,” in large part due to the closeness of the people involved, and the fact that Carol sings during the recording of instrument tracks.

The cheerfulness is in part maintained by the unwillingness of the engineer, Hank, to put equipment problems on the shoulders of the musicians.

Rather than tell a bass player that his amp sounds bad, or that there is something wrong with his sound, he’ll explore every avenue open to him and try to solved the problem for the musician before even mentioning it.

Hank dislikes “button freaks” who feel the need to constantly prove that they are aware of everything happening in the studio.

Usually, if a musician plays a bad note, he’s more than aware of it. Jumping on him immediately and telling him so destroys the atmosphere. He feels it’s wiser to let the track run and retake that instrument later.

Sweetening is Lou Adler’s responsibility, but his decisions in that realm are a result of the artist’s music, rather than his own likes and dislikes. The sound of the final product is the artist’s.

Still, these things seem mostly to go unspoken. “There are no confrontations as far as sweetening goes,” he says. “If that happened, it would be time for us to go our separate ways.”

Overdubbing goes just as swiftly as the basic tracks. Once again, experience and openness seem to be the key.

Lou works like he knows a lot about it, and his track record certainly confirms this conclusion: “I was at the beginning of independent production, where most of the rules just came out of trying. I’ve learned a lot about overdubbing, especially when it comes to vocals. The training I had with The Mamas & the Papas you can’t buy. There hadn’t ever been any vocal groups with the amount of counter melodies that John Phillips had running through his head.”

Mixing requires as much, if not more, skill than overdubbing. It is interesting to note that in the midst of all the discussions these days about proper monitoring, using several systems for listening, and several mixes before choosing a final one, Lou Adler is remarkably unconcerned with the difficulties pointed out by many others.

“I mix by the speaker I’m listening to,” he notes. “If I listened to more than one speaker system, I’d go crazy. Whatever speaker it is, I’m mixing off of that speaker. I mixed Carole’s albums on small speakers.

“Mixes are very personal things, the most personal part of a producer’s role in recording. How could I do several final mixes, and choose one? You can only mix your best possible mix. It’s like saying, “now I’ll make a bad mix’.”

A good mix only comes from good tracks. In Hank’s words, “I have that saying, we’ll fix it later’. You can’t fix it later. You can touch up, but the basic stuff you have to get up front, or it’s never going to sound right.

“I never like to do things that really lock me in. If I compress, limit, or whatever, I’m always careful about doing it to a degree.

“You have to be open to new ideas. Some engineers aren’t, and that’s a hassle. Some guys have got one set up and they’re not going to change it. They’ve got to be insecure.

“For instance, we don’t have many leakage problems so we don’t need a Kepex for that, but we do use it for effects. You can get a tremelo sound off of it by keying it with an oscillator. Have the oscillator at five cycles, which is inaudible. By putting an organ though it, and beating the music against it, you get a very unusual tremelo effect.”

In making those good tracks, the choice of mikes is up to Hank. Limiting and compressing usually happens without even a request from Lou.

All of these things give testimony to an easy rapport which exists during these sessions.

FIRST COMES THE ARTIST, THEN THE PRODUCER, THEN THE ENGINEER. IT’S GOT TO BE A MARRIAGE OF ALL THESE PEOPLE.”

This triangle is more than just Hank’s words. It is working.

There are a lot of pros in this business, and a lot of perfectionists. Carole King, Lou Adler, and Hank Cicalo are certainly among them. But they have the added beauty of not only being good, but being easy about it as well.

A sign on the console reads, “Anything that is not quite right, is wrong.”

The philosophy is not wholly unique. What is more unique is the lack of anxiety and tension that normally accompanies so absolute an approach. If something is not quite right, nobody gets upset. They just change it and make it better.

Maybe they’ve got something there.

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Thursday, July 07, 2011

RE/P Files: Signal Feed Techniques For Recording Electronic Musical Instruments

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is an interesting look back at techniques for recording electronic instruments. This article dates back to April / May of 1970. (Volume 1, Number 1). The text is presented unaltered, along with all original graphics. A pdf for a print-out of the original article is located on page 2.

There are variations of three basic methods which seem to satisfy most requirements…that is, those requirements which don’t demand instant audio annihilation…for getting a signal out of an electronic musical instrument and its amplifier.

Assuming that the sound to be picked-up is generated by a fundamental electronic instrument, say, an electrified guitar, one without built-in reverberation, wah-wah or the like.

Then, there is no particular problem in coming directly off of the magnetic pick-up on the instrument into a mult-jack, with the dual feeds then going, on the one hand, to the guitar amplifier, while the other line, then, goes to the microphone input of the mixing console through an impedance matching transformer . . . Direct Box. (See figure 1)

The obvious advantages, here, are that the player has complete monitoring capability through his own amplifier in the studio, while the mixing engineer retains complete control of the output volume of the instrument in the control room.

Electronic instruments with built-in special effects; the fuzz tones, wah-wahs, reverbs, etc. are picked up directly in two additional ways.

If the amplifier being used by the musician in the studio has either a line-output or a pre-amp output the mult-jack approach is still where the process starts. 

One line from the jack goes out through the impedance matching transformer (sometimes called a bridging transformer) straight to the microphone input of the control console. 

The mult feed from the jack goes back into the amplifier.

As in the previous example, the player still has complete liberty to monitor his own performance at any volume level in the studio. The use of any of the special effects originating in the instrument or the amplifier remains the choice of the artist. The engineer, on the other side of the glass, still has absolute control of the volume of the sound being recorded.

Although less desirable from the control-of-volume point of view of the engineer, the third method of direct pick-up is used because of its simplicity. This method looks pretty much the same as the immediately preceding set-up, except that a pair of clip leads are used to clip onto the voice coil of the amplifier speaker before going back into the bridging transformer and then on into the microphone input of the mixing console.

