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Wednesday, December 14, 2011

The 10 Most Frequently Asked Questions About Mastering: Part II

In this second installment of our three-part series, Tom Volpicelli of The Mastering House answers three need-to-know questions about mastering.

In this second installment of our three part series, Tom Volpicelli of The Mastering House answers three need-to-know questions about mastering.

4. What are some tips to help ensure the best possible master?

Audio quality can be very subjective. Before hiring a mastering engineer for a project you should have a clear objective on how you would like the finished project to sound.

Communication of these objectives between client and engineer is a key component to the success of a project. The language used to describe the character of audio can be ambiguous.

Terms like “brassy,” “fat” and “present” mean different things to different people.

One of the skills of a great mastering engineer is to able to translate this loose terminology into the specific technical processes required to achieve the client’s goals in a non-obtrusive way.

Some mastering engineers find reference tracks from clients to be helpful. Reference tracks can be worth a thousand words, because they serve to demonstrate the sonic objectives of the client.

My personal preference is to receive mixes that are as close as possible to what the finished product should sound like, but with enough leeway to be able to manipulate the sound in order to mold a cohesive album. Some general tips toward achieving this are:

Knowing your room and monitors. If you are using smaller nearfield monitors for mixing, be sure to listen to the mixes on a system that has extended bass to ensure that there are not low end bass problems.

If your monitors or room “color” the sound in any way be sure to compensate to ensure that the mix will translate well on other systems.

Fix track related issues before mastering. Listen for issues like excessive sibilance, uneven or exaggerated frequencies, phase or polarity problems, bad edits, depth and width of the sound field, and the relative levels of instruments and vocals.

I recommend listening to a mix in mono in order to hear if anything disappears or becomes exaggerated as well as listening to the mix at different levels and positions within your room. This can sometimes make an issue more obvious due to a different perspective.

Leave enough of the mix dynamics intact so that the engineer can make adjustments not only in the overall level but in the punch and clarity of the transients.

Don’t use any processing on the master bus that will interfere with processing that is best performed while mastering. This may include exciters and harmonic enhancers, EQ, normalization and limiting used to achieve a higher overall volume.

Leave the heads and tails of a mix intact so that there is room ambience before the music starts and enough of the music at the end to be able to tailor the fade out.

Having a bit extra at the start and end can also be useful so that a “noise profile” can be created for noise reduction systems that use this as a technique for removal of broadband noise.

Use mute and volume automation to remove extraneous noises from the individual tracks.

Noises include: headphone bleed when the vocalist is not singing, hum from electric guitars during breaks, and a drummer who may lay down his sticks after the cymbals fade at the end of a song, but before the final fade out of other instruments.

5. What should I send to the mastering engineer?

Mixes should be delivered in a format that alters the sound by the least amount.

For digital mixes, an uncompressed format (AIFF or WAV) should be used rather than compressed formats like MP3 or AAC.

You should speak to the mastering engineer that you will be working with to verify the formats that they accept.

I recommend staying with the same sample rate used in the original tracks, unless mixing through an external converter.

In that case, increasing the sample rate has its benefits. The bit depth should match the one used during the mix session rather than supplying tracks on audio CD where truncation and optionally dithering of the original tracks is applied.

I also prefer that digital mixes be sent as a single stereo interleaved file rather than split stereo files in order to help ensure phase coherence.

While a standard when sending analog tape for mastering, reference tones are becoming a lost art with digital.

If mixing through an analog board or to an external device, having an unaltered 1k reference tone (corresponding to 0 VU on the console) can help to identify issues where left and right channels are not calibrated or set properly.

If you are not attending the session, be sure to send all documentation regarding the sample rate, bit depth, format, and a listing of the filename with the full name of the song for each file.

Also note if there are alternate mixes of the same track (e.g. vocal up/down). A listing of the song order is also necessary along with requirements for song spacing and fades if not printed on the original mix. If CD text, UPC/EAN or ISRC codes are to be added to the final CD they must also be included in the listing.

Documentation may include information about your audio chain such as equipment and processing used (particularly if applied on the overall mix), what you feel are some of the enhancements that you would like to hear in each mix, along with any other information that you feel will be useful to the mastering engineer.

6. How much will mastering cost?

Prices vary depending on the profile and experience of the engineer, previous credits, along with studio costs and overhead. Typical rates are based on:

     - Flat rate per album usually tiered based on the total number of tracks,
       sometimes with a total hourly cap.
     - Flat rate per track or number of minutes per track.
     - An hourly rate that can include additional costs for media due to time spent
       verifying and listening to the disc.

Some studios may also charge more for attended sessions versus non-attended sessions where the final product is delivered and approved by mail or Internet.

Costs for mastering vary anywhere from $10 per song to $500 per hour for well regarded professionals. Given that mastering is a subjective service-based business, as opposed to a commodity which can more easily be compared, caveat emptor applies.

Assuming that both quality and cost are considerations, set a realistic budget for mastering at the start of your project. Sometimes independent artists will not have anticipated the costs for mastering until a project is completed.

This forces them to use lower quality alternatives that are not necessarily best for their project. It’s a good idea to research the studios that will work within your budget. Call them to discuss the details of project and their approach.

In addition to gaining a better understanding of their process you will be getting a feel for the quality of their customer service. Some studios provide a demo of your material to ensure that they meet your expectations; others may charge for this service.

In either case, this is a good way to hear the quality of their work before committing to the cost of an entire album.

Tom Volpicelli is the president and founder of The Mastering House and has an extensive list of mastering and mixing credits to his name.

Editor Note: This article is Part II in a series of the 10 most frequently asked questions about mastering. Stay tuned for Part III where we’ll cover the remaining 4 questions and be sure to check out Part I in the series.

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API 1608 Console Chosen For Canyon Hut Recording Studio

Two famous studios were the inspiration for Canyon Hut’s “2001 meets the 1950s” design

Constructed in rocker Alice Cooper’s former house in 2008, Canyon Hut Recording Studio has recently acquired and installed a fully analog, all-discrete API 1608 console.

According to co-owner Tim Hutton, picking the right console was an easy decision once he heard the 1608.

“When making the decision to buy a console, I knew it had to feel ‘right’ in my gut,” says Hutton. I tested many and most of them were fairly linear. The 1608 was the only console that was un-darkened, incredibly warm and all embracing. Its design is flawless and I felt right at home when I first sat down and started tracking.”

Hutton is a songwriter, producer and bassist, and co-owns the Canyon Hut with his brother, Dash. The brothers were born into a musical family, as their father, Danny Hutton, was a founding member of classic rock band Three Dog Night.

Touring as children with the group, they were able to interact with influential musicians such as Brian Wilson, Van Dyke Parks, Glen Campbell and America. After attending the Hamilton Music Academy and later touring with his band, the Telacasters, Tim Hutton started to record and produce tracks for some of his friends’ bands.

The recording hobby later turned into a full-time gig, and Hutton knew he needed to find a professional soundboard.

“Things really took off at that point,” he says. “I decided I needed ‘the best’ console. I was already very happy with my API 554s and 525s, so I decided to test the 1608. I was floored. It was exactly what I needed, wanted, and demanded to make the Canyon Hut one of the best studios in Los Angeles.”

Two famous studios were the inspiration for Canyon Hut’s “2001 meets the 1950s” design, with the control room situated so that it looks down into the live room, similar to Abbey Road and Motown. Canyon Hut’s live room also shares dimensions close to those of Motown.

The studio offers an extensive microphone selection, a 1959 B3 Hammond organ and a 1929 Parlor grand piano. Canyon Hut’s past clients include Film School, TS and The Past Haunts, Three Dog Night and HB Surround Sound.

API

Established more than 40 years ago, Automated Processes, Inc. is the leader in analog recording gear with the Vision, Legacy Series and 1608 recording consoles, as well as its classic line of modular signal processing equipment.

 

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Tuesday, December 13, 2011

The Polls Are Open! Vote Now For Your Favorite Products In The Readers Choice Awards

Many of last year's races, with more than 35,000 votes cast, were extremely close - so your vote does really count

The third annual Readers Choice Awards - where you can vote on your favorite sound reinforcement products - has just launched here on ProSoundWeb.

Readers Choice is unique for a number of reasons, chief among them (and as the name says), all voting is the exclusive domain of the readers of PSW.

The first two editions of RCA featured tens of thousands of votes cast. The races in each category were close and competitive, owing to the overall strength of every product entered combined with the distinct yet varied preferences of the pro audio industry’s largest online community. Click here to see the roster of last year’s winners.

Some of the races were so tight, in fact, that they were decided in the final days of the contest, separated by just few votes. So every vote really does matter.

The polls are open, and voting is simple - just go to HERE to get started.

 

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Thursday, December 08, 2011

In The Studio: Audio Effects Explained, Part 2 – Reverb

Exploring the different types of reverb, with audio samples, as well as tips for using reverb

 
Continuing my series on effects, I’m talking about reverb. (See part 1 about modulation here.)

The different types and methods, and I’ll also explain the most important parameters.

I’ll mostly be talking about the kinds you will be using when mixing and what is available as plugins.

Digital Reverb Technology
There are two ways of creating a reverb effect in the digital world, by using mathematical calculations to create a sense of space, which is called algorithmic. And, by creating an impulse response, a snapshot of a real space, and applying that to the sound, which is called convolution.

Reverb is essentially a series of delayed signals, and algorithmic reverbs work pretty well to recreate this. Most reverb plugins, stomp boxes, and racks are algorithmic style.

When you want really realistic reverb, then convolution can not be beat. To create an impulse response the creator goes into a room and records the sound of a starter pistol going off and the natural reverb of the room.

The recordings are then deconvolved in software which is removing the sound of the starter pistol from the recording, leaving only the reverb.

Sine wave sweeps can also be used for the impulse creation. This is a more accurate way of creating reverb because it also captures the character of the room, and the way different frequencies react in the room.

The same process can be used to create impulse responses of speaker cabinets, guitar amps, vintage rack gear or basically anything that can make a sound.

Analog Reverb Types

In the analog world there are a few other ways, most of which will not be available to the home studio musician, except for their recreations in plugins.

Analog reverbs come in three flavors - plate, spring, and chamber.

Invented in 1957 by EMT of Germany, the plate reverb consist of a thin metal plate suspended in a 4-foot by 8-foot sound proofed enclosure. A transducer similar to the voice-coil of a cone loudspeaker is mounted on the plate to cause it to vibrate.

Multiple reflections from the edges of the plate are picked up by two (for stereo) microphone-like transducers. Reverb time is varied by a damping pad which can be pressed against the plate thus absorbing its energy more quickly.

This is what a plate reverb sounds like: platereverb.mp3

A spring reverb system uses a transducer at one end of a spring and a pickup at the other, similar to those used in plate reverbs, to create and capture vibrations within a metal spring. You find these in many guitar amps, but they were also available as stand alone effect boxes. They were a lot smaller than plate reverbs and cost a lot less.

This is a spring reverb: springverb.mp3

The first reverb effects used a real physical space as a natural echo chamber. A loudspeaker would play the sound, and then a microphone would pick it up again, including the effects of reverb. Although this is still a common technique, it requires a dedicated soundproofed room, and varying the reverb time is difficult.

This is a chamber: Chamber.mp3

These three types of reverb are all available in digital form in addition to a few other styles simulating real spaces, and others not found in nature.

