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Tuesday, February 07, 2012

Church Sound: How To Transition From Analog To Digital Mixing

A digital mixer is a whole new way of doing the same old things

 
I’m in the process of helping one of my churches transition from an analog mixer to a digital mixer.

They were in need of more channels than their Allen & Heath 16-channel MixWiz with some outboard gear (front of house EQ, couple of compressors, effects unit) could provide.

Based on the maximum number of channels that they anticipated needing over the next five years, I recommended the PreSonus StudioLive 24.4, one of the least expensive 24-channel digital mixers on the market.

The church has two audio volunteers that are pretty much average in their knowledge of sound and sound systems so this would be a typical transition for a lot of churches in the 100-400 person attendance range. Volunteers selected more for their willingness to serve than their knowledge of audio. I know that nothing has been touched with the front of house EQ, compressors and FX since I helped them set it up about a year ago.

Some things that you need to consider in this transition is how uncomfortable the volunteers are going to be until they make the paradigm switch from the analog WYSIWYG (what you see is what you get) to the digital layers.

Depending on the digital board, layers control everything from different grouping of faders (1-8, 9-16, etc) to control over the aux sends, FX, etc. Outboard gear usually goes away and everything is now handled with the digital mixer. It’s a big transition and you shouldn’t minimize it, but treat it with care and planning and the transition will go smoothly.

Getting Started
What I recommend is that the digital mixer not be put into service immediately but be brought into a two-to-four-week training duty cycle. It requires some mics and cables as well as a couple of speakers for monitors and front of house stand-ins. If you have instruments that you can plug in that helps as well. Keep the existing analog system going as the production system until everyone has been trained and is comfortable with the digital board.

Before you start with the digital mixer, make sure everyone has reviewed the user manual. A digital board is a computer with knobs and faders and is significantly more complex than an analog mixer. While they are pretty robust, you can still mess them up and repairs can be costly.

An Investment of Protection
One thing to invest in if you haven’t is a top-line power conditioner like those from Furman. I also recommend a computer UPS (battery backup) from a company like APC or Tripp Lite. Get a decent capacity one. The reason is that because a digital mixer is a computer, when power is interrupted you can’t just switch it back on like an analog mixer. You need to boot it up and, depending on the mixer, that could take anywhere from a minute to several minutes.

Having a UPS unit, the mixer will stay powered on, so even if the rest of the system is knocked offline by the power interruption, when the power comes back on, the mixer will still be up.

Unboxing The Mixer
Once you get the mixer unboxed, check for any damage. If everything looks good bring all faders down to minimum and turn on the mixer. I like to let the mixer “burn in” for about four hours with nothing going on or plugged in just to let all the electronics warm up to full operating temperature. This will check to ensure that nothing is shorting out. Be aware of any burning electrical smell or smoke. If you detect either one shut the mixer down immediately and unplug it. Contact the vendor.

Preparing For Training
The StudioLive is close to an analog board in that all the channel faders are on one surface as opposed to layers. This makes the transition somewhat easier. All effects, aux send levels are controlled through the center “Fat Channel.” That will be where most of the confusion is going to come in so be prepared to spend a lot of time going through this area.

The StudioLive is set up pretty easy so I was able to figure 85% of the board out without looking at the manual. There are also a ton of video tutorials on the PreSonus site and YouTube that can help with anything to do with the board. But for volunteer sound techs it will be a bit of a challenge.

Building A Mini-System
Hook up a mic to channel 1 on the mixer and hook up a speaker to aux send 1 and to front of house. This will be the basic training setup.

Once you get it hooked up, bring up the gain to an appropriate level. A digital board is less forgiving about exceeding the 0 level than an analog board before going into clipping so run the level less than needed for training until you get comfortable with the way the board handles signals.

Don’t worry about EQ settings or FX yet. All you want to do is to learn the signal flow from the channel to the aux send and FOH.

Once you’ve figured out how to adjust the aux send levels for the channel and you can adjust FOH level you’ve gotten over the initial hump.

Using EQ
The next thing you’ll want to learn is how to adjust EQ’s for each channel. Depending on the digital mixer you’ll either have a screen that will have a parametric equalizer, or in the case of the StudioLive, you’ll have the knob adjustments for high, high mid, low mid and low bands. As with all digital mixers you are able to set the frequency points for all these bands as well as the Q, which is the width of the frequency adjustment. This is a lot more adjustability than what an analog mixer has and is worth spending some time practicing.

After the channel EQs get figured out you’ll want to adjust the front of house EQ. On the StudioLive it’s set the same way that the individual channel EQs are set. One nice advantage about digital mixers is that most of them have a library of preset EQs that you can start with. The StudioLive has built in a nice set of professional quality EQ presets that are good enough to leave alone and assign to each channel.

The other nice feature of digital boards is the ability to save all your settings to a scene. So you are able to set up multiple scenes for different worship teams or different instruments and recall them just by dialing up the scene and pressing the load button. So no more needing to reserve channels based on who’s playing that day.

Enter Effects
The power of digital mixers means that you can assign FX to each and every channel, both to auxes and to front of house, so you’ve got a lot of flexibility. Just remember that just because you can doesn’t mean you should. Less is more, at least in the beginning. Some boards give you more FX capabilities than others. The StudioLive offers two channels of FX, others more.

Multi-track Recording
Another advantage that digital mixers have is that they usually provide some form of multi-track recording capability. In the case of the StudioLive, it’s provided by a FireWire port into the provided Studio One software. This means you can record each channel separately into your computer, as long as it has a Firewire port.

One very cool reason for doing this for the worship team is the ability to do what’s called a Virtual Sound Check. What that means is that you don’t need the worship team there to set up the board. You can play back the individual tracks back into their respective board channels and use those tracks as the sound check.

Then, once the band gets in, sound check is very minimal. It’s also a great way for the sound team to train on the board and allows them to massage settings without needing the musicians.

Saving Scenes
Once you get everything set the way you want it remember to save your settings to a scene. I usually recommend naming the scene with the church name and 1. That way you can always recover your baseline settings.

Sound techs should create their own “sandbox” scene, which allows them to manipulate settings and save it to their own scene without affecting the master scene. Make sure that no one other than the lead sound tech saves to the master scene.

Once you’ve got the master scene saved it won’t matter what changes people make to the board during the week. Bringing back the master scene will only require a quick push of a button, and in the case of the StudioLive, resetting the gain and adjusting the faders. In other digital boards, gain settings and fader positions are saved within the scene.

Making The Switch
Once the sound techs are comfortable with the digital board then it’s time to switch out the old analog board with the new digital one. Check all your settings. Be sure any settings you change are saved to the master scene once you’re happy with how everything sounds.

Finally, when you shut things down, do NOT shut things down by just turning off the power conditioner. This WILL damage the digital mixer. Follow the shutdown procedure in the manual. It can be anything from just powering off the mixer with the mixer’s power switch to a shut-down procedure on the screen.

Summary
A digital mixer is a whole new way of doing the same old things. It’s exciting as well as terrifying for volunteers, so go slow. Take it one step at a time and ensure they are comfortable with the new system before putting it into production. You’ll achieve a seamless transition and have fun doing it!

Brian Gowing has helped over 30 churches meet their technology requirements. Brian works towards shepherding the church, analyzing their technical requirements, sourcing the equipment, installing the equipment and training the volunteer personnel.  As he likes to say, “equipping the saints with technology to help spread the Good News.” Contact Brian here.

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Thursday, February 02, 2012

Church Sound Files: The Reason For “Bad Sound” May Not Be The Sound System

Three factors, roughly equal importance, play the key role in good sound - and “two out of three” isn’t good enough

Many things around us are getting better. Computers are faster, televisions have more resolution, and dishwashers are quieter and more powerful than ever.

But with all of our digital wiz-bang processors, technology has been unable to eradicate “bad sound.” Why is this so? This short piece is an attempt to shed some light on three possible causes, two of which have been completely unaffected by the technological revolution.

The goal of most sound reinforcement systems is to deliver high quality sound reproduction to the listener. While we would like to think that a high quality sound system guarantees this, it does not.

The quality of the reproduced sound will only be as good as the weakest link in the reproduction chain. Let’s examine some of the major “links” individually.

The Room
The room is a major factor in the reproduction chain. Most large spaces are hostile environments for sound systems, unless they have received special attention from a professional and a considerable financial investment from their owner. Good acoustics doesn’t just “happen.” It is the by-product of careful planning.

A quality sound system may radiate an exceptionally high-fidelity sound field into the room. Unfortunately, most of the radiated energy will create acoustic events that detract from the listening experience. While small rooms have their share of acoustic problems, these problems pale next to the late reflections, reverberation, and energy build-ups encountered in large spaces.

If your sound system doesn’t sound good, ask yourself the question “What have I done to provide a good acoustic environment?” If the answer is “nothing,” then you got what you paid for.

The Sound System
Of course, a good sound system is a vital link in the reproduction chain. But this doesn’t just mean expensive equipment. It means that equipment that is suitable for the environment has been selected and implemented by someone who understands the compromises involved in large room reinforcement systems. Money can be wasted on “features” that offer no real benefit for the large room environment.

The vast majority of auditoriums that I have visited are not suitable for multi-channel formats such as stereo, surround sound, etc. since each channel must be delivered to all listener seats. Loudspeaker placements that are optimal for stereo reproduction are horrible choices for single-channel systems.

Even with monaural systems, “first choice” loudspeaker placements often create problems with sight lines and aesthetics, and are therefore ruled out by venue owners. Multiple loudspeakers must overlap somewhere, and there will be sound problems in these areas.

A properly designed system will often sound bad in the aisles – the very place where casual onlookers will stand to evaluate it. We all have good sound at home, but the rules change as the listening space grows. Intuition that is not filtered through the proper large-room principles leads to errors.

Sound system designers are often forced to compromise away the performance of the system to make it fit aesthetic concerns, budget limitations, and fashion trends within the industry.

The Operator
I’ve intentionally saved this one until last. The most overlooked link in the chain is the end user of the system. This includes the mixer operator and any supporting staff, such as those who run the monitors and place microphones.

A monitor system that is too loud will dump excessive energy (usually low/mid frequency) into the audience area. This excess energy will upset the spectral balance of house sound system, tempting the front-of-house operator to compensate by over equalizing (usually in the form of high frequency boost). This results in a reduction in gain-before-feedback and an unnatural sounding system. Microphone placement is equally critical, as is an understanding of the shortcomings of various miking techniques.