In this situation the player has the opportunity of “playing” with the amplifier volume controls, thus affecting the volume of sound fed to the mixer. To the degree that the performer might want to do this, the absolute control over the volume being fed to the tape machines is no longer vested completely in the engineer doing the mixing.

These techniques can be applied to almost every electronic instrument; electronic piano, electronic harpsichord, etc. In each case the signal must be fed through an isolating or bridging device (impedance matching device) into the mixing console, while at the same time allowing the musical signal to also get to the performer’s amplifier in the studio.

Direct signal pick-up eliminates distortion from both the amplifier and the speakers, which in musical instrument amplifiers are nowhere near the quality or balance of the studio monitoring system. Too, the recording system is not exposed to any extremely high sound power levels. Those remain safely isolated out in the studio.

Conventional Micing
Especially as it applies to ‘rock’, the biggest problem in picking-up an amplified instrument sound through conventional microphones is that the acoustical power coming out of the amp speakers can very easily overload the microphones.

However, in order to record the electronic instrument and its amplifier as faithfully as possible to the sound which the combination is putting out, using conventional micing methods would mean that the microphone must be placed only inches from the amp speakers.

Where this is attempted, the use of dynamic microphones is recommended because of their ability to withstand extreme sound pressures, of between 110 and 140 dB before ‘CO’

Still, there may be times when the producer/mixer might want the best of both the direct and conventionally miced sound.

If there are enough inputs in the console, then both the microphone line and the one coming in from the ‘Direct Box’ (bridging device) can be run into separate ‘pots’ for recording on the common track.

As the engineer seeks the brilliance and clarity of the instrument sound fed direct, or the sound of the instrument plus the ambient of the room (studio) as the sound comes from the conventional micing procedure, he can switch from input to input, or blend both of the signals together.

The Direct Box
The primary impedance of the matching transformer should, of course, be high enough so that it does not disturb the match of the output of the magnetic pick-up from the instrument . . . and, so that it attenuates the high end, or doesn’t drop the level too much ... so that the signal comes out of the ‘Direct Box’ at approximately microphone level.

It should be a nominal impedance of, say, 30,000 ohms to 50,000 ohms. The primary impedance should be high enough so that it doesn’t disturb or load the instrument’s magnetic pick-up and delivers enough signal at the console for control.

The matching transformer should be mounted in a small, well-shielded box. Careful attention should be given to ‘grounds’ or shielding of both input and output cables. Appropriate connectors on each cable should be compatible with the output of the magnetic pickup-on the instrument, and the input connector to the mixing console.

 
At the time of publication, William Robinson was of Engineering Director Sunset Sound Recorders Hollywood, California

 
Downloadable Media
Original Article (pdf)

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Thursday, June 30, 2011

Real World Gear: Subwoofers - All Hail The Maestros Of The “Boom Boom”

It’s quite useful to be up to speed on the plentiful assortment of subwoofer options – you never know what the next gig may bring

It’s pretty obvious to anybody that listens to popular music that bass is featured prominently, but most styles of music also have important low-frequency content.

The smooth glissando of a plucked upright bass in a jazz quartet, the dramatic staccato roll of a tympani in an orchestra, and the melodic rhythm of a piano player’s left hand are all examples of this.

Accurately reproducing these and other low notes through a sound reinforcement system often requires at least one subwoofer.

Even when I’m doing smaller speech-heavy gigs, I’ll bring a sub or two along because the “speakers on a stick” can’t produce much below 100 Hz.

They’re well suited for the “talking head” portion of the event, but not music playback. Modern audiences expect to hear those bottom octaves, and I certainly don’t want to disappoint.

There’s an embarrassment of riches when it comes to subwoofer choices.

Do you want an active subwoofer with built-in amplification and processing, or are you looking for a passive one that needs an external amp? Seeking a small portable box that one person can easily transport, or will there be extra hands at the gig to help you move and position large cabinets?

What frequency range do you need to cover? What box type will work best in your application (horn loaded, front loaded, vented, band-pass, etc.)?

Once you’ve narrowed down the choice of box, there are still more questions. Will you run the subs in line with the mains or from a separate aux send feed?

Is the subwoofer going to be flown or ground-stacked on the floor? How are you going to position or array the subwoofers (left/right, center cluster, end fire, cardioid, etc.)?

The options are almost endless, and there are no true “best” answers to these questions. It depends on the needs and requirements of the gig, as well as your own preferences. In my company, I keep it relatively simple, stocking three different types of subs, mostly defined by their footprint: small, medium and large.

Some corporate gigs need extra low end, but the event planner doesn’t want to see those “ugly black boxes.”

So I use my small subs and hide them under the stage, or conceal them behind things like planters in the corner of the room.

The mid-sized single-woofer cabinets are my go-to subs, used for mains, side fills, delays, and even low-end support in the drum fill. Add a pole and they can elevate my compact loudspeakers to create a clean-looking full-range ground-supported rig.

When there’s need to “move a lot of air” (one of the best reasons for being in this business!), my big guns get the call and are rolled into the truck.

This month’s Real World Gear look at numerous current subwoofer models reflects the reality that the types and choices are vast and diverse.

It’s quite useful to be up to speed on the plentiful assortment of subwoofer options – you never know what the next gig may bring.

Enjoy our Real World Gear Tour of a wide range of subwoofers from across the pro audio industry.

Craig Leerman is senior consulting editor for Live Sound International, and has worked in professional audio for more than 25 years. He is also the owner of Tech Works, a regional production company based in Las Vegas that focuses on live corporate events.

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Tuesday, June 21, 2011

Real World Gear: The Latest Developments In Modern Power Amplifiers

Meaningful power comparisons are best made with both channels driven to reflect real-world situations

Most modern power amplifier designs have abandoned traditional “linear” power transformer-based supplies in favor of digital switching supplies, shedding about half their weight in the process, getting into the range of 10 kilos or 22 pounds.

