The Magnetic Tape Head
Most professional analog recorders use three magnetic tape heads, each of which performs a specialized task:
• Record
• Reproduce
• Erase.
The function of a record head (Figure 9) is to electromagnetically transform analog electrical signals into corresponding magnetic fields that can be permanently stored onto magnetic tape.
In short, the input current flows through coils of wire that are wrapped around the head’s magnetic pole pieces.
Since the theory of magnetic induction states that “whenever a current is injected into metal, a magnetic field is created within that metal” … a magnetic force is caused to flow through the coil, into the pole pieces and across the head gap.
Like electricity, magnetism flows more easily through some media than through others. The head gap between poles creates a break in the magnetic field, thereby creating a physical resistance to the magnetic “circuit.” Since the gap is in physical contact with the moving magnetic tape, the tape’s magnetic oxide offers a lower resistance path to the field than does the nonmagnetic gap.
Thus, the flux path travels from one pole piece, into the tape and to the other pole. Since the magnetic domains retain their polarity and magnetic intensity as the tape passes across the gap, the tape now has an analogous magnetic “memory” of the recorded event.
Fig. 9: The record head.
The reproduce or playback head (Figure 10) operates in a way that’s opposite to the record head. When a recorded tape track passes across the reproduce head gap, a magnetic flux is induced into the pole pieces.
Since the theory of magnetic induction also states “whenever a magnetic field cuts across metal, a current will be set up within that metal” … an alternating current is caused to flow through the pickup coil windings, which can then be amplified and processed into a larger output signal.
Note that the reproduce head’s output is nonlinear because this signal is proportional to both the tape’s average flux magnitude and the rate of change of this magnetic field.
This means that the rate of change increases as a direct function of the recorded signal’s frequency. Thus, the output level of a playback head effectively doubles for each doubling in frequency, resulting in a 6-dB increase in output voltage for each increased octave.
The tape speed and head gap width work together to determine the reproduce head’s upper-frequency limit, which in turn determines the system’s overall bandwidth.
The wavelength of a signal that’s recorded onto tape is equal to the speed at which tape travels past the reproduce head, divided by the frequency of the signal; therefore, the faster the tape speed, the higher the upper-frequency limit. Similarly, the smaller the head gap, the higher the upper-frequency limit.
The function of the erase head is to effectively reduce the average magnetization level of a recorded tape track to zero, thereby allowing the tape track to be re-recorded.
After a track is placed into the record mode, a high-frequency and high-intensity sine-wave signal is fed into the erase head (resulting in a tape that’s being saturated in both the positive- and negative-polarity directions).
Fig. 10: The playback head.
This alternating saturation occurs at such a high speed that it serves to confuse any magnetic pattern that existed on the tape.
As the tape moves away from the erase head, the intensity of the magnetic field decreases, leaving the domains in a random orientation, with a resulting average magnetization or output level that’s as close to zero as tape noise will allow.
Equalization Equalization (EQ) is the term used to denote an intentional change in relative amplitudes at different frequencies.
Because the analog recording process isn’t linear, equalization is needed to achieve a flat frequency-response curve when using magnetic tape.
The 6-dB-per-octave boost that’s inherent in the playback head’s response curve requires that a complementary equalization cut of 6 dB per octave be applied within the playback circuit (see Figure 11).
Bias Current
In addition to the nonlinear changes that occur in playback level relative to frequency, another discrepancy in the recording process exists between the amount of magnetic energy that’s applied to the record head and the amount of magnetism that’s retained by the tape after the initial recording field has been removed.
As Figure 12a shows, the magnetization curve of tape is linear between points A and B, as well as between points C and D.
Signals greater than A and D have reached the saturation level and are subject to clipping distortion. Signals falling within the B to C range are too low in flux level to adequately magnetize the domains during the recording process.
For this reason, it’s important that low-level signals be boosted so that they’re pushed into the linear range. This boost is applied by mixing an AC bias current (Figure 12b) with the audio signal.
This bias current is applied by mixing the incoming audio signal with an ultrasonic sine-wave signal (often between 75 and 150 kHz).
The combined signals are amplitude modulated in such a way that the overall mag- netic flux levels are given an extra “oomph,” which effectively boosts the signal above the nonlinear zero-crossover range and into the linear portion of the curve. In fact, if this bias signal weren’t added, distortion levels would be so high as to render the recording process useless.
Monitoring Modes
The output signal of a professional ATR channel can be switched between three working modes:
• Input
• Reproduce
• Sync.
In the input (source) mode, the signal at the selected channel output is derived from its input signal.
Thus, with the ATR transport in any mode (including stop), it’s possible to meter and monitor a signal that’s present at a channel’s selected input. In the reproduce mode, the output and metering signal is derived from the playback head.
Fig. 11: A flat frequency playback curve results due to complementary equalization in the playback circuit.
This mode can be useful in two ways: It allows previously recorded tapes to be played back, and it enables the monitoring of mate- rial off of the tape while in the record mode. The latter provides an immediate quality check of the ATR’s entire record and reproduce process.
The sync mode is a required feature in analog multitrack ATRs because of the need to record new material on one or more tracks while simultaneously monitoring tracks that have been previously recorded (during a process called overdubbing).
