Tuesday, October 27, 2015

Answers To Common Questions About Dynamics Processing

Dynamic range, compression, noise gates and more, explained

What is dynamic range?
Dynamic range can be defined as the distance between the loudest possible level to the lowest possible level.

For example, if a processor states that the maximum input level before distortion is +24 dBu and the output noise floor is -92 dBu, then the processor has a total dynamic range of 24 + 92 = 116 dB.

However, the average dynamic range of an orchestral performance can range from - 50 dBu to +10 dBu on average. This equates to a 60 dB dynamic range. 60 dB may not appear to be a large dynamic range but do the math and you’ll discover that +10 dBu is 1000 times louder than -50 dBu!

Rock music on the other hand has a much smaller dynamic range, typically - 10 dBu to +10 dBu, or 20 dB. This makes mixing the various signals of a rock performance together a much more tedious task.

Why do we need compression?
Consider the previous discussion: You are mixing a rock performance with an average dynamic range of 20 dB. You wish to add an un-compressed vocal to the mix. The average dynamic range of an un-compressed vocal is around 40 dB.

In other words, a vocal performance can go from -30 dBu to +10 dBu. The passages that are +10 dBu and higher will be heard over the mix, no problem.

However, the passages that are at - 30 dBu and below will never be heard over the roar of the rest of the mix. A compressor can be used in this situation to reduce (compress) the dynamic range of the vocal to around 10 dB. The vocal can now be placed at around +5 dBu. At this level, the dynamic range of the vocal is from 0 dBu to +10 dBu. The lower level phrases will now be well above the lower level of the mix and louder phrases will not overpower the mix, allowing the vocal to ‘sit in the track’.

The same discussion can be made about any instrument in the mix. Each instrument has it’s place and a good compressor can assist the engineer in the overall blend of each instrument. This brings our discussion to another good question…

Does every instrument need compression?
This question may lead many folks to say “absolutely not, over-compression is horrible!”

But this statement can be qualified by defining “over-compression.” The term itself must have been derived from the fact the you can hear the compressor working.

A well designed and properly adjusted compressor should not be audible! (Of course this can make a well designed compressor difficult to demonstrate.) Therefore, the over-compressed sound is likely to be an improper adjustment on a particular instrument.

Why do many consoles put compressors on every channel?
The answer is simply that most instruments need some form of compression, often very subtle, to be properly heard in a mix.

Why are noise gates needed?
Consider the compressed vocal example above and you now have a 20 dB dynamic range for the vocal channel.

Problems arise when there is noise or instruments in the background of the vocal mic that became more audible after the lower end of the dynamic range was raised. (air conditioner, loud drummer, etc.) You might attempt to mute the vocal between phrases in an attempt to remove the unwanted signals, however, this would probably be disastrous.

A better method is to use a noise gate. The noise gate threshold could be set at the bottom of the dynamic range of the vocal, say -10 dBu, such that the gate would “close” out the unwanted signals between the phrases.

If you have ever mixed live you know well the problem cymbals can add to your job by bleeding through your tom mics. As soon as you add some highs to get some snap out of the tom the cymbals come crashing through, placing the horn drivers into a small orbit.

Gating those toms so that the cymbals no longer ring through the tom mics will give you an enormous boost in cleaning up the overall mix.

Thanks for PreSonus for this information.

Posted by Keith Clark on 10/27 at 12:20 PM
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Monday, October 26, 2015

Church On The Move Upgrades To Solid State Logic

CCI Solutions provides SSL L500 consoles and a Live Remote Expander for the Main and Oneighty auditoriums.

The main Tulsa campus of Church on the Move (COTM) occupies around 300 acres and is home to several venues including the 2500-capacity Main Auditorium, a 1500 capacity space in the Oneighty building (youth ministry), and an outdoor amphitheater.

To compliment its high-energy events and professional productions, the church has recently purchased two Solid State Logic L500s and a Live Remote Expander for its Main and Oneighty auditoriums from system integrator and SSL Partner, CCI Solutions.

Andrew Stone has been the production manager at COTM for nearly 11 years and has been a professional front of house engineer for 25 years.

He has a reputation for an uncompromising approach to preparation, quality, and ambition.

“We like to do things big and with tons of quality,” he notes.

“Most church services are a three-camera shoot - we use 11 cameras… My ‘A’ crew are mostly professional touring people that I have been fortunate enough to recruit… The PA systems are full-blown line arrays, and now we have the SSLs for front of house…  That is what sets us apart.

“Musically, we love organic, real, precision, clarity… It’s all about quality… In rehearsals for the weekend we’ve taken five hours to work on four songs - and we’d done those songs before! We’re very committed to getting it right.”

Stone has been well known as a devoted analog fan for as long as he has been working, and in COTM’s main auditorium he had been using two large-format analog live consoles for some time. However, after considering how to better serve the needs of the ever more complex productions, he decided to audition a variety of digital live consoles - including the SSL L500.

“The SSL sounded the best - plain and simple,” he states. “And it wasn’t just a little better - it was night and day better… It was the only console that I thought could exceed the beautiful analog sound that I had been used to…. Not just match it, but exceed it. There’s a clarity and a transparency that comes from the SSL platform that I have never experienced before.

“I now get fantastic, positive comments from people that normally don’t pay attention to the sound… All they know is that they are sitting in the seat, listening, and it’s giving them a different kind of experience. That’s a huge win… It really is the only digital console I would replace an analog console with.”

A recent increase in the number of Sunday services means that the Stone and his crew now alternate 9am, 10am, 11am, and noon services on a Sunday between the main auditorium and the Oneighty building, allowing time for audience entrance and exit at each venue. Those are in addition to other events and activities that continue at COTM seven days a week. Both the main auditorium and the Oneighty venues now have L500s installed, and a Live Remote Expander will be going into the main auditorium to provide an additional 36 faders and large multi-touch screen.

Stone has quickly mastered the new platform.

“I’ve become very fast at moving on it - I don’t even think about it anymore. I never thought that would happen after an entire career of analog consoles… I’ve got one thing on the small screen, I’m using the overview monitor to watch all my fader levels, and I’m using the main touch screen to maneuver… And I’m really starting to enjoy duplicating faders and VCAs, and using the independent layer and bank switching on fader tiles…”

These control aspects help Stone put the emphasis on his creative role: “That was my biggest problem with most digital platforms,” he says. “They made me feel like I couldn’t be creative… Like I had to be an IT genius rather than a Mixer…. I wouldn’t dream of doing what I want to do, at the level I’m doing it at, without a certain kind of console at my fingertips.

“The SSL Live platform has made me think more about that part of it… I feel I am being more creative now than I have been in the last 10 or 15 years…”

One special aspect of COTM’s live production workflow is a simultaneous broadcast mix that is used for the Church’s streamed output - an increasingly significant part of all services. “We have thousands of people listening to the stream every time we do a service - it’s like another church out there.

That broadcast mix is created at front of house, and requires a specific solution.

“We don’t have a separate broadcast facility that we’re re-mixing in,” explains Stone. “...For a church service I might have music at 100dB, but then I’ve got big parts of the event that are all spoken, and those will be closer to 70dB. For a live broadcast mix that’s a big level disparity between segments - if you’re streaming on your computer and the talking is 30dB lower than the music you won’t have enough volume available to hear it.”

Stone mixes the spoken section/MC microphones to a ‘Stem’, which is a type of sub-group unique to SSL Live consoles. Stems are available in Dry or Full Processing types and with independent send levels, flexible onward routing, and more feed point options than auxiliaries.