If a lapel mic could sound like a hand-held, then no one would use hand-helds. The overhead drum mic that captures the cymbals also captures the stage monitors and “spill” from other instruments, as does the vocal mic used at arm’s length. And that “mellow” bass guitar sound that the musician likes in the practice hall turns to “mush” in a large space, where increased definition provided by the use of a pick and brighter strings may be required.

These factors and many more “eat away” at the sound quality of the system as a whole. A good mixer operator will evaluate and optimize the sound of the instruments individually before allowing the band to perform as an ensemble. There’s no room for democracy here – effective mixer operators learn to say “no” and “be quiet.”

A question that I recommend for an interview of prospective mix personnel would be “What will you do if something starts to squeal?” If the answer is anything other than “Turn the offending channel down slightly until I figure out what the problem is” move on to your next applicant. Filters implemented in desperation do nothing to preserve sound quality.

Modern mixing consoles pack a considerable “wow factor.” It’s fashionable to sit behind a large one and move knobs all of the time. But doing so doesn’t make one an engineer. Completing an accredited academic program or piloting a locomotive does. The decision as to which console to purchase is often made with no consideration as to whether anyone at the facility will be able to operate it. The result? Bad sound.

I have personally witnessed the performance of many good sound systems ruined by bad rooms and incompetent operators. I have also seen skilled operators “salvage” the sound reproduction in situations where the room and system were less than optimal.

The performance of a sound system is only as good as its weakest link. Unfortunately, all of the links that I have mentioned are of roughly equal importance, meaning that “two out of three” isn’t good enough. Good sound requires all three.

Experienced, well-trained audio people realize this and are there to help you find your weakest link. Pay for their advice and follow it.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. Synergetic Audio Concepts (SynAudCon) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, SynAudCon is dedicated to teaching the basics of audio and acoustics. For more information, go to http://www.synaudcon.com

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Thursday, January 26, 2012

A Look At Microphones Of The Past With Recording Legend Bruce Swedien

An excerpt from his book, "Make Mine Music," rife with need-to-know history and personal stories.

This excerpt is the first in a series from Bruce Swedien’s book Make Mine Music by Hal Leonard

Microphone Design Technology And Microphone Technique
Along with this development of a more live sound and hi-fi in the popular recorded music of the early 1950’s, a great deal of experimentation and improvement in microphone placement and technique was going on at the same time.

Much energy and effort were put into the development of innovative microphone design.

American microphone design technology and microphone technique were handed down from the broadcast industry to the recording industry, and were definitely ready for experimentation and improvement.

Many of the so-called unidirectional and bi-directional mikes of the time were actually omnidirectional in the low-frequency range of the audio spectrum.

Of course, this only accentuated the problem of too much reverb time in the low-frequency end of the spectrum in the day’s major recording studios.

To further intensify this low-end “coloration” of recorded music, the off-axis response of most of these older mikes caused very unpleasant and unmusical-sounding time and spectral coloration of the sound.

As microphone placement technique underwent radical and welcome experimentation and improvement in the early 1950’s, the introduction of exotic, new microphones, such as the Telefunken U 47 from Germany greatly improved the recorded sound of music.

In the fall of 1951, I was attending classes at the University of Minnesota. Walking from class to class on the campus, my schedule took me close to beautiful Northrup Auditorium.

A large concert hall with wonderful acoustic qualities, Northrup Auditorium was, at that time, the home concert stage for the Minneapolis Symphony (then under the baton of Antal Dorati).

As a kid, I had attended Minneapolis Symphony concerts almost every Thursday evening, with my mom and dad, at a time when Dimitri Metropolis conducted the orchestra.

The sight of that big, lovely concert hall reminded me of the fantastic sound of the orchestra in such wonderful acoustics that I had heard as a youngster.

Telefunken U 47
While at the University of Minnesota, I worked part-time at KUOM, the University radio station. Every Sunday afternoon, KUOM broadcast the Minneapolis Symphony Orchestra in concert (in mono, not stereo).

One day, in the KUOM studios I met a man by the name of Bob Fine, a recording engineer from New York who was in Minneapolis to record the Minneapolis Symphony for Mercury Records. He had in his hand a black box that resembled a miniature coffin.

This important-looking little black case was about 10” long, 2-1/2” wide, and about 2” high. Bob opened the case, and resting in it on a little bed of dark blue velvet was an absolutely gorgeous German microphone. Bells went off in my head!

I had never seen anything like it in my life before! It was the Telefunken U 47 microphone! I was most definitely in love!

I was, of course, very impressionable at the time, but I will never forget the sight of that exotic-looking microphone with its handsome chrome top and impressive machined metal-and-rubber shock-mount.

I couldn’t wait to hear how it sounded! Every time I look at my Telefunken U 47s now, my mind flashes back to that moment. Bob took the mike out of its case and showed it to me. He explained a bit about how it worked and how he was using it suspended 10’ above Mr. Dorati’s head.

That way, the microphone “heard” the orchestra in virtually the same balance as Mr. Dorati did.

This concept in microphone design, with its extremely wide and smooth frequency response, was almost like a miracle to me!

I had used condenser microphones before, but the Altec 21b condenser mike that we had at Jay Kershaw’s little basement studio didn’t sound anything like this!

Bob let us use the mike for a few days while he was there, to broadcast the Sunday symphony matinees on KUOM. The sound was absolutely fantastic!

I recall that there were also some television broadcasts of the Minneapolis Symphony Orchestra from Northrup Auditorium using that Telefunken microphone at about the same time. The use of this incredible mike was explained in detail on the TV program, and I remember watching and listening in rapt delight.

Here are a couple of interesting facts about one of my most cherished microphones, the Neumann U 47:

I later learned that the Germans had been experimenting with and had actually produced microphones of close to this fantastic quality 15 years earlier.

The Neumann U 47 was the first post-war mike produced by Georg Neumann GmbH in West Berlin. It was designed around a World War II military radio tube (that probably was in great supply at very low-cost) with a capsule design from 1929!

About 10,000 U 47s were made. It became the “benchmark” expensive (at $390) microphone in the early 1950’s, and engineers found out quickly that the sensitivity of the U 47 greatly enhanced the detail of their recordings.

The U 47 was a very popular vocal mike. There were many U 47s (and U 48s) used for the famous Beatles recordings, and George Martin, the Beatles’ producer, wrote that the U 47 is his favorite mike.

U 47s are pictured in abundance in the Beatles’ recording studio photos. Their aggressive sound makes them an excellent choice for lots of rock applications. Drums, guitars, amps, and brass instruments shine when sitting behind a U 47!

I bought my first pair of Telefunken U 47 mikes in 1954, from American Elite in New York, while I was still living in Minneapolis (I still have all the original paperwork).

They were a bit unusual in that they are the long-body, nickel-grille version. When those fantastic mikes arrived, it was a very big day for me. I was only 20 years old, and my two Telefunkens were the only U 47s in Minneapolis (I’m sure Bill Putnam had some Telefunken microphones in Chicago).

One of these precious mikes was stolen while I was working on Michael Jackson’s Thriller in 1982 (it’s one of his favorite vocal microphones). To this day, I use my remaining Telefunken U 47 on almost every project I am involved in. Isn’t it incredible that even today, this wonderful microphone is still often the first choice for miking many sound sources?

Now, let’s take a bit of a journey back in time with my Neumann U 47. It was in the early 1950’s that we began in earnest to attempt to improve the actual studio set-up of the musicians, singers, and microphones.

We abandoned many of the handed-down studio and microphone techniques of the past that had come from the radio broadcast industry. In the early 1950’s, microphone placement technique underwent radical and welcome experimentation and improvement.

At the same time, the introduction of exotic, new microphones, such as the U 47 greatly improved the sound of recorded music. The U 47 was probably the first microphone designed specifically for ultra-high-quality sound recording, with music recording as its primary intended use.

The Neumann Model M 49
A few years later, Telefunken introduced the Neumann model M 49 condenser microphone, another exotic model from Germany.

This is another of those wonderful microphones that is still in use today in the best world-class studios.

Designed in 1949, the M 49 was introduced to the buying public in 1950 as the answer to the question, “U 47?” It is a continuously variable multi-patterned, large (approx. 1”) dual-golddiaphragm microphone using the Telefunken ac-701 or ac-701k tube as its hub.

It had three different stand mounts, and it came in a variety of boxes. There were various models, including the M 49, M 49b, M 49c, M 249b, and M 249c.

The M 249b and M 249c were designed with an RF (radio frequency) suppression-type screw-on connector designed for the German broadcast industry. They usually utilized a “cassette system” power supply known as the N-52.

The M 49 is a superb vocal mike, but may be used for many other applications, from miking an electric guitar amp to recording French horns! Because of its adjustable polar patterns, it can be used in everything from omnidirectional mode for room miking to figure-8 for background vocals.

The Different Colors Of The Neumann Logo
Here are some small, but highly interesting facts. I love little historic details like the following:

There is a significance and meaning of the different colors of the Neumann logo on the various models of Neumann microphones.

Beginning with the Neumann “Bottle” microphone, the CMV 3, in 1928, Neumann microphones sported a logo with a black background.

This was used with all vacuum-tube-equipped microphones. For this reason, the microphones from the ’40s, ’50s, and ’60s feature the black logo, as well.

Beginning in 1966, the first transistorized (solid-state) microphones were offered by the Neumann company.

This was the 70 series for 12-volt A-B powering (also known as T-powering), with the models KM 73 through KM 76, plus the U 77 switchable-pattern model. For this series, the black logo was retained.

With the introduction of the microphones of the 80 FET series in the mid 1960’s, the 48-volt phantom power system was launched, and these microphones were identified by their purple Neumann logo.

The prime example of this series is the U 87. My personal favorite of this series is the U 47 FET. I have two U 47 FETs, very close in sequence numbers, that sound simply fabulous! I absolutely treasure these lovely mikes.

The currently used red Neumann logo signifies microphones with transformerless electronic circuits of the 100 series (e.g., the KM 140, TLM 170 r, RSM 191, TLM 193, or the KM 184) and TLM series.

Here’s Something To Think About:
“No sound system, no sound product, no acoustic environment can be designed by a calculator. Nor a computer, nor a cardboard slide rule, nor a Ouija board.

There are no step-by-step instructions a technician can follow. That’s like Isaac Newton going to the library and asking for a book on gravity.

Design work can be done by designers, each with his own hierarchy of priorities and criteria. His three most important tools are knowledge, experience, and good judgement.”

That quote is from Ted Uzzle, a Harvard-educated design consultant for motion picture facilities in Hollywood from 1973 to 1980 who joined Altec-Lansing in 1980, was made a fellow of the Audio Engineering Society in 1984, and became editor of S&VC (Sound & Video Contractor) magazine in 1992.