Here’s the deal … using the record head to lay down one or more tracks while listening to previously recorded tracks through the reproduce head would actually cause the newly recorded track(s) to be out of sync with the others on final playback (due to the physical distance between the two, as shown in Figure 13a).
To prevent such a time lag, all of the reproduced tracks must be monitored off of the record head at the same time that record tracks are being laid down onto the same head. Since the record head is used for both recording and playback, there is no physical time lag and, thus, no signal delay (Figure 13b).
To Punch Or Not To Punch
You’ve all heard the age-old adage … “$%& happens.” Well, it happens in the studio—a lot! Whenever a mistake or bad line occurs during a multitrack session, it’s often (but not always) possible to punch-in on a specific track or set of tracks.
Instead of going back and re-recording an entire song or overdub, performing a punch involves going back and re-recording over a specific section in order to fix a bad note, musical line—you name it.
This process is done by cueing the tape at a logical point before the bad section and then pressing Play. Just before the section to be fixed, pressing the Record button (or entering record under automation) will place the track into record mode.
At the section’s end, pressing the Play button again causes the track to fall back out of record, thereby preserving the section following the punch.
From a monitor standpoint, the recorder begins playback in the sync mode; once placed into record, the track switches to monitor the input source. This lets the performers hear themselves during the punch while listening to playback both before and after the take.
When performing a punch, it’s often far better to “fix” the track immediately after the take has been recorded, while the levels, mic positions and performance vibe are still the same.
This also makes it easier to go back and re-record the entire song or a larger section should the punch not work.
Fig. 12: The effects of bias current on recorded linearity: (a-top) magnetization curve showing distortion at lower levels; (b-bottom) after bias, when the signal is boosted back into the curve’s linear regions.
If the punch can’t be performed at that time, however, it’s generally a good idea to take detailed notes about mic selection, placement, preamps and so on to recreate the session’s setup without having to guess the details from memory.
As every experienced engineer/producer knows, performing a punch can be tricky. In certain situations, it’s a complete no-brainer … for example, when a stretch of silence the size of a Mack truck exists both before and after the bad section, you’ll have plenty of room to punch in and out.
At other times, a punch can be very tight or problematic (e.g., if there’s very little time to punch in or out, when trying to keep vocal lines fluid and in-context, when it’s hard to feel the beat of a song or if it has a fast rhythm). In short, punching-in shouldn’t be taken too lightly … nor so seriously that you’re afraid of the process.
Talk it over with the producer and/or musicians. Is this an easy punch? Does the section really need fixing? Do we have the time right now? Or, is it better just to redo the song?
In short, the process is totally situational and requires attention, skill, experience and sometimes a great deal of luck.
Before committing the punch to tape, it’s often a wise idea to rehearse the punch, without actually committing the fix to tape. This has the advantage of giving both you and the performer a chance to practice beforehand.
Some analog decks (and most DAWs) will let you enter the punch-in and punch-out times under automation, thereby allowing the punch to be surgically performed automatically.
If you’re recording onto the same track, a fudged punch may leave you with few options other than to re-record the entire song or section of a song. An alternative to this dilemma would be to record the fix into a separate track and then switch between tracks in mixdown.
The track(s) could be transferred to a DAW, where the edits could be per- formed in the digital domain.
Tape, Tape Speed, And Head Configurations
Professional analog ATRs are currently available in a wide range of track- and tape-width configurations.
The most common analog configurations are 2-track mastering machines that use tape widths of 1/4 inch, 1/2 inch, and even 1 inch, as well as 16- and 24-track machines that use 2-inch tape.
Figure 14 details many of the tape formats that can be currently found. Optimal tape-to-head performance characteristics for an analog ATR are determined by several parameters: track width, head-gap width and tape speed.
In general, track widths are on the order of 0.080 inch for a 1/4-inch 2-track ATR; 0.070 inch for 1/2-inch 4-track, 1-inch 8-track, and 2-inch 16-track formats or 0.037 inch for the 2-inch 24-track format.
As you might expect, the greater the recorded track width, the greater the amount of magnetism that can be retained by the magnetic tape, resulting in a higher output signal and an improved signal-to- noise ratio. The use of wider track widths also makes the recorded track less susceptible to signal-level dropouts.
The most common tape speeds used in audio production are 15 ips (38 cm/ sec) and 30 ips (76 cm/sec).
Fig. 13: The sync mode’s function. (a-top) In the monitored playback mode, the recorded signal lags behind the recorded signal, thereby creating an out-of-sync condition. (b-bottom) In the sync mode, the record head acts as both record and playback head, bringing the signals into sync.
Although 15 ips will eat up less tape, 30 ips has gained wide acceptance in recent years for having its own characteristic sound (often having a tighter bottom end), as well as a higher output and lower noise figures (which in certain cases eliminate the need for noise reduction).
On the other hand, 15 ips has a reputation for having a “gutsy,” rugged sound.
Print-Through
A form of deterioration in a recording’s quality, known as print-through, begins to occur after a recording has been made.
This effect is the result of the transfer of a recorded signal from one layer of tape to an adjacent track layer by means of magnetic induction, which gives rise to an audible false signal or pre-echo on playback.