The band is mixed to a Master Bus and that, plus the MC Stem, are routed to one of the console’s four matrices. One set of Matrix outputs is used for the PA and one set is for the broadcast feed, with ‘MC’ contributions to the PA output set at -30dB. This allows Stone ultimate flexibility during a show. He can adjust the MC mix, as well as both band and MC contributions to the PA and broadcast feeds, with full processing.

“It does put another level of complexity on how you have to mix,” says Stone. “But the L500 made life easy, so now it’s just not a big deal.”

Duke DeJong of CCI Solutions, which supplied the consoles to COTM, comments: “Church on the Move has amazing musicians and top notch audio engineers; they demand the best.  The L500 has not only met the needs of their team, but has far exceeded all of their expectations… Every little adjustment pays huge dividends in the mix.”

SSL’s George Horton adds: “Solid State Logic is very happy to have the opportunity to work with Church on the Move. Their commitment to quality and technology leadership in the House of Worship Market is truly second to none.”

Andrew Stone also contributes to the COTM Seeds blog with articles about live sound production.

Solid State Logic

Posted by House Editor on 10/26 at 10:43 AM
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Monday, October 19, 2015

The Basics & Essentials Of Power Amplifiers

A power amplifier is a device for making a larger, more powerful signal out of a small, weak signal.

It was on February 18, 1908 that Lee DeForest was granted U.S. Patent No. 879,532 titled “space telegraphy,” but in actuality, the patent was for a vacuum tube triode amplifier that DeForest called the “audion amplifier.” For the first time it was possible to amplify signals.

The first amps were developed to support radio. Later, other applications were developed. All electronics depend in some way on amplification. Our industry uses Amps for audio.

An amp consists of two basic elements:
1) A power supply that supplies a large energy source and
2) An amplifier section that modulates the large energy source in accordance with the small signal

The result is a large, powerful signal.

Classes of Amps
Class A: A small signal modulates a larger current. This larger current is present when the small signal is not present. Efficiency up to about 26 percent. Excellent quality sound.
Class B: Uses a push-pull arrangement where one amplification device operates on the positive side of the waveform and another operates on the negative side. Efficiency up to 75%. Sounds bad because of distortion caused by switching from one device to another.
Class C: A small signal turns a larger signal on or off. There is no in-between state. Efficiency up to 90 percent. Not usable for audio. Audio requires accurate reproduction of all levels—not just no power and full power levels.
Class D: A variation of Class C. Class D is a way of modulating a Class C amplifier to allow it to carry audio information. Sounds very good with latest technology. Efficiency up to 90 percent. Produces Electromagnetic Interference (EMI)
Class AB: A variation of Class B. Always has a small current flowing (class A operating region) and this eliminates the switching distortion inherent in Class B. Efficiency up to 65 percent. Sounds excellent—if well-designed
Class H: A variation of Class AB. Changes the power supply voltage to the amplifier depending on the signal level. Improved dynamic efficiency. Requires complex power supply. Single tone efficiency up to 65 percent Sounds excellent—if well-designed

Power Supplies
Type 1: Standard (Analog) Power Supply: Efficiency to 80 percent. Heavy, for large amounts of power. Components are large

Type 2: Switching Power Supplies: Efficiency to 90 percent. Lightweight—even for large amounts of power. Components are small. Can support universal line input voltages. Can support regulated output w/no power loss. Produces EMI

Efficiency is the power output of a device divided by the power input to the device. Power input that is not outputted by a system is dissipated as heat from the system. The greater the efficiency of an amplifier, the less AC power is required to deliver the same output power to the load.

A typical Class AB amplifier with a 65 percent efficiency, used with a standard analog power supply running at about 80 percent efficiency yields an overall amplifier efficiency of approximately 50 percent. Efficiency is typically rated at full power, continuous tone levels.

At lower power levels, efficiency is much worse. It is at low signal levels where Class D and Class H designs offer significant improvements. Class D maintains almost the same efficiency for all power levels while Class H switches to a lower power supply at lower signal levels in order to maintain good efficiency.

Switching Supplies
A switching supply consists of four basic building blocks:
1) A DC supply that operates directly off the AC line
2) A power oscillator which converts the DC supply to a very high frequency, typically 50 kHz—200 kH
3) A transformer that changes the high frequency power signal to the various outputs needed
4) Rectification and filtering stages that produce the DC outputs needed

Smaller & LIghter
Because the transformer in a switching supply is operating off a very high frequency power signal—instead of 50/60 Hz—it can be much smaller and lighter.

This is the major advantage of a switching supply. Since switching supplies are more complex than a standard supply, their circuitry usually costs more. In addition, more parts have to be added to control the EMI produced by switchers.

But these costs are usually more than compensated for by the reduction in size, weight (and cost) of the transformer. In addition, costs of the chassis can be reduced because the weight it has to support is much less. Since the power supply is the heaviest part of an Amp, the entire product is much lighter and easier to handle.

Amp Design Trends
For many years, the standard amp was a Class AB design with a standard analog power supply. Manufacturers have been experimenting with other types of both amplifier design and power supply design in order to support customer needs better.

The primary customer needs today for amps are:
- High Power
- Good Sound
- Low Cost
- Lightweight Construction

“Good sound” is very controversial. There is so much confusion about technical terms for rating good sound—and how these terms actually relate to the listening experience—that the focus today for most consumers is power rating and cost.

Power Ratings
Twenty-five years ago the rated output power of an amp was the continuous tone output level of the amp; a 300 W amp could produce a 300 W tone all day long. Then it was recognized that most audio Amps are not used on continuous tones, but are used on audio signals.

Audio signals consist of many tones of different power levels. So customers didn’t need an amp that produced continuous tones all day; they needed an amp that could produce audio signals all day. This is easier and cheaper to do. Manufacturers today have adopted various methods for rating the power output of their products. This is causing much customer confusion.

Manufacturers, industry and standards groups have contributed to defining how to rate power output for audio Amps. There have been many methods proposed for rating amplifier power. Many of these proposals—such as tone bursts tests—attempt to rate the instantaneous or short duration power levels.

Because audio signals vary in duration and level, the validity of these rating methods depends on how the Amp is used and the characteristics of the signals it is amplifying. Some methods use non-audio signals, such as square waves, to determine the amplifier power rating; this causes the number to be higher.

Today, the primary standards are dictated by the Federal Trade Commission (FTC). Many manufacturers also use standards developed by Industry associations—such as the Electronics Industry Association (EIA). Some manufacturers use other methods.

Safety agencies (particularly those in the European Union, and Underwriters Laboratories here in the US) have developed standards for measuring average continuous power of an amplifier. These are used in turn to measure maximum AC line power consumption and to confirm maximum temperatures. Safety groups have determined that typical worst-case power for an amplifier amplifying an audio signal occurs at one-eighth of the non-clipped output power (measured with a 1 kHz tone).

The amp is then “cooked” with a bandwidth-limited (20-20 kHz) pink noise signal whose power is equal to 1/8 of the tone full power. Measurements are then made for temperature and AC line power consumption. Thus 1/8 of the maximum tone power before clipping represents realistic worst-case continuous power levels for an audio amplifier.

Standards bodies are always reviewing and updating industry standards. We can expect standards for audio amplifier testing and ratings to change as our customers’ needs and usage change—and as our knowledge and technology improves.

Good Sound?
Good sound is achieved by good design tailored to the application and user preferences. But good sound is a subjective thing; different users prefer different characteristics to their sound. Different applications may require a different sound character. A presentation application typically requires good sound accuracy, also called sound clarity.

The industry has developed various solutions to satisfy different user preferences and different applications. The primary way of rating sound clarity is by rating distortion. Distortion compares the accuracy of the large output signal to the small input signal. The lower the distortion, the more accurate the sound.