Nearly all amps are two “rack spaces” high for a total of 3.5 inches in a standard 19-inch wide equipment rack, with depths ranging from roughly 10 to 20 inches.

Various circuit designs have emerged over the years. Class AB designs provide low distortion at medium efficiency for high fidelity at moderate power levels and are still popular choices for compression drivers.

However, AB uses a full-power DC supply at all levels, generating substantial heat at less than full output. Class D designs employ Pulse Width Modulation, using the input audio signal to switch the output devices at ultrasonic frequencies and then filter those frequencies with a low-pass filter. They’re very efficient, with a slight cost to frequency response, distortion and damping factor.

Class G uses a second output stage that’s turned on when louder input signals require it, while Class H uses a single stage whose DC power supply is controlled by the input audio signal.

Several other classes have been introduced by manufacturers that are proprietary modifications of Class D that address its shortcomings, combining high efficiency with high audio performance.

Meaningful power comparisons are best made with both channels driven to reflect real-world situations. Four-ohm operation is preferred, though most modern designs provide stable operation with two-ohm loads as well.

Subwoofer applications usually require many thousands of watts to drive today’s high-power 18- and 21-inch voicecoils. A rule of thumb is to match amps with at least twice the continuous power rating of the loudspeaker loads they’re driving.

The ability of an amplifier to control woofer cone inertia is referenced by a numerical factor which is the ratio of the loudspeaker load impedance divided by the amplifier’s output impedance. The lower an amp’s output impedance, the better it can control the cone’s “ringing” or “time smearing.” Higher damping factors are associated with tight bass.

Amplifier sensitivity, quoted in either volts or decibels, refers to the input voltage needed to reach full power. Most amps provide the user with two or more choices, allowing it to be matched with various sources or other amplifiers. Lower voltage or decibel values make the amp seem louder, though it will still clip at the same level.

For amplifiers with a 0.775 V setting, this corresponds to a 0 dBu level. 1.4 volts relates to a +4 dBu level. Many amps have a 26 dB “fixed gain” position, which often provides the lowest gain, and usually works well with output levels of 2.5 volts and higher (+10 dBu), allowing systems to be mixed across models and even manufacturers.

Frequently misunderstood is that amplifier front panel rotary controls don’t change the available power, but rather simply reduce the input sensitivity. Other common rear-panel switch functions include parallel or bridged-mono input, high-pass filters and limiters.

Neutrik Speakon output connectors are a nearly standard feature, and not just on tour class amplifiers. It’s standard to find a pair of Speakon connectors, plus sometimes a third across both channels, for bridged mode applications.

Some manufacturers have eliminated “five-way” binding post outputs, and models with Euroblock connectors – a modular terminal block connector favored by installers for their ease of use – are becoming more common on amps and other types of audio equipment that can be networked.

Take our Real World Gear Photo Gallery Tour of the latest available power amplifiers for sound reinforcement applications.

Mark Frink is Associate Editor for Live Sound International.

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Monday, June 20, 2011

Tech Talk: Building Directional Subwoofer Arrays

Working toward consistency.

Directional subwoofers are one more tool that can be used by sound system designers in their quest to achieve consistent sound throughout the intended listening area.

When using traditional, more or less omni-directional bass reflex (a.k.a., “vented,” “ported,” or “front-loaded”) subs arranged left and right of a stage, there is a build-up or “power alley” created in the center, where the energy from each source location shows up at the same time, with no phase difference, and sums quite nicely.

Moving left and right off of the center line, this area of addition is followed by alleys of cancellation.

Wavelengths of 40 to 100 Hz are roughly 11 to 23 feet long. At any frequency in this range, as you move away from the center line and change the path length difference between the two sources by half a wavelength (about 5.5 to 11.5 feet) there will be a cancellation, with higher frequency “nulls” encountered first.

To alleviate this there are three methods that have been employed: line arrays of subs, end-fired sub arrays, and cardioid subs, which are sometimes combined.

Line Arrays
Lines of subwoofers are one application of what Harry Olson discussed in the 1957 text Harry F. Olson, Acoustical Engineering, when he described a straight line source; using omni-directional elements, in a line, all reproducing the same signal, with relative close spacing compared to the wavelength, pattern control can be achieved.

Imagine a row of subs is assembled across the front of a stage. If it’s longer than the wavelength of the lowest frequency for which pattern control is desired (25 Hz is about 45 feet) and if the elements are close enough to one another, within two-thirds of a wavelength of the highest frequency produced (100 Hz is about 11 feet, so 2/3 is about 7 feet), cancellation at the ends of the line and addition in front of the array (and behind the array!) will be achieved.

Observed from the audience area, from one end of the line to the other, enough of the energy from each of the elements of the array arrives within +/- 120 degrees, at about the same level and sums.

Observed from the end of the array, enough energy from each of the elements arrives enough out of time but at similar enough level, causing destructive interference and level loss.

The use of a line array (yep, that’s what it is) of subwoofers can avoid horizontal differences in frequency response and deliver more energy to the audience area, while avoiding those nasty side wall reflections at lower frequencies.

Further, maximizing spacing can reduce the level differences from the front to back. In the interest of making sound where the audience is and not making noise where they are not, this is one option.

Remember, though, that the energy is the same in front and behind the array.

These arrays can also be assembled vertically, though space between the elements is not easily achieved with most rigging systems, so they are generally closely spaced arrays.

In amphitheater and arena situations where coverage to the sides is desirable, incrementally delay-tapering the horizontal array - so that moving away from center, each sub is slightly later than the one before it - can spread the coverage out towards the sides.

End-Fired Arrays
The end-fired array can be made up of two or more subs, spaced closely together, one facing the rear of the next, in a row along the “z-axis,” facing the audience and the direction of coverage.

Yes folks; it looks like it won’t sound “right.” Each cabinet needs its own drive line because we are going to incrementally delay all but the rear-most.