Innovative Placement Of The Musicians And Microphones
In early 1950, in an effort to improve the separation of musical instruments in music recording, the reverb time of modern studios was reduced.

A concerted effort was made by major-label and independent music recording studios alike to reduce the reverb time in the middle and lower frequencies.

It was also at this time that we began to use acoustical separation screens or isolation flats, or “gobos” – or whatever we wanted to call them.

These acoustical isolators were placed between instruments or whole sections of the orchestra to improve the definition and separation of the sound sources in a recording.

Using acoustical dividers in this manner made it possible for the microphone (or microphones) to be focused on a single musical instrument or group of instruments, and thus minimize the acoustic interference of other instruments playing at the same time.

In the 1930’s and 1940’s, the musicians and singers were arranged in the recording studio in an almost concert-like setup, and little or no effort was made to achieve clarity or apparent separation of sound sources in music. As a result, the sonic images of the musicians and singers in many old recordings is rather blurred and indistinct.

The year 1950 was the beginning of a very important decade for recorded music. With the release and incredible success of Les Paul and Mary Ford’s “How High The Moon” in 1951, it seemed as though a big section of the record-buying public was no longer interested in cold reality in popular music.

As the 1950’s came to a close, we in music recording found that reality in sonic image was not necessary, and perhaps not even desirable.

This innovation and improvement in technique actually began in the very early 1950’s, although when I began my work at Universal in Chicago in 1957, this renaissance in mike technique and studio setup was still very much in evidence. It was a wonderfully exciting time to be learning.

As a youngster in my early twenties, every minute of every day was full of new experiences in the studio. The big bands and musical artists that I worked with every day were very much in love with the recording process.

This is the first in a series of excerpts from Bruce Swedien’s book Make Mine Music by Hal Leonard

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Friday, January 13, 2012

Properly Cleaning Your Microphones

Advice on cleaning and maintaining microphones to ensure their continued reliability

You’ve finally invested in a high-quality vocal microphone and your voice has never sounded better.

Unfortunately, the keyboard player in your band decides he wants to use your mic during his featured rap. You cringe as he practically eats the microphone.

You can barely watch as he encourages audience members to scream into the mic.

Afterwards he returns your mic, still operational but considerably wetter and unhygienic.

Microphones are subject to an inordinate amount of abuse, especially in live music. Grilles and foam windscreens can become saturated with saliva, clogged with lipstick, and will absorb the smell of cigarette smoke prevalent in most clubs.

Regular cleaning of your microphone will not only improve its performance, but is also good hygiene. This document provides several simple yet effective techniques for cleaning microphones.

Dynamic Microphones
The best way to clean a microphone is to remove the grille. Most vocal microphone grilles simply unscrew, e.g., SM58, BG3.1. If the grille doesn’t slide off easily, gently rock it back and forth while pulling it away from the cartridge. Do not pull sharply or with excessive force, since that could damage the cartridge or separate it from the microphone housing.

Once the grille is removed, it can be thoroughly cleaned without damaging the mic. Since most of the offensive material on the grille comes from the human body, plain water should be a sufficient cleanser. Adding a mild detergent (dishwashing liquid) to the water will act as a mild disinfectant and remove odors absorbed by the foam windscreen.

To remove lipstick and other material stuck in the grille, use a toothbrush with soft bristles. In some models, the foam windscreen can be removed from the grille, but this is usually not necessary since water will not damage the grille. Most Shure microphone grilles have a nickel finish that makes them resistant to rust, and replacing the foam windscreen can also be difficult and time-consuming.

The most important thing to remember is: let the grille dry completely before reattaching it to the microphone! Microphones don’t like water, and although dynamic mics can withstand small amounts of moisture, a soggy foam windscreen will introduce more than is acceptable.

Air drying is the best way to dry the grille, but a hair drier on a low-heat setting can be used. Care must be taken not to get too close to the grille as excessive heat can melt some windscreen material.

Cleaning must be done more carefully for microphones that do not have removable grilles, e.g., SM57, 545.

Using a damp toothbrush, hold the microphone upside down and very gently scrub the grille.

Holding the mic upside down will prevent excess moisture from leaking into the microphone cartridge.

This technique is also useful for cleaning the foam that covers the diaphragm inside an SM58.

Again, keep the mic upside down, and be very gentle.

In live situations with multiple acts, it may be desirable to clean the microphones between acts. Use a diluted solution of mouthwash (Listermint, Scope) with water. Using a toothbrush and holding the microphones upside down, scrub the grille of the microphone.

At the very least, this technique will make the microphones smell more pleasant to the performer. Also make certain the sound system is turned off before the cleaning begins!

Condenser Microphones
Due to the more delicate nature of condenser microphones, never use water or any other liquid for cleaning purposes. Even a small amount of moisture may damage a condenser element.

For microphones with removable grilles like the Beta 87 or BG5.1, the grille and foam windscreen may be washed as described above.

Again, the grille and windscreen must be completely dry before reattaching it to the microphone. To clean a microphone with a permanently attached grille like the SM81 or BG4.1, use a dry, soft bristle toothbrush and gently scrub the grille.

Keep the microphone upside down so that loosened particles fall away from it. Take care not to let stray bristles get caught in the grille. This technique also works well for lavaliers and miniature gooseneck mics.

For condenser microphones that will be subject to harsh conditions, such as vocals and theater applications, it is advisable to use a removable external foam windscreen.

This will protect the microphone from saliva and make-up, and can be removed and cleaned with soap and water after the performance. Remember, never get water near a condenser element!

(Provided by Shure Incorporated.)

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Thursday, January 12, 2012

RE/P Files: Construction Of A Live Echo Chamber

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine comes a wealth of knowledge about echo chambers which first appeared in the July / August 1979 issue.

A live echo chamber can be a considerable asset for any recording studio, that is providing that it is a good one.

That’s the problem — how do you construct a good echo chamber? When someone builds a chamber, they hope it will turn out great and pray it won’t turn out absolutely dreadful and good for nothing but storing echo plates.

The truth is there are a number of complex variables which will make each chamber unique.

These factors which effect the chamber include the type of wall construction and the selection of materials used on the inside surface.

Probably the most important consideration is the cubic volume and physical proportions of the chamber.

This leads to the first question to be asked before a chamber can be built. What space is available?

Space
Most times echo chambers occupy surplus space. The more space that is available to start with, the easier the construction job since the builder won’t have to deal with odd angles or cramped building conditions.

The size of the chamber can generally vary from 1,000 cubic feet to about 2,000 cubic feet of internal volume. 1,500 cubic feet seems to get excellent results within a workable space.

A small chamber won’t get optimum results and the largest chambers are a luxury since they occupy a space large enough to be usable for other purposes.

When determining if enough area is available, it is necessary to remember to allow for figuring the wall thicknesses.

There is also a need to consider a wide enough passageway around the structure for a hammer to be swung. If this isn’t planned for putting on the exterior sheeting is going to be a difficult proposition.

It is also suggested that the space be large enough to accommodate a chamber with a minimum interior dimension of 7 feet. A chamber with a side shorter than this will usually give unsatisfactory results.

The Floor Plan
Once it has been decided that there is enough space, the next step is to design a floor plan (Figure 1). When laying out the sides of the room none of the walls should be parallel or even near to parallel. The ceiling should not be parallel to the floor. This is very important if the room is to have maximum random reflections and a smooth decay. It’s at this stage that a bit of math should be introduced (1).

T = Const X M/α S

where:
T = Reverberation time
V = Total volume of the room in cubic units
S = The total combined surface area of all sides in square units
α = The average absorption coefficient Constant:
.049 if measurements are in feet.
.161 if measurements are metric.

The American National Standards Institute (in S-l. 1-1960) defines reverberation time of a chamber as the time it takes for the mean square sound pressure level to decrease to 60 dB after a steady state signal has ceased (2). Generally this level is referenced to 500 Hz although some information relates the level to 1 kHz.

The following equation for figuring decay time was developed by Wallace Clement Sabin (1900). Since his time there have been a number of alternate equations developed but the original equation continues to be the most popular. This is due partly to the simplicity of the computation and the similarity of the resulting data.

Sabin determined that the reverberation time was related to the volume of a room, its surface area and the total amount of absorption.

As the formula would indicate, the time is directly proportional to the volume and inversely proportional to the surface area of the room. With this being the case, the longest echo will be obtained when the required volume is achieved with a minimum of interior surface area.

Trying to achieve more diffusion in the chamber by building accordion type splays will have the effect of cutting down the delay time because the total surface area will have been increased.

Another point to be remembered when figuring out the length of the sides is that none of the dimensions between any two opposing walls should be the same or a multiple or fraction of any other two opposing walls.

There are a number of ratios which serve as guides when figuring acceptable proportions, including the Golden Section; (5^1/2 + 1): 2 :( 5^1/2 - 1).

This relationship was proposed by the Greeks and divides a line in such a manner so that the smaller dimension is to the greater as the greater is to the whole. There are four or five other ratios which have been proposed and accepted to varying degrees, (3) but the one most often used is Sabin’s 2:3:5 relationship. (Figure 2) (3)

Wall Angles
The angles used for the intersecting corners should not be severely acute. In practice, the simplest way to arrive at the wall angles is to build the sides so that two of the joining sides meet at right angles.

Their opposing walls are constructed similarly, but without permanently nailing down the floor plate. Once these walls have been framed, the entire unit can be angled inward. For the average chamber moving one end of each of these two walls in by a foot or so should be sufficient.

Rigidity And Isolation
Two things which are important in making a good live echo chamber are maximum interior wall rigidity, and the total isolation of the chamber from its surroundings.

The more rigid the wall is, the less energy dissipated when a sound wave hits it, hence the surface is more reflective. The isolation is important since it’s essential that the rooms broadband ambience be very low.

Echo chamber wall systems which achieve these goals can be constructed from a variety of materials. There are a few chambers that have been built with walls, floors and ceilings of poured concrete. Such a design if properly executed will get very good results, but will be expensive to build, very permanent and extremely heavy. It will be there forever, providing you haven’t underestimated the strength of your sub- floor.

The second most popular approach is concrete block walls. They are easier to build, and a bit easier to tear down, but once again you have the weight problem.

The most popular and cheapest type of construction is a wood frame design made of 2” x 4” and 2” x 6"s. It is the easiest to build with hand tools and more importantly is relatively light compared to concrete and block.