THD Distortion is rated by specifying THD (Total Harmonic Distortion). A harmonic is a tone whose frequency is an integral multiple of another tone, called the fundamental (pure) tone. If a pure tone is inputted to an amp, the output should be the same pure tone. In practice, the output contains small levels of tones that are integral multiples of the input pure tone. These extra tones are distortion.

The THD rating indicates the percentage level of distortion tones in the output relative to the input signal. Today’s THD measurement equipment measures the output signal only. It measures the amount of harmonic frequencies relative to the fundamental frequency in the amp output. It is therefore important that the measurement input signal be a pure tone—a tone with only a single, fundamental frequency and no harmonics.

Some measurement techniques include some noise—hence the parameter THD + Noise. It is important for the equipment setup to minimize noise as well as have a very low distorti input tone in order to correctly measure the THD of the amp.

It is generally accepted that 1 percent THD is the maximum acceptable distortion for high-fidelity sound reproduction.

Odd or Even Distortion is not always bad. This is because some user preferences for sound character imply certain types of distortion. If the distortion tone frequencies are multiples of odd numbers (3,5,7 for example) the distortion is said to be odd-order.

If the distortion tone frequencies are multiples of even numbers (2,4,6 for example), then the distortion is said to be even-order.

Odd order distortion sounds very bad (dissonant). Even order distortion is like hitting octave keys on a piano; it sounds good—and some users may prefer to have some of this type of distortion. Tube amplifiers produce mainly even-order distortion, even when they clip. Solid-state amplifiers produce mainly odd order distortion—especially when they clip.

A very low THD rating is therefore more important for a solid-state amplifier than for a tube amplifier in order for some users to feel the Amp sounds good. Some users who operate their amps at clipping levels a lot (musicians, for example) may prefer tube amplifiers.

Transient Distortion is a way of rating how quickly an amplifier can react to changes in the input signal. If an Amp takes a little time to react to a change (as always happens), then its output is not faithful to the input signal for the time it takes to react.

Damping Factor
Damping Factor is a way of rating how well an Amp can control the movement of a loudspeaker. Since a loudspeaker is a mechanical device, it will follow the basic laws of physics: when it is put into motion by some stimulus it will tend to stay in motion after the stimulus is removed. This extra motion produces distortion of the sound. A high damping factor enables the Amp to better control the speaker and minimize these extra movements.

Good (high) damping factor is important to achieving good transient response—especially transient bass response. A low damping factor will result in the speaker not reacting quickly to a bass signal, causing the bass to be “mushy”. But some users and applications may require a “mushy” characteristic to the sound.

For example, if the Amp is used for background music, the user may want the sound to be “mushy” or “mellow” and non-dominating. In a presentation application, the sound is dominant.

“Mellowing” of the sound can be achieved by degrading transient response and damping factor. Tube amps have much lower damping factors than solid state amps. Therefore, users who prefer a “mellow” sound may prefer a tube amp.

Users with very old source material (records or 78’s) may prefer an Amp with a “mellow” sound. This is because the signal information from these old sources does not contain fast transients in the program material. An Amp that has good transient response will only reproduce more recording defects and surface noise — not additional program information.

Users that want accurate, realistic sound are likely to prefer a solid-state amplifier; good solid-state designs achieve the best accuracy.

Lower Cost
The basic principle in designing for low cost is to not over design! Products must be designed so that they are adequate for their intended application—and no more. Products must be protected from stresses that will exceed their intended application in order to stay reliable.

Making products just adequate for their application saves size, weight and cost. Additional savings in size, weight, and cost are possible by using the new technologies of Class D and H amplifiers and switching power supplies.

Supplied by Technologies for Worship Magazine. For more information visit

Posted by admin on 10/19 at 05:54 AM
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Tuesday, October 13, 2015

Haybale Studio Selects SSL To Capture Bonnaroo Festival

With the help of a local farmer and an SSL AWS 948, The Toy Box Studio becomes Haybale Studio at the Bonnaroo Festival.

Haybale Studio spends four days every summer capturing Tennessee’s Bonnaroo Festival - with a little help from a Solid State Logic AWS 948 SuperAnalogue console.

Music and podcast producer Lij Shaw is normally found in residence as the owner of The Toy Box Studio in East Nashville, TN.

However, every summer he packs his essentials into a trailer and heads 60 miles South of Nashville to a 700-acre farm, along with 85,000 other people.

There, with the help of a local farmer and an SSL AWS 948, he creates Haybale Studio at the Bonnaroo Festival.

The Bonnaroo Festival is known for putting on a wide variety of music and comedy acts for an equally wide variety of music fans - all bound by a love for live music, and the Bonnaroo Code, which in essence says ‘be nice’.

The humor that courses through this celebration can be spotted even in its venue names: The What Stage, Which Stage, Who Stage, This Tent, and The Other Tent.

The Haybale Studio is equally individual, with a particular mission to record original arrangements and performances of music from the bands that are playing at the Festival. These recordings are a kind of In-Session supplement that aim to capture the mood of the fans, the bands, and the moment.

The ‘Haybale’ name originates from a practical solution to the need for sound isolation. The Studio is set up behind the Which Stage, so 85,000 fans and a very big PA need to be supressed somehow.

Sean O’Connell of Music Allies, who originally invited Shaw to set up a recording studio at Bonnaroo in 2005, was inspired by some large bales of hay in a nearby field, and he asked Shaw if these could be used to soundproof the studio.

“I said ‘yeah, that’s a perfect sound insulator.” explains Shaw. “Now we come and we park the trailer here, and then we bring in hundreds of bales of hay… They go all around the trailer… right up above the sides. When you look at the outside you just see a giant bale of hay with a couple of doors. “

What started as a small idea, has grown beyond all expectations. Originally a two-man operation, recording a few bands on limited equipment, now Haybale Studio aims to record three-song sessions with 40 bands over the Festival’s four days.

Shaw: “In the festival we’ll have everything from all-night DJs like Deadmau5, to Scrillex, to Kanye West, to rock bands, to bluegrass… All sorts of stuff. And that is reflected in the studio.” In any hour, on any of the Festival days, Haybale studios could be hosting anything from a couple of mics on a single guitar/vocal act, to a full-on rock band, to a classical ensemble.

At the center of this is an SSL AWS 948 console - equipped for the Haybale challenge.

“This console does some incredible stuff,” says Shaw. “This is like having a rocket ship in the studio… The sound, the tone of it, and the flexibility - and the fact that I can mix on it in real time. I’ve tried this with other mixers and I’ve tried to emulate this by mixing in the computer somehow…It’s just not possible…

“The sound is awesome ...It’s got punch, it’s got attack to it, the detail is incredible, and it’s so quiet. For me a lot of time is spent keeping everything else quiet to match how good the console sounds.”

To give Shaw the flexibility and speed he needs in Haybale, tracks get routed to eight busses which are then sent for parallel compression and brought back into stereo returns, created by using the Stereo Mix mode available on all the AWS 948’s dual-path channel strips.

The mix gets passed on to Joe Hutchinson (Garage Masters, Nashville) who actually masters the tracks as the sessions are happening.

Shaw: “It’s just analog live signal through the microphones, through the wires, through the console, through the outboard, straight down… There’s a quality and purity to that sound… I’ll be damned if I can ever figure out another way to create that…”

Once the mastering is done, the tracks get uploaded to a server, and are then available to radio stations and other outlets all over the country. “...It’s almost real time,” says Shaw. “Within an hour that stuff is accessible and it’s getting aired on radio stations around the country… It’s a really cool way to bring people into Bonnaroo…”

For many recordings there’s a video crew as well, creating a series of in-session videos, available via the Haybale website and on the Bonnaroo YouTube channel (Haybale Sessions playlists).