The rear, upstage, sub is delay time zero.

Moving towards the audience, each sub needs delay added corresponding to its distance from “sub zero.”

Let’s say the spacing is 3.5 feet: the delay time would be 3.1 ms (speed of sound = 0.9 ms/ft) for the next element.

The end-fired array produces gain in front of the array because the energy from each of the elements arrives in time at all frequencies being reproduced.

Cancellation behind the array is the summation of the energy produced by each source that is out of time and arrives at almost the same level.

There are a number of dips in frequency response based on the number of signals that have 180 degrees of phase difference. The level difference between front and rear is about 18 dB with a four-element array.

Cardioid Arrays
A few manufactures make multi-driver, single-cabinet cardioid enclosures, but they can be created with simple arrays of two or more cabinets.

The physical arrangement can be one of two options, both speakers facing the audience, one upstage of the other, lined up on the ‘z-axis,’ or one sub oriented facing backwards next to one or more facing forward. Again, people will question the appearance.

When both subs are facing the audience, one upstage of the other, delay and a polarity flip are applied to the signal going to the rear speaker.

In the rear, the energy from both loudspeakers arrives in time, at almost the same level, but with reversed polarity, resulting in broadband destructive interference and reduced level. In front of the array, the two signals arrive with polarity different and out of time.

This is a little tricky, but the first dip in the comb filter in this example is going to be at 160 Hz, out of band of the sub. If the spacing between the subs is 3.5 feet and the delay time is 3.1 ms, the two signals arrive 7 feet apart in front.

The wavelength of 160 Hz is 7 feet. With the polarity flip, the first dip of the comb filter will be at 160 Hz, not 80 Hz. The two signals in front are also at about the same level, so the dip will be significant.

The cardioid arrangement using forward and rearward facing subs can be assembled vertically or horizontally, subs stacked one on top of another or laid side by side, in a line, some facing the audience and one or more facing backwards.

Talk about looking like it won’t sound good. Behind the array, the output of the front and rear facing elements of the array need to match in time and be very close in level, but polarity backward to create cancellation behind the array.

A polarity flip and delay of the rear facing loudspeakers achieves this.

Determining the number of forward and rearward facing elements depends on model and how much energy needs to be created behind the array to cancel the energy from the forward facing subs.

The delay time will vary too, depending on model, and dimensions of the array, both vertically and horizontally.

Measurement is needed to determine level and time relationships between the front and rear subs.

An FFT transfer function can quantify this accurately. In front of the array, the summation of the rear facing loudspeakers is out of time and polarity different from the energy being produced by the forward facing subs.

The problem in frequency response, that first dip in the comb filter, must be kept out-of-band, higher in frequency than the operational range of the subs.

Alternative Methods
A hybrid approach, combining cardioid pairs, arranged in a line across the front of the stage, results in cancellation left, right, and to the rear. Alternatively, combining end-fired arrays and line-arrays also achieves additional directional control.

Using a directional array left and right affords the opportunity to join -6 dB down points in the middle of the audience and minimize the interaction between the arrays by minimizing the area where they are level similar, moving quickly into isolation of one or the other arrays. This would lend itself to very wide audience areas, such as amphitheaters and festival sites.

Directional arrays are often misconceived, mis-assembled, or are faulty in their operation. They require a knowledgeable operator, good equipment, and proper implementation.

The benefits can be substantial and are sometimes worth the risks. Avoiding some reflections in rooms, decreasing the amount of low-frequency energy on the stage (turn the floor monitors down, folks), and making the coverage smoother in amplitude and frequency response in the audience area are the substantial benefits when considering the use of directional low-frequency arrays.

There are several critical factors of performance that must be considered when assembling these types of arrays. Control of low frequency directivity is only possible when using exceptionally linear systems, precision-manufactured to perform identically.

The relationship between individual components must be consistent. What is sent electrically to the array elements needs to be turned into acoustic energy, without distortion or changes in frequency response as signal level changes.

New Tools
Historically, directional low-frequency loudspeakers have been in existence for some time.

Meyer Sound developed the first commercially available design, the PSW-6, a dozen years ago.

The PSW-6 uses a four-channel amplifier and signal processing built-into an enclosure that houses dual 18- and 15-inch drivers facing the audience, plus two more 15-inch drivers mounted in its rear.

This self-powered subwoofer provided cardioid vertical and horizontal polar response, serving as a new tool in the challenge of designing sound systems.

It eliminated 15 to 20 dB of the energy from the rear that would have bounced around and arrived in the audience area late.

Another advantage was the ability to place these loudspeakers in front of large walls without having to consider boundary reflections.

These and others continue to be advantages over omni-directional designs.

The PSW-6 design was a result of field experiments using the SIM (Source Independent Measurement) FFT measurement platform, along with prediction results from Meyer Sound’s then new MAPP Online (Multipurpose Acoustical Prediction Program).

MAPP, among its many uses, has become a tool that many practitioners use to design low-frequency directional arrays. Users are able to apply signal processing, arrange elements, and observe the results graphically as a narrow-band pressure plots or as broad-band Virtual SIM transfer functions, all predicted from the interaction of measured data sets of real loudspeakers.

“Measure twice and pile it up once.” Let’s face it, moving subs around in a parking lot is a lot of work and requires a substantial investment of time and effort, plus there’s tinkering with signal processing and measurement, as well as additional DSP and multiple drive lines.

On the other hand, moving subwoofers around on a computer screen is a two-finger event, and without the need for real subs, signal processing, and measurement platforms, a real time and money saver.

Not having to build and measure subwoofer arrays in the physical world as a first step has allowed users to design arrays that they might not have spent the time to experiment with in real life.

Steve Bush is a technical support representative for Meyer Sound.