Natural Reverb Types

Room – A room is anything from a classroom to conference room. There is generally a short decay time of about 1 second: room.mp3

Hall – A hall is larger than a room, it could be from a small theatre with 1 second of decay up to a large concert hall with a decay time up to 2.5 seconds: hall.mp3

Church – The decay time of a church can vary between 1.5 seconds to 2.5 seconds: church.mp3

And Cathedral decay times can go above 3.5 seconds: cathedral.mp3

Remember, the sound of a room is not just the decay time. The materials it was built with make a huge impact on the character of the sound. Stone, wood, metal and tile all sound drastically different.

There’s also a few other types of reverb that are not natural - these are Non Linear, Gated and Reversed.

Non-Linear has a decay that doesn’t obey the laws of physics: non-lin.mp4

Gated was a popular effect in the 1980s, but it’s sounding pretty cheesy these days: gated.mp3

Reversed sounds like this: reverse.mp3

Reverb Parameters

Reverb Type – What kind of reverb emulation it is. There are Halls, Rooms, Chambers, Plates, etc…

Size – What the physical size of the space is. This can range from small through large.

Diffusion – How far apart the reflections are from each other.

Pre-Delay – Sets a time delay between the direct signal and the start of the reverb

Decay Time -Also known as RT60, which is how long it takes for the signal to reduce in amplitude 60 decibels.

Mix (Wet/Dry) – Sets the balance between the dry signal and the effect signal. When you have the reverb effect on an insert you need to adjust the wet and dry ratio, when you are sharing the reverb in a send and return configuration you want the mix to be 100 percent wet.

Early Reflection Level – Controls the level of the first reflection you hear. Early reflections help determine the dimensions of the room.

High Frequency Roll Off – Helps control the decay of high frequencies (as it is found in natural reverb).

Tips For Using Reverb

—Using pre delay can help keep your vocals up front, while still giving them space.

—Try to keep decay times short for faster tempo music.

—Filter out low frequencies before the reverb to keep it from sounding muddy

—Try de-essing the reverb to reduce harsh sibilance.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.

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Monday, November 28, 2011

Selecting The “Right” Direct Box (DI) For Bass Guitar

Both are useful, depending on the desired outcome

What type of direct (DI) box works best for bass guitar?

The answer is easy: it depends. In fact, more than anything else, it depends on the type of bass that the DI is going to be used with.

When it comes to signal flow, there are two types of bass guitars: passive and active. The first electric basses, i.e., the original Fender Precision, were passive, and in fact still are today.

They employed magnetic pickups to generate the signal - as the string moves in and out of the magnetic field, a low-level alternating current is generated.

The signal from the bass travels through the cable to the amplifier, which in turn increases the voltage level so that it is sufficiently powerful to drive another electromagnetic device: a loudspeaker. In essence, the signal is amplified by a series of buffers that work together to increase the voltage and/or current as needed.

For years this worked well, until bands like the Beatles messed everything up!

The problem was that the fans at those concerts were so loud that the bass amp was unable to produce enough ‘thump’ to overtake the screaming. The solution: send the bass guitar signal through the PA system.

Eureka! The amazing direct box was born. The first direct boxes were basically hand-made black boxes that had transformers inside.

These passive devices would tap a signal off the bass and split it so that part of the sound would go to the bass amp on stage, and the rest of it would go to the PA system some 50 to 100 feet away.

Origins Of Active
As the PA systems got larger, so did the performance venues (or vice versa). Eventually, things escalated to the point where concerts moved to arenas and stadiums.

And bass players complained because they noticed that when their bass was connected to all of the long cable runs in these larger systems, the sound changed. It was not as beefy, and there was no more thud.

This shouldn’t have come as a surprise - if you take the signal from a magnetic pickup and ask it to drive hundreds of feet of cable in addition to the bass amp on stage, the level will be weaker. And it will not sound the same. This effect is known today as “loading.”

The solution: buffer the bass signal. In other words, incorporate a small amplifier inside the direct box so that 99 percent of the signal is directed to the bass amp and 1 percent is split off to drive the PA. And thus the active DI box was born! Ye old Fender P-Bass was happy - the thud had returned.

Bring The Mayhem
So for the next bunch of years, everything worked just ducky, until one day, some guy decided to put a 9-volt battery inside the bass and buffer the signal.

Now all of the sudden, instead of the bass producing around 1 volt, the battery powered preamp inside the bass was kicking out 5 to 7 volts.

Then the CEO of the Acme Bass Company had a revelation: “We can do even better - let’s put in a second battery!”

A modern 6-string bass could now deliver a whopping 18 volts of mayhem, and bass players rejoiced. They could overload the front end of “ye old SVT” and finally out-blast that pesky lead guitarist and his lowly Marshall!

All good, expect for one problem: that 18-volt output now overloads the direct box, resulting in a distorted, muddy, no-punch sound in the PA system. Or, if you prefer, it just plain sounds bad.

The solution? Dust off the old passive direct box, connect it up and bingo, great tone - the thud is back.

Phantom Solution
Here’s the deal. Early active direct boxes were powered by batteries and in fact, some still are. But the problem with batteries is that they go dead… usually right in the middle of the second set.

So some years ago, DI manufacturers started to use phantom power as a means to supply the needed voltage and current to the active DI box (buffering amplifier).

But phantom power, invented by Dr. Neumann as a means to supply a polarizing voltage to his condenser microphones, was never intended to be a power source for an amplifier. And without current, you do not get headroom.

Think of a bass playing through a miniature guitar amp - turn it up, and it distorts like crazy. DI boxes do exactly the same. Without headroom, high-output bass signals will cause the buffering amplifier in the DI to distort.

But remember, back then, basses were all passive so for the most part, so they worked fine with regular phantom power as the buffers only had to process 1 to 3 volts. The advent of active basses with their huge output levels changed the rules.

Two Groups
So the rule of thumb is that for a highoutput bass that already has a built-in buffer, a passive direct box will likely do a great job - the bass will produce the drive.

On the other hand, for a lowoutput passive bass, an active DI will leave the bass sound unaffected while generating the drive for the PA system.

Keep in mind that the sound quality of DI boxes depends on the circuit design and parts that are being used. Better designs focus on eliminating all types of “bad” distortion such as harmonic, phase and inter-modulation distortion.

These designs are then categorized into two groups.

Some direct boxes are designed to transfer the signal without artifact or distortion so that the original sound of the bass is delivered as purely and naturally as possible, while others, such as tube DI boxes, tend to be designed to “color” the sound with “good” distortion to create new bass tones and exciting textures. Both are useful, depending on the desired outcome.

Peter Janis is president of Radial Engineering.

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Wednesday, November 23, 2011

The 10 Most Frequently Asked Questions About Mastering: Part III

In this third and final installment of our series, Tom Volpicelli of The Mastering House answers four need-to-know questions about mastering.

In this third and final installment of our series, Tom Volpicelli of The Mastering House answers four need-to-know questions about mastering.

7. How much of a role does gear play versus the talents of a mastering engineer?

As the saying goes, “It’s the driver not the car”. A good engineer can work around limitations while a bad engineer will likely produce poor results, great gear or not.

This does not entirely discount the aspect of the gear. Having gear which is made specifically for mastering makes a big difference, not only in the quality of the sound, but in how quickly and easily the engineer can perform his work.

This includes equalizers, compressors and the usual components that most associate with the term “gear” as well as quality converters, monitors, and the room where the mastering engineer works.

Any of these can skew the perception of what an engineer hears potentially causing them to make decisions that wouldn’t happen given better accuracy.

There are many hardware and software companies claiming the ability to allow anyone without prior experience to use a particular preset, match frequency curves with references, or use other methods which will allow them to master their own music.

These “cookbook” approaches really miss the point of what the mastering process should be about. This approach cannot replace the skill acquired by an experienced engineer.

The processing performed should bring out the elements of the mix that are most important to each song. This requires both an artistic and technical evaluation.

Using a generic EQ or compressor setting to try to achieve this doesn’t address the individual characteristics of the song that make it unique or the specific problems that it may have in translating those elements.

8. What is the best (fill in the blank) for mastering?

This is a question that is often asked within mastering forums. The simple answer is that there are no “best” or one-size-fits-all solutions.

If there were, mastering houses would look more like a chain of department stores with the same type of room, monitors, and gear.

Just as the processing chain used for a particular piece of music will vary according to the character of that track, the hardware and software chosen by an engineer is based on his workflow and tastes.

There are however some common characteristics among mastering studios. The following are what I would consider the universal set of tools ranked in order of importance.

     - A discriminating pair of ears. The ability to critically analyze issues that will
       interfere with the enjoyment and translation of a piece of music is the most
       fundamental tool of a mastering engineer.

     - Knowledge and taste. Having the technical knowledge to be able address the
       problems heard in a mix and the taste to know whether or not to use a given
       technique.

     - An accurate room and monitors. A good pair of monitors in a bad room can
       misrepresent a mix as much as a bad set of monitors in a good room.

       Both room and monitors work together to produce a listening environment which
       will not distort the presentation of a mix causing an engineer to potentially make
       bad decisions.

     - A transparent processing chain. As with physicians, one credo of the mastering
       profession is to “do no harm”. Mastering engineers go through great effort and
       expense to ensure that their processing chain is as distortion and noise-free as
       possible.

       Everything from the type of cables to the software and hardware used is analyzed
       and potentially modified to reduce any ill-effects caused by the processing chain.

     - Processing which provides additional “color”. What would seem like a
       contradiction to a transparent chain is the addition of hardware or software which
       actually adds distortion in order to enhance a track.

       This includes both new and vintage hardware which adds tube distortion,
       transformer or tape saturation, along with software based modeling algorithms.
       The intent of these effects is to add warmth, thickness and depth to mixes that
       would otherwise sound thin or too “digital.”

9. Should you choose an engineer based on their “style”?

Ten different mastering engineers working in the same room with the same equipment will create ten totally different masters, each sounding great on their own.

If you ask those same engineers to go back and reproduce any given master, you are likely to get ten almost identical masters back.

While each individual mastering engineer has his own style, it is important that he is able to separate himself from his style when needed.

An engineer should never let his personal taste interfere with the goal of the artist he is working with. Again, this is where communication with the client is a crucial element.

A good mastering engineer should be well versed in a variety of different categories of music. In general, there is no reason why an engineer known for creating great Country albums cannot produce a great Rock album.

While an engineer’s work should be able to transcend musical genres, if a mastering engineer has a certain style that is appealing to you as the artist, you should consider working with him.

It is important that both the engineer and the artist can communicate in a way that is complimentary to both individuals.

10. Which is more important, a technical background or musical one?

A mastering engineer should be well versed both technically and musically. The craft of the engineer is to be able to know good music and know how to make that music sound better.

Still, while a technical background is extremely important in the mastering world, that background should not interfere with the aesthetics.

Likewise, any personal feelings an engineer has about the stylistic choices of the music he is mastering should ultimately be discussed with the musician. It is because of this that an engineer’s musical background should not hinder his craft.

Given a technical background, some mastering engineers are capable of making modifications to equipment to create a more transparent sound, or provide color according to their taste and needs.

Having a musical background, particularly in the area of pitch, allows an engineer to identify frequency issues relating to musical notes and can speak directly to the musician about these issues in their terms.

An engineer should make sure that he strays away from favoring either background. While most engineers come from one or the other, their craft is in combining the two.

A mastering engineer should remain as objective as possible while still providing necessary feedback and insight from both a musical and technological perspective.

Tom Volpicelli is the president and founder of The Mastering House and has an extensive list of mastering and mixing credits to his name.