However sophisticated the distribution though, Shaw has to make sure he always captures spirit of Bonnaroo - keeping the source true to the performance. The AWS 948 is key to that goal: “You can get really intense and fancy within the computer,” he says. “It interfaces with Pro Tools if you want it to… But man, if you just want to run mics through it, and you want a mixer that is just a workhorse, that sounds killer, and works when you need it to… It works great for that too.”

Solid State Logic

Posted by House Editor on 10/13 at 01:14 PM

Monday, October 12, 2015

Sound Productions Names New Southwest Regional Sales Manager

Jay Dominguez brings more than 25 years of experience in focusing on the retail operations of pro audio, musical instruments, DJ and stage lighting

Production and pro audio product supplier Sound Productions, based in Irving, TX has named Jay Dominguez as its new southwest regional sales manager.

Dominguez has more than 25 years of experience in focusing on the retail operations of pro audio, musical instruments, DJ and stage lighting. His experience spans a range of market segments, including recording studios, recording artists, regional and national touring bands, house of worship, EDU and contracting. 

Prior to joining Sound Productions, Dominguez spent 19 years with Hermes Music in the southern region of Texas, serving as artist and public relations liaison as well as NAMM Show operations manager. He is based in San Antonio.

In addition to pro audio products, Sound Productions provides backline rental and production services on a regional basis. Headed by CEO Charles Kitch for more than 40 years, the company also recently opened new, state-of-the-art training spaces as well as a new showroom.

Sound Productions

Posted by Keith Clark on 10/12 at 12:23 PM
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In The Studio: A Tale Of A Project-Saving Monitoring Technique

As guitarist Jeff Baxter once said to me during the “heat” of a Rod Stewart recording session: “We can do anything—the impossible just takes a little longer.”

Along with talented musicians like Jeff, engineers, producers, and live mixers often are called upon to solve problems that are at first glance, flat impossible.

To a technically challenged client, all the flashy and complicated gear, computers and the alacrity at which a pro uses them to produce nearly instant, seemingly magical results (I think) hypnotizes or lulls people into a state of “anything is possible”—even though what they want defies the basic laws of physics!

One such situation occurred to me a while back when I was in Sydney Australia recording an album with an R&B band called The Rockmelons. (A rockmelon is Aussie for a cantaloupe if you were wondering) Australia is a wonderful and mystical place especially out in the middle of the country—a U.S.-sized desert.

So perhaps the laws of physics are suspended in parts of the “down under” but they were not for us at EMI 301 Studios in downtown Sydney.

The lead singer in the band, at that time, had the worst case of “red light” fever I’ve ever encountered—actually more like a severe headphone phobia. As soon as he heard the track and his voice in the cans, he acted intimidated and overwhelmed; he would stop, not sing at all, or sing terribly.

All of us were puzzled because at live gigs in front of an audience, he was wonderful—the main attraction. The most peculiar thing was that if we suddenly stopped the track’s playback, for a few measures he would sing the most spectacular soul riffs and melodies all acapella.

Robin Smith, the producer was obviously extremely concerned because if this guy couldn’t sing, we would not have an album—they were not an instrumental group.

The first flash of “can do” brilliance came from Smith when he had the second engineer setup a two-track tape deck so that it recorded the same audio feed from my vocal recording chain at the session’s multi-track received. This machine was to be kept it in record, rolling at all times.

The second engineer and the producer also worked out a system of hand signal routines—a rating system where the assistant would jot down the two-track’s tape counter number and a 1 to 5 rating whenever the producer heard a piece of a vocal he liked. These chunks of vocals might be used in the final vocal compilation process.

So for the rest of the day, whenever I would stop playback, we captured all these cool riffs and lyrics on the two-track. Later, after the singer left, the producer and I would “fly” in all the good bits into the multi-track master vocal take. To say the least, this was not a satisfactory record production method that grew very old very fast.

So after a couple of days when the singer did not get any better, we decided to tackled the problem at the root cause: what was it about the phones that put this otherwise great singer off? We played with volume; mix, compressing the phone mix, reverb and other effects, different brands of headphones—everything we could think of.

We determined that the singer was a sensitive fellow who just felt physically uncomfortable and a little paranoid wearing headphones—and it was worst when music was coming out of them.

So I remembered a trick I saw Crosby, Stills and Nash used ‘back in the day’ at a Hollywood studio called Sound Labs. I was working on another project in studio 1 and they were banging away in studio 2.

Of course everybody in those days would occasionally check out “who was in the other room” and what it sounded like. It was a great learning atmosphere with a very rich and free exchange of remarkable ideas from very talented people—artists, engineers and musicians who sometimes were not that technical but always open to any sonic experiment no matter how ludicrous-sounding.

I went in and saw they were using two Yamaha NS10ms or Auratone monitor speakers mounted on mic stands instead of headphones!

I think it was Graham Nash who said they had always harmonized as a group around a single microphone listening to each other more than the track. I went into the studio during a playback to hear how low in volume the speakers were. They were very quiet.

But the big revelation to me was when I stood equal-distant between the two speakers and discovered they were flipped in polarity—out of phase from one another.

Apparently I (being a recording engineer trying to be vigilant for such serious problems) was much more sensitive to this than CSN. It was also true that none of them ever stood exactly equidistant between the speakers.

Further, this “wrong” hookup was anathema to everything I was ever taught or had experienced. But this was situation where practicality outweighs technical correctness.

So I used that idea for my singer in Australia and it worked! The main point of this trick is to place the vocal mic exactly—dead on—in the null-point of the two speakers. I did this by playing the cue mix and listening to the mic channel in solo.

I put headphones to hear only the mic’s signal and would just move the mic around until I got minimum speaker spill or leakage.

Is the leakage a problem? No, not unless you want to do some wacky dance remix where the lead vocal is solo’d or you produce and record an entirely new track under the vocal.

The other ‘detail’ is to make sure you play only the most minimal track mix out on the speakers and try to keep it mostly monaural. You might have to play with panning positions etc. Play just enough elements of your track’s production for the singer to sing well.

I’d probably leave out most of the sweetening ideas, fancy percussion playing and vocal effects off. I’d also try to keep the bottom end not too big and the Yamahas will help in the regard.

Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Be sure to visit his website

Posted by admin on 10/12 at 06:43 AM

Tuesday, October 06, 2015

Analog Snake Systems And Their Applications

Back in 1996, I predicted that the good old analog snake would be gone, soon to be replaced by digital counterparts. But something strange happened…the world did not change.

As both a manufacturer of snakes and a distributor to the Canadian market, we sell more analog snakes today than ever before. With that in mind, let’s have a look at analog snakes and their application. There are basically two formats: traditional floor boxes or rack-mount snakes.

The floor box is less expensive, generally has fewer features, and is maybe a little less sexy. It’s less expensive to eliminate expensive multi-pin connectors by attaching the cable directly to the snake head. This is usually stored together with cabling in a large case.

Rack-mount snakes are usually split into two, with the head stored in one case and the cable in another.

These are generally outfitted with multi-pin connectors and more “concert specific” options. System techs find the upright positioning of rack-mounted snakes to be easier to use with less spaghetti to sort through in times of need.

Most concert snakes are set up with a 200- to 300-foot (75- to 100-meter) multipair cable (trunk) for front of house and a 50-foot (15-meter) trunk for monitors. These are “Y’d” via simple hardwire or “split ” via isolation transformer. A ground lift switch is usually on the monitor out.

Applications of analog snake systems.

Some trunks have the XLR break-out splay attached, while others employ a multi-pin at the breakout that allows the splay to remain inside the console dog-house. The size and magnitude of the snake can vary immensely depending on the situation, and they can be scaled depending on individual needs.