Walls
In recording studio construction it is very popular to use 2” x 4” staggered stud construction since the results achieve a better transmission loss than a conventional wall. (Figure 3) A staggered wall uses two rows of studs 8” apart (2” x 4” alternating 16” centers) on 6” top and bottom plates. Every other stud is flush to the opposite edge of the plates. In this way the wall sheeting of the two sides is only connected at the top and bottom plate.

A standard stud wall uses a 2 x 4 plate. The sheet rock covering both sides is connected through each of the common studs. The transmission loss between these two types of construction with insulation is STC 36 dB for the standard wall and STC 49 dB for a staggered wall.4

The staggered wall would be the preferable wall design if transmission loss were the only consideration, however, rigidity is a more important factor in the design of the inside shell, hence standard construction is preferred. (Figure 4)

The reason for this is that staggered construction does not allow enough space for cross bracing. The interior shell should have a cross brace splitting the span of every stud, joist, and beam. There should be no unbraced span longer than eight feet.

The preferred layout is to use a staggered exterior wall, and a standard interior one. These two walls should not be coupled in any way and hopefully will not only be decoupled from each other but from the rest of the building.

Floating Walls and Floors
A very important part of the de-coupling of an echo chamber is isolating it from the building it sits in. What sort of isolation is needed will depend on the specific situation and the funds available.

The drawings shown use Celotex for de-coupling. As can be seen, the outer shell is de-coupled from the floor of the building, while the echo chamber has a completely floated floor.

If the location requires extreme de-coupling then it might be necessary to construct the outer shell on its own completely floated floor. It is also likely that something other than than Celotex will be needed.

Machine rubber is a good alternative to Celotex. It seems to work acceptably when used in a quiet environment but it does seem to compress a great deal and might break down with time. A thickness of or more is necessary.

In an extreme isolation situation involving low frequency vibration more severe measures will be necessary. The worst case may need a floated concrete floor on spring isolators, but once again you have considerable weight and expense.

The best material which has been found for most situations are Fiberglas decoupling blocks. As shown, they are two inch cubes, covered with latex (to keep the moisture out) and are specially designed for floating floors. They can be used for both concrete and timber designs.

When they are used with concrete, a sheet of plywood is laid over them and a border runs around the plywood to create a pouring form. Be sure all cracks in the form are sealed so that none of the concrete will seep and re-couple the floating floor to the structural one.

When the blocks are used with wood construction, they should be set under three or four 2” x 6” headers. The timbers form the base for the echo chambers floor joists. Isolators should be placed about a foot or so apart along the entire length of the headers.

The only problem with the block spacers is the gap it leaves between the floating floor and the structural floor. The solution is to fill the space with Fiberglas and run the sheetrock down to %” away from the floor caulk.

These blocks are available from a number of suppliers. The blocks can also be used to isolate ceilings from walls. For this application the blocks seem to work best if they are split in two.

A chamber built on a concrete ground floor can be further de-coupled by using a concrete saw to cut a slot around the perimeter of the chamber. This helps quite a lot in isolating the rest of the building from the chamber. The newly formed slit can be stuffed with Fiberglas but a hard sealer should be avoided since it will slowly harden and compress and re-couple the slabs.

Insulation, Sheetrocking and Sealing
All the walls should be liberally stuffed with foil backed Fiberglas insulation (3%” #R-11). Note: when working with the glass, be sure to always wear gloves, goggles, a mask, and clothes that won’t allow the glass to touch your skin.

Both sides of each wall are covered with sheet rock. Two layers of y2” sheet rock is specified for the interior wall but two layers of 5/8” will work just as well or better. When two layers are used one should overlap the other so that none of the seams coincide.

All seams should be taped and sealed so the room will be airtight. In addition to standard seam Hydroseal which is a black gummy adhesive is highly recommended.

It is normally used as a roofing sealer and will stick to anything. It should be used liberally at every structural intersection of the sheeting or the stud construction. You end up using gallons of this stuff and practically glue the building together.

The purpose of the Hydroseal is to close every crack or seam. The smallest crack should not be ignored. If you have a 1/32” crack along the floor, the actual total area of that hole is considerable.

A final note on Hydroseal. It is suggested that trowels are unnecessary for the application. Wood shims made from construction debris works a lot better. These are good for only about one or two applications. It is almost impossible to clean Hydroseal off anything, including the applicator.

The Chamber Surface and Its Application
After the walls have been built and the last inside layer of sheet rock has been taped and sealed, the chamber is ready to have its reflective surface treatment applied. The reflective walls of the echo chamber can be made from a number of different materials.

The key to how good a particular material is for this application depends on how rigid it becomes after installation and what its absorption coefficient is. Referring back to the earlier formula, the length of the echo is inversely proportional to this coefficient.

The absorption coefficient of any material is defined as “the ratio of sound energy absorbed by a given material to that which arrives at the surface from the source.” (5)

Hence, a porous surface will have a much higher number than a reflective one. This figure will not only be different for each material but vary widely depending on the frequency. Figure 5 shows an absortion coefficient chart of materials versus frequency. As can be seen, plaster has a remarkably low number and is relatively flat. The plaster which Scott uses is Keen Cement (absorption coefficient of .015). He says, “when it is properly applied, it is as smooth as a baby’s bottom.”

Before plastering the surface, the sheet rock has to be prepared so that it will hold the plaster. This is done by nailing on the actual buttonboard.

The actual plastering involves the application of two different layers. The bottom one is a brown coat layer and is put on as a preparation for the top coat of Keen cement. It is strongly suggested that the actual plastering be done by a very good professional.

The mixing of the two cements is nothing more than following the directions on their respective bags. One of the most important things to remember about the plastering process is the amount of time required for proper curing.

As often happens the chamber is complete after the final coat of plaster is applied and there is a desire to use it right away. If the chamber doors are closed prematurely, ventilation will stop as will the drying of the walls. How long it takes for a wall to properly dry changes with the temperature, ventilation, and humidity of the environment, and can vary from a few days to a few weeks.

The longer you can continue to ventilate a new chamber, the better. Scott suggests hooking up a dehumidifier in the room during the drying. He added, “you’ll be surprised how much water comes out of those walls.” This is a crucial part of getting the best end results.

“I have been in chambers that were built years ago that have walls that have never completely cured.” Needless to say such a room has a poor reverberation.

Some chambers sound bad only because the plaster coat was improperly applied. In such cases all that might be necessary is a re- plastering. It is also likely that an additional layer of buttonboard will be necessary so that the new plaster will have a firm base.

Connections And Doors
The chamber should have an IN and OUT opening so that the speaker and microphone lines can be kept separate. The pipe used should be flexible and have an I.D. of 3/4”.

Increase the diameter if there is a need for more than two or three lines per pipe. The smaller it is, however, the less possibility there is for any leakage. Plastic hose works well because it is flexible, has gentle curves and there are no ridges inside for wires to hang-up on.

Make the hole through which the hose is run as small as possible since you want to maintain the integrity of the wall system as much as possible. Once the hose has been installed, any gap between the hose and the wall should be completely caulked with Hydroseal.

A door is obviously needed somewhere in the chamber. Be sure the access to the door and the passageway leading to it is large enough to get a good sized speaker in and out of the chamber.

The two doors used should be of solid construction, not hollow. They are mounted on completely separate frames and jambs to coincide with the de-coupled inner and outer walls. (Figure 6) Machine rubber should be used between the two frames where they almost touch (%” to between the walls.

Rubber should also be used completely around the jambs of the two doors. When they close the rubber should compress and make a tight seal. Since these doors are generally used infrequently, an elaborate closing is unnecessary.

Additional Construction Notes
A light should be installed somewhere inside the room. The switch for it can go anywhere convenient including right on the fixture. Be sure that de-coupling practices are maintained while running the AC conduit and mounting the fixture.

All the walls are built with studs on 16” centers. When laying out the studs, be sure to take into consideration that the drywall is 8’ or 12’ high and 4’ wide and the centers of every third stud needs to line up with the edges of the drywall. #16 nails are used for all end nailing and #8 for toe nailing.

Drywall nails are used on the first layer of drywall, but because of the added thickness #8 nails are used on the second layer. The buttonboard should be nailed on with #8 nails.

Be sure to estimate lumber lengths thoughtfully, to limit the amount of scrap.

However what is left should be used for blocking. The greater the waste, the more the chamber (and for that matter any type of construction) will cost.

Speaker(s) and Microphone(s)
Changing the speakers and microphones or altering their placement will change the sound heard in the control room. Deciding what type of speaker sounds best or what microphone should be used becomes a matter of taste.

This is equally true as to where they are placed in the room.

Providing there was adequate space to start with and care was taken with the design, construction, and isolation considerations, it is probable that the chamber will end up sounding very good.

With a little bit of luck, it might turn out to be the sort of chamber that gains a reputation for itself as well as the studio that has it.

References:
1 - Sound Recording Practices, J. Borwick; “The Acoustics,” by Alex Burd, Oxford Press, page 27.
2 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 25.
3 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 87.
4 - Handbook of Multi-Channel Recording, F. Alton Everest, Tab Publications, page 261.
5 - The Audio Encyclopedia, H. Tremaine, Sams Publications, page 44.

Scott Putnam has been extensively involved in studio construction, having designed and built a number of echo chambers including two for Kaye- Smith in Seattle, and a pair for the Record Plant. As a builder he has collaborated on various projects with a number of acoustic consultants and architects including Jack Edwards, George Augspurger, and Scott’s Father, Bill Putnam.

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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Monday, January 09, 2012

In The Studio: The Mysteries Of Dynamics Processing Revealed

A focus primarily on compression, because that’s going to be the most commonly used dynamic processor

 
What is dynamic processing?

A dynamic processor is something that outputs a signal, where the level of the outgoing signal is based on the level of the incoming signal.

In other words, a loud signal coming in will come out differently than a quiet signal coming in.

Basic Types of Dynamic Processors

Compressors: The most common – the louder the signal is coming in, the less level it provides going out. In a compressor, a target level is set – called the “threshold” – and any signal coming in that exceeds that level will be reduced. The higher the level is above that threshold, the more reduction will occur. More on this later.

Limiters: Limiters are like super compressors. The idea is to ensure that the level does not exceed the threshold. Because this amount of compression is extreme, a limiter relies on certain functions and design that regular compressors do not have.

Expanders: The quieter the signal is coming in, the less level it provides going out. In other words – it makes quiet signals even quieter. Much like a compressor, the threshold is set at a certain level. Any signal that does NOT exceed that threshold is reduced, and the quieter the signal, the more reduction is done.

Gates: Gates are like super expanders. Anything that does not exceed the threshold is reduced to inaudible. Again, because gates are extreme, they often require a slightly different design than a regular expander.