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Wednesday, June 08, 2011

Digital Audio Transport: Versatile, Flexible, Scalable – And Plenty Of Options

While analog wiring is still a viable solution for many applications, digital audio, or more specifically digital transport systems, are coming of age

Way back when I started in professional audio, hooking up a sound reinforcement system was a pretty straightforward affair.

Signals ran one way, point-to-point, and every signal had its own cable.

Larger systems included a multi-channel snake, which allowed us to position the mixing board away from the stage and hear what we were actually putting through the system.

Over the years I’ve watched as analog systems have progressed from simple cables to complex routing solutions.

These interconnect systems can feature quick cable disconnects, splitters, ground lifts, sub stage snakes, and even stage drop boxes that have built-in direct boxes (DIs).

But no matter how complex the analog system has become, it still follows the old rule that signals run one way, point-to-point, and every signal has its own cable - except now those cables might be part of a larger multi-conductor bundle.

When digital technology first came to the live audio scene, it was used mainly for signal processing and effects, eventually making its way into control and transport systems. These early systems suffered from a few problems – there was a relatively high initial investment required, as well as issues with latency.

Until fairly recently, the latency was far too great for some of these products to be used in a live setting. Instead of using dedicated wires for each signal channel, digital signals share a common fiber optic, coax, or Ethernet Cat 5 and Cat 6 cable.

The days of needing three or four stagehands to wrestle a large, heavy multicore snake to connect a larger system will eventually come to a close. One stagehand can easily carry a reel with 300 feet of digital cable.

Scalable Solutions
Today many manufacturers offer digital solutions for signal processing, distribution, monitoring and control.

Some of these are simple point-to-point multichannel audio transfer systems, but some are scaleable network solutions with a main “trunk” and several subsets as the “branches.”

Not everything is as rosy as it may seem. There are still a few issues to be sorted out, among them the fact that networking solutions from various manufacturers remain largely proprietary.

There is no “standard” as of yet, although AVB (Audio/Video Bridging) may help foster a solution.

AVB could do for audio what MIDI did for instruments, or what DMX has done for lighting, and that is basically to allow equipment from various manufacturers to communicate with each other.

It’s something that we really need in the audio world. In the meantime, several manufacturers have adopted protocols that allow their own gear to interface.

These fall into a few groups: Ethernet Layers 1, 2 and 3, and MADI (Multichannel Audio Digital Interface).

Layer 1 protocols use Ethernet cables and switches but primarily use a proprietary media access control instead of the native Ethernet MAC . Some of the systems using Layer 1 include Riedel Rocknet, Aviom A-Net, and Roland REAC.

Layer 2 protocols process the audio by using the standard Ethernet packet system. Two of the most well known digital transfer systems, CobraNet and Ethersound, use Layer 2, and it’s been adopted by numerous manufacturers.

Layer 3 protocols use IP packets to transmit the audio data over Ethernet cables. QSC Q-LAN, Axia Livewire, and Audinate Dante are a few of the protocols that use Layer 3.

MADI (also known as AE S 10) fosters serial digital transmission over coaxial cable or fiber optic lines of 28, 56, or 64 channels, with sampling rates of up to 96 kHz and resolution of up to 24 bits per channel. We see an increasing number of digital consoles offering MADI interfaces and cards.

Many Advantages
In the most simplistic form, a digital transport system takes audio signals from one place and sends them to another, just as analog does.

But the more you look, digital systems offer a myriad of advantages over hard-wired, point-to-point lines.

Digital consoles can accept digital inputs directly, either by way of their own proprietary digital network or by using cards that allow a user to choose different networks.

Built-in or third-party A/D converters at the stage replace the need to run analog signals down a long copper cable out to the console, eliminating potential problems like ground loops and line loss.

Digital signals can be sent anywhere along the network using switches, allowing multiple sends for monitor consoles, remote recording, broadcast feeds, sends to intercom systems or remote monitoring.

Some systems allow these signals to be sent to personal monitoring systems that allow a performer to tailor their own monitor mix using multiple source inputs.

Recording devices can be easily interfaced into many digital transfer systems, allowing convenient multi-track recordings to be made of every show, and allowing the playback of the multiple tracks to be used for “virtual sound checks.”

Some transport systems even allow both audio and video signals to be transmitted down a single cable, simplifying cable runs on corporate events to overflow rooms, breakout rooms, and feeds to and from video systems.

While analog wiring is still a viable solution for many applications, digital audio, or more specifically digital transport systems, are coming of age.

Let’s have a look at several leading system options - click here and take our Photo Gallery tour to check them out.

Craig Leerman is senior consulting editor for Live Sound International, and has worked in professional audio for more than 25 years. He is also the owner of Tech Works, a regional production company based in Las Vegas that focuses on live corporate events.

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Tuesday, June 07, 2011

The Current State of Near-Field Studio Monitoring Systems, And Recent Developments

Important considerations when selecting a new monitoring system, and a walk through some of the currently available options

It’s been said the world over to engineers with mixing chops that far exceed my own “We’ll just fix it in the mix.”

That’s not only a poor idea, but it’s unlikely it will ever be accomplished.

However, if the control room monitoring environment isn’t accurate, that really isn’t going to happen!

According to Bob Katz, the monitor system permits hearing inner details in the music that otherwise might cause problems for the end listener. (Bob Katz, Mastering Audio: The Art and The Science, Oxford: Focal Press 2002)

In other words, your monitors should be a transparent and critical window through which you can asses your mix.

Whether you’re setting up a basement project studio for the very first time or upgrading your room in the hopes of one day becoming the next Bob Ludwig, ensuring an accurate and critical listening environment is crucial. I’m here to help.

First we’ll take a look at important considerations when selecting a new monitoring system, and then we’ll walk through some of the currently available options for near-field monitors.

Where To Start?
OK, so, you know you need a pair of monitors, but where to begin?

The first important step is assessing your needs:
—What style (or styles) of music do you work with?
—What do you dislike about your current monitors?
—Is space a factor, or can you go big?
—Active or Passive? Preference?
—What is your budget?