Types Of Cable
Most snakes employ twisted pair balanced wire, the same type of cable that telephone companies have used for 100 years to transmit phone calls thousands of miles across the continent. Twisted pair is highly immune to electro-magnetic fields and well suited for these relatively short runs.

These are usually shielded with a foil wrap (and drain wire) to further reduce noise from RF contamination. Better snake systems are equipped with RF filtering circuits to further reduce noise that is emitted from lighting dimmers, power transformers, and electrical systems.

Trade-offs with cable are size and performance. Back in the early days, folks used to use Mogami - a very flexible recording cable - for concert touring. Their idea was to get lower capacitance cable for maximum high-frequency performance. But this cable corkscrewed in no time because it was too soft.

At the other extreme, we’ve built snakes for use in the arctic (literally). These are quite stiff when warm, but would remain somewhat flexible in sub-zero temperatures, and as you might imagine, this type of cable is very difficult to deploy. A nice balance between ruggedness and flexibility is preferred.

Today, most touring companies opt for 50 or so audio channels with individual PVC jackets that hold the shield in place.

The classic multi-pin connector.

This is extruded “hot” so that it embeds itself into the outer jacket to resist corkscrewing. A cable with a 1-inch (2.5-centimeter) OD (outside dimension) is usually preferred as it is relatively easy to handle while having sufficient channel capacity.

Most concert touring companies employ a separate drive snake, which keeps high level signals away from more sensitive mic levels which helps reduce cross-talk.

Multi-pin Connectors
Multi-pins connect the snake trunk cable to the snake head and splay, with the two most popular connectors being Mass and Veam.

Originally developed by Cannon for the oil exploration industry, the Mass connector is a screw-on connector that comes in 122 pin (40 channels) and 176 pin (56 channels) formats in a hermaphroditic configuration with half of the pins male and female.

The advantage with the hermaphroditic design is that you can deploy the cable from either end. The downside is that the pins are relatively small and one has to be careful when mating as misalignment will cause damage.

The C5015 Mil Spec developed by the U.S. military employs a quartertwist bayonet connection with a rugged circular metal body in a variety of pin configurations that often range from 37 pins (12 channels) to 201 (66 channels). The most common used on concert snakes is a 150-pin (50 channels) design originated by Veam and now offered by a number of companies.

The benefit of the circular bayonet is that the pins are larger and easy to replace due to the rubberized pin insert. The 201-pin option allows for more channels, but cable size and pin fragility should be considered.

Custom shops build snakes using all of these. Considerations when selecting a multi-pin connector includes compatibility with older snakes, crossrental opportunities with other sound companies in your area and personal preference based on experience with a given product.

Snake “heads” come in a multitude of sizes, varieties and capabilities.

With more than 150 wires crammed into such as small space, multi-pins are time consuming to wire and repair. Outfitting the snake with a proper Kellems Grip strain relief will protect the sensitive wiring inside the connector and reduce field service.

Snakes & Splitters
Most concert snakes are in fact splitters. What this means is that the signal from the stage microphones and direct boxes are split to feed the house mixer and monitor mixer.

Often these are set up with an auxiliary output that can be used to feed a broadcast truck or recording system. And with more bands using backing tracks and computers, we find that 48 channels is now the minimum used for most productions.

As the XLR connector is the only ‘standard’ in balanced professional audio, it’s a really good idea to outfit the snake with an XLR panel or a spare trunk with XLR outputs. Broadcasters are well familiar with the problems of hum and buzz that plague most stages and therefore will generally have a transformer isolated input panel at the truck.

This leads to a discussion on transformers. Audio transformers allow audio to pass while blocking DC. They also allow the ground to be disconnected between consoles, thus eliminating ground loops. Good quality transformers are expensive but they sound much better (less distortion, more linear, less phasing) and can save countless hours of trouble shooting as they eliminate noise.

Normally, the house console is connected directly to the microphone, thus providing a return path for 48-volt phantom power for condenser mics and direct boxes while the monitors will be isolated. To save money, some sound companies will opt for a “Y” or hard-wire split whereby the microphones wires are simply wired in parallel to feed the monitors and the ground is disconnected. This can work, but is prone to noise and ground loops.

Passive Versus Active
Transformers do not require any form of power to make them work. They are passive. Mic bridging (or splitter) transformers are available with 1, 2 or even 3 isolated outputs.

The more splits, the more expensive. This sometimes leads to companies opting for active splitters, which employ preamp circuits (like the ones in your mixer) to buffer the microphone signals. When an active splitter is employed, the preamp essentially moves from the mixer into the snake.

This also applies to digital snakes. In other words, the sound of your expensive mixer has now been replaced by low-cost ICs. Transformers and active buffers react very differently when they’re overloaded. When active circuits are hit hard, they distort causing a very harsh sounding square wave (clipped output).

When transformers are hit hard, they saturate. Good ones such as those made by Jensen round out the sound in a more gradual way, rendering a warmer tone.

A point of contention with digital snakes is “who” is in control. With a passive snake, each mixing desk is directly coupled to the electrical output from microphone. Many system techs prefer passive snakes because it allows each engineer to set the mic levels based on their needs.

For instance. when controlling levels for in-ear monitors, having the house engineer turn up the trim level without notice can cause serious ear damage to the artist on stage. (And sometimes the monitor engineer’s job.)

A serious gamut of analog snake and other interconnect found on typical live sound applications.

And although one cannot deny the appeal of deploying a single fiber versus and a big fat snake cable, when a snake cable fails it is usually only one channel that goes down, not the catastrophic problem of a complete system failure.

Analog snakes can be repaired with a soldering iron while digital alternatives require the band to carry a spare and extra logistical planning to return the down unit to the factory. For many, field serviceability is a primary concern.

It’s important to note that most signals traveling around a stage are mic level, including the output from the many direct boxes that are used to feed bass, acoustics and keyboards to the PA system. By ensuring all levels are the same, should a channel fail, it is simply a matter of re-patching to a spare. This “standard” practice makes it easy and quick to resolve issues, especially when 20,000 fans are waiting to be entertained.

Modular Versus Flat Panel
For decades, snake manufacturers have been producing flat-panel snakes that mount all of the XLR connectors on the front panel with multi-pins below. This is the most economical method of producing a snake.

More recently, there has been a shift towards modular designs that enable the user to reconfigure the system to adapt for future needs. The benefits with modular designs include being able to add extras such as sub-snakes, cross-patching capabilities, or extra outputs for broadcast and recording feeds.

Being able to quickly replace an input strip should field service be required is an added benefit. The down-side is a higher cost at the outset due to added metal work and electronics employed to produce the frame and individual channel strips.

When on tour, 99 percent of all problems can usually be traced to cold solder joints, faulty wiring or second rate connectivity due to bad cables. Truck vibrations are particularly hard on equipment. And the biggest monster of all is the snake system. If your interconnect system fails, you’re late for dinner or worse, you may lose the gig.

The snake system is the umbilical cord that brings it all together and should not be skimped on.

Peter Janis is president of Radial Engineering and has worked in professional audio for more than 30 years.

Posted by Keith Clark on 10/06 at 06:30 AM
Live SoundFeatureBlogStudy HallAnalogInstallationInterconnectSignalSound ReinforcementSystemPermalink

Monday, October 05, 2015

ASP8024 Returns To Isle Of Wight Studio

Studio Humbug upgrades by replacing the Audient console they had previously sold.

Studio Humbug owners Jim and Rob Homes have taken delivery of a 24-channel Audient ASP8024 analog mixing console, complementing their growing collection of vintage mic pres.

“The pres we have are from 1961 and VERY colored, this is a wonderful characteristic - but not on everything,” says Jim.