Editor Note: This article is Part III in a series of the 10 most frequently asked questions about mastering. Be sure to check out Part I and Part II where the previous 6 questions were covered

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Monday, November 07, 2011

Middle Atlantic Products Debuts RackLink Suite of Power Management Products

RackLink products maintain AV system reliability through automatic actions including automatic reboot, turning on emergency fans and shutting down equipment for safety in overload or over-temperature conditions

Providing installers with flexible, intelligent power monitoring and management for AV systems, Middle Atlantic is announcing the RackLink suite of products that feature extensive monitoring and preemptive support functions. 

RackLink products are capable of maintaining AV system reliability through automatic actions that include automatic reboot, turning on emergency fans and shutting down equipment for safety in overload or over-temperature conditions. 

Designed from the integrator’s point of view, the RackLink system can be configured and managed for any level system ranging from simple to complex: it can be managed locally or remotely via its standard browser-based user interface and integrates easily with control systems via its open-architecture API.  Accessibility ranges from local to RS-232, remote IP and web browser. Free mobile apps that provide instantaneous insight and control of any installed RackLink system are available for iPhone/ iPad and Android devices.

In addition to offering essential functions expected of any power management system, including local, remote and automatic reboot of individual outlets and dry contacts, the RackLink system also provides proactive power management.  These practical features include input voltage and temperature monitoring, as well as user-defined monitoring of thresholds and logging and issuing of email alerts if monitored values breach thresholds.  All this functionality ensures that system downtime is minimized and system performance is optimized.

Key to RackLink’s practical use is its simple, plug-and-play functionality and communication flexibility.  “Building the open platform architecture into RackLink was a priority,” said Murray Williams, Middle Atlantic Electrical/Electronic Product Manager “It was important for us to develop a system that offers cloud functionality without being dependent on it. The key advantage of this is its ability to integrate seamlessly into any environment regardless of the platform or communication requirements.”

RackLink units are available in 15A and 20A rackmount and in-line models, with controlled or monitor-only capabilities.

Middle Atlantic

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Friday, November 04, 2011

RE/P Files: The ‘Planet Waves’ Sessions - Recording Bob Dylan At The Village Recorder

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine come interviews with Rob Fraboni and Dick LaPalm on recording a legend which first appeared in the March / April 1974 issue.

R-e/p: Dick, how did you choose an engineer for the Dylan album?

Dick LaPalm: I left the decision up to Rob. I asked him who should do it.

At the time we had 3 guys. Rob came back after a couple of days and said, “I should do it.” I said, “Fine.”

R-e/p: Rob, why did you decide to do it?

Rob Fraboni: Mainly because I was really familiar with Bob’s music, as well as The Band’s.

I’ve been listening to them both since their first albums. I talked to the other guys, and it seemed like I was the most familiar.

R-e/p: Dick, do you feel that familiarity with the music is essential for a mixer?

Dick LaPalm: Engineers are much like a medical specialist. I just don’t think that every engineer can do every kind of music.

I think this guy might be a hell of a lot better to do an R&B date, as opposed to a Country & Western date.

And one engineer might be a hell of a lot better to do a Dylan and a Stones. I’m not taking anything away from him; I’m sure he could do a Willie Hutch.

I’m sure he could do a Little Milton or a Chuck Berry. But I don’t know that he could do it as well as someone else who’s really into that kind of music.

I think there’s a hell of a lot more to it than just knowing that board. I think it has to do with gut feel, and feeling for the music itself.

R-e/p: Rob, did you listen to their stuff before the sessions? Did you go home and prep on it?

Rob Fraboni: No, I didn’t. I make sure not to do that. You’ve got to approach things fresh; that’s the way I feel.

After we mixed the album and it was all done, then I went and listened to his records.

I didn’t want to be influenced before the sessions. I just wanted to do it fresh, and that was what they wanted, too, Dylan and The Band.

R-e/p: Was there anything unusual about the way Dylan and The Band work which would affect the choice of an engineer?

Dick LaPalm: We talked about engineers. The one thing they wanted was a guy that not only knew the equipment and respected it, but someone who could work really rapidly.

Knowing how a Dylan works - - the guy says, “Let’s do it now,” and he expects the engineer can do it, just like that, without fumbling.

R-e/p: Why did Dylan and The Band record at Village? What did you have that made it just right for them?

Rob Fraboni: One thing, the room was right for them. As far as the size, they really liked that. And as far as the control room is concerned, they just wanted something that sounded good.

It could have been done at a number of places, but we had a combination of things: the room, the security and the location.

They liked the idea of being out of town (The Village Recorder is situated in West Los Angeles, about ten miles from Hollywood).

When we actually got down to the mixing, Robbie was comfortable with what he was hearing, and that was the really important thing.

R-e/p: When you say Robbie, you are talking about . . .

RF: Robbie Robertson, (guitar. The Band).

RF: There was no producer on this record. Everybody was the producer.

Robbie is the one who gives a lot of direction, although they all have something to say about the music, and are all really involved.

DL: He seems to be the one that has the most knowledge as far as engineering is concerned.

He has tremendous knowledge about what equipment can do - - what a board can and can’t do.

R-e/p: Let’s get back to the room. You told us that studio B was used for the album. What is it about this room that made it attractive!

RF: For one thing, you can work in here for hours and hours and not get fatigued.

And you can turn this room up very loud and it won’t hurt. Numerous people have commented on that.

R-e/p: What kind of monitors are you using?

RF: The room was conceived by me and designed by George Augspurger, and the monitors arc custom built using JBL components and custom crossovers.

Each enclosure has two 15” 2220 woofers, which are thin-cone units.

They’re also efficient, so our amplifiers aren’t working so hard on the low end. It gives us a punchier bottom than a 2215, with a different coloration.

The 2215 has a more rubbery sound. While the curve of our room might look like another room, it has a certain character.

The 2405 tweeters are also part of the picture. I just really like the way they sound in this installation. The overall system has a very low fatigue factor, or whatever you’d call it.

R-e/p: What kind of a curve does the room actually have?

RF: Well, it was originally flat, but we tailored the high end a little differently. I found that having a flat monitor system was a terrible hype.

The way we finally decided on the curve was that I went to a lot of studios and to a lot of people’s homes and played music on different systems.

I took notes and gathered the information.

R-e/p: Since the room is equalized, you could probably have achieved similar frequency response with other speakers.

Was there another factor involved in the choice of these particular speakers?

RF: Well, I like 604’s with the Mastering Lab crossover. But they still have a beaming effect.

That’s one thing you just can’t get away from, and that was the reason we decided to switch to units with better dispersion.

R-e/p: Without the beaming, what kind of coverage do you get? Where is the best sound in the control room?

RF: Realistically, the working area is the length of the console. You can sit at the producer’s desk and hear well, although there is some difference from behind the console.

As far as quad sound, it’s surprisingly good for a small room. It sounds very large and open in here.

R-e/p: We’ve talked a lot about the control room. Let’s discuss the studio for a while.

For example, how many mikes were used in the sessions?

RF: As it turned out, I used about 28 microphones.

R-e/p: That seems like quite a few mikes for a relatively small studio. Why were so many mikes necessary?

RF: 7 were used on the organ. Garth (Hudson) has got this elaborate Lowrey organ with a Leslie on each of two keyboards.

One Leslie is a model 103, of which very few were made. It has stationary speakers with a phasing device in the tube-type amplifier, as well as 2 rotors.

There was also a Hammond organ with a Leslie. Sometimes Garth would play both organs at one time, so we were miking three Leslies.

R-e/p: How about the other instruments?

RF: I often use a lot of mikes on the drums; I used about 7 or 8. I wanted to mike everything kind of tight in this case.

Bob had an electric and an acoustic guitar, as well as his vocal mike. And it all had to be ready to go because they would just say “OK” and boom, you go.

R-e/p: We’d like to know a little more about the miking, and the diagram you’re doing will help. But you just raised an interesting point.

That is, what kind of a recording artist is Bob Dylan? What was it like working with him?

Dick mentioned and you are also hinting that Dylan needs an engineer who’s on his toes.

RF: Right. Robbie came in that first morning and said to me, “There are going to be no overdubs. We’re doing it live. This is it, what’s happening here is it.” Bob doesn’t overdub vocals.

R-e/p: It sounds like Dylan was in the studio to perform, period.

RF: That’s really true. The record was really a performance, as far as I’m concerned. It wasn’t like we were “making a record.”

It was more of a performance, and Bob wanted it to sound right, to come across.

When he starts playing, there’s nothing else happening but that, as far as he’s concerned. I don’t think I’ve seen anyone who performs with such conviction.

R-e/p: Maybe we can back up a little and get some information on how the album was first conceived. And how long did Dylan work on it!

RF: I can tell you what I know, although I don’t know everything. A few weeks before we started the album, Bob went to Now York by himself. He stayed there for two to two and a half weeks and wrote most all the songs.

One of the classic- songs, “Forever Young,” he told me he had carried around in his head for about three years.

He gets an idea for a song sometimes, he said, and he’s not ready to write it down. So he just keeps it with him and eventually it comes out.

R-e/p: When did he get together with The Band for this album?

RF: I’m not exactly sure but I know they had started rehearsing for the tour before we began recording.

They only knew two of the songs on the album before coming in. The balance of the songs on the album they never heard until they were right here in the studio.

R-e/p: It appears The Band are pretty good musicians.

RF: They’re really something. And it’s got such character the music sounds like it’s all arranged.

Bob would just run it down, and they’d play it once. Then they’d come in to the control room and listen. That’s another thing that really astounded me.

Nobody was saying, “You ought to be doing this,” or “You ought to be playing that.”

They just all came in and listened to hear what they should do, and then they’d go out into the studio. That would usually be the take, or the one following. That was pretty much the way it went.

R-e/p: Were the takes run straight through from the top?

RF: Yeah. Almost all of them were complete. The other thing was, if that wasn’t the take, they’d do a few more. Sometimes, they would change the arrangements from take to take because it was still so fresh.

Then they’d choose the one that felt best.

R-e/p: How many days did it take to do all the recording?

RF: They initially came in on Friday, November 2, to get set up and to get a feel for the studio. We did use one song that we recorded that day.

They cut three or four things for the album on Monday. Just came in and knocked them off. Then on Tuesday they cut about four more things, and we used about three of them.

We took two days off. Then they came in Friday and we cut the balance of the album that day.

R-e/p: So you really cut most of what you used in about three days.

RF: Yeah. Then we were assembling on Saturday, the next day, and Bob, myself, Nat Jeffrey (assistant engineer on “Planet Waves”) and Bob’s friend were here. We put together the master reels.

Then around noon, Bob said, “I’ve got a song I want to record later,” and I said fine.

He said, “I’m not ready right now. I’ll ‘tell you when.” We were doing what we were doing, and all of a sudden he came up and said, “Let’s record.” So he went out in the studio, and that was “Wedding Song,” the cut that ends the album.

R-e/p: You mean he just walked out and it was a one-take?

RF: He just went out and played it. It was astounding. I hadn’t heard him do anything that sounded like his early records.

Lou Kemp, his old friend from Minnesota, was there. He also came on the tour with us.

Anyway, Bob went out to record, and I put up some microphones, and I was going to get a sound. But usually he wouldn’t sing unless we were recording.

That’s the way he was. You couldn’t get him to go out and just sing, unless he was running something down with The Band.