Now – I’ll focus primarily on compression, because that’s going to be the most commonly used dynamic processor.

Compression

Every signal you hear is compressed??? Yes, every signal you hear is compressed.

Bare with me. Imagine you have a rapper in front of a microphone. The rapper raps, you record. You play it back. You haven’t used any processing – you’re just playing back the raw vocals.

You are listening to a signal that has gone through at bare minimum 3 stages of compression – and more likely than not – closer to 6.

—The microphone capsule gains tension as the rappers voice pushes it – in other words – it pushes back. The more the rapper’s voice pushes in – the harder the capsule diaphragm pushes back. In other words, the louder the signal is hitting the capsule, the more reduction the capsule does to the signal. That’s compression! (It’s mild compression, but it’s still compression).

—Along the way through the microphone, you may hit a tube. Tubes have a non-linear response to voltage – the response is quite curved, and also changes the frequency balance of the signal. This is called saturation – which will tend to “round out” a signal, by reducing the loudest peaks. Compression! And before leaving the microphone, the signal may hit a transformer as well, which will saturate in a similar way… more compression.

—The preamp is going to have multiple stages of saturation – and often times, the more gain you give something – the deeper that saturation curve goes. In other words, the more you drive the signal at the preamp, the more compression the signal experiences.

—Then the sound has to actually come out of the speaker cones. Well, those speaker cones are going to build up tension when pushed further. See where this is going? This is called “cone compression”.

OK – so this is a bit of a simplification – but there’s a point here. The point is that “compression” is always part of the signal. Some mics have less of it, some have more – same with speakers, tubes, transformers, etc. And they all do it in different ways.

With tubes, people will talk about their saturation curves and %THD (total harmonic distortion – or frequency alterations). With mics, people will refer to how it “grabs” a sound – or more specifically – the sound’s shape.

Frequency & Shape

Instead of thinking of a compressor as compressing – think of it as something that changes the shape of a sound.

If you start listening for “shape,” the mysteries of compression will reveal themselves to you, and fairly quickly.

It may help to think of shape in terms of a sound’s envelope: it’s attack, decay, sustain, and release.

Setting a compressor is like setting a mold for the signal to fit into:

—Threshold determines at what amplitude the compressor starts working.

—Ratio is how hard it’s going to work.

—Attack is another way of saying how sharp will the transient sound be.

—Release is how much tail or sustain you want to emphasize.

Transients

A transient is a very fast signal – in other words the “attack” of the signal. Drums have transient attacks. Strings have gradually rising attacks. So the attack control on the compressor is really like saying – how much emphasis on the attack of the signal do you want?

Do you want the attack to be really rounded out and diminished? Set the attack fast.

Do you want the attack to be prominent and stick out? Set the attack slower.

Of course, this works directly in conjunction with the threshold. Try it yourself, set the threshold low, and the attack short. Suddenly, the attack sound of your snare is gone.

Set the threshold low and the attack long. Suddenly the punch of your kick is very round and bouncy.

Set the threshold high and the attack short. Now the snare is a little fatter and rounder, and not quite as spikey (but possibly a little duller).

Set the threshold high and the attack long – the change is hardly noticeable, the attack is just a little bit “rounder.”

Maximum Punch

There is a thin line between a transient sound, and a sustained sound. A sound that holds for any noticeable amount of time is sustaining. A sound that moves by too quickly to register as it’s own moment is transient. But transients can vary in length. A transient can be half a millisecond or it could also be ten milliseconds; they won’t sound the same.

A big factor in punch is how long that transient exists. A quick transient sounds “spikey” – but a long transient sounds “punchy.” You want to find the point that makes the transient exist as long as possible before “flattening out” or becoming a sustained sound. Only your ear can tell you where that point is.

Good samples are already shaped to have that kind of impact – and any additional compression may actually soften that. Of course, punch has a lot to do with frequency as well – but that’s for another article.

Now what about the release? The release is super elusive. It determines how long it takes for the compressor to let go.

If the release is too short for the signal you are going to get a disjointed sounding shape which usually results in distortion. If it’s too long, your signal never really returns to its natural shape, and you generally lose tone (or you just get permanent drive on the compressor’s output, giving the whole signal a new bit of tone).

So the idea is to find a point that emphasizes the sustain (which is where most of the signals tone lives) properly.

Lastly, when the attack and release are set in a way that seem to argue – the compression can become very audible. You either hear the decent or the ascent of the signal level. This is called pumping. It’s generally annoying, but can sometimes be used an effect. If audibly desired, consider the rhythm of the release time, and ask yourself if it’s groove is complimenting the song.

———————————-

So, rather than thinking of a compressor as something that effects the “level” of a signal. Think of a compressor as something that effects shape. Why? Because level can be controlled with the volume fader more accurately and transparently. A fader doesn’t really control shape, unless you are being extremely meticulous. Conversely, compression will always effect the shape of the sound it is working on.

Once you start hearing shape, you will understand compression.

Matthew Weiss records, mixes, and masters music in the Philadelphia, New York, and Boston areas. Find out more about him here.

Be sure to visit the ProAudioFiles for more great recording content.

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Friday, January 06, 2012

In The Studio: Four Reasons To Invest In Analog Gear

I try to ask one simple question: What will help me make better recordings?

 
As you may already know, I’m in the process of doing a few upgrades in my studio.

Whenever I’m looking to buy a new piece of equipment or upgrade an existing piece of equipment, I try to ask one simple question: What will help me make better recordings?

My goal isn’t to marginally improve the recordings. I’m looking for holes in my system. I’m looking for weak links.

For example, I don’t currently own an actual guitar amp (I’ve been using a pretty killer amp simulator pedal, but still don’t have an actual amp). A good amp would definitely improve my guitar recordings.

Also, I’m working on upgrading my cabling. It’s not a sexy upgrade, but it makes a noticeable difference. (More on that in a future article.)

Another deciding factor when I’m buying gear is whether or not the gear is “digital” or “analog.” Buying software, plug-ins, computers, is a lot of fun, and I have nothing against those things, but software inevitably has to be updated. Computers will eventually be too slow…or will simply die.

You know what lasts a lot longer and will work with ANY recording system? Good, analog equipment.

Examples of good analog equipment:
—microphones
—preamps/channel strips
—compressors
—EQs
—summing mixers
—acoustic treatment
—studio monitors, monitor controllers, headphones
—cables/stands
—guitar pedals/direct boxes
—guitars (and other instruments), amps

The following don’t qualify:
—audio interfaces
—AD/DA converters
—DAW software
—plugins/virtual instruments
—MIDI controllers

You’ll find that the list of analog gear can easily outnumber the list of digital gear. But aren’t we all guilty of drooling over the latest piece of software? The latest audio interface?

Again, there’s nothing wrong with these things, but I’ve put together a small list of reasons why you should (as much as it makes sense) invest in good, analog equipment.

1. Analog gear can last decades.
A lot of major studios around the world still use the same outboard gear they used thirty years ago. Sure, they might need to be repaired occasionally, or you might need to swap out tubes, etc.

But these studios have gone from analog tape to digital tape to full-on Pro Tools HD systems. All these systems can easily use the same analog equipment.

A good piece of analog hardware is timeless.

2. Analog gear is HOW you “get it right at the source.”
You hear it all over the place. When recording, you must “get it right at the source.” The idea of “fixing it in the mix” is absurd, if you aren’t diligent about first capturing the audio properly.

Analog gear will always be the only way to properly capture an analog source.

3. Analog gear never has compatibility issues.

Have you ever purchased a new computer, only to find you needed to upgrade a few pieces of software to get them to work? Yeah, me too.
I’ve never heard of somebody needing to upgrade his LA-610 when he updated his operating system, have you? Me neither. Analog gear will always be relevant and useful.

4. Analog gear (for the most part) maintains its value.
If at some point in the next 10 years you decide you want to sell your nice solid-state preamp for a high-end tube pre, you’ll get a lot more money for a good preamp than you will for a good audio interface. ESPECIALLY if the piece in question is over 5 years old.

Imagine buying a $1,000 interface and a $1,000 microphone today. In ten years, you could get a lot of money out of that microphone. You might not even be able to GIVE the audio interface away. It would probably be obsolete. A good mic is never obsolete.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

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Thursday, January 05, 2012

A Conversation With Audio Pioneers, SynAudCon Founders Don & Carolyn Davis

The life, times and contributions of two individuals who dedicated their lives to improving sound quality through education

When noting the contributions of Don and Carolyn Davis to the professional audio industry, it’s hard to know where to even start. Their book, Sound System Engineering, originally published in 1973 (and since updated), remains a standard audio and systems resource.

Founders of SynAudCon, Don and Carolyn established the industry’s pre-imminent and most respected (and independent) educational resource, teaching thousands the essential concepts of audio and acoustics that in turn has led to remarkable advancements in systems and sound quality that we all enjoy. Now consider that these accomplishments just scratch the surface of their crucial role in leading the industry to its current modern era…

I had the privilege of spending an afternoon with Don and Carolyn while attending a SynAudCon seminar and workshop in southern Indiana. They were gracious enough to travel to meet me, with the warm and at times reverential reception they received from attendees standing as a testament to the tremendous respect they’ve tirelessly earned in service. Our conversation was fascinating, spanning a wide range of topics and touching on crucial historical landmarks that lend perspective and understanding to the current state of the industry.

Now “retired,” they continue to travel extensively, staying in touch with an ever-growing network of friends and exploring new places. Like many long-married couples, they have the endearing trait of often finishing each other’s sentences or interrupting to take the conversation in new directions. Frankly, I didn’t have to interject much as the two shared the fascinating tale of their lives in pro audio. So without further adieu, let’s roll tape and simply say, “go”.

Keith Clark: Don, I understand you worked with Altec Lansing prior to the founding of SynAudCon.

Don: I worked with Altec from 1959 through the early ‘70s, marketing and, really, managing mostly. I was a field rep based in Chicago serving a big chunk of the Midwestern U.S. We weren’t exactly sales reps, but more comprehensive in scope. Prior to this point, Altec Lansing products were distributed through Graybar, and major installations were often headed up by the Altec Service Company, the theater service division.

Just at the time I joined the company, they decided to set up their own distribution with sound contractors. A guy named Mo Morris had seen the vision that sound contracting was a viable thing, that it was a good way to move inventory out of the factory and into the warehouses of the contractors, and that it was a good way to respond quicker to needs.

So my job was to go out and identify potential contractors, and then to set them up as dealers and make sure they were supported, providing any encouragement possible.