“The ASP8024 has allowed us to integrate the sound of the pres whilst using the headroom, EQ and reliability of the Audient. The two worlds marry together perfectly.”

Audient wasn’t an uninformed choice of desk; Jim & Rob had previously owned a much bigger ASP8024 that they’d sold on, due to “moving about a lot. Normally things that leave our studio never come back,” confesses Jim.

“The Audient is the first time. After a while, we realized that it solved a lot of the problems we had. We also re-listened to some of the mixes we did on the Audient and they sounded great.”

The return of the Audient - albeit in a smaller form - seems to have worked out just right.

“It’s perfect for us. The ASP8024 is a fantastic central hub for our studio,” continues Jim.

“The built in routing makes selecting signals easy and also encourages us to make decisions to tape. It also allows us to work quickly. Its high fidelity lets us make better decisions at the tracking stage and hear more of the mics we are using.”

Based in a Victorian water tower on the Isle Of Wight, Studio Humbug boasts a 900sq ft live room and a certain eccentricity, which suits the boys’ passion for vintage gear rather well.

“We are slowly working towards a sidecar. We have 16 channels and in a few weeks’ time will take delivery of the first 8 EQs to complement them. In time they will be split across the two rooms we have, eight up and eight down.”

There are a few projects in the pipeline for Studio Humbug, including albums for Champs, Wolf People and Eilmer Reed, lots of single mixes for other acts “….and our ongoing work with the EMI music library team,” details Jim. “Then there are more studio improvements, some interesting live sessions and our continuing threat to start a small label.”

With so much on their plate, Audient recommends they keep it simple, and hang onto the desk this time.


Posted by House Editor on 10/05 at 07:25 AM

Tuesday, September 29, 2015

Kenny Larkin Adds API BOX Console To Tedra Studio

Electronic dance music studio in Detroit returns to analog roots with API console.

Many people know Kenny Larkin as an artist—but few know that he is also a producer and studio owner, and his studio just got some familiar new gear.

After his older digital mixer broke down, Larkin knew it was time to return to analog, and knew API would be just the ticket.

“I had to look for something new, and I had that analog itch again. I wanted more warmth in my recordings without having to keep going through outboard gear. What I got after buying the BOX was warmth and PUNCH. I call it my ‘mojo BOX. I’ve had a few analog consoles during my career, but nothing sonically on this level.” 

Larkin owns Tedra Studio in Detroit, Michigan. He opened Tedra twenty years ago at the beginning of his career.

“Techno started in Detroit back in the late 80’s, so my style or genre is specifically Detroit Techno. EDM is the commercial side of dance music, which is totally different than our sound. DJs from the EDM scene often grew up listening to our stuff.”

Two decades later Larkin knows what works for him and what doesn’t, which is why he turned to API to move beyond his old mixer. “The BOX is the centerpiece of my little studio, which is comprised of the typical DAWs and a small amount of high-end outboard pieces.”

Tedra is used exclusively to record electronic dance music, and Larkin has managed to keep the studio for his own private use.  When he saw the BOX announced by API in 2013, it was not long before he was ordering one of his own.

“I did a demo at Vintage King, and I was sold.” Although he knew about API for years, it was the first-hand experience that closed the deal. “That’s when I first heard the classic API ‘punch’ everyone talks about. I loved the form factor, how it was small yet powerful. Not a toy, but not a monster to learn to use.”

And since then, Larkin has continued to find things he likes about his console. “Everything about the board is amazing. Any pro user would be giddy knowing that each channel has inserts and effect sends. The compressor is also fantastic across the stereo buss. I also like that I can assign it to different input channels. With the BOX, I find it’s quicker and much easier to get things sounding fat.”

While his work takes him around the world, Larkin was pleased to return to Detroit and finish off two long-term projects. The difference in his workflow, he says, is undeniably the BOX. “The most important thing it did was get me excited about recording music again. It inspires me to make amazing recordings.”


Posted by House Editor on 09/29 at 03:14 PM

Monday, September 28, 2015

Audient Announces ASP800 Worldwide Shipping

Offering eight console mic pres, Burr-Brown A-D converters, two discrete JFET instrument inputs and two brand new ‘Retro Channels."

ASP800, the new, eight-channel mic pre & ADC from Audient is now shipping worldwide and is available at all Audient dealers.

Following the release of the bus-powered audio interface iD14, the ASP800 offers eight Audient console mic pres, Burr-Brown A-D converters, two discrete JFET instrument inputs and two brand new ‘Retro Channels’ featuring tone controls HMX and IRON.

HMX is a custom harmonic distortion designed to emulate the sweet, thick tones often associated with 1960s tube designs, whilst Audient’s brand new variable British transformer saturator, IRON instantly adds sparkle, replicating the coveted transformer ‘zing’ of British audio in the 1970s - both giving new dimensions to a recording.

“Not only does ASP800 boast an incredible feature set, Audient is one of the only companies to offer custom transformer saturation at this price point,” explains Tom Waterman, Audient’s technical director.

“It’s the perfect addition to your existing setup, offering eight additional Class-A inputs, making it ideal for recording drums - or if you simply need more channels.

“As with all mic pres across the Audient range, these are the same design as those found on the flagship ASP8024 console designed by David Dearden. Audient is certainly known for its natural & open mic pres, but that doesn’t mean we don’t understand the importance of adding colour,” he adds, highlighting the unit’s ‘Retro Channels’.

Key features also include Burr-Brown A-D converters, ADAT connectivity, a Word Clock input and balanced analog line outputs - all packed into a convenient, 1RU rackmount.

ASP800 has an RRP £599 inc VAT or $799 MAP and further details - including videos - available on the Audient website.

Feature set:
· 8 x Audient Console Mic Pres
· 2 x Retro Channels with Dual Stage COLOUR Saturation Controls - HMX & IRON
· 2 x Discrete JFET Instrument Inputs
· 118-dB Burr-Brown A-D Converters
· Balanced Analog Line Outputs
· ADAT Output for Expandability
· Word Clock Input
· All-Metal Enclosure


Posted by House Editor on 09/28 at 08:12 AM

Friday, September 25, 2015

Wez Clarke Adds SSL AWS 948 Console To Refurbished London Studio

Mix engineer makes the decision to upgrade his studio, and his sound, with Solid State Logic analog console.

Mix engineer Wez Clarke has mixed top-sellers for Naughty Boy, Beyoncé, Jess Glynne, Rudimental, Little Mix, Duke Dumont, Tinie Tempah, Ella Eyre - and this year, along with Clean Bandit and Jess Glynne, collected a Grammy for Best Dance Recording of 2015 (Rather Be).

Clarke recently refurbished his London studio, with acoustic design by Munro Acoustics, ME Geithain monitoring, and a Solid State Logic AWS 948 console with δelta-Control.

After eight years, working in an increasingly cluttered space, Clarke made the decision to reassess his studio, and his sound.

“It was more to keep things fresh for me - to have a nice environment to go into every day… It’s really important. Now I’m surrounded by quite a lot of wood and it’s nice and calm in there.”

The breadth of Clarke’s work has expanded somewhat in recent times, as his talent in the mix room has become more widely recognized.

“For some reason I’d got pigeon-holed into dance music - before that there was a lull in dance music and everything I did was Hip Hop and Grime. These days I do much more than that - ballads, guitar-based… All sorts, really.”

Clarke needed a space (and equipment) that would work in any genre, and for him one of the primary strengths of the AWS is its sonic versatility.

“Because it’s so clean, you don’t have to imprint the analogue desk sound,” he explains. “People don’t necessarily want a sound printed on a project at the mixing stage. They might like their sound - they just want it sharper and punchier, and so on - so I need to be quite careful. I think the desk gives you the best of both worlds; you can get either sound you want.”