These are all important factors in beginning to shop for monitors. Focusing a moment on size as the other criteria are fairly personal, I’m going to assume that you’re in the market for near-field monitors unless you also have a room large enough for a fairly serious large frame console.

If you happen to be that lucky, well, we’ll be getting to far-fields in a few months. Until then, happy mixing!

Anyway, you may be wondering: what exactly are near-field monitors?

Simply put, the term near-field refers to the placement of small to medium sized loudspeakers within the critical distance. (David Miles Huber, Modern Recording Techniques Fifth Edition, Oxford: Focal Press 1997)

Some feel that the term near-field is misleading and prefer close-field. No matter what term you use, it’s going to refer to a pair of “bookshelf” style speakers which can range in size from 6 inches to 10 inches. (Roey Izhaki, Mixing Audio: Concepts Practices and Tools, Oxford: Focal Press 2008)

OK, Now What?
Well, certain aspects of picking new monitors are, as I said before, very personal and can only be determined after some careful listening at another local studio or music retailer.

For instance, while monitors need to be transparent, if you mix Hip Hop you may still have different monitor preferences than an engineer who mixes predominantly Classical music.

Similarly, only you can know if there was something irksome about your last monitoring environment.

However, while certain aspects of picking monitors are very personal, other aspects are less so. For instance, active or passive?

Personally, I tend to prefer passive loudspeakers, as they remove the potentially noisy amplifier electronics from the monitor enclosure.

On the other hand, using passive loudspeakers does mean that I’m saddled with buying both speakers and an amplifier, which can often be too complex a setup for a first time buyer.

Your decision of active vs. passive could, on the other hand, be decided for you by the manufacturer should you pick an active or passive only monitor.

Can I Pick A Pair Already?
Absolutely! I’ve compiled for you a list of just some of my favorites, ranging in price from the ridiculously affordable to the amazingly expensive.

What do they all have in common? Every single option offers pristine audio quality and accuracy that is sure to provide you a transparent and critical window through which you can asses your mix.

So, what are you waiting for? Get picking! (Take Kyle’s Gallery Tour of a wide range of recent near-field studio monitors.)

One Final Thought
Also be sure to talk with friends and colleagues, listen to as many different monitors as possible, and in the end remember that if the prospective monitors are showcasing flaws in your mix, that’s OK.

After all, isn’t that what you’re buying them to do?

Kyle P. Snyder is an audio engineer with innumerable credits in the public and private sector, and he also recently created the Technologist blog here on ProSoundWeb.

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Thursday, June 02, 2011

Real World Gear:  A Look At The Latest Power Amplifiers

Amplifiers have never enjoyed a reputation for being particularly sexy, but now we see a different story

Until fairly recently, the audio power amplifier was almost as synonymous for excessive heat and weight as for its primary application. This reputation was the result of early inefficient output stages.

Early designs dissipated most of the power drawn from the wall as heat, which then had to be sucked away from internal electronics with large metal heat sinks.

Amplifiers have never enjoyed a reputation for being particularly sexy, being largely black boxes sporting a couple of big rotary dials and an LED or two, and mounted in racks that are usually concealed from audience view.

But now we see a different story. The brutes of the past have been tamed, exceedingly efficient and delivering far more audio power with far less heat and weight.

Amplifier design topology has evolved from Class A to several advanced topologies:

Class D uses an on-off switching method for its transistors called Pulse Width Modulation. Because its output devices are either on or off, the efficiency of the amplifier it greatly increased, as very little power is lost to heat. Class D designs do this without jeopardizing the integrity of the audio waveform by switching positive and negative output transistors on and off many times per waveform cycle.

It’s analog but similar in theory to digital sampling, where a 44.1 kHz sampling rate is used to accurately capture a 20 kHz signal. This rapid switching creates a square wave that is then low-pass filtered to recreate the audio waveform.

Class G takes a post-AB Class signal and switches it between two power supplies rather than switching between positive and negative transistors in the amplifier’s output stage. One supply is for softer output levels and the other for louder. Efficiency improves as the power amplifier draws full power from the wall when higher a level of amplification is needed.

Class H also works on the two-power supply output concept but is a bit more precise in terms of what AC power it draws, and when. Instead of simply switching between lower and higher voltage power supplies, the second power supply’s voltage level is controlled via the audio input signal. If the signal increases, so does the power supply’s voltage.

Class I, a topology design patented by Crown, is an advancement of the Class D switch-mode design. The risk in the purely Class D design is the potential for distortion in the moment that one transistor turns off and another turns on. If both transistors are momentarily off during the switch, audible distortion results. If both transistors are momentarily on during the switch, excess current can cause physical damage to the circuitry.

The big factor in the continued evolution of the power amplifier is multitasking, with most premium units now outfitted some kind of signal processing control, external or internal, providing EQ, filter, delay, compression, limiting, routing and monitoring in real time.

Let’s take a look at a wide range of recently debuted amplifier models and technologies. (Click on slideshow directly below.)

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Monday, May 16, 2011

Real World Gear: The Latest In Medium Format Line Arrays

The advantages are many.

It was acoustical pioneer Harry Olson who first demonstrated the line array effect of the narrowing of the beam with increasing frequency.

This approach was published in Olson’s book, “Acoustical Engineering,” in 1957, and these concepts were used to develop column loudspeakers comprised of vertically aligned drivers in a single enclosure that teamed up to produce mid-range output in a wide horizontal and narrow vertical pattern.

These column loudspeakers proved popular in the 1960s and 1970s, primarily for spoken word and public address applications in reverberant spaces.

Not much significant happened with the technology until Christian Heil and L-Acoustics took the pro audio market in the mid-1990s by storm with the introduction of V-DOSC , showing the concert sound world that more level and smoother frequency response can come from fewer drivers in a line array.