This led to doing a little bit of everything. I enjoyed this role a great deal, and in the process, I worked with some of the “old-time” guys who had been Western Electric contractors. They were superbly trained people and quite used to top-of-the-line equipment – a piece of Western Electric equipment cost more than anything else, yet they invariably got all the better jobs.

Nate Reese in Detroit is a good example of this. It was said that during his first couple of years in business, he lost almost every job he bid on. But then he followed up with these same customers a bit later, knowing that most would be unhappy. He’d say ‘hello, I’m Nate Reese and I was high bidder on your project. Are you happy with the work?’ And, of course he got most of them on board as permanent customers.

After a while he didn’t really have to be too involved with the bidding process, because if they wanted it done right, they came to him. Nate was probably the first guy to make himself a millionaire in audio. It was his integrity, and that of Western’s gear, that did it.

So in the background was Western Electric, and you went out and tried to find people that fit that mold. When Altec was formed after the dissolution of Western Electric in the late 1930s, a lot of the Western personnel came on board. They bought up the rights to the best Western products for pennies on the dollar and then proceeded to make themselves wealthy men.

KC: You were one of the pioneers of equalization…

Don: At Altec, I constructed a seminar program in 1968 to show people how to equalize systems. The initial problem was that while even the early equalizers worked very well, the systems in general didn’t. People put in EQs and discovered they hadn’t planned enough power, for example, because now they could raise the levels. And what had been adequate before in feedback constraints wasn’t even close to adequate any more. A 10 dB increase in acoustic gain meant a 10 dB increase in power.

This emphasis in training people for equalization is exactly what Pat (Brown) is doing here with SynAudCon. You’ve got to look at polarity, you’ve got to signal align, you’ve got to clean up all of the impedances, match all the levels, and so on.

The way I found out about the problems, initially, was that we had franchised a bunch of contractors to handle equalization, and they had to spend about $10,000 on specialized equipment – GenRad and Hewlett Packard test gear. But nothing good in the way of progress and improvement seemed to be happening, so Carolyn and I loaded the first HP Real-Time Analyzer (RTA) ever made into the trunk of our car -

Carolyn: - Don had talked HP into building the RTA for him, the first one ever -

Don: - and we went on the road to find out what was going on. We quickly saw that even the best contractors were building inadequate systems – not that they weren’t great compared to most others, but they still weren’t adequate in terms of the extra power that could and should be delivered.

We learned to look at a space and to understand that what it presented acoustically was the challenge. Fit and match the space with an array that could meet the criteria of the space, and then work backwards through the system to fill it out with power and other components needed to do the job right. At that point, system design was being done just the opposite, from the microphone out, rather than speakers back.

Carolyn: And you should mention at that time that HP had also just introduced the desktop computer –

Don: – and that was a huge help.

Carolyn: Yes, Don bought into it quickly.

Don: I was looking at all the “gimmicks” of the time. But in this case, specifically, I was always lousy with a slide rule anyway, and the ability to be able to program everything on this portable computer was great. These early computers were really nothing more than a big programmable calculator, but they were very helpful.

In the earliest computer, we had to go through about 2,000 steps to attain calculations. Reverberation time, noise control, acoustic gain – all of this and more was plugged in for calculation. Of course, we hadn’t discovered how to do intelligibility yet, this was still intuitive only.

A bit later, V.M.A. Peutz of Holland and some other smart people figured out that intelligibility could be designed into a system ahead of time. Peutz was a real genius, unlocking the whole intelligibility problem. While there are current “gods” of intelligibility, this is where it all came from, where it all started.

When Peutz took one of the early TEF analyzers and programmed it to measure intelligibility, essentially - everybody objects to the term “measurement” in this regard but its an estimate taken off the data, it provided a place and explanation as to why so many systems of the time were falling short. The numbers really proved it.

KC: I’ve also read that you were instrumental in bringing the first TEF analyzer to market.

Don: Cal Tech (university) came to us and asked if we’d take over the licensing of Time Delay Spectrometry. They had only one licensee at that point, after a decade, and we got them 120 or so licensees within a year. That was kind of an interesting experience, and when they said, ‘OK, now it’s going good and we want it back’, we gave it right back to them. We weren’t in the business to be wheeler-dealers.

Carolyn: Getting back to equalizers, in March of 1968, Don went to a convention and came back with this idea for equalizers. He went straight to Art Davis (an Altec engineer) and told him about it. Art wanted to do it a little differently, and Don said fine, I don’t really care, and he and Don were on the original patent.

Don: I spaced out what the filters had to do, and Art made a contribution I hadn’t thought about, to make frequencies combining, summing -

Carolyn: – and we had a prototype by September and went to the AES Convention that year and presented a paper on it.

Don: The chairman of the session had been involved in early equalization work as well, and when he read the title of the paper - “One-Third Octave Broadband Equalizer” - he kind of stopped and raised an eyebrow on the word “broadband”.

To him, what we were calling “broadband” was actually very narrow. Now, there’s nothing wrong with a filter being exactly the shape of whatever your problem is, but you can’t go after anything that isn’t the middle of the phase realm. There are things in there - “bumps” - that if you put an EQ on it, you only make the problem worse.

But if you put it in the minimum phase realm, then the EQ clears everything – it corrects amplitude, it corrects phase, it even corrects time. But it must precisely meet, and any divergence causes problems. There was a great deal to be said for a parametric equalizer, only nobody really knew how to make them at the time. Dr. Paul Boner was making these real narrow filter devices, trying to make the intrusion as minimal as possible. But one-third octave shaping filters could shape to the broadband nature of the problem beautifully, and they didn’t introduce any major phase anomalies as a result. You follow the general shape of the curve.

Now there might have been a little individual narrow-band anomaly, but these were so narrow that they were inside critical bandwidths, and thus they didn’t much matter.

Nowadays we have the correct parametric process and equipment, and there are also these beautiful programs that invoke the house curve and let you match to it. If you know what you’re doing you can get very refined equalization.

But in the meantime, one-third equalization dramatically improved loudspeakers of that time, and it also led to discovery of problems with signal alignment. This is still something no one has really pursued fully yet, at least that I’m aware of. I don’t think the equalization field and issues have been fully worked out yet.

Right now, with most of the current devices, you get further by improving the audible quality of sound systems with signal alignment than you ever do with anything else, particularly with the newer array concepts. It will always be a tough job to have more than one of anything in an acoustical system – nature doesn’t like that. So, you make your compromises.

The contribution that I felt like I made is that prior to this work, the acoustic environment was almost totally ignored. Yet all along that was the major tool to play with. And in fact, most rooms ought to be corrected by people doing sound systems. There’s an optimum match for every system to every room, so that you don’t add any more power than needed for maximum intelligibility and you don’t add any more absorption than necessary for maximum control of energy. This is what a good acoustical consultant should do, but it’s surprising how many of them don’t.

KC: What are the roots of SynAudCon?

Carolyn: By 1972, we could see that things at Altec were not going so well due to some management problems. About that time, Don was asked to establish the European market for them, and he said we’d go over and check things out before agreeing to do it. But at that time, the economy was under some dramatic changes and it just wasn’t feasible -

Don: - well, we had an acquaintance named Mr. Vorwig who had been in charge of truck production during the war (World War II), on the German side, and who also had been the engineer that originally tested the Volkswagen for Hitler. Mr. Vorwig had a party that we attended, and he and some of the guests, including a banker in Frankfort, laid out for us what exactly was going to happen with the economy, the deflation of the U.S. dollar that would occur. I had to tell Altec that I wouldn’t take their offer.

Carolyn: Don and I used to work for a few years and then take time off and go to Europe and travel for months at a time – we didn’t have children so we could do that. Through the ‘50s, the economy was great, but by ’72, we found that prices were already 10 times more than in the ‘50s. And, things had changed with Altec –

Don: - when a company is being torn apart by bad management, the talent leaves first. The ones that hang in there may be great workers, but that’s not where the talent lies and where the future and insight is. There were a lot of strange contracts coming across my desk that I didn’t want to sign, and this is what happens… I’ve often sworn I was going to write a book on mismanagement with all of it I’ve seen over the years. I resigned from Altec in December 1972.

Carolyn: Altec offered Don a year’s salary if he would not go to work for the competition -

Don: – Which I had no intention of doing anyway –

Carolyn: – we took six months to write our book, Sound System Engineering, because we had an income from Altec. Sams Publishing printed it at no cost and allowed us to buy it at $10 a copy. It was loose leaf at that time, and about three years later they decided to publish it as a book. Then a few years later, we revised it.

Don: We had a lot of lovely people help us with this, just like Pat (Brown) does now with SynAudCon.

Carolyn: GenRad and HP loaned us thousands of dollars worth of equipment for our seminars.

Carolyn: In 1973, the oil crisis started and things were not good in terms of starting a business, but we decided to anyway.

Don: We set out on the road with a Dodge three-quarter ton truck and a camper shell to house all the gear, towing a trailer behind it to live in. We toured the country and taught audio.

Carolyn: Don could see that the only way we would really be able to make it in doing this tour would be to set up a sponsorship program. He went to Shure - or they came to him, I can’t recall – and they were great in terms of support. That first year, Shure, UREI and Sun Music were our first sponsors.

Don: The point is that there were several of these engineering folks and their companies who were very supportive, who understood what we had and wanted to give.

Carolyn: Another interesting and critical thing at this point in time is that Altec pretty much owned the contracting business. RCA had a service company and could still do some things at that point. And, some other names that aren’t even around anymore were the big entities. At the time, companies like Electro-Voice, Shure, JBL and so forth were really still just independent gadget makers.

What we did that was unique at the time was to put together all of the elements offered by these companies into proper systems. These pioneer sponsors of SynAudCon could provide the quality components, individually, and then that equipment could be formed into quality systems.

Don: UREI, for example, was one of the first to make the equalizer, and they were a sponsor. Emilar would make the drivers that were needed. So we “filled the chain” with sponsors so that people would know where to go to fill out an entire system. That was a piece of serendipity that worked out well for both us and the sponsors. It wasn’t really a deliberate thought-out thing, but just something that happened.

Carolyn: The next year after we started the sponsorship program, Don wanted something to bind SynAudCon “grads” together, so he started a newsletter subscription, free for one year to everyone who attended a seminar, and then renew for $25, later raised to $35.

KC: I understand you had settled in California by this time?

Don: Well, we owned property there, up in the mountains. It was a place to park the trailer and basically camp out. We’d be on the road for nine months out of the year and then go back and spend part of the winter in California.

Then in the summer, when everyone was busy putting in school systems, we’d park the trailer out at the (family) farm in Indiana. The old house hadn’t been rejuvenated at that point. We were living a gypsy life.