The effects of print-through are greatest when recording levels are very high, and the effect decreases by about 2 dB for every 1-dB reduction in signal level. The extent of this condition also depends on such factors as length of storage, storage temperature and tape thickness (tapes with a thicker base material are less likely to have severe print-through problems).
Because of the effects of print-through, the standard method of professionally storing a recorded analog tape is in the tails-out position.
Remember:
Professional analog tape should always be stored tails-out (on the right- hand take-up reel).
Upon playback, the tape should be wound onto the left-most “supply reel.”
During playback, feed the tape back onto the right-hand take-up reel, after which time it can again be removed for storage.
If the tape has been continuously wound and rewound during the session, it’s often wise to rewind the tape and then smoothly play or slow-wind the tape onto the take-up reel, after which time it can be removed for storage.
So why do we go through all this trouble? When a tape is stored tails-out (Figure 15), the print-through will bleed to the outer layers, a condition that causes the echo to follow the original signal in a way that’s similar to the sound’s natural decay and is subconsciously perceived by the listener as reverb instead of as an easily-audible pre-echo.
Cleanliness
It’s very important for the magnetic recording heads and moving parts of an ATR transport deck to be kept free from dirt and oxide shed.
Oxide shed occurs when friction causes small particles of magnetic oxide to flake off and accumulate on surface contacts.
This accumulation is most critical at the surface of the magnetic recording heads, since even a minute separation between the magnetic tape and heads can cause high-frequency separation loss.
For example, a signal that’s recorded at 15 ips and has an oxide shed buildup of 1 mil (0.001 inch) on the playback head will be 55 dB below its standard level at 15 kHz.
Denatured (isopropyl) alcohol or an appropriate cleaning solution should be used to clean transport tape heads and guides (with the exception of the machine’s pinch roller and other rubber-like surfaces) at regular intervals.
Fig. 14: Analog track configurations for various tape widths.
Degaussing
Magnetic tape heads are made from a magnetically soft metal, which means that the alloy is easily magnetized … but once the coil’s current is removed, the core won’t retain any of its magnetism.
Small amounts of residual magnetism, however, will build up over time, which can actually partially erase high-frequency signals from a master tape.
For this reason, all of the tape heads should be demagnetized after 10 hours of operation with a head demagnetizer. This handheld device works much like an erase head in that it saturates the magnetic head with a high-level alternating signal that randomizes residual magnetic flux.
Once a head has been demagnetized (after 5 to 10 seconds), it’s important to move the tool to a safe distance from the tape heads at a speed of less than 2 inches per second before turning it off, so as to avoid inducing a larger magnetic flux back into the head.
Before an ATR is aligned, the magnetic tape heads should always be cleaned and demagnetized in order to obtain accurate readings and to protect expensive alignment tapes.
Backup And Archive Strategies
In this day of hard drives, CDs and digital data, we’ve all come to know the importance of backing up our data. With important music and media projects, it’s equally important to create a tape backup copy in case of an unforeseen catastrophe or as added insurance that future generations can enjoy the fruits of your work.
Backing Up Your Project
The one basic truth that can be said about analog magnetic tape is that this medium has withstood the test of time.
With care and reconditioning, tapes that have been recorded in the 1940s have been fully restored, allowing us to preserve and enjoy some of the best music of the day. On the other hand, digital data has two points that aren’t exactly its favor:
Data that resides on hard drives isn’t the most robust of media over time. Even CDRs (which are rated to last over 100 years) haven’t really been proven to last.
Even if the data remains intact, with the ever-increasing advances in computer technology, who’s to say that the media, drives, programs, session formats and file formats will be around in 10 years, let alone 50!
These warnings aren’t slams against digital, just precautions against the march of technology versus the need for media preservation.
For the previously listed reasons, media preservation is a top priority for such groups as the Recording Academy’s Producers & Engineers Wing (P&E Wing), as well as for many major record labels—so much so that many stipulate in their contracts that multitrack sessions (no matter what the original medium) are to be transferred and archived to 2-inch multitrack analog tape.
When transferring digital tracks to an analog machine, it’s always wise to make sure that the recorder has been properly calibrated and that reference tones (1 kHz, 10 kHz, 16 kHz and 100 Hz) have been recorded at the beginning of the tape.
When copying from analog to analog, both machines should be properly calibrated, but the source for the newly recorded tones should be the master tape.
If a SMPTE track is required, be sure to stripe the copy with a clean, jam- sync code.
Fig. 15: Recorded analog tapes should always be stored in the tails-out position.
The backing up of analog tapes and/or digital data usually isn’t a big problem … unless you’ve lost your original masters. In this situation, a proper safety master can be the difference between panic and peace.
Archive Strategies
Just as it’s important to back up your media, it’s also important that both the original and backup media be treated and stored properly. Here are a few guidelines:
As stated earlier, always store the tapes tails-out.
Wind the tapes onto the take-up storage reel at slow-wind or play speeds.
Store the boxes vertically. If they’re stored horizontally, the outer edges could get bent and become damaged.