Well, I said I was going to get a sound. He asked, “Is the tape rolling? Why don’t you just roll it.”

So I did, and he started singing, and there was no way in the world I could have stopped him to say, “Go back to the top.” It was such an intense performance.

If you listen to the record, you can hear noises from the buttons on his jacket. But he didn’t seem to care. Lou and I were both knocked out by the song.

We listened to it a few times and didn’t think about it again until we got down to mixing.

I mentioned re-cutting it to eliminate the button sounds, at one point, and Bob said, “Well, maybe.” But he never said yes, so we let it go.

R-e/p: Was that the last song they cut?

RF: Actually, the final recording happened during the mixing. We had mixed about two or three songs, and Bob, Robbie, Nat and I were there.

Bob went out and played the piano while we were mixing. All of a sudden, he came in and said, “I’d like to try ‘Dirge’ on the piano.” We had recorded a version with only acoustic guitar and vocal a few days earlier.

R-e/p: Were you ready for it?

RF: We weren’t ready at all, we were mixing. But we put up a tape and he said to Robbie, “Maybe you could play guitar on this.”

They did it once, Bob playing piano and singing, and Robbie playing acoustic guitar. The second time was the take. It was another one of those incredible, one-time performances.

R-e/p: Was anyone else involved in the mixing?

RF: Robbie Robertson has a good ear for mixing, knows what he wants to hear. So it was pretty much him and Bob when it got down to mixing. Robbie and I mixed the record together, and Bob was there commenting and making suggestions.

R-e/p: Can you describe Bob’s concern with the mixing, or at least the kinds of things he picked up on?

RF: Well, for one thing, he wanted certain types of sounds. He wanted a kind of bar room sound from the piano on “Dirge” rather than a majestic sound. He also wanted a raunchy vocal sound. We actually mixed “Dirge” immediately after we recorded it that night. Robbie and I listened to it once and I said, “Let’s mix it right now.”

So we took a mix and that’s what’s on the record. It had a unique character. The sound of that particular mix made a lot of difference and was important to him.

We did another mix later going for a more “polished” sound, but didn’t use it. That’s the kind of stuff he was sensitive to, how the mixes affected the character of the music. That might have been more important to him than the sound quality

R-e/p: Did it take a long time to mix the album ?

RF: We came in and mixed a few songs. We would work a day or two and take a few days off. And we always worked from noon to about eight, really good hours. One of the songs, “Hazel,” we used the way we first mixed it. But we remixed the other two because we felt we could do better.

Once we got into doing them, we mixed the whole album in about 3 or 4 days. But then we spent more time than it took to record or mix just to sequence the record. Bob wanted to live with a few different sequences, until he found one that was just right.

R-e/p: How far did you go with the project, Rob? Were you involved in the Mastering?

RF: After the mixes were done, they virtually turned the whole thing over to me.

They let me decide on the spacing between songs, and everything regarding mastering. I cut sets of refs for them for approval when I was satisfied, and then they gave me the final go-ahead.

R-e/p: We see the record was cut at Kendun. What made you go to that particular mastering facility?

RF: I did some checks, actually. I cut flat parts at a few places, and put 700-cycle tone at the front to get accurate comparisons of the cutting. From there I decided on Kendun. So I went out there, cut it, and that was it.

R-e/p: Kendun’s room was done by Westlake, wasn’t it?

RF: Yes. It sounded a bit bright in there.

R-e/p: That isn’t surprising, considering the different monitor systems involved. Did you have any trouble adjusting to the difference and getting the right EQ?

RF: I suggested that we do nothing to it, and Kent Duncan, who did the cutting, agreed. I just relied on our previous checks of the mixes.

R-e/p: You mean when you got back to the Village with the refs, it sounded right?

RF: Yes, when we cut it flat. But we tried some EQ on the critical refs, a little on this and a little on that, and we couldn’t do anything to really improve it.

R-e/p: So you think it’s no problem to mix on one system and cut on another?

RF: No, I’ve done that. An even better example was the album I did with Richard Green before we did Bob’s album.

Our studio was booked so heavily that we had to go outside to Sound Labs (Hollywood). It sounded very similar and was easy for us to adjust.

R-e/p: That’s a 604 system with the Mastering Lab modification.

RF: Right. The bottom end is different in here, it goes lower - - down to 40 cycles almost flat. It just didn’t sound like it was doing that at Sound Labs.

Our bottom end has a certain feel to it, as well as a sound, which is different over there. But the high end sounded very similar, which surprised me.

R-e/p: What about people who like a different sound?

RF: Of course we’re talking about taste. That’s pretty much what it comes down to. Some people like 604’s, and you can’t argue with it.

What we do have in all our rooms is a speaker switching system. We have a rotary selector switch, with other speakers on custom made stands.

They have small bases, telescoping height adjustment, and heavy-duty casters. They’re sturdy enough to hold a 604E or 4320 and roll around.

R-e/p: You brought up the subject of taste, and it reminds me that we were going to discuss the mikes used for the album.

I wonder if you can describe Dylan’s vocal mike, to begin with.

RF: We used a Sennheiser 421. But we went through five or six mikes to find out which would be best.

R-e/p: Did Dylan have a favorite mike?

RF: He preferred a 421 because he had used it before and liked it. Robbie suggested the 421. To tell the truth, it didn’t cross my mind because I hadn’t used it for vocals before.

R-e/p: Which one would you have used?

RF: As I said, I was experimenting, although there wasn’t much time for it. The first day, we tried an SM-53, 57, an 87 and a 47.

I figured the condensers weren’t going to work because of leakage problems.

We also had to consider popping, which was a problem with the 421, especially because Bob doesn’t like to use a wind screen.

R-e/p: What did that do to the sound?

RF: It worked out OK. He’s always popped and seems to be used to it.

R-e/p: Did you use any de-essing or correction on the mix?

RF: No de-essing. We had a Pultec filter we would click in for the p’s. We usually shelved the vocal at 50 Hz. Nat would sit over there and switch to 80 Hz just for the p’s.

On one song, “Dirge,” I got Bob to use a wind screen, He used it, and it really worked well. So, to answer your earlier question, that was how we chose the vocal mike — experimentation, with an ear to leakage.

R-e/p: What are the leakage characteristics of the 421?

RF: Well, The Band was playing fairly loud and I was limiting Bob slightly, 3 to 5 dB. Live, we were getting - 15 dB, tops, on the leakage, and that was incredible. I couldn’t believe it.

I’d look at the meter, and it was just barely moving. I was immediately sold on the mike. Plus, what leakage there was, sounded good.

R-e/p: Would you mind getting into more detail on the instrument miking?

RF: On the drum kit, I used quite a number of mikes: a Shure SM-7 on the bass, Sennheiser 421 on the snare, KM-84 on the high hat, and 87’s for toms and overheads. I experimented with the set a little bit.

R-e/p: Was there anything you particularly like in that combination of drum mikes? Is it a favorite set-up?

RF: It just worked. The Band likes a thick torn sound, and the proximity effect of the 87’s worked to our advantage in this respect. And I like the sound of condenser mikes on drums, so that’s why I chose them.

On the high hat, I have found the 84 just works well on almost any set. I’ve got about three or four different mikes I use on snares, based on the kind of sound the drum set has.

R-e/p: So you try to get a sound tailored to the specific situation?

RF: Yeah. I don’t have a set up that I use on every drum set.

R-e/p: You really seem to be enthusiastic about the drums.

RF: That’s probably because I play drums. I feel they’re really an important part of a good sounding record. I have a feeling for musicians, having played myself. I always go out in the room and listen. They’ll run through something and I’ll stay in the studio.

When the musicians come in initially I always ask, “What’s the most comfortable way for you to set up?”

I tell them we’ll start from there, and if there are any problems, we’ll rearrange things. It helps a lot – when you give musicians that kind of room, they feel better.

R-e/p: Let’s run through the rest of the miking. The diagram you prepared shows a lot.

What about the choice of piano mikes?

RF: We used two KM 84’s. I tried a couple of things. I miked both facing the hinge.

One of them was almost to the end of the harp, and about 12” toward the hammers about a foot to 18” from the hinge.

The body of the mike was parallel to the soundboard, about 2” up.

The other mike was in the same basic position, but angled a bit toward the soundboard - about 30 degrees.

It was in the high end section of the piano, nearer the holes. It worked really well, with practically no leakage at all.

R-e/p: Did you have the top open?

RF: I had it on the short peg, with it really covered. We were all surprised how low the leakage was.

But when I did “Dirge” with Bob, we used a completely different set up, mainly because he wanted it that way. I had it open all the way, no covers, nothing.

R-e/p: Did the piano get into his vocal?

RF: No, he sings so loud. Interestingly enough, the one thing that leaked into the drums was Bob’s vocal. That’s one reason the leakage was so low. He really sings hard. In fact, he was leaking so badly into the uncovered piano that I had to experiment.

I used RE15’s. I faced them toward the back of the piano, instead of the hammers, and it worked really well. It took a bit of EQ, but as far as leakage went, it was really excellent. Plus, as I said, lie wanted a more “far away” sound for that number.

R-e/p: Were there any other unusual or special miking techniques?

RF: Let’s see. We used a special direct box for the bass. Our maintenance man, Ken Klinger, built it. It’s a solid state, discrete, FET type. We used that on the bass, and miked the amp - a twin reverb, I think with a 56.

R-e/p: It’s becoming easier to see where all the mikes were used. According to the diagram, there seem to be quite a few more instruments than there were players. Were they all used in the same session?

RF: Yes, sometimes. There was a pianet and clavinet both were direct. Rick (Danko), who played bass, also played fiddle a bit. And there was an accordion.

There was also a Dobro guitar. I had extra mikes up for these instruments, for whatever might happen. The Band didn’t do any singing on the album. And that’s it.

R-e/p: With all the close miking and the experienced musicians, did the actual levels in the studio tend to be low? And, if so, did everybody wear phones?

RF: The levels were medium-loud, and they could hear each other in the room. They would occasionally wear phones.

R-e/p: What kind of mixes would you give them? Heavy on their own instruments, just the other guys, or what?

RF: A stereo mix of the whole thing, and they loved it. They had Sennheiser 414 phones, and the stereo worked out very well, especially for Garth.

I could put one Leslie in one ear, and the other Leslie in the other ear, and it gave him the perfect effect because that’s what he does. He puts the Leslies on either side of the Lowrey so that when he uses the different keyboards, the sound goes back and forth.

R-e/p: As far as your monitoring was concerned, did you listen in mono at all?

RF: Yes, a lot. That’s a sure-fire way to acoustically catch phase problems.

R-e/p: But what do you do with something like the Leslie, where the phase is all over the place?

RF: That’s a whole different circumstance. You just do your best to make it sound good.

Downloadable Media
Original Article (pdf)

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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Tuesday, October 25, 2011

Generating Seismic Activity: Big Sound For Metal Masters Slayer & Rob Zombie

System tech Andrew Dowling of Sound Image worked closely with both front of house mix engineers to ensure the system was up to the task of reproducing each band’s brand of live metal throughout the tour’s successful run

Although Slayer and Rob Zombie have crossed paths on the road in recent years, the recent Hell on Earth co-headline concert tour of North American arenas and sheds marked the first time the two metal heavyweights have toured together in over a decade.

Slayer’s musical style is defined by fast tremolo picking, double bass drumming, and shouted (or chanted) vocals, and they’re recognized as one of the “big four” thrash metal acts, joined by Metallica, Megadeth and Anthrax.

After first gaining fame as the long time frontman for metal superstars White Zombie, Rob Zombie has built a thriving solo career in his own right, and is noted for his work with iconic artists such as Alice Cooper as well as his burgeoning career in horror films.

The sound reinforcement system, along with technical support, was provided by Escondido, CA-based Sound Image. System tech Andrew Dowling of Sound Image worked closely with both front of house mix engineers, Joel Lonky (Zombie) and Tim Quinby (Slayer), to ensure the system was up to the task of reproducing each band’s brand of live metal throughout the tour’s successful run.

Horsepower Matters
The choice of Adamson Systems line arrays as the system’s main loudspeakers came about late in the sound design process, but proved to work out well. “I’d never toured with an Adamson rig, but I’ve used it, so I had no qualms – I’m a firm believer in the fact that there are a bunch of great boxes on the market,” says Quinby. “It depends on the guy setting it up.”

That guy was Dowling, whose concerns were straightforward. “Headroom,” he says. “Horsepower really mattered on this one.”

That’s one of the reasons Sound Image specified Crown Audio I-Tech HD12000s amplifiers, he adds – three racks of eight I-Tech 12000s and another rack with four per side to drive 24 Adamson Y-Axis Y18 modules (3-way, dual-18) for mains, another 32 Y10 modules (3-way, dual-10) as side hangs, and 32 Adamson T21 SUB subwoofers (dual-21) per side.

Both engineers were pleased with the overall sound of the arrays, also crediting the efforts of Dowling and Adamson applications engineer Ewan McDonald, who traveled in support for a portion of the tour.

“It was pretty much on the money every night,” Quinby says. “It had the gas it needed to have and, most importantly, it transferred the impact from the band to the audience.”

That Big Rumble
Since ample headroom was most important in the low end, the subs (two stacks of eight T21s per side) were set up in an end-fire configuration.

“In end-fire, the rejection behind the array isn’t quite as good as a cardioid setup,” Dowling admits, “but it gives me a little more positive forward gain and cuts down on rumble where we don’t need it, which helps us toward more accurate low end.”

The T21’s Kevlar drivers also helped in that respect, he adds. “They’re more rigid, so if you hit an impact with the driver, it puts that into the room. I think it translates to a more accurate transient response, which is exactly what Tim is looking for.

In a mix like his, where there’s so much fast kick, that for it to not sound like it’s washing together, he needs a tight, punchy sub, but it still needs to be full frequency because he wants that big rumble at 40 Hz and below.”

“I use a tremendous amount of sub, but it’s very accurately tuned,” Quinby notes, adding that while the end-fire configuration did the job, he normally uses a cardioid arrangement. “I need a controlled pattern for the sub. We’ve got to focus it away from the stage, or the back wave from the left and right PA ends up meeting directly on our drummer’s seat and he ends up nauseous.”

Way More Detailed
While both Rob Zombie and Slayer offer up high-impact, high-volume shows, Quinby and Lonky’s needs, in terms of low end, were somewhat different. “The number of subs we had was plenty for me,” Lonky says. “Tim wanted to double them, but then he’s all about knocking the rivets out of these arenas.”

Quinby doesn’t disagree. “If I can register seismic activity with a Slayer show then we’re having a good day. That’s what the band wants me to do. That’s what their fans come to see. And I’ve become pretty detailed about it, but I can get way more detailed with more subs.

“We’ve done shows previously with Adamson rigs – the same amount of top end boxes in the air – but with 40 subs,” he continues. “I need a lot of output in order to cut problem frequencies using very narrow cuts, and maximum wattage to get a 60 Hz kick drum that feels like someone is beating you senseless. For what Joel was doing, it was great. For me, they sat on clip.”

The I-Tech amps internal processing handled all loudspeaker crossover, with the exception of the subs, which were crossed over with XTA DP Series processors. Lake Mesa Quad networked processors (with wireless tablet interface) were applied to overall system processing and management.

I-Tech amplifiers also drove the large-scale stage monitoring system, which was comprised of Sound Image (dual-12) wedges, L-Acoustics ARCS loudspeakers for side fill and dB-SUBs (triple-15) for drum fill. The monitoring system was headed by a Yamaha PM5D-RH console manned by Jared Woods, who did double-duty as monitor engineer for both bands.

For lead vocals, Rob Zombie uses a Beta 58A wireless capsule and a Shure wireless system. Beta 58As are also used for background vocals. Lonky mics guitarist John 5’s amps using a Heil Sound PR 31BW, a Shure SM57 and a Shure KSM44. Bassist Piggy D is taken direct.

“I use three different mics on guitar because they all have different tones,” Lonky explains. “I’ll pan them 3 o’clock and 9 o’clock, then put the SM57 up the center. I can pull it back and the left and right become less pronounced, or if I want to go with a big, in your face thing, I can bring the 57 up. I tune the left and right mics a little differently on the EQ to get a layered sound, like there’s not just one guitar playing.”

The drum kit of Ginger Fish gets the most mic attention - Shure Beta 91A and Heil PR 40 on kick, Audio- Technica ATM350 condensers on rack toms, Heil PR 28 on floor tom, and Shure Beta 56a (top) and Neumann KM 184 (bottom) on snare. More KM 184s handled hi-hat, ride, and overhead.

“You can’t go wrong with Neumann, it brings a lot of life to things having all them up there,” he says. “I love the ATM350s - they’re a great sounding tom and floor mic. The Heil PR 40 seems to translate the kick drum the way the artist wants it.”

For Slayer, Quinby notes that “kick drum is the main source for Slayer.” His approach is to outfit drummer Dave Lombardo’s kit with two Shure Beta 91As for the interior and two Heil PR 48s for the exterior of the kick drum. Heil PR 31s (top) and PR 22s (bottom) are the choice for snare, with dual Sennheiser e609s and PR 31s for toms and PR 40s for floor toms. KM 184s do hi-hat and ride duty, and Shure Beta 98s are applied to each cymbal.

Guitar rigs for Kerry King and Jeff Hanneman have the same treatment: dual Radio JDX DIs, two PR 40s and two PR 31s. Bassist Tom Araya has a Countryman DI for pre-signal and another for post. Araya’s lead vocals are captured with a Heil PR 35.

“I close-mic everything, making sure that the diaphragm of the mic is parallel to the source. The only mics that are away from the source are the external kick mics, and that’s only because I can align them with the 91s in the kick drums using Radial Phazers,” he explains. “Each of our guitar players has three heads and six live cabinets a side, so each head gets its own signal. I also align my guitar DIs and mics with the Radials, bringing them in to get a fuller mid-range sound.”

Console Viewpoints
Transferring impact was also a key driver in each engineer’s choice of consoles – a Midas PRO6 for Lonky and a Midas XL4 for Quinby – which led to an ongoing discussion of the merits of digital versus analog. While it’s an old debate, the co-headlining bill offered an interesting setting for it to take place in.

“You get to hear the exact same system, from basically the exact same starting place,” Dowling says. “And because they both funnel through Mesa, you could hear the flavor of each console when they tuned. There is no right or wrong. They both played to their strengths.”

“The PRO6 is one of my only choices,” Lonky states. “You can drive the front end like an analog desk without the audio penalties of a digital front end. Put it in the red - it doesn’t care.” Automation also plays into his choice. “The Zombie show has a lot going on. I’ve got 40 scenes on the console.”

Lonky utilized the onboard EQ to tweak the house as well as five or so other onboard plug-ins for most of his effects needs, including Midas’ pitch shifter as a means of emulating the sound of an Eventide 533 Voice Doubler. He does carry some outboard gear, including an “old favorite” TC Electronic 2290 delay as well as two Eventide H3000s applied to Zombie’s vocal because “he’s been using them since the White Zombie days and that’s his sound.”

Lonky also couldn’t resist attempting to convert Quinby to the PRO6. “I was like, ‘Look, I’ve got a 300-pound control surface, my stage rack, my little side rack, and I can do 90 channels. You’ve got a double 20-space rack and a thousand-pound console.’

“And the snake - I run two Cat-5 lines from monitors to front of house. One day our monitor engineer left the snake in the venue and I walked back in and carried it out in one hand – a 180-channel, 350-foot snake, and I’m taking it out to the bus in one hand.”

Pretty Good Aim
Quinby was unmoved, however, and insists he does travel light when necessary, confident an XL4 will be at the venue when he gets there.

The only thing he refuses to leave to chance is his compliment of Radial Phazer phase alignment tools.

“Even for festivals where I’m not carrying anything, I still walk in with six channels of Radial Phazers. I’ll put them in my luggage if I have to,” he says. “We have two kick drums, so I align each kick drum microphone with them, and also align my guitar DIs with their mics. Slayer’s a guitar and drum driven band, so the things that need to be the most accurately translated are guitars and drums. Radial Phazers and Empirical Labs Distressors are my favorite things in the rack, by far.”

If transportation costs or availability were to become an issue, Quinby admits he might consider another console. “I will use an XL8,” he notes, “because it has 24 faders, but on the PRO6 there’s 12, so when I have 23 inputs of drums I can’t see my whole kit.”

The issue isn’t necessarily a matter of sound quality, he adds, but workflow. “It’s not right for Slayer. I’ve carried the same setup for 10 years – the exact setup their previous engineer used. It’s been touring with them for 18 years. Realistically, as fast as they play, and running things at the gain structure that I do, I need to be able to make adjustments in milliseconds. The PRO6 sounds great, but it doesn’t have everything laid out in front of you.”

All that said, Quinby was prepared to switch over to the PRO6 if necessary, and carries a backup for whatever the second desk at front of house happens to be. “Kids at Slayer shows throw things. I’m going to make a show happen no matter what. I’m very aware somebody could take my desk out with a beer, but if someone takes out my XL4, I switch to the other console, the band still plays, and we still get paid.”

Given the potential for beer-related console damage, it’s worth pointing out that a desk like the PRO6 offers the additional benefit of presenting a much smaller target. “Maybe,” he concludes, “but Slayer fans have pretty good aim, and it’s getting better.”

Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.

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Monday, October 24, 2011

Extreme Makeover: A Look At Modern Power Amplifiers

It’s funny the difference that innovation can make...

The common “wisdom” just a few years ago was that the emergence of self-powered loudspeaker lines with enclosure-mounted amplifiers and protection circuitry tuned specifically for the loudspeaker meant the days of rack-mounted power amplifiers were numbered.

But it’s funny the difference that innovation can make. Today’s amplifier is a powerhouse, light years removed from the huge, heavy and inefficient designs of the recent past. Current designs are also far lighter, and in some cases, come in a compact package that more resembles a rack-mount processor than a device capable of generating several thousand watts of audio power.

The addition of DSP into the package hasn’t hurt either, offering convenience, space and cost savings, operating efficiencies, performance advantages and more. It also seems that some users will always prefer their amplifiers on the ground, where they can be quickly and easily accessed if there’s need for service or repair. This is particularly true for live events, where the show must go on.

The primary advancement has been to make the brutes more efficient. Greater output efficiency means less heat, and therefore, less weight, as well as more AC power making it to the loudspeakers.

The most common amplifier topology now is Class D (and variations), which 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 is greatly increased, and this is done without jeopardizing the integrity of the audio waveform by switching positive and negative output transistors on and off many times per waveform cycle.

This method is 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.

ProSoundWeb offers dozens of articles regarding amplifier classes, designs, data, testing, applications and more. In the meantime, take our Real World Gear gallery tour of audio innovation, the modern power amplifier.

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Wednesday, October 19, 2011

GC Pro Announces New Features & Functionality For Neve Genesys Mixing Console

Neve continues to expand its Genesys with enhanced functionality and new features

Guitar Center Professional (GC Pro) has announced new features and functionality for the Neve Genesys, a custom-crafted, expandable analog recording console that incorporates digital workstation control.

Exclusively distributed in the U.S. by GC Pro, the Genesys was designed to address new realities of modern professional audio industry for music, post production and gaming, while building on Neve’s 40 years of accumulated technical heritage.

New features being unveiled at AES include:
—New and updated Routing Implementation (L-C-R/Stereo)
—Pro Tools track arm/disarm control from the console surface
—Apple Logic Pro Integration:
——- Control of up to 6 banks of 8 faders each (48 Channels)
——- Extensive control of Pans/Sends
——- DAW Metering
——- Ability to ‘Flip’ Pans/Sends onto faders
——- Channels Encoder in DAW simulates V-Pot
—EQ/Dynamics processing can be placed in any order in the audio path using the keyboard/mouse with a graphical representation of the audio chain
—EQ/Dynamics Link/Copy/Paste feature
—Digital Line Enable on Channels with AES/FireWire
—Ability to Lock Monitor Level
—Improvements for Gangs/Trims, MTC operation, FireWire clock sync

The Neve Genesys base configuration offers 16 channels of mic/line preamps, 16-channel DAW monitoring, 32-channel analog summing at mixdown, DAW control for Pro Tools, Logic, Nuendo and more, 8 auxiliary buses, 8 group buses, 2 main outputs, 4 effects returns, comprehensive metering, 5.1 monitoring, 2 cue mixes, talkback services and an internal power supply.

It can be expanded to over 60 channels in a straight or articulated frame, with options including motorized fader automation, recall, mastering-grade A/D and D/A converters, digitally controlled EQ and dynamics, remote mic amp control and much more.

GC Pro

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Tuesday, October 18, 2011

Audio? Confusing? Learning Is A Life-Long Process

Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop...

Almost every Syn-Aud-Con seminar has attendees from other technical fields that need to learn about sound systems and audio.

These fields include networking, telephony, lighting, electrical and others.

Many tell us that audio is the most confusing thing they have encountered in their technical careers – and it is no wonder.

Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop.

There are nine (9) analog topologies and twelve (12) digital topologies. Each serves a purpose. Each works fine for its intended application. Each is defensible from a technical and practical point-of-view. Each will likely remain in use as other connector types and topologies emerge.

Consider also that some of these have several variations, such as the polarity convention on an XLR connector or AES3 on a DB25 connector.

It is ironic that digital I/O is often touted as making things easier, yet there are more digital connector types than analog! Add to this the confusion caused by the need to configure digital I/O for the correct sample rate, bit-depth, etc.

It’s no wonder that noise and distortion remain the weak links of most sound systems. They often result from feeding the wrong signal to the wrong jack.

We have all heard a DVD player over-driving a microphone input. Yes, you get sound, but in audio the presence of sound does not necessarily mean that you hooked it up right.

A modern digital mixer may be able to convert between any of these formats.

The signal may come in as some form of analog and go out as some form of analog or digital.

The user must often choose based on the required cable length, input options on the next device or some other criteria.

So not only does the audio tech have to understand the connectors and interface topologies, he must also know the characteristics of the devices at the other end of the cable.

The interconnect is the easy part. Consider the knowledge and experience required to configure a DSP for a 3-way loudspeaker.

Technical complexities aside, the most perplexing part of audio for non-audio people is the artistic side. While some levels can be set with voltmeters and analyzers, many other adjustments are based on subjective criteria – you just turn the knob until it sounds right. But what is right? There can be any number of “rights.” How confusing is that!

The really good audio people have strong theoretical and practical backgrounds. They have “must have” tools in their toolbox that you can’t buy anywhere. They have the ability to diagnose many system problems by just listening to a speech track played over the system. They can often make it sound way better by turning one knob a little bit.

Their personal study time may be divided between Sound System Engineering, product manuals and Einstein’s papers on relativity, and they understand that all three are completely relevant to what they do.

Most importantly, they know that they will never know it all.

Yes, audio is confusing. Background in another technical field can help, but learning audio is a life-long process.

Pat and Brenda Brown own and operate SynAudCon, conducting training seminars around the world in addition to providing in-depth web-based training.

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Wednesday, October 12, 2011

History Files: The Genesis Of Clair Bros To Today

The building of one of the most significant entities in modern touring sound reinforcement, and still going strong

The story of Clair Brothers starts in 1954, when a grocer decided to purchase a PA system as a Christmas gift for his two sons, Gene and Roy Clair.

“He had no knowledge of electronics or anything!” exclaimed Roy Clair of the extremely unusual present.

“I like to think my father was ‘Clair’-voyent in choosing this as a gift.”

The two brothers enjoyed using their PA to provide sound reinforcement for local dances, Easter egg hunts, etc.“The PA bug had bitten us!”

In 1963, Gene and Roy had purchased a loudspeaker re-coning business from a local music store.

This allowed them to acquire loudspeakers at the dealer level, granting the opportunity to build them for a local music store in Lancaster.

When musicians would visit from out of town to purchase loudspeakers, such as Baltimore’s Billy Joel Royal, it allowed Roy and Gene to go hear their products in use at local clubs.

“It was extremely gratifying, and I believe it was then that we realized that working with musicians would somehow be a fun career.

It was the same time we realized that having fun while making money was possible.”

F&M, a local liberal arts college in Lancaster, PA soon requested the brothers’ services to support headlining acts.

Now working in a 4,000-seat facility, one of the largest in the area, the duo would see their first brush with fame in 1966 when Dionne Warwick performed at the college.

“At the time, we had a Bogen MX-60, a few Shure microphones, and two column loudspeakers containing six 8-inch full-range loudspeakers each.

The concert went well, but looking back, we were lucky to start with an easy listening performer or things may have gone entirely different!

Timing and luck is something that has stayed with us our entire careers.”

Not long after working with Warwick, the brothers’ path would cross with Frankie Valli and the Four Seasons at the F&M venue.

Valli showed a vested interest in the duo’s Voice of the Theater A7-500 loudspeakers, particularly since the group had just performed in Miami, FL, and were denied the use of another artists’ sound reinforcement system.

“They were second on the bill to Herb Alpert and the Tijuana Brass at the Fontainebleau Hotel.

Alpert was not only a musician, but also a sound fanatic. It was no surprise that they were carrying their own sound system.

Unfortunately only Alpert would be allowed to use his system, while opening acts would have to settle for using the house PA system – even the Four Seasons’ wives and girlfriends noticed how much better Alpert sounded.

Timing and luck would strike again as our A7’s helped to make their F&M show incredible.

Valli felt they needed their own system if they were going to be successful on the road, and these two young lads were available – and cheap too!”

The brothers were working for $100 per show, including transportation, per-diem and hotels. They obviously weren’t doing it to make money at that point.

“If I remember correctly, after our first tour, in Ohio, we ended up with approximately $40 profit.”

Hardly a profitable tour, even for those days. The brothers weren’t aware of other sound companies touring like they were, but they assumed that they were one of the first to do so.

Touring with the Four Seasons and their continuing work at F&M necessitated a second sound system.

“In the beginning, I think we did a lot of begging and borrowing to do both accounts. Eventually we saved up enough and bought more A7’s.

However, with musical tastes changing as bands got progressively louder, we realized that our A7’s weren’t adequate enough anymore.

We used some of our A7-825 cabinets, and added more power by inserting two loudspeakers in the same-sized box. That seems pretty straightforward by today’s standards, but back then, it was innovation.

We had a slight advantage because we had a double-woofer, horn-loaded cabinet which was portable.

We added power with the first 300 watts per channel Crown DC300 directcoupled amplifier, purchased at the AES Show in 1968 from Clive Moore. It made us unique at the time.”

In 1968, a Cream concert at the Spectrum in Philadelphia, PA, was the now named Clair Brothers’ first large concert with 18,000 people in attendance. “Cream was big luck for us!” states Clair. “Luck and timing rides again!

Bob Kirnan, a sound and lighting technician from New York City who we met while touring with the Four Seasons, was contracted to do the show but was too busy. He recommended Clair Brothers to the show’s promoter.”

With their new Crown amplifiers, Altec Lansing cells with 288-C drivers, paired with Clair Brothers’ bass bottoms containing dual Altec woofers, they seemed to be the perfect fit for the in-the-round performance.

“Coming from Lititz, PA, we were extremely low-profile up to this point. That show in Philadelphia would soon change that…”

The Philadelphia promoter, the Electric Factory, soon started hiring Clair Brothers for shows, in addition to introducing them to many of the San Francisco bands that were successful at the time.

They also worked for the Belkin Brothers in Cleveland, OH, doing one-off shows.

“Their particular sound was instrumental in our company’s next step.

We started appearing on riders as one of the qualified sound companies for concerts, including Hanley from Boston, Kirnan from New York, McCune from San Francisco, and Swanson from Oakland.

Needless to say, Clair Brothers from Lititz didn’t get a lot of attention.”

As business started to increase, the brothers quit their day jobs and focused on building Clair Brothers full-time. They hired their first fulltime employees.

“We were lucky to have incredibly talented people from a rural area that wouldn’t normally be associated with the sound industry in larger cities.

Donald Gehman was our first employee, who did amazing things with Clair Brothers and went on to be one of the recording industry’s best engineers (R.E.M., Still, Mellancamp).

Ron Borthwick, with an EE degree from PENN State, is one of the best engineers in the industry, is still working for Clair Brothers to this day.

Dave Hendel, EE from Lehigh University, who moved on to a computer company. These were some of the few that gave Clair Brothers its start.”

The next four years, from 1968 to 1972, would see product development expand within the company.

Many “firsts” were built by Clair Brothers, including slant monitors, four-way sound systems, electronic crossovers (built by SAE), and the Elvis aluminum hanging system.

“The fourway systems contained W boxes for low end, a double-12 cone for the mid-range – built by Clair Brothers, JBL radials for the high frequency, and JBL for the super-high frequencies. Somewhere in between 1969 and 1970, Clair Brothers switched from Altec Lansing to JBL.”

The year 1970 also saw Bruce Jackson join the company, who would bring new design ideas to the company.

“We also added a lot of accounts at this time, both American and English. Blood, Sweat and Tears was a full-time account that gave us some financial stability.

We then later added Elton John, Moody Blues, Yes, Billy Joel, Bruce Springsteen, the Jacksons, etc., as accounts.”

In 1974, a large leap forward was made by the company with the creation of its S4, single-box loudspeaker system (the first all-in-one four-way box), with its hanging grid system.

Previewed on Rod Stewart’s tour that year, the S4 created industry buzz, to the point that when Mick Jagger came to Stewart’s show, Clair Brothers was hired for the Rolling Stones 1975 tour after he heard the system.

The S4 included high frequency drivers from JBL (2 x 18-inch, 4 x 10-inch, 2 x 2-inch, and 2 x 2405). Truck dimensions played an important role in the sizing of the S4, to allow them to fit two across in a standard trailer.

The S4 has lasted over 36 years, with updates as needed, allowing it to continually serve the touring industry.

The loudspeakers were even used in 2008 for the closing of the NY Mets stadium in New York City.

Clair Brothers’ patented i4 system along with its engineering digital processing continued to drive the company to the forefront of the audio industry. The Lake I/O originally designed by Clair Brothers, which was sold to Lake was a very important part of this next step in the history of innovation.

The dream that Gene and Roy started when they formed Clair Brothers is being kept alive with the second generation, namely Troy and Barry Clair.

The company has formed two divisions, with one son handling each one: Troy handles the touring side, while Barry has run the systems installation side since 1989.

Nearby Manheim, PA, is the site of the new facility has been constructed to house the systems division of the company, Clair Brothers Audio Systems, as the company has outgrown its headquarters.

The freed up space will allow the touring division to continue its own expansion.

Clair Brothers is proud of its rich history, from supporting just one show per night with the Four Seasons in 1966, to now delivering high quality sound systems to a multitude of world class acts, night after night, the world over. The story continues.

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Tuesday, October 11, 2011

How To Build Your Own Plate Reverb: A Concise Step By Step Process

Back by popular demand, here's a discussion of the plate reverb, as well as precise instructions on building a plate reverb unit.

As Larry Crane, editor of Tape Op magazine, noted, “Plate Reverb. Many people ask me about this and I usually tell them to listen to some records from the ‘70s and ‘80s and look for reverb with a thick, pillowy sound that doesn’t obscure the vocal yet doesn’t quite sound like an actual room.”

In 1983, I was the owner of a 16-track studio. One of the things that really separated the sounds of the recordings we could get from the sounds of the recordings made in major facilities was the quality of the reverb.

Spring reverb was the only affordable system for small studios at that time, since EMT plate reverbs ran almost $9,000!

EMT’s patent was about to expire, and when it did, competitors came out with similar products. While they were cheaper, they still averaged $2,500!

So an engineer who worked with me, Joe Errico, and I researched plate reverbs and came up with an affordable way to build one.

This article presents our plans for making a plate reverb unit, which won’t require any electronics other than your mixer and a headphone amp. (If you don’t have these items, you’re not ready for a reverb plate anyway.)

The construction cost will be between $100 and $500, depending upon what components you already own - a lot less than the $2,500-$8,500 for commercially available units.

Later in the article, I’ll also detail how to find and evaluate the materials needed, construct the frame, mount and tune the plate, fit the driver and pickups, and add dampening to the plate.

It concludes with some “tricks” and techniques for enhancing plate sound.

Almost everyone with a knowledge of recording is familiar with spring reverbs, or at least with their sound. (They were the most common type reverbs used in studios when this article was originally written. Now digital reverberation units are the type most often used.)

Most low-end or semi-pro reverb units were based on the spring principle, as are most musical instrument amps or accessories with reverb. That “spring sound” can range from excellent to “under water,” depending on the unit and the way it is used.

The reason spring units sound the way they do is because that is exactly what they are; springs. There are usually several rows of them, possibly with two or three strung in a series. Just like the springs on your screen door, they will “twang” or “boing” when plucked.

However, instead of being plucked, the reverb springs are excited at one end by a driver and mic’d at the other end by a pickup - and so are the twang and boing, especially on transient material.

Although some designers have used tricks to smooth out their sound with excellent results, they may still have spring characteristics inherent in their sound, as well as a limited bandwidth, especially at high frequencies (8 kHz+).

Plate reverb has none of these drawbacks, although it can go from sounding like a true concert hall to an oil drum being banged with an ax in the subway, again depending on its application and who’s using it.

Typically, the plate is a large (one by two meters, or 39.37 by 78.74 inches) sheet of steel suspended in a tubular steel frame.

In theory, the plate simulates a large concert hall, or church, with a decay time (RT-60; the time required for the level of the reverb to diminish by 60 dB) of approximately five seconds at approximately 500 Hz. A driver attached to the plate excites it, and as the sound waves travel through it, the plate flexes.

The plate’s motion is then picked up by one or two contact mics, and added to the dry signal at the mixer. Transients do not twang or boing, but behave much as they would in a reverberant room, sounding smooth and natural.

As an additional feature, incorporating a damping plate to change the decay time of the reverberated signal can be included in the design.

It was at the Broadcast Technical Institute in Nuremberg, and later at the Institute for Broadcast Engineering in Hamburg, West Germany, that the first reverberation plate using these principles was developed.

EMT (in Germany) patented and made the only available units until the patents ran out a few (now 25 or so) years ago. Since then, several American and foreign companies have come out with newer units.

The plans presented here are of a hybrid unit that can be optimized to the design of any of the commercial units you may favor.

Construction
As mentioned in the introduction, the design of this unit will incorporate your mixer and cue (headphone) system as all the electronics that are required.

We will mostly concentrate on the construction of the mechanical system and the transducers-the frame, plate, driver, and pickups.

This is probably the most critical of all the steps involved in the process, so be careful. The plate is actually “the instrument” used for the reverb, so it should be chosen as if it were a fine acoustic instrument.

EMT used a 1-meter by 2-meter cold-rolled steel plate approximately 1/64-inch thick. Lawson, who manufactured “The Plate” (LP1 and LP2), used basically the same size plate, but it’s a little thinner.

On the other hand, some manufacturers used stainless steel. The Ecoplate by Studio Technologies used approximately the same gauge in stainless, as did Audi-ence, while DB Cassette of Sweden, who manufactured the Stocktronics Plate, used a stretched, hardened piece of cold-rolled stainless approximately 0.03 inches thick.

The question of what kind of steel to use is totally subjective. Reasons claimed for using stainless steel include consistency, high density, and the fact that it’s tarnish proof, while regular steel users claim smooth, more natural sounding reverb and a less “metallic” decay. Only you can decide what sound you prefer.

Befriend your local steel warehouse owner, bring two associates, and prepare to listen. Most steel sheets come in 3-foot wide sheets; this is close enough to 1 meter for our purposes.

The length, however, is usually 8 feet, and cutting charges to make it 6 feet might be added to the price of the steel. Some places also have minimum orders, so try to buy your plate and frame materials from the same source to save added expense.

If the owner of the shop will allow (and it’s worth a healthy tip to have him help you out), have your two friends hold the sheet of steel horizontally as tight and still as possible, such that it doesn’t “thunder.”

Tap it in the center with a key and listen for a “sizzle” and long decay in the high frequency, as opposed to a “clangy” sound.

The delicacy and length of the high frequency decay are what you are really after, since the bottom and mids can be dealt with more successfully by tensioning.

Try several pieces of different types until you find what you want. Be selective and take as much time as possible, because this is the heart of your system and you must be happy with it.

Including the cutting, the steel sheet should run between $50 and $100, depending on the type you choose. Reinforcing the corners by spot welding a triangular piece of steel on each one is the recommended procedure.

For corner-cutting by the cost conscious, however, it’s not totally necessary, since this can run an additional $25 to $50.

However, it really should be done if at all possible, because the plate will be put under heavy tension, and holes will be drilled in those corners later in the plate-preparation procedure.

The holes should be drilled after the frame is completed, so a more “custom” fit may be made.

The Frame
The frame is simply 1 to 1–1/2 inch tubular steel, either round or rectangular-shaped, and welded together at the (preferably mitered) corners.

Simple angle iron can be used, if absolutely necessary.

The frame should be reinforced by three transverse beams (Figure 2). Near both sides of each of the four corners (eight all together), weld flat pieces or slats of steel, which may be channeled for extra strength.

Note that Joe and I enhanced the design by adding 2 extra tensioning slats in the center of the long dimension of the frame, to assist in the tuning of the plate.

The slats should extend 1-1/2 to 2 inches beyond the frame, and be about 1 to 2 inches from the corners. Holes will also be drilled in the center of each of the slats.

To determine the exact placement of the holes in the slats and in the plate, as well as the exact measurement of the length of the tubular steel for the frame, you must make sure the plate and the frame line up together.

Make the inside measurement of the frame 1 to 1-1/2 inches larger than the dimensions of the plate. Then lay the plate on top of the frame.

On the plate, mark the eight spots where the holes will be drilled. Then mark the frame where the eight slats will be welded. Next, mark the slats where the holes will be drilled.

When all the holes are drilled and the slats welded in place, paint the frame to stop rust and corrosion.

The next step is to suspend the plate in the frame. EMT used spring clips that held the plate in place and were also used to determine tensioning. These are weak, and often snap.

One of the improvements made by most plate manufacturers was to use stronger, heavier clips or hooks. Ecoplate used clips similar to those that secure fiber straps used on packages.

We will use simple, tempered, hardened-steel hooks, threaded on their shafts. (Joe and I now utilize fiberglass clevis yokes with machine shop quality allen bolts to better control the tensioning and Plate/Frame isolation.)

If the hook is plastic-coated, and hard-rubber and metal washers are used, the plate and the frame can be totally isolated as far as direct metal-to-metal contact goes.

To suspend the plate, you will probably need help getting the hooks through the holes in the plate.

Slip the shaft of each hook through the holes in the slat; thread the washers and a nut of the correct size on the hook shafts, and hand-tighten all nuts (Figure 3).

The plate can now be suspended from the frame.

Now comes the subjective and fun part of the project-mounting the driver and pickups, and tuning the plate.

The Driver
EMT, Ecoplate, and Lawson all used similar drivers. A bullet-shaped metal moving-coil “slug” is screwed into the plate. Two wires carrying the signal go to the coil and it is suspended in a large, heavy, circular magnet (Figure 4).

It is important to be sure the moving coil assembly does not rub or touch against the sides of the magnet. The coil and magnet are aligned using a plastic alignment disk.

The procedure is delicate, and transporting the unit sometimes misaligns the driver/magnet assembly.

Stocktronics and Audi-ence both used a wire rod attached to the plate on one end, and to the voice coil of a speaker on the other.

It can be moved with no realignment, since there is plenty of “play” in the movement of the rod, and this is restricted to within limits by a rubber guide.

The system we use is similar to both, but unique unto itself. It is also one of the main reasons that this plate can be built so reasonably.

This design uses a specially designed driver-similar to what used to be offered as a “coneless speaker” several years ago.

Whatever the driver is attached to becomes the “sounding board” and vibrates enough to reproduce sound.

Therefore, if screwed into a door, that door would become a “speaker.”

The specially designed driver (Figure 5) is an improved version of the coneless speaker.

To install the driver, simply drill a small hole, the size for the screw on the driver, in the center of the plate 2-1/2 inches off to one side of the center of the beam of the frame.

Screw the driver in the side of the plate, with the frame reinforcements toward you, about half way until tight. Attach a speaker cable to the two terminals on the driver.

Be neat and run the cable down the reinforcement with “ty-raps” or tape.

Now the fun begins.

Move the plate into your studio. Make sure it is standing upright. Connect the other end of the speaker cable attached to the driver to the output of your cue (headphone), or dedicated tube, class A, MOSFET, etc., amplifier.

Put on a tape with a steady snare-drum track or a constant vocal track. Send only the selected track to your cue and, voila, the signal will be heard on the plate.

Assuming it is the snare track, what you should hear is a thunderous snare sound similar to “Bridge Over Troubled Water” or “The Boxer” by Simon and Garfunkel (although I think they used an elevator shaft for their reverb chamber).

But, anyway, congratulations! You have a working plate reverb!

Now comes another fun part - using your opinions, taste, and ego to get it to sound just the way you want. This will require your choice of pickups as well as tuning and equalization.

Pickups
All the commercial units use piezoelectric pickups or accelerometers. These are basically contact mic/pickups and are available from several manufacturers.

As a matter of fact, you probably already own one, or at least know someone who does. Examples of available units are those from Barcus-Berry, Fishman, and Frap.

Some pickups need no preamp and can be plugged into the echo return(s) on your board.

You can also return the output of the pickups through two mic inputs on your mixer, using a tube, discrete class A, transformer or transformerless (differential), FET, IC DI.

These are some of the variables that you must work out depending on the console you own, and the pickups you use.

Try as many pickups as you can borrow until you find one that you like and that easily interfaces with your mixer.

For a mono reverb, place the pickup near one of the side frame reinforcements. Experiment by moving the pickup up around, and down, on both sides of the reinforcement, until you find the spot you like.

Then secure the pickup by epoxy, wax, putty, or whatever the pickup manufacturer recommends. Run the pickup wire down the reinforcement, again using “ty—raps” or tape.

For a stereo unit, do the same thing on the other side (Figure 6).

Tuning The Plate
With the pickups in place, the plate itself now comes into focus for tuning. In theory, think of the plate as similar to a drum head. Also, correct tuning means all the lugs are equally tensioned.

So, start by holding the hooks suspending the plate in the frame with a pair of “vice grips” or similar pliers, and tighten the nut on that hook with a ratchet wrench. Do this evenly around all eight hooks.

How do you know when the plate is tuned? Good question. You don’t, really, because every manufacturer used their own method for tuning.

EMT shipped units pre-tuned except for four nuts, which were supposed to be tightened by exactly 1/4 turn when installed. Most independent EMT servicemen will tell you to tighten each one until a spring suspension clip breaks, and then replace it and tighten until 1/8 of a turn before it breaks again!

Lawson and Audi-ence shipped their units pre—tuned, no adjustment necessary. Ecoplate supplied a spring gauge and specified pushing the gauge against the plate at all eight plate tensioning points, until there was approximately 150 pounds of pressure at each point.

As a total contrast, Stocktronics used no tuning at all, claiming the steel was pre-stretched/tuned during manufacturing. In fact, their plate was simply suspended by six springs in a very light aluminum frame.

Which method is correct? Any/all/none, depending on your point of view. One thing is certain, though. If you like the way it sounds, it’s right. So I suggest tuning by ear.

Remember, the tighter the plate, the tighter the bottom. It is usually better to over-tension than under-tension. Also, listen for “flutters” or “beats” (like two slightly out-of-tune guitar strings) on the decay of the reverb, and even-out the tension until they disappear.

EMT warned about the “oil can effect,” a very metallic sound that is heard on an obviously out-of-tune plate.

What I suggest is to find an existing plate you like in a studio near where you live. Rent an hour of time, and bring along a tape of various tracks-snare alone, drum kit, congas, tambourine, voices, piano, and run it through the plate.

Record the reverb return on one track of a two -track or cassette, and your original dry signal on the other. Bring it back to your place and pan the dry signal to the center of your monitors, and the reverb send from their plate on the right.

Send the dry signal to your plate, return it to your mixer, and pan it to the left. Now you can directly compare your plate to theirs, and tune and equalize until the you sound of is equal to, or better than, theirs (again, subjectively speaking).

A Case For The Plate
Theoretically, you’re done - but you really need a place to put your unit and something to put it in.

The best place would be a separate quiet room or closet so that no outside vibrations will affect the plate. Even so, a case for the unit is suggested. The case is simply a wooden box that the frame can sit in.

EMT, Audi-ence, and Ecoplate used pressboard, Lawson used plywood, while Stocktronics used only paneling.

The frame can be placed in the case on rubber feet, or better yet, suspended in the case using rubber straps with hooks, such as those found in automotive stores for holding down luggage.

The straps can be wrapped around the frame and the hooks hooked to holes or eyelets in the case. This way you can literally pound the case with little vibration.

Eyelets can also be put on the outside of the case on each side so that rods can be inserted for carrying.

If you’re only using the plate during mixdown, the studio isn’t a bad place for it. It probably has the best isolation from your monitors and has easy access to your mic inputs and headphone jacks.

The case only has to be a few inches bigger than the entire unit on each side, unless you plan on using the next step - damping.

Damping
The decay time for the reverb as it now stands is approximately 5 seconds at 500 Hz.

This is fine for most applications, but is easily altered by fitting a damping plate, which can be a piece of plywood the same size as the plate and covered with an absorptive material (such as compressed fiberglass, styrofoam, or foam rubber) that can be moved closer to or farther from the plate to alter the decay time of the reverb.

EMT, Lawson, Audi-ence, and Ecoplate all moved the damping plate in parallel to the steel plate, from almost touching (1/8 inch) to 6-8 inches away. This is accomplished by forming a parallelogram type set-up where two metal arms attach to the frame and to the damping plate so that when the damping plate is moved, the arms travel sideways and move it closer to the steel.

Stocktronics simply hinged their styrofoam damping plate at the bottom and then pulls the top closer to, or farther from, the steel, claiming this gives a more uniform frequency response in the decay characteristics. A handle or lever on the damping plate facilitates moving it.

It can also be remote-controlled using servo motors and cams, but this is beyond the scope of this article. The choices of materials, method, or even use of damping at all is left up to you.

Plate Tricks
Using equalization will help you get the reverb characteristics that you are after much easier than tuning alone.

In fact, all the commercial units have some sort of equalization in their electronics, either a bass cut-off on the pickup amps, a high-frequency boost to the drive signal, or both.

EMT cuts the bottom at 80 Hz, but many engineers use a 700 Hz high-pass filter to accentuate the top.

If you have a few equalizers to spare (i.e.: tube, Class A, graphic, parametric, etc.), it would be a good idea to patch one to the send and one to each return. This will allow you to match the sound of almost any of the commercial plates or any plate sound you have heard. (We have designed a drive signal response shaper that we feel emulates the sound of our favorite EMTs.)

Take the send to the plate and first put it into your delay line. Use a full-bandwidth setting so that you don’t lose any top end.

The effect is that you’re in a large hall where the first reflection isn’t heard until milliseconds after the initial dry signal. The longer the delay, the bigger the hall.

A good example of an extra long pre-delay is the reverb on the snare at the end of “It Keeps You Running” by the Doobie Brothers. You hear the snare hit first-and the reverb later. Sort of “boom ... cha!”

This will also bring out the deficiencies of a unit, and if you try it on a twangy spring, the time delay doesn’t let the program mask the boing of the snare transient. But with a plate, this is no problem.

To shorten the decay without damping, a noise gate comes in handy. Placed on the return, the release time can be shortened.

When the attenuation and threshold are properly set, the decay will be gradual and smooth, only shorter. If the controls are set to dramatically attenuate the decay, it can be rhythmic.

For example, if hand-claps are done on the downbeat, the reverb decay can end sharply and completely on the upbeat.

You can also gate the send to the plate such that you only reverberate certain signals. For example, if you want reverb only on the snare track, and it wasn’t gated when recorded, gate the snare track to the send, and you will only get the reverb on the snare beats, not on any tom toms, bass drum, or cymbals that might have leaked onto your snare mic.

If you drive the plate a little harder, the effect will sound like a series of fading repeating reflections analogous to what it looks like when you drop a pebble in a pond.

Early reflections can be achieved by using a digital delay, in combination with the plate, as well.

Experiment and you can get any sound you’ve heard, and some you haven’t.

“So, if I use one plate reverb with a lot of top end and a gate for the snare, and another one with a lot of bottom for `thunder toms,’ and one more with a long pre-send delay and high frequency boost for that `sizzly vocal’ sound; maybe one for the strings… with maybe a little flange on the return ... and maybe one more….”

Plate Reverb Required Parts
1 x Steel Sheet (your choice)
Approx 30 feet x Tubular Steel or Angle Iron (for frame)
10 x Threaded Rubber Coated Mounting Hooks
10 x Nuts (to fit Hooks)
10 x Rubber Washers
10 x Fender Washers
1 x Driver (your choice)
1 or 2 x Pickup Transducers
1 or 2 x Preamps, DI boxes, Transformers for Pickups (if needed)

Optional Parts
1 x Case (your choice of style and material)
1 x Damping Mechanism with Absorptive Material (your choice)
Suspension Method (to isolate plate and frame from case)

 
Editors Note: Since this articles original publishing, DIY plates have become even more popular. While the parts Bob referenced are increasingly difficult to source, a good starting place for those interested is his website.

Bob Buontempo has more than 30 years of professional recording experience, and has been the president/owner of Buontempo Entertainment Services since 1976. He has also taught numerous recording and audio educational courses over the years.

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Monday, October 10, 2011

Aerosmith & Producer Jack Douglas Use Endless Analog CLASP On Sessions For Upcoming Album

"It has that rich, fat sound of analog and we have the convenience of being able to edit in Pro Tools." - Producer Jack Douglas

Sessions for Aerosmith’s new album has them collaborating again with producer Jack Douglas, who helmed sessions for the band’s classic 1970s LP’s Get Your Wings, Toys in the Attic and Rocks, and indicating that the band is trying to capture an old-school, classic vibe on the new set.

Helping Douglas, engineer Warren Huart and the band achieve the right sound is an Endless Analog CLASP (Closed Loop Analog Signal Processor) system, which integrates analog tape into the digital audio workstation environment. In fact, they are using four CLASP units on the sessions.

The band began sessions in July at two Aerosmith-affiliated Boston-area studios: Pandora’s Box, which the band owns, and the Boneyard, guitarist Joe Perry’s personal studio.

Two CLASP units are present at Pandora’s Box – one connected to a 16-track two-inch Studer A800 analog tape machine for recording drums, and one connected to a 24-track two-inch A800 MkIII for recording everything else.

At the Boneyard, another CLASP is connected to an additional 24-track two-inch A800 MkIII for overdubs.

A fourth CLASP will adorn a yet-to-be-announced Los Angeles studio when sessions move to the west coast later this year.

Douglas, who, aside from his tenure with Aerosmith, is noted for his work with John Lennon, the New York Dolls, The Who and other classic acts, states, “CLASP is revolutionary. We’re sticklers for big, fat analog sound, so this CLASP system was just perfect for us. And we’re so happy that Chris (Estes, CLASP inventor) came along and was able to hook us up with the stuff, and we really love it.

“The album’s gonna sound amazing,” Douglas continues. “It has that rich, fat sound of analog and we have the convenience of being able to edit in Pro Tools, so we really love CLASP and what it’s let us do. Normally, if we were using tape, we’d be using reels and reels. But CLASP lets us use the same reel over and over again. And then normally we’d have to break for a few days to dump everything into Pro Tools for editing, but CLASP allows us to do real-time transfer. And we can record at any tape speed we want.”

Huart, best known for his work with top-selling acts like The Fray, Augustana, Kris Allen, Better Than Ezra and many others, adds, “The reality is that recording with tape just makes things sound better. Every single piece of digital equipment you buy tries to make it sound like tape. CLASP is fantastic because it does what digital is trying to do for you by actually using real tape. It’s been a godsend in the studio as Aerosmith is capturing such a powerful, classic sound.”

Brad Whitford, one of Aerosmith’s two guitarists, notes, “Absolutely amazing sound, the CLASP. You forget what it sounds like to hear these instruments actually going to tape. It’s the only way to go. You hit that tape, and you get all those rich harmonics. People are going to hear this album, and they’re going to say, ‘There’s something different going on here,’ and the difference is CLASP. CLASP all the way.”

Aerosmith’s upcoming LP is slated for release in spring 2012.

Endless Analog

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