Another aspect of the console that suits Clarke’s work is the flexibility of the dual-path input channels. Each AWS 948 channel has two input paths and three main routing modes: In-Line Tracking (switched input and return), In-Line Mix (two mix inputs per channel - one on fader, and one on V-Pot), and Stereo (Stereo input channel, including width control). This means the AWS 948 can be configured with 24 stereo channels to mix.

“I think the stereo channels on the AWS were a massive selling point for me,” says Wez. “Everything that comes to me is stereo - literally everything - all the stems… It makes life a lot easier to move one set of knobs rather than two.”

Clarke’s working method has evolved significantly recently, as he used to do most mixing in-the-box with Pro Tools.

“I basically have everything in Pro Tools at zero and do it all on the desk in the analogue world… Once you get the bug - once you hear the sound - it’s difficult to go back; and now I’ve got δelta-Control and I can automate using that, I hardly do anything in-the-box.”

δelta-Control is a plug-in that enables automation of an analog console (SSL AWS or Duality) as if it were a DAW plug-in. The automation system in the DAW is used to record and playback control data from the faders and switches on the console as an alternative to using the SSL legacy automation system.

According to Clarke, this console-centric approach has actually allowed him to work considerably faster than before: “It really does speed things up. I know everyone says that, and you wonder how, but it does. When you’re using a mouse you can only deal with one thing at a time, but on the desk you’re thinking ahead - going all over the place; The EQ controls are right in front of you – straight away.”

In addition to the AWS 948, Clarke has also installed an SSL X-Rack - a 4U modular system with a variety of analog modules available. This one has five Stereo Dynamics Modules, one Stereo EQ module, and a Stereo Bus Compressor (G-Series console centre-section compressor).

“I use the bus compressor over the rec bus, which sums to the master bus,” Says Clarke. “I slam it hard and punch any tracks in on the desk that I want to parallel compress - mainly percussion elements.”

Total Recall on the X-Rack module means Clarke can come back to mixes that are in progress at any time, without losing the analog outboard settings.

Solid State Logic

Posted by House Editor on 09/25 at 08:00 AM

Thursday, September 24, 2015

For The Record: The Past Tells Us Much About The Future Of Live Recording

Looking back at the historical trends of our industry to stay ahead of the curve and keep providing the gear and the services that our clients need

Many of us make our livings providing concert-goers with the best live music experience possible. We deploy high-fidelity loudspeaker systems and microphones with the latest in digital effects and studio-quality processing in an effort to make the live show sound “just like the record.”

Only better, of course, because the excitement, visual elements, crowd response and performance spontaneity are impossible to reproduce in someone’s living room. Or is it? Let’s step back in time and examine our progress in the effort to capture the live experience for the fans to take home.

Live recording has taken many forms over the years. In the big band era, it was typical to put a single microphone out in front of the performers and hope for the best. Early refinements consisted of adding a second mic for the soloists to step up to. Performances were largely acoustic, with the possible exception of a lead vocalist, so there was no interface with the live reinforcement system.

Recordings were monophonic, and the only options available to the recordist for influencing the outcome were mic choice and location. By the way, the delivery system was usually 78 RPM vinyl records. Given the limitations, it’s amazing how many vibrant, exciting examples exist from that era of music.

Through the 1950s and 60s, there were huge changes in performance, recording and playback technology. On the performance side, the invention of the electric guitar changed everything. (In fact, a case can be made that the electric guitar spawned our entire industry.) The concept of an amplified performance where the audience heard an electronic representation of the instrument rather than the instrument itself was revolutionary in many ways.

It wasn’t long before the bass joined the ranks of amplified instruments, and all of the other musicians (with the possible exception of the drums) were using mics. This allowed shows to be staged at much larger halls than was possible in the “acoustic era,”  enhancing exposure for the artist - and revenue for everyone.

This also shrank the size of performing groups as well. Previously, if the trombones needed to be louder, more trombone players were added. Now the trombone sound could simply be turned up.

On the recording front, the big news was multiple tracks. Two- and even three- track recorders were invented. This created a need for mixing consoles, and most were built by the studio owners themselves. Some even sported advanced features like equalization.

This technology then migrated over to sound reinforcement, and it required operators. We all got a job!

Big things were happening on the playback scene as well. The hi-fi craze swept many parts of the world. Playback systems with wide frequency response and low distortion became available. The 33 RPM Long Play (LP) record allowed much longer playing times.

Meanwhile, stereophonic sound finally gave recorded music more of the spatial impact of a live performance. With stereo playback, the instruments could be spread across the soundstage to simulate sitting in front of a real band. The elements required to bring the live performance experience into the listener’s home were falling into place.

As the music business roared into the 1970s,  the capability grew to duplicate the recording techniques for live events. Record companies wanted to be able to issue as many LP’s as possible from their hottest bands. One way to do this - without taking them off the road – was the live album.

As budgets became available for quality live recording the first studio trucks were created. A recording studio control room was crammed into a box truck and trundled off to the gig. Either using splits off the sound reinforcement mics or double mic’ing everything. a quality multi-track recording an actual concert could be made. A few audience mics were added, and voila, the record company had their new release.

The best part? No new songs had to be written. The same songs could be sold to eager fans twice! Soon, no self-respecting band was without a live album. Of course, the other advantage was that if the house mix or sound system was substandard, or the acoustics were bad, a multi-track master tape provided some ability to “fix it in the mix.”

And on more than a few instances the band would nip into the studio to fix “green notes” in the vocals or a botched guitar lead.

Live recording had started to generate its own revenue stream, which supplemented the box office receipts from the show. Eventually someone got the bright idea of bringing a movie camera into the proceedings. Between the audio recording truck, the camera operators, directors and miscellaneous technical personnel, it could turn into a huge undertaking. For some events, it was worth the money.

The Woodstock movie made far more cash than the festival itself. If you couldn’t go to the concert, the concert would come to your local movie theater. But only the biggest bands or the most high profile events could justify the expense of the production and pack the fans into theaters.

As technology continued it’s relentless march, many acts wanted to record every performance. It started innocently enough with the ubiquitous “board tape.”  At first this was just a stereo cassette coming right off the same stereo pair feeding the mains. These tapes were generally used by the band and their management to review the night’s performance.

Of course, sometimes this led to some mix criticism as well. It was hard to explain to a guitar player that the reason he couldn’t hear himself on the board tape was because his stage amplifiers were on “11” and his mic was off.

So eventually we started doing sub mixes for the board tapes. I’ve done tours where I had a combination of pre-fader and post-fader stereo aux sends, and used delays to time align an X-Y stereo pair of room mics into a DAT machine – all just to make the troops happy with their review tapes.

And inevitably some bright soul would say, “We could release this as a live album!” or maybe give their copy to their girlfriend, which later appeared as a bootleg causing great consternation and finger-pointing within the ranks. 

But that’s another story.

I saw one act that even carried a 24-track recorder in a huge flight case and a maintenance technician on tour so they could record every night. They even organized their set list to give the tech time to change tapes. A sound company I worked for owned a Midas Pro 5 board reputedly built for Harry Belafonte (and of course christened the “Day-O” board), and it had an extra 24 output buses to feed his recorder. It also weighed a ton.

But once again technology came to the rescue.

In the 1990s, digital recorders utilizing tape cartridges were introduced. Each unit recorded eight tracks and several could be synched up. They were rack mountable, reasonably light and low maintenance. A portable rack could now hold enough recorders to run a direct out from every board channel and record every night for future use.