Media storage facilities exist that can store your masters or backups for a fee. If this isn’t an option, store them is an area that’s cool and dry (e.g., no temperature extremes, in low humidity, no attics or basements).
Store your masters and backups in separate locations. In case of a fire or other disaster … one would be lost, but not both (always a good idea with digital data, as well).
Meyer Sound Installation At Denmark’s Kronborg Castle Brings The Past Alive
The soundscape created by this unique installation helps visitors to understand what life was once like for villagers.
Kronborg Castle - a UNESCO World Heritage site in Denmark best known by most as the setting for William Shakespeare’s famous tragedy Hamlet - recently underwent a technical upgrade, with a soundscape adding a new dimension to the visitor experience.
Using a range of low-profile, weather-protected Meyer Sound loudspeakers distributed throughout the fortress, visitors journey through its centuries of cultural legacy, from the heavy battle conquests, power struggles, to its transformation into one of Europe’s most forbidding prisons.
“We call the sounds at Kronborg Castle ‘Echoes of the Past,’” says Exhibition Manager Jesper Gottlib Wik. “The aim is to create sounds that appeal to our guests’ imagination and encourage them to form their own interpretation of what they hear.”
The tour begins at the main entrance, where a left-center-right configuration of UPM-1P loudspeakers plays back prerecorded sound effects composed by Stephen Schwartz.
Military music and sounds of marching soldiers and stonemasons at work combine to help visitors form their own imagery of an era gone by.
As the trail leads visitors to the gunpowder house, two MM-4XP miniature loudspeakers and an MM-10XP subwoofer camouflaged in granite blocks depict villagers’ everyday life. Finally a UPM-1P loudspeaker tells the story of the 1658 battle of Sweden, Denmark, and Holland.
Sound quality, treacherous weather conditions, aesthetic preservation, and coverage control were all considerations that complicated system design for the audio supplier, Stouenborg.
“Denmark endures the extremes of seasonal weather, from snow, rain, wind, to very hot summers,” says Anders Jørgensen of Stouenborg. “We were drawn to Meyer Sound knowing that their systems are installed on cruise ships.”
“And if a system can sustain the rough conditions at sea, it’s the system that we wanted for the Kronborg Castle. Some of the loudspeakers are hidden in granite enclosures to stay out of sight and to be protected from vandalism.”
Feedback to the soundscape at Kronborg Castle has been overwhelmingly positive, according to Jørgensen. “When people hear the sound of soldiers giving command at Württembergs Ravelin next to the main entrance, they turn around to look for where the sound is coming from,” says Jørgensen.
Project consultant Lars Holst adds that Kronborg is planning to install sensors that can automatically regulate the volume of the prerecorded material based on the level of environmental background noise. “There is quite a difference in the volume requirement between a quiet summer day and an autumn gale.”
Q: At the studio where I intern I’ve noticed something kind of interesting.
When I get to use the studio occasional at night, I let the band setup however they want in the room because I figure they should be comfortable.
However, I’ve noticed that the chief engineer is really picky where everyone is in the room when he’s tracking, especially the placement of amps and drums.
Does the placement of the instruments in the studio really affect the overall sound of the recording?
A: A very interesting question, and a good follow-up to yesterday’s question regarding the miking of guitar amplifiers. Todays answer also comes to us from multi-platinum engineer/producer Keith Olsen on recording.
As you observed being done by other engineers at your studio, always use the shape of your room to your advantage when tracking.
Here’s the idea: when recording a directional source, such as a guitar amplifier, don’t place the amp/speaker so that it is parallel or perpendicular to the room’s walls.
Rather, turn the amp at an angle to the walls so that reflections bounce around the room, rather than directly back at the amp. Likewise, angle the amp upward so that the reflections from the ceiling are directed away from the amp.
Think of a flashlight shining into a mirror; rather than have the light reflected straight back into your eyes, angle the flashlight so the light reflections bounce around the room. Trust me, you really will notice a difference.
As always, we welcome input from the PSW community and would love to know your thoughts on instrument placement. Feel free to let us know in the comments below.
The VisionMount VXF220 is intended for indoor / outdoor use with even the largest TV's.
Sanus Systems has announced the shipment of an all-weather mount for 42”-75” flat-panel TVs weighing up to 175 lbs.
The VisionMount VXF220 All-Weather Full-Motion Mount is manufactured with a special rust-resistant coating and stainless steel hardware and has been rated to withstand rain, snow and salt, making it an ideal choice for indoor/outdoor use.
The mount offers robust, dual extension arms that support even the largest TVs and allow the user to extend the TV up to 20” from the wall, tilt, swivel and pan it in any direction.
Some features of the VXF220 include Virtual Axis tilt technology for easy viewing angle adjustment with the touch of a finger. As well, the mount includes FollowThru channels for complete cable concealment and protection the entire length of the mount’s dual extension arms.
The VXF220 utilizes the QuickConnect system, allowing the mounting head to snap onto the arm assembly and lock into place, as well as the unique ProSet for post-installation height and leveling adjustments
The VXF220 also offers easy installation with the capability to shift the TV left or right on the wall plate for perfect centering on the wall and wall plate mounting holes that accommodate both 16” and 24” studs.