As frequent Live Sound contributor John Murray noted a few years ago, “After everyone realized that for a given listening area, the drivers have no destructive interference in the horizontal plane and combine mostly in phase in the vertical plane, the race was on.”

Today, medium and small format represent the largest market for line arrays, whether in performing arts centers, worship facilities, ballrooms, convention centers or auditoriums.

We define “medium format” line arrays as those with 10-inch low frequency drivers, and we’ve created a “small format” category for those with 8-inch LF drivers, and will be covering those later this year.

When the modern version of line arrays hit the market more than a decade ago, the vast majority were of the large format variety (12-inch and larger LF drivers), with subsequent introductions primarily defined by ever-decreasing footprints.

Compact enclosures are not only less expensive, but also weigh less and can bend more rapidly without breaking their coupling, due to the smaller diameter of their woofers. The physics of coupling dictates a limit to the angle from one enclosure to the next, beyond which beaming and spotty coverage occurs.

While a line array with 15-inch LF drivers has a limit of about 5 degrees, enclosures based on 10-inch cones can bend by 10 degrees from one cabinet to the next. Lesser scaled line arrays can therefore provide a greater angle of vertical coverage in a shorter height, especially important in smaller venues.

In addition, line arrays lose pattern control at frequencies whose wavelengths are longer than the array’s height.

To provide pattern control down to 100 Hz, for example, an array must be 11 feet tall and, with a typical cabinet height of a foot or less, control down to 100 Hz requires 11 or more cabinets in an array.

Hanging half a line array provides omni-directional low frequency coverage that creates both a puddle of mud on stage and thin-sounding response at the back of the listening area, often where the mix position is located.

Suffice to say the advantages are many when talking about line arrays of a reduced scale for numerous applications. Enjoy this Real World Gear tour of the latest medium format models.

Enjoy our PSW Photo Gallery Tour look at 14 medium-format options.

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Friday, April 29, 2011

Historic Costa Rican Theatre Undergoes A Modern Sonic Transformation

Inside the new sound reinforcement system recently implemented at National Theatre of Costa Rica

The National Theatre of Costa Rica (Teatro Nacional de Costa Rica) is one of the country’s most beloved buildings, brimming with cultured operas, concerts, and dance, and it now features a new high-end sound reinforcement system carefully integrated within its historic infrastructure.

Located in the central section of San Jose, the National Theatre opened to the public in 1897 with a performance of Johann Wolfgang von Goethe’s Faust.

Since then it has hosted many of the great theatre companies of the world, as well as famous musicians, artists and classical orchestras, along with attaining national monument status.

Designed by architects from Italy, Spain and France, the interior is filled with incredible art and souring ceilings, highlighted by a three-story horseshoe-shaped auditorium that seats 1,040 people.

It offers a classic “opera house” configuration, with a main floor of seating surrounded by second floor private viewing boxes, balcony, and gallery (second balcony).

The wide front stage is framed by a proscenium, with orchestra seating beginning immediately in front of the stage.

The theatre had been outfitted with a small sound reinforcement system several years ago, but it wasn’t designed to support the needs of modern, non-operatic performances, and it also supplied rather poor coverage.

National Theatre Technical Director Claudio Schifani began discussions about an upgraded system with Virgilio Azofeifa and Elias Arias of RSTV, a leading sound company based in San Jose.

Small Profile
The primary challenge facing the system design and installation team was integrating the new, full-range system within the architecture and aesthetic of the auditorium, while attaining the desired coverage and full-range performance to all seating areas.

With this in mind, the team favored an approach utilizing compact line arrays with a small physical footprint flown to the left and right of the proscenium to deliver the majority of coverage, supplemented by strategic application of compact loudspeakers for specific areas.

The Adamson Systems Metrix Series of compact line arrays held promise in meeting these needs, with Azofeifa noting, “Metrix cabinets were chosen as they offer most power and best coverage of all cabinets in their category, in addition to not creating an undesirable image as an install in a theatre where minimal visual distraction is required on top of a harmonic aesthetic.”

Daniel Fernandez of the Adamson Latin America technical support group was brought aboard to lend his expertise to the design process.

Specifically, he proposed the use of Metrix-I (installation version) line array modules for several reasons, including their purpose-built installation rigging system that also helps save on costs, in addition to providing additional flexibility by offering two different vertical coverage patterns.

“The flexibility of different vertical coverage patterns allowed us to obtain the ideal coverage pattern with a minimal number of boxes, thus adapting the design to the limitations of the architectural aspects of the venue,” Fernandez explains.

Each 2-way Metrix cabinet incorporates an Adamson-designed 8.5-inch Kevlar neodymium mid-low frequency cone driver optimized for the enclosure, as well as a 1.4-inch high-frequency compression driver on a proprietary wave-shaping chamber. Horizontal coverage pattern is 120 degrees.

The aforementioned vertical pattern is available in either 5 degrees (Metrix) or 15 degrees (Metrix W - Wide Angle Vertical Enclosure) in a total cabinet footprint measuring 8.5 inches high, 21.2 inches wide and 16 inches deep, and weighing just over 40 pounds.

Modeling Insight
The design took shape with left and right arrays comprised of 10 Metrix modules, with the tighter vertical dispersion models at the top providing extended coverage to more remote areas, transitioning to the wider vertical dispersion models to cover the main seating areas.

To further fine-tune and optimize the design, Fernandez utilized Adamson Shooter array modeling and configuration software, which provides 2-dimensional vertical and horizontal calculations and 3-dimensional SPL calculations, in addition to a mechanical view of the array structures.

“We needed to be very careful and accurate with the splaying of the arrays, and this is where help from Shooter was invaluable.”

“The software provided vertical and horizontal predictions in several frequencies and at different locations,” Fernandez says, who was supported on the project by Adamson’s Benoit Cabot.