Some enterprising engineers even used the previous night’s show routed back to the console to do a preliminary sound check. The only downfall was that you had to spend every spare moment formatting tapes for the recorders, and archiving was a pain. Depending on the length of the show and the number of tracks required, a single performance might use 30 tapes or even more.

By this time, almost every home had at least a decent stereo and a VCR. More and more tours were filmed, whether a theatrical release was realistic or not. Home entertainment technology had created an alternative market for video concert releases.

Although live records were still being released, the concert experience had much more impact if the visual elements were included. Most top tours and almost all major festivals had an audio and video recording element to document the event and provide a revenue stream long after the actual show. The concert experience was now as close as your local video store.

The 21st Century has only expanded this paradigm. A combination of forces has created a “perfect storm” supporting concert recording. On the recording technology front, digital audio workstations are smaller, lighter, more robust, and in fact, are often the same machines being used in the recording studio.

An entire show can be recorded on a single hard drive. Digital consoles can easily provide audio streams to the recorders without multiple analog to digital (A-D) conversions or analog signal splits.

The advent of the DVD and home theater systems provide a delivery medium with the quality and impact to really bring the concert experience into the home. Large high-definition screens and surround sound can do a remarkable job of reproducing the feeling of being at an event. They also provide new ways to make money from a live performance, and in a day and age where file sharing and piracy have eaten away at the traditional money flow in the music business.

It used to be common for record companies to provide tour support from record sale receipts. Now, it’s more common for touring and the recorded products that come from touring to be the largest source of income for performers.

Some bands have taken it to the next level by selling recordings of the actual show to attendees on their way out. “Jam bands” are still popular, and no two performances are alike. So getting a recording of these performances show may have more significance than whether the band says,  “Good night, Seattle” or “Good night, Detroit”. Concerts are being staged for the sole purpose of producing a DVD or even a pay-per-view broadcast.

Nothing can really replace the adrenaline, the excitement and the immediacy of being at a great concert. Our jobs are going to be around for a long time.

But we should always remember to look back at the historical trends of our industry. It’s the only way we can stay ahead of the curve and keep providing the gear and the services that our clients need.

And anything that enhances the revenue stream from live performances for the artists, promoters - and especially for us - is a very good thing indeed.

Bruce Main has been a systems engineer and front of house mixer for more than 35 years. He has also built, owned and operated recording studios and designed and installed sound systems.

Posted by Keith Clark on 09/24 at 01:49 PM
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Mark Ziemann Upgrades To API BOX Analog Console

German recording engineer returns to analog roots to improve studio workflow.

Mark Ziemann “got tired of mixing with the mouse”. As a result, he upgraded to an API BOX console this June.

Ziemann made the switch from digital mixers and calls the console a “complete API system” for his home studio, “now I know I needed analog faders.”

Ziemann works in Garbenteich, Germany, where he has been involved in pro audio for just over a decade.

“As a guitarist, I was never quite happy with the productions of my music, so I got interested in pro audio recording in 2004. My first professional mic pre was API.”

Since then, Ziemann steadily expanded his API gear collection to include a 2500 compressor, 525 compressor, and an 8-slot lunchbox before deciding that what he really needed was an analog console in his studio.
Ziemann was recording big projects before adding a BOX to his workflow, but without a console of his own, he had to take his outboard gear with him to record in bigger studios. It wasn’t until working on his “Oktoberfest Version” of Pharell Williams’ chart-topping track “Happy” that he realized a console was the solution to his problems.

Mixing “Happy”, a project he completed with engineer Christian Wahl, was a challenge, but Ziemann says his API gear made it much more possible. “It was quite hard to build this new version around the original vocals, but with a little help from my 525, it worked.”

Now that he has an API console, Ziemann doesn’t have to leave home and he can bypass the plugins he once relied on.

“The BOX is a real centerpiece and it’s easy to use. It has all the functionality I require for my work. I mix pretty straight using all 20 channels with the internal 527s on drums or vocal applications, while using my 2500 on the main bus. The BOX sped up my workflow so much. Everything sits in the mix better right from the start.”

Ziemann purchased his BOX from API Distributor ES-Pro Audio, based in Germany.


Posted by House Editor on 09/24 at 10:29 AM

Church Sound: When Is It The Right Time To Upgrade A System?

This article is provided by Gary

In this ever evolving world of more, more, more, better, better, better, when is good enough, good enough? Is it really an absolute necessity to update or upgrade my 15-year-old sound system? 

The working life of a sound system can extend well beyond 20 years, and I’ve personally seen systems 30 to 40 years old still in use and functioning quite well.

The question deserves considerable thought, and begs a slew of additional questions:

1. Has your programming changed (added a keyboard, drums, bass……..)?

2. Have you added any additional seating, like additional rows of seats in the front or back?

3. Are you experiencing intermittent problems or shall we say, surprise noises?

4. Has the expectation of your congregation changed?

I not a person who promotes technology for the sake of technology.  However, I do enjoy thinking of myself as “hip” and an early adopter.

In fact I owned the original Palm Pilot, one of the first Windows Mobile PDA’s and a Palm Treo Pro phone all purchased right when they were released. 

Oh yeah, I forgot the Apple Performa 405 PC I purchased in the early 90’s, the iMac, and now the Macbook pro…..  Okay you get it, I love new technology and am not afraid to be one of the early one that jumps in, with some caution. 

I am usually not the first, but reside comfortably in the early pack that makes a purchase. I prefer to wait and see the viability and stability of the product.

So what does my personal love of technology have to do with upgrading your sound system? Not much, other than show that I am not an anti-technology kind of guy.

Getting back to the issue at hand. I would like to introduce a fifth question.

5. What is the expectation of people who come and visit your church?

Between 1913 and 1920, Thomas Edison did more than 4,000 “blind listening tests” to promote his Phonograph equipment and Diamond disc recordings. Edison would rent theaters and concert halls to do a comparison. 

He would hire some of the prominent musicians of the day and have them behind a curtain. The musician would sing a song and then a recording would be played on the Phonograph. 

Believe it or not the audience could not distinguish any difference between the two. In other words, they could not decipher if it was live or recorded. If you are like me you have to be saying - Hold on! People had to hear the difference between a scratchy, frequency limited recording and a live person. 

I guarantee if we took the exact same equipment and repeated the test today, the majority of people would easily be able to point out what was live and a phonograph recording.

Other than almost 100 years, what is the difference? Reference! What did people in the early 1900s have with which they could compare the recording? 

Okay, by now I think you get the point.  There are a number of great reasons to upgrade your audio system; however, if it’s still in “working order,” then managing and dealing with the expectations of your congregation (also known as questions #4 and #5) are great reasons to update. 

So, how often should a system be updated? Ignoring changes in programming, seating, and any potential issues you may currently be experiencing with your system (do it today if these reasons apply, you are overdue), here’s my answer: if the system is older than your car, it’s time to update. 

This is my answer primarily because our family drives a 1998 Suburban and two other late 1990s-vintage vehicles. Maybe a better answer is every time the second number in your age repeats (10 years for those not good with numbers and abstract concepts), it’s probably at least time to begin considering upgrading.

Gary Zandstra has worked in church production and as an AV systems integrator for more than 35 years.

Posted by admin on 09/24 at 06:09 AM
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Tuesday, September 22, 2015

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

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 online and around the world. For more information go to

SynAudCon is now offering “Audio Applications – System Optimization & EQ” as web-based training. Click the link to see the related article.

More Church Sound articles by Pat Brown on PSW:
How To Illuminate The Audience With Beautiful, Consistent Audio Coverage
Proper Loudspeaker Placement: How To Avoid Lobes and Nulls
Ten Reasons Why Church Sound Systems Cost More
What Makes A Quality Loudspeaker?

Posted by Keith Clark on 09/22 at 01:25 PM
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