KC: So how long did you operate SynAudCon as a “road show” concept?

Carolyn: Well, in 1992 we were still doing classes in the U.S., Canada, Europe, Japan and Australia. We were in Japan on one of these trips when Don woke up one morning and said ‘this isn’t the way I want to spend the rest of my life’. So we canceled everything at that point. Travel had gotten old.

We had moved to the farm in 1987, so we decided to take the “old farmhouse” - built in 1883 - and fix it up so we could hold classes there for 10-12 people at a time. This allowed us to keep teaching, because we still loved that part of it. We did this through 1995.

A good consultant and/or contractor – someone who worked daily in the industry - would present the hands-on, and Don would teach the theory. Now Pat can do both the theory and the practical. Don has more of an interest in the theory, never quite as interested in the hands-on side of things.

KC: So outside of your absolute dedication, why do you think SynAudCon thrived?

Don: The fascinating thing is that in the 25 years we ran SynAudCon, we hardly had a conflict with any sponsor about anything, and almost all of them are still with Syn-Aud-Con to this day.

We always tried to have a sense of integrity about our relationships with sponsors, and this was reciprocal. One time we did have to “fire” one prominent loudspeaker company as a sponsor, because they were unfairly attacking another party and presenting grossly incorrect information. This just couldn’t stand, and we refunded their money. So we always did our best to have a sense of integrity about what we were teaching.

Carolyn: We limited sponsorships to 20 and had a waiting list, and Pat has expanded that.

Don: The point is that you’re out there trying to teach people about what’s right and wrong from a technical standpoint and they’re being told so many other things making it that much harder. We’ve had people accuse us of being prejudiced, and that’s not the case.

Carolyn: We’ve always had a special appreciation of new ideas and talent, and have so much enjoyed the promotion of that talent. So much of the ‘70s was an accumulation of a lot of information, and then in the ‘80s, all of this began to be focused into new ideas and products.

Don: We got to the stage when we could recognize talent when it wasn’t perhaps all that obvious to others -

Carolyn: - Richard C. Heyser, Peter D. Antonio, V.M.A. Peutz, Dr. Eugene Patronis, Gerald Stanley, Ed Long, Ron Wickersham, Ken Wahrenbrock – these were the people that developed the concepts that were so important to us: TEF, QRD Diffusors, %Alcons, LEDE control rooms, PZM, signal alignment, etc. They conceived the ideas. We often brought their concepts to the attention of manufacturers. I was mentioning this idea recently to a friend, and he said that the ‘80s was an outpouring of everything we had learned. But this was more on an individual basis, and now Pat and Brenda are taking the entire industry upward in the same way.

KC: How did Pat and Brenda come to take the reins?

Carolyn: Each seminar that was scheduled at the farm had a consultant or contractor to work with us. A consultant scheduled to work with us in a seminar had to cancel at the last minute due to health reasons, and we asked Pat to come in and teach on an emergency basis – Pat lives only an hour from the farm. And he was great, pretty much teaching just as he does now, explaining things so clearly and so well and feeling very comfortable in front of a group of peers.

Don: He loves it -

Carolyn: - and a little later, Janine Masten, who was with EV at the time –

Don: - sharp lady –

Carolyn: - this was in 1995, and she called to ask if Don would break his rule and go to Europe to teach for them. I was sure Don would say no, but instead he turned around said, “I’ll do it if Pat Brown comes with us.”

Don: And they said yes -

Carolyn: - and when they finished the classes, Larry Frandsen (head of Mark IV Audio Europe at the time) invited us to come back the next year. Don declined, but as he did so, Larry immediately turned to Pat. Pat accepted – which was what Don had in mind when he asked for Pat to be included in the tour.

That trip lasted about three weeks, and during that entire period I didn’t call to check in with the office, and it was such a relief. It felt like it was time for us to move on, and we asked Pat if he would take over. He talked it over with Brenda, who was a very successful nurse at the time, and she gave her support. Gradually, she worked into the business more and more and now has taken on a full partner role with Pat.

Don: Well, Brenda’s very sharp, very on top of things, understands the technical part in addition to her business talents. They also have a very spiritual side to them, that we love -

Carolyn: - they don’t talk about it much. They’re so ethical and we take a lot of pride in that.

KC: What’s the biggest difference in SynAudCon now, in comparison to what you handed off to Pat and Brenda?

Don: Pat has computerized the teaching process, has brought it the rest of the way into the digital age.

Carolyn: At a seminar or workshop, everything you see Pat doing with the computers and video screens, Don used to do with slides and overhead projectors.

Don: We have preached digital revolution for 20 years, that it would be the way to go, the way of the future. It’s interesting to look at the space race – a lot of people think all of their money was just shot to the moon, but actually a very small amount of hardware went there. The big thing to come out of it is the ideas, the outflow of technical creativity.

Don: Pat and Brenda have done another vital thing, and that is to go places where we had never gone. Mexico, South America, Jordan, India, Dubai -

Carolyn: - and he’s invited to China -

Don: - and that’s invaluable. He’s spreading the knowledge. In the late ‘50s, Carolyn and I worked at the American National Exhibition in Russia, an exchange fair between them and the U.S. We were showing audio equipment.

Recently, I was interested to read a book written by a former top KGB agent who noted the most subversive thing that ever happened between the U.S. and Soviet Union was this exchange, that it changed more things in Soviet Russia than anything else. He was kind of tongue in cheek about the subversive part, but what he was saying is true.

As the SynAudCon attitude gets around, the idea that you share the information rather than hold it close, as that philosophy gets into new places, it’s fascinating to see what comes about. SynAudCon became a society, a family really, without meaning to, based on this idea of sharing information.

Carolyn: Along these lines, the web site and what Pat does with the list serve is unbelievable, and the newsletter keeps going strong. This all goes with being a society. It’s just amazing that top professionals in this industry will gladly tell everything they know through these channels, unselfishly and for the benefit of anyone willing to learn.

Recommended Reading:
Sound System Engineering
If Bad Sound Were Fatal, Audio Would be the Leading Cause of Death

{extended}

Monday, December 26, 2011

Ensuring You’re Recording At The Proper Levels

You're probably tracking too hot. Here's why...

 
I’m going to try to keep this very “fool proof”.

This was born out of the rantings of hundreds and hundreds of posts on a dozen or more audio forums exploding like a volcano recorded with lots of headroom.

I hope to instill a basic understanding of why certain trends and common beliefs are just plain bad.

And by the time you’re done reading, and perhaps doing a little experimentation based on this, you won’t need me to prove it. You’ll know it yourself.

Is this a “miracle cure” for bad recordings?

Normally, I’d say no.

But with the dozens and dozens - easily now into hundreds of e-mails, phone calls, letters, forum posts and other forms of communication I’ve received about how this advice has completely changed a persons view of recording recording, I figured this information is worth sharing with a wider audience.

The sad part is this should be common sense. To anyone that grew up “on tape” it probably is. To those brought up in 1’s and 0’s, it might not be so obvious.

So, if you’ve been struggling with recordings that sound “weak” or “small” or too dense or “just not ‘pro’ enough” then please, read on. If this is about you, you might think differently soon.

As a mastering engineer, I work on recordings from pretty much every level of experience. A few years ago, I noticed something unusual.

“Ultra rookie” recordings, that is those made by people with little or no experience, sounded fine. They didn’t know any better, so they didn’t have enough rope to hang themselves with.

Then, at the other end of the spectrum, “Pro” recordings sounded fine. They know what they’re doing and/or are using gear with obscene amounts of usable headroom (explained later).

The “middle of the road” engineers with a year or two - or much, much more experience—Those are the recordings that sounded “small” and spectrally challenged. So after quizzing these people over months and months, I came up with the following conclusion…

You’re probably recording too hot.

And it’s absolutely ruining recording after recording after recording. And it’s the simplest thing in the universe to correct.

I know, I know - “It says in the manual to record as hot as you can without clipping.” Well, I’m going to flat-out call that B.S. and I’m going to back it up with a simple (if not somewhat time-consuming) experiment.

Also as a mastering engineer, let’s get something straight - I don’t like the “loudness war” going on.

However, I’m as guilty as the next in contributing to it. I can’t fight it, as much as I try. Hopefully it’ll be over some day. Unfortunately, with the quest for loud, there are a lot of engineers out there shooting themselves in the foot before they even know how to aim.

They think that tracking loud and mixing loud contributes to a louder recording after the mastering phase. This is absolutely untrue and it’s generally the best way to make sure that your recording will not have the “loudness potential” of the average commercial release.

Clean recordings, or those made with low distortion and good spectral balance, are the ones that handle the “abuse” of the mastering phase with flying colors.

This article isn’t intended to give you some secret way of making louder recordings. But it will almost undoubtedly give you the ammo needed to make better recordings. And those better and cleaner recordings are the ones that can be louder recordings in the end.

First, let’s get through a little nomenclature,

dBFS
Deci-Bel (one tenth of a Bel) Full Scale, on the digital recording scale, -0dBFS is the hottest signal you can have.

“All ones.” Top of the scale, can’t get hotter, etc. Always “minus” as you can never go higher - So the reading will always be a specified amount below 0.

Line Level / 0dBVU
Just what it says. Line level. 0dBVU on an analog VU (volume unit) meter. Pro (+4dBu) or consumer (-10dBv) level, it’s line level.

We can also refer to this as RMS (Root Mean, Squared), or a level over a specific amount of time. You can go above or below 0dBVU.

It’s simply a nominal level to which basically everything audio is related to.

Headroom
The space between a nominal signal (in this case, line level) and the point where the circuit fails.

In digital, basically anything under full scale (-0dBFS) would be considered headroom.

In analog, it’s the space between 0dBVU and the point where the circuit clips (failing completely). In analog, there can be a big difference between “headroom” and “usable headroom.” We’ll get into that in a bit.

Steak
From the old Norse “steik” meaning “roast”. A slice of meat, typically beef, usually cut thick and across the muscle grain and served broiled or fried (thank you, Wikipedia).

So…

You have a microphone and a preamp going into a converter or sound card. Those converters are calibrated at line level.

In most cases, over the last several years, most I’ve seen are calibrated to -18dBFS = line level (or 0dBVU).

In other words, if you run a steady signal (a sustained note on a keyboard for instance) through a preamp and turn up the preamp gain until the VU meter reads 0dBVU, at the converter, it will read -18dBFS (or -18dBFS(RMS) - full scale, but measured over time).