“On top of that,” he continues, “we had to carefully account for some limitations with the array placement, since we were not allowed to open holes in the main structure of the proscenium in flying the arrays.”

The system team created and fabricated a special “arm” in order to fly the arrays from an upper support bean so that they did not touch the proscenium and surround structures.

Further, the Metrix cabinets were outfitted with Enclosed Installation Rigging (EIR) system, providing all of the flying/aiming precision and safe structural support found in the rigging for touring Metrix enclosures, but at a lower cost.

Four Metrix Sub compact subwoofers, each with dual 15-inch Kevlar neodymium mid-low frequency woofers, can be utilized to extend low-frequency response for applications requiring it.

These can be rolled out to the wings of the stage via dollies or placed under the stage on the main floor level, with cabling infrastructure facilitating quick connection at both locations.

Several Adamson CB1 compact 2-way loudspeakers providing 90-degree by 60-degree dispersion are mounted horizontally in cutouts on the face of the stage to supply front fill support.

Additional CB1 loudspeakers are flown singly, using the supplied mounting hardware, at the far sides of the proscenium to bolster presence in the boxes and gallery seating located nearer to the stage.

All of these loudspeakers are carefully time-delayed in reference to the main arrays to insure synchronous arrival.

A Higher Level
Just two Adamson M Series 4-input by 8-output digital loudspeaker processors were required for all loudspeakers, providing all filtering, limiting and delay. One of these processors feeds the main arrays and subwoofers, while the other is applied to the front frill and side fill satellite loudspeakers.

Lab.gruppen power amplifiers, recommended by Adamson to drive these loudspeakers, are rack-mounted in a secured room with the digital processors. Two FP 10000Q power the arrays, with a single FP 7000 for all four subs and a C 28:4 offering four channels for the remaining fill loudspeakers.

The former system’s Midas Heritage 2000 mixing console was retained because it is still performing at a very high level and affords plenty of mix facilities for any application hosted at the theatre.

The console resides on a cart so it can be easily rolled into position when needed, with infrastructure for quick interconnection built into the main floor mix position. It is also capable of serving as the monitor console when small wedges from the existing system are needed.

The RTS team performed the new system installation quickly and efficiently during a “dark” period for the theatre, and it debuted to positive reviews that have not ceased. “The new system has exceeded all the expectations, offering the coverage, sound quality, and aesthetic ‘look’ the client was seeking,” Fernandez concludes.

Don’t miss the PSW Systems Measurement Photo Gallery.

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Putting Safety First In Live Sound

While not often considered so, it can be a dangerous job and it's critical to take appropriate precautions.

We don’t often think of concert and event production as being a dangerous profession, but far too many accidents and injuries - and sometimes even deaths - occur each year in our chosen profession.

Most of these accidents are caused by human error and can be avoided if we simply pay attention to what we’re doing and follow basic safety rules.

Particularly as we ramp up for the busy summer season, here are a few things (and more) to keep in mind:

• Always wear eye protection when using tools or working in an area where others are using tools. We only get one set of eyes in our lifetime.

• Our ears are our livelihood, so hearing protection is a must.

• Wear gloves when loading in/out, and especially when working with ropes, aircraft cables, or chain.

• Remember to lift with your legs, not your back! And get help when moving heavy and/or bulky items.

• Wear closed-toe footwear at gigs. Even better, go with steel toe boots.

• Make sure electrical power is off before connecting or disconnecting power and/or feeder cables.

• For feeders, always connect ground wires first, then neutral wires, and finally, hot legs. Disconnect in reverse order (hot legs, neutral, ground).

• Protect power cords from damage and avoid creating trip hazards with cable covers or ramps, or by using a cable bridge and running cables overhead, out of harm’s way.

• When using a portable generator, make sure that a ground rod is in place and connected properly to the generator.

• Keep a first aid kit in your vehicles, and one at the event site. Now is a good time to check your kits and restock any supplies.

• Never block a fire exit with equipment or cases.

• Check fire extinguishers to make sure they’re in good operating condition. Repair, recharge or replace them as needed.

• Be sure all portable ladders are set up correctly and are stable before using.

• Wear a correctly sized harness when working off the ground or operating lifts. Now is a good time to check harnesses and lanyards to make sure they’re in good condition.

• Always tie off with lanyards when working off the ground. Wearing a harness does no good if you’re not tied off to a good anchor.

And check anchor points before relying on them with your life!

• Only qualified people should design rigging systems or perform rigging.

That said, anybody who sees any problem with rigging (or any other safety issue for that matter) can call “stop” and point out the issue so it can be addressed and corrected to avoid an accident or injury.

• Only properly trained and certified persons should operate lifts and material handling equipment like fork trucks. Be especially careful when operating machines around people.

Many machines have dead spots where the operator’s vision is hindered. Use a spotter to help guide machines when needed.

And when working outside, always check for overhead power lines before raising a load, ladder, or lift!

• Stay hydrated when working, especially when outside in the summer heat.

Drink plenty of water throughout the day, and drink it even before you even feel thirsty, because by the time the feeling of thirst kicks in, your body might already be low on fluids.

• Pay attention (to yourself and others) for signs of heat cramps, heat exhaustion, and heat stroke. Again, stay hydrated, take plenty of breaks, and cool off periodically in the shade or inside to avoid heat induced problems.

Don’t forget the most important safety equipment that we possess: our brains!

Get a good night’s rest so you can be fresh and alert at the gig the next day. Drowsiness and inattention to details cause of a lot of accidents and injuries on shows.

Last, if you see something unsafe on any event you are working, stop and make sure the problem is corrected, even if the problem is not audio related.

Even though many different trades work on a show, we’re all a team and safety is everybody’s job.

Craig Leerman is senior contributing editor for Live Sound International, and has worked in professional audio for more than 25 years. He is also the owner of Tech Works, a regional production company based in Las Vegas that focuses on live corporate events.

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