Sound Devices Upgrades Recorder Firmware Bringing Welcome Features
Firmware update 2.10 for the 788T Recorder and Wave Agent to be released at IBC 2010.
Sound Devices is unveiling the 2.10 firmware update for its 788T and 788T-SSD multitrack digital audio recorders, along with the 1.15 upgrade to its Wave Agent software at IBC 2010.
The 788T update, available to all 788T users on the company’s website, allows for the linking of multiple 788T recorders, as well as other Sound Devices 7-Series recorders. Wave Agent 1.15, meanwhile, features a new Control Mode.
Thanks to a new linking capability, multiple 788T digital recorders are easily interconnected for applications requiring higher track counts or synchronized backups.
The 788T’s latest firmware update enables, via its C. Link connection, multi-unit linking, locking all connected 7-Series recorders to one 788T world clock, time code and transport.
The new 788T update also supports editing metadata from Wave Agent software. Sound mixers can directly edit 788T file metadata from a Mac OS or Windows computer running Wave Agent 1.15 or greater.
The newly released upgrade for Wave Agent, 1.15, features this new Control Mode (formerly known as Meter Mode). Metadata available for edit includes scene, take and notes values.
“Production audio recording and mixing continues to require more tracks as films, documentaries, episodic TV and reality TV keep accelerating the need to record sound sources as isolated tracks,” said Jon Tatooles, managing director for Sound Devices.
“The new multi-unit linking capability in the 788T makes it easy for the sound engineer to run the 788T simultaneously with any other 7-Series recorders and keep all recorders sample-accurate from the start using the same word clock and time code source.”
“Additionally, the ability to edit metadata directly in Wave Agent adds efficiency to their workflows.”
The Sound Devices 788T is an eight-input, 12-track digital audio recorder designed for sound engineers mixing on location for films, documentaries, episodic television and reality TV.
The 788T’s eight inputs ensure high audio quality, accept either microphone or line-level signals, provide 48 V phantom power for condenser microphones, offer peak limiters for microphone inputs, and feature fully adjustable high-pass filters in a compact package.
The 788T, like all Sound Devices products, is designed to withstand the physical and environmental extremes of field production. The front panel is gasketed for water resistance.
Russ Berger To Present at SynAudCon Audio and Acoustics for Conference Systems Seminar
Berger to discuss the importance of acoustics in teleconference applications.
Russ Berger, president of Russ Berger Design Group, will be sharing his industry insight on acoustics for corporate installations at SynAudCon’s (Synergetic Audio Concepts) “Audio Acoustics for Conference Systems” seminar.
The event will take place September 16-18, 2010 in the Performance Theatre at Sweetwater’s headquarters in Fort Wayne, IN.
“Audio Acoustics for Conference Systems” will discuss how to approach the design and commissioning of audio conferencing systems, and how architectural acoustics and associated noise plays into these plans.
The principles presented will be demonstrated in simulated conference rooms using equipment provided by a variety of manufacturers, offering attendees with a true hands-on experience.
Berger has been a longtime SynAudCon supporter and has participated in conferences and workshops for more than 25 years. His co-presenters for this workshop include Mario Maltese, CEO of Audio Visual Resources in New York, and Jay Paul, Vice President with integrator AVI-SPL in Maryland.
“Intelligibility is key when designing a teleconferencing environment. Even using top-of-the-line gear won’t guarantee that everyone is heard properly,” says Berger. “Acoustical design, when used effectively, can actually enhance the different audio equipment being used.
This helps ensure that the teleconference is about the ideas and topics being discussed, rather than whether or not everyone can be heard.”
Berger’s key contribution to the industry has been in the acoustical design development of broadcast and recording studios as well as other technical spaces, specifically the behavior of sound in acoustically small spaces and the properties of acoustically coupled spaces.
Attendees of the seminar will see firsthand the results of Berger’s work as architect of record for and designer of the Sweetwater Performance Theatre where the seminar is being held.
For more information on the “Audio Acoustic for Conference Systems” seminar, please visit the SynAudCon Website.
First UK Renkus-Heinz IC Live Rental System Deployed By Stage Audio Services
Selected for size vs. coverage, ease of rigging, and sound quality
Midlands based Stage Audio Services, which supplies live audio production for events ranging from TV shows and sports events to touring, has become the UK’s first rental purchaser of the Renkus-Heinz IC Live digitally steerable array system.
The company describes its first four clients’ reactions to the diminutive system, which was rated an “impressive performer” in a recent loudspeaker shoot-out, as “bowled over by its sound and looks.”
Polar Audio, UK distributor for Renkus-Heinz, signed the deal after a week of trial runs by Stage Audio Services that included the Courtney Pine-headlined Mostly Jazz Festival in Birmingham and four large corporate events.
Its first commercial outing was the quarter final of the Twenty20 cricket tournament at Edgbaston on 16 July.
Since then the system has, company founder Kevin Mobberley said, been in virtually continuous use, including HMV’s stand at the Global Gathering electronic music festival.
“It’s the ideal system for a lot of the corporate, comedy and outdoor work that we do; it ticks all the boxes. We are amazed by what it can do, and long may it continue to make us money.”
Stage Audio Services, formed by Mobberley in 1982, provides audio production for a broad mix of touring, live TV and corporate events. Clients include the BBC’s Top Gear Live shows and the Gadget Show Live.
Live music work includes smaller stages at the Download, Global Gathering and V festivals and regional and national rock, pop, comedy and theatre tours. Birmingham radio station BRMB, the NEC and Birmingham Council are among its regional clients.
“We do a lot of festival and other varied work,” said Mobberley, “but it’s predominantly touring, and work for promoters who come to us for a complete package. We ship gear and crew all over the world.” The ‘corporate rock & roll’ side of the business, he says, is what attracted his attention to the IC Live system.
“We’ve bought a lot of equipment in the last four or five years, and this is the first piece of equipment we’ve bought for a very long time that I felt very excited about. I just fell in love with it after first seeing it.”
“Normally we buy equipment out of need; with this, it’s not a replacement for anything, it’s a totally new tool for us. It will do these corporate events and smaller venues in a much nicer way, with a better presentation, a better sound and in a much more cost effective way.”
The deployment at Edgbaston, home of Warwickshire County Cricket Club, was designed to meet a singular challenge - providing sound to spectators seated around three quarters of the 21,000-capacity cricket ground. In keeping with the carnival atmosphere of Twenty20 the content comprised musical stings and a live compère, provided by a BRMB production office.
The spectators were covered by four IC Live stacks spaced around the ground, with the upper and lower seating tiers covered by separate beams from the mid/high units, each locked onto the matching subwoofer using the integrated hardware. Mobberley commented: “We brought in the IC Live to do quite a challenging job and it did it fantastically.”
He adds: “When we first heard it demonstrated, I knew we were onto something special. Over the next week we put it through its paces in our typical show environments - Forces Day for Wolverhampton City Council, a festival in central Birmingham for 5,000 people, a similar event in the grounds of a stately home, and the two-day Mostly Jazz Festival in Birmingham.
The latter site was surrounded by houses, so containing the sound was vital, which the beam steering did fantastically.
“The last trial was at Wolverhampton Civic Hall, a venue every rock & roll sound engineer knows. An amazing result came from flying a single sub and two of the mid/high units upside down, which covered the whole of the 3,100 capacity room at around 101dB in every seat including the balcony; it was phenomenal.
“The main benefits to us compared to conventional systems are the small size compared to the coverage, the ease of rigging, and the sound quality. It’s very expensive to transport systems around, so small is good, and the ease with which you can physically set it up means that once the engineers know how to use it, you can be set up and line checking in minutes.”
“It’s also the only system I can remember where I’ve had the clients commenting on how good it sounds and how good it looks, and I had that from our first four clients; it’s got a lot of people talking about our company in a very positive way.”
Commodity, Schmodity: The Need For Differentiation
If the ubiquitous Sharpie can stand out in its market, then audio professionals and companies most certainly can in theirs.
More and more, I hear professional audio product categories referred to as “commodities”:
“Wire is a commodity.”
“Connectors are commodities.”
“Small mixers are commodities.”
“(Insert component here) is a commodity!”
The point seems to be that there’s little distinction between certain types of products.
Or, perhaps more accurately, that these products are being purchased based more upon price because performance is perceived to be similar, if not the same. Is this really true? I tend to think not.
A favorite analogy of my “anti-commodity” philosophy is the ubiquitous Sharpie, or “fine-point permanent marker” as it’s called in non-brand-specific terminology. Several companies make fine-point permanent markers.
Do these other markets write as well as a Sharpie? Do they hang from a lanyard as well as a Sharpie? Do they slip out of a sweaty hand less often than a Sharpie? And do they sign T-shirts as well as a Sharpie?
I don’t have answers since I’ve never used anything but Sharpie (and, for the record, haven’t ever signed a T-shirt). But if anything should be a commodity, one would think it would be a fine-point permanent marker.
Whether due to brand recognition or a true difference in performance (or a combination thereof), the Sharpie is the predominant choice when it comes to fine-point permanent markers.
And what of gaff tape? Surely since its something unceremoniously thrown away after one use, gaff tape must be a commodity! Well, isn’t it? Somehow I think the folks at Permacel would beg to differ.
So how is it that a fine-point permanent marker or gaff tape isn’t a commodity, yet some in our industry have come to think just the opposite of small mixers?
You’ve probably heard something like this: “Our customers don’t buy mixers based on quality anymore. They’re just looking for the cheapest one. They’ve become a commodity.”
I repeat - is this really true?
In a sense, these comments reflect a change in the state of mind of customers.
More than ever before, there are numerous inexpensive mixers available, and all are designed to perform similar tasks.
And typically, many customers can’t properly evaluate performance or build quality, so in this sense, the issue does come down to a matter of price, albeit within a range of features.
When the lower end of the market becomes crowded, a certain ignorance kicks in, and price is the primary method used by the consumer to differentiate products.
In reality, small mixers are not all created equal. Therefore, they’re not commodity products. But customers can and will perceive them as such without special knowledge and understanding. Let’s go a step further.
Couldn’t the actual service of providing a sound system be perceived as a commodity? How many of your customers have the ability to evaluate a sound company? How many times do you get phone calls asking the price of providing a system, only to be told that there’s already someone offering to do it for half that price?
As with small mixers, this thinking largely occurs at the lower end of the market, where the most “bottom feeders” reside. And yes, they’re more than happy to do a gig for half of a realistic price.
But, you argue, they don’t provide the same level of service, their crew is inexperienced, their gear is really not commensurate for the gig. While you know, and I know, does the customer know?
Let’s shift responsibility for this thinking where it really belongs. Simply, it’s up to us - all of us working on the front lines of pro audio, and not the customer - to make sure we’re not perceived as commodities.
It’s called differentiation: providing a higher level of service, presenting an organized, professional image, making sure we don’t take the easier/cheaper path on any gig.
Think about your own purchase decisions and how this might apply to your work, and a customer’s decision to hire you and/or your company. The one that best differentiates itself from its competition, that stands above the rest, typically wins.
If the ubiquitous Sharpie can stand out in its market, then audio professionals and companies most certainly can in theirs.
Based in the Seattle area, Ivan C. Schwartz is a pro audio industry veteran providing market and product development services to equipment manufacturers. He has been involved for over 30 years in mixing and both portable and installed sound system design and implementation, and currently works with Tannoy.
d&b Audiotechnik J-Series Makes Many Appearances On The Latest Jackson Browne Tour
House engineer Paul Dieter has fun mixing on his PA in a variety of environments
With a career in sound spanning well over two decades, mix engineer Paul Dieter is now enjoying himself mixing for one of rock and roll’s more enduring, if understated, legends, Jackson Browne.
“I started mixing live in the 80s, then spent maybe 15 years in the studio; I was just so frustrated with the live sound systems available then,” he states.
“When line arrays came along I was eventually tempted back to live and I’m having much more fun.”
“It’s a subtle consideration,” says Dieter, “Jackson’s song-writing is rock and roll for the discerning ear. I try to be manufacturer and sound company agnostic, I don’t want politics getting in the way of making the best choice for my client.”
“The combination of J8, J12 and J-SUB loudspeakers must be absolutely correctly balanced; I expect to be able to achieve nothing less than great vocal clarity throughout the house.”
Dieter’s system technician, Tom Laveuf, is the man responsible for creating that correct balance day to day. “Truth is, the system is very well behaved,” he begins.
Greek Theater, Berkeley. Photo: Tom Laveuf (click to enlarge)
“We’re touring through a mix of theaters, arenas and sheds which we can adapt easily, ArrayCalc makes that very straightforward,” Laveuf continues. “The 180-degree amphitheater of The Greek (Theater) in San Francisco was a delight.”
“The only EQ on the system is in the R1 file (d&b’s proprietary remote control software); there’s nothing in the desk or in the Lakes (processors), which I think says it all. The only challenge has been to figure out what to do with the subwoofers.”
Cuthbert Amphitheatre, Eugene, OR. Photo: Tom Laveuf (click to enlarge)
Laveuf has only recently joined Schubert, having cut his teeth as a system tech at Beachsound, another d&b advocate, in Florida.
“They taught me well; I’ve modeled many B2 and J-SUB arrays and find them very well behaved. Although the J-SUB is inherently cardioid, we quickly found that putting this type of defined Sub-array arc across the front of stage worked best for Jackson, leaving his mic totally uncolored by the low end.”
It’s a liberating condition for Dieter’s gain structure and a pleasing experience for his studio sensibilities.“That’s what I mean about correctly balanced; I don’t have to pull the subs at all, Tom got that right away,” Dieter notes.
Fabulous Fox, St Louis. Photo: Tom Laveuf (click to enlarge)
“Last night we played our first closed in arena, I was a tad anxious but Tom set it up and I was immediately struck by just how even and well distributed it was.
“Mixing on the d&b J-Series is the closest I have found to listening to great studio monitors.”
Q: I’ve noticed that when I’m recording guitar, often the bottom end is really muddy and it just sounds terrible.
I’ve thought about getting a stand or something to raise the amp up off the floor, but I always thought those stands were just so the musician could hear better.
Would something like that help the sound of my recordings?
A: Though some people may be surprised, a stand often will help in the case you’re describing.
In fact, it’s one of many tips courtesy of multi-platinum engineer/producer Keith Olsen on recording.
To achieve tight, controllable bottom end when miking a guitar amp, get the amp up off of the floor on a stand, case, or anything else that’s handy (and that’s sturdy enough to support the amp and damped enough not to resonate in response to sound from the amp).
In his opinion, elevating the amp a bit reduces floor reflections that can interfere with the low-end coming from the amp.
As always, we welcome input from the PSW community and would love to know your thoughts on this situation. Feel free to let us know in the comments below.
Symnet At The Heart Of The Historic Rose Theater’s Much Needed Audio Upgrade
Auto-mix and feedback suppression provided by SymNet made the system fully automatic and foolproof.
The building that houses The Rose Theater in Omaha, Nebraska originally opened its doors in 1927 under the name The Riviera.
Lavish in every respect, the building boasted ornate tapestries and Oriental rugs, sculptures, a mosaic floor, decorative fountains, Mediterranean-style murals, and a ceiling painted with sunset clouds and dotted with electric stars… not to mention stunning acoustics.
Like many other theaters built in that era, hard times fell upon The Riviera during the depression, and the theater changed hands numerous times in the ensuing decades.
Each time, the new owners changed several aspects of its original architecture and charm until what remained in 1981 bore little resemblance to what had once been so magical.
That year, furniture magnate Rose Blumkin purchased the building to save it from the wrecking ball and generously deeded it to the Omaha Theater Company for Young People, along with the first $1 million towards its full restoration and an endowment for upkeep in perpetuity.
More recently, a parallel tale played with regard to The Rose Theater’s sound system. Installed in the mid-1990s, the system was passable, though hardly exemplary, when new. Coverage, in particular, was mediocre.
After a brief maintenance-free run, console channels began failing, amps stopped working, and speaker components lost the crispness and intelligibility they once had. The theater was moved to action, spurred by the volume and quantity of complaints about the sound system from patrons and renters.
Julie Walker, managing director of The Rose Theater, turned to the Designed Systems Group at Omaha’s prominent Midwest Sound & Lighting, who had been dutifully applying band-aids to the old equipment.
Initially, David Walters of Diversified Design in Lincoln, Nebraska drew up a plan for the new system using equipment other than SymNet. However, Tim Burkhart knew that SymNet could do the job better with more flexibility and eventually convinced Walters that SymNet was a better choice for this project.
“Walters didn’t have anything against SymNet per se, rather he simply felt most comfortable recommending the manufacturer that he was familiar with,” said Burkhart. “We lobbied hard for SymNet, as we have had several very successful SymNet installations in the area and we felt that the SymNet ARC remote control panels would give The Rose Theater the sort of simple, intuitive user-interface that they were hoping for.”
Burkhart invited Walker and the other interested parties at The Rose Theater to visit the Omaha Community Playhouse (the largest community theater in the country), where a recent Midwest Sound & Lighting installation had been a big hit. “The group was impressed by the system’s performance, and there were several bits of functionality with respect to the ARC controllers that they wanted to go directly into The Rose Theater renovation.”
Everyone, including Walters, was sold on SymNet and so the project went forward. Working with speed and flexibility around a non-stop performance schedule, Midwest Sound & Lighting gutted the old system and replaced it with something vastly better.
A SymNet 8x8 DSP forms its heart, with a SymNet BreakIn12 and a SymNet BreakOut12 bringing the I/O count to twenty on each side. Three Crown CTS600, four CTS1200, and two CTS2000 amplifiers power an array of JBL loudspeakers. Left, center and right arrays, composed of one JBL AL6115, two JBL ASB6118s, two JBL AM6315-95s, and three JBL AM4212-00s for the balcony, provide elegant coverage of almost all the seats in the theater.
A handful of JBL Control-25 loudspeakers complete it with front fill. A new Yamaha M7CL-48 upgrades the system’s capabilities at FOH, and a Lectrosonics Venue wireless microphone system provides the system with a high-end input.
Two SymNet ARC-2 wall panels greatly simplify control of both large-scale performances and smaller-scale events. “A lot of the time small events will rent The Rose,” said Burkhart. “Before, they had to wrestle with the FOH console, even if they simply wanted a few microphones for speech reinforcement. With the SymNet DSP and remote control, they can put mics up in simple situations and bypass the console entirely.”
“SymNet’s auto-mix function and feedback suppression makes the system fully automatic and foolproof. An interface at the stage manager’s position makes simple volume tweaks intuitive. A second ARC-2 at the FOH position allows an engineer to send the program mix or a submix to anywhere in the building.”
“We used to get so many complaints about the sound system,” said Walker, “It really compromised the quality of our productions.”
“But after this fabulous renovation, not only is there an absence of complaints, but a steady stream of compliments, which we like! I also want to add that the professionalism of Midwest Sound & Lighting made this the smoothest project I’ve ever been involved in.”
Kramer Electronics’ Announces New VP Of National Accounts
Industry veteran and accomplished employee Mike Lewis promoted from national account manager.
Kramer Electronics USA has announced that it has promoted Michael Lewis as its new Vice President of National Accounts.
Lewis has a long history of association with Kramer Electronics.
He spent 15 years at Comprehensive Video Supply in New Jersey selling cables and Kramer Products, and then spent the next 6 years selling cables for CompuCable before joining Kramer as Regional Sales Manager of the western region in 2003.
For 4 years, Lewis was consistently one of Kramer’s top performers in regional sales before being promoted to National Accounts Manager.
As National Accounts Manager for the past 3 years, Lewis has increased sales to Kramer’s several large accounts with multiple locations across the U.S., at a higher growth rate than the average double-digit growth rate Kramer has enjoyed for the last 12 of 13 years.
Mike Lewis has been an integral part of Kramer’s ongoing success, resulting in his newly deserved promotion to Vice President of National Accounts.
Within this new position, Lewis will be responsible for managing a group of sales personnel tasked with continuing to grow Kramer’s National Accounts business at a rapid rate, as well as building and actively promoting Kramer’s ever-expanding line-up of cable, wall