This is where your gear is designed to run. This is where it’s spec’d at. You will have a decent amount of headroom, the lowest distortion, the best signal to noise ratio, etc., etc., etc. around this level or lower.

Some gear - usually very high-quality stuff, has a good amount of usable headroom above this level.

A lot of “budget friendly” gear does not. So all of this advice is more important if you’re using “okay” gear at the input. Even your digital converters are analog components up to the converter itself. They don’t want to be “beat up” all the time either.

Let’s look into headroom. Above that 0dBVU/-18dBFS range, digital headroom is simple - perfect, perfect, perfect, perfect, CLIP.

The signal is “what it’s supposed to be” until the point of failure. Analog gear (your preamp, compressors, outboard signal processors, etc.) isn’t like that…

It’s more like “Perfect, a little noisy, “tight” sounding, spectrally distorted, CLIP.

The converter’s job is simple - reproduce the signal it’s fed digitally, whether that signal is clean and dynamic or distorted and squishy.

The analog chain’s job is anything but - typically, you’re adding 20, 30, maybe 50dB of gain to the incoming signal.

The preamp is working - not just “passing” the signal.

And that signal can start to suffer from noise, distortion and dynamically dependent (varying along with volume) spectral imbalance (a skewing of the overall spectrum from an EQ standpoint).

In other words, a nice, thick, chunky guitar tone (for example) might have different characteristics depending on how hot the signal is.

The highs might be open and airy and then the signal gets loud for whatever reason and the highs either get swallowed up, or perhaps get very harsh and strident.

In any case, it’s an inconsistency that isn’t’ there when the levels are more “normal.”

Even though the analog gear probably has spec’d headroom well above digital’s full-scale, it doesn’t mean that signal actually has the integrity it should up to that level.

So what happens is simple, a signal is recorded that’s too hot (usually to “use all the bits” which again, is a bunch of BS).

It overdrives the input chain not unlike a guitar preamp overdriving a Marshall stack (well, not that much, but the premise is the same).

Now, after all the other tracks are recorded, all of them need to be attenuated by 12, maybe 15dB or more so the mix doesn’t clip. Those distorted, spectrally questionable, squishy, noisy tracks all get turned down.

Are you seeing my point yet?

When you take a steak and cook it until it’s burnt, it’s burnt.

If you pour ice cubes all over it, it doesn’t make it more rare - it makes it a cold, wet, burnt steak. No matter what you do, it’s still burnt. Just like if you record too hot.

But if you cook a steak a little too rare, you can always heat it up a bit later.

You can microwave it without it turning into leather. You can pan-fry it for a few minutes and it’s still a tasty, savory piece of steak.

When you use up all your headroom right away, you don’t get it back by turning it down.

It’s gone forever. Sure, you can increase mix headroom or the headroom at the buss - but it’s not going to make the track less distorted or fix the skewed S/N ratio.

Here’s Your Experiment
You’ll need a few Y-cables (let’s not get into the technical aspects of splitting a mic signal - It’s an experiment) and at least one stereo (2-channel) preamp.

Record a song using as many tracks as you feel fit. The more, the more apparent. You’re going to split the mic signal and record each twice simultaneously.

On one channel of the preamp, set the gain so it peaks between -18 and -12dBFS at the converters and record them to odd numbered channels.

On the other, set it as high as possible without clipping and record them to even numbered channels. Record some guitars, drums, maybe piano, of course some vocals, keyboards, go nuts.

Set all the odd numbered (“normal” level) channels to unity and toss up a rough mix to a stereo buss - which should be a piece o’ cake.

Switch over to the even numbered channels and figure out how much you’re going to have to attenuate them all so the main buss isn’t clipping constantly.

It might be a lot. Could be a 10-15dB cut on all channels before you can even think of starting to do anything else. Send those to a stereo buss. Solo the busses, one at a time, and try to match the levels between the mixes.

You’ll probably immediately notice that the “normal” mix is much more open, dynamic, airy, clear, clean, with much more “sonic space” between the instruments than the “hot” mix.

Now, add a limiter on the main buss. Run the “hot” mix into it and bring the level up until it starts to obviously distort and fall apart sonically.

Then switch over to the “normal” mix - which should now be “rammed” by the same amount. If your experience is pretty much like everyone else’s, the “normal” mix is still much more open, airy and dynamic with less distortion and more “crankability” than the other.

The “Dumbed Down” Version
Stop recording so hot. Instead of trying to get your tracks to peak at -2dBFS, have them peak between -20 and -12dBFS and your recordings will almost undoubtedly sound better.

Mixing will be easier. EQ will be more effective. Compression will be smoother, more manageable and predictable. You’re in the age of 24-bit digital recording - relax and enjoy the headroom.

Even if your only concern is the volume of the finished product (which would be a shame, but it happens), recordings made with a good amount of headroom are almost undoubtedly better suited to handle the “abuse” of excessive dynamics control.

Quieter recordings have more potential to be loud later. It’s because they’re usually better sounding recordings in the first place.

John Scrip is the Owner and Engineer at MASSIVE Mastering.

{extended}

Friday, December 23, 2011

The 10 Most Frequently Asked Questions About Mastering

In this, the first in a three part series, Tom Volpicelli of The Mastering House answers three common questions about mastering.

In this, the first in a three part series, Tom Volpicelli of The Mastering House answers three common questions about mastering.

1. What is mastering and the role of the mastering engineer?

Mastering is essentially the step of audio production used to prepare mixes for the formats that are used for replication and distribution.

It is the culmination of the combined efforts from the producer, musicians, and engineers to realize the musical vision of the artist.

Each stage of the audio production process, from pre-production to mastering, builds on each other and is dependent on the previous process.

Mastering is the last opportunity to make any changes to positively affect the presentation of your music before it evolves from a studio environment to the outside world.

An awareness of the differences between the roles of mixing and mastering engineers should be noted.

While the tools may be similar, the perspectives between mixing and mastering are very different. When mixing, the focus is on the internal balance of individually recorded tracks and effects used both sonically and creatively for a single piece of music.

An album cannot be heard in its entirety until the job of a mix engineer is completed. The mastering engineer picks up where the mix engineer leaves off. Mastering is geared toward creating the balance required to make the entire album cohesive. The mastering engineer is most concerned with overall sonic and translation issues.

A mastering engineer works with the client to determine proper spacing between songs and how songs will be ordered on the CD. The flow of an album must appeal to the listener; it should engage them and take them on a musical journey as determined by the artist. Any final edits will be addressed during the mastering process as well.

Finally, the role of the mastering engineer is to provide preparation and quality control of the physical media send to the plant for replication.

This includes listening to the premaster CD to verify integrity, along with the more technical aspects such as encoding text, UPC/EAN and ISRC codes, checking for errors within the media and providing any necessary documentation such as a PQ list.

2. Is mastering always necessary?

A writer’s words are not complete until the editor approves them. A painter’s work is not complete until it has been matted and framed.

A musician’s work requires the same treatment. Audio production should not be rushed, finished haphazardly or completed “just to get it out there”. A finished product should reflect all of the work of the artist, producers and engineers that carry that vision forward.

Even a “perfect” mix needs mastering to a degree. In this case, you want the mastering to be as transparent as possible so that the original sound is maintained while preparing it for the final media.

As mentioned earlier, it is difficult for a mixing engineer to know how an entire album will sound in its entirety while mixing an individual track. In some cases a given track may be perfect on its own.

However, when that track is placed within the context of an album, slight adjustments in level or frequency balance may be required.

Given the amount of music distributed online, an album needs to stand up from start to finish to be noticed in such a competitive market. If the final goal is to create a product that is ready to be played on the radio, distributed online, or sold as a physical product, it should be mastered.

Mastering helps say something about the professionalism of the artist, from the arrangement of certain styles of songs to the volume of the recording to the pacing of the tracks. If an artist is serious about their music, they should make sure that someone with experience signs off on the finished product.

3. What kind of improvements can be expected from mastering?

Mastering can help to achieve the correct balance, volume, and depth for a style of music. It can add clarity and punch to music, giving it more vitality.

The idea behind mastering is that a product will sound better after it is treated by the mastering engineer. The degree with which a mastering engineer can achieve this is dependent on the given mixes. In some cases there may be limitations or compromises that need to be made.

One limitation of mastering is the inability to restore severely distorted material. Distortion in a mix is like corrosion; once present it cannot easily be removed and has permanently destroyed a part of the material.

While mastering can mask the effect of some types of distortion, it is essentially covering blemishes that should be addressed before the mastering stage. A common misconception is that mixes should be as “hot” as possible. With the advent of 24 bit digital technology there is no reason why mixes have to “go into the red.”

Most mastering engineers recommend a cushion of anywhere between -6 to -10 dBFS from peak level to help ensure that clipping does not take place and to allow room for processing.

In addition to peak level, the crest factor (peak-to-average ratio) is very important. While dynamic range can always easily be reduced, it is very difficult to undo the effects of over compression or limiting.

If the internal balance of a stereo mix is off, there may be compromises in the sound of the mastered track that will need to be made. For example, if cymbals or a vocal is very sibilant and bright while other parts of the mix are dark, it can be difficult to balance the overall sound in a way that enhances all elements.

In addition to frequency, levels between tracks may also be an issue. If the mastering engineer is given a stereo mix (as is usually the case) specific individual components of the mix cannot be completely isolated and processed separately.

While there are techniques such as de-essing, mid/side processing, equalizing or compressing for a specific imbalance, the results will likely not be as good as with a mix not having these issues and allowing the mastering engineer to address the balance on the whole.

One method of getting around internal balance issues is to provide alternate mixes. Some examples are vocal up/down mixes or mixes where one EQ is favored over another. Another method is supplying the mastering engineer with “stems” or sub mixes of the stereo track.

These might include a separate stereo mix of the vocals or instruments that when summed together are the same as the stereo mix minus any stereo bus processing.

In this case the mastering engineer is placed slightly in the role of a mix engineer and can make adjustments that wouldn’t be possible with a stereo mix alone. Another advantage with using stems is that alternate masters can easily be created such as radio edits, instrumental and vocal-only masters. 

Another area where “fixing it in the mix” is better than “fixing it in mastering” is when dealing with the issue of noise.  Mute automation on individual tracks should be used where there are noises during sections of a track that are not contributing to the mix.

Some examples are electric guitar hum/buzz on intros, outros, and breaks, bleed from headphones on the vocal track when the vocalist is not singing, drummers laying down their sticks after cymbals have faded but while other instruments are still playing at the end of a track.

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 I in a series of the 10 most frequently asked questions about mastering. Stay tuned for Parts II & III where we’ll cover the remaining 7 questions.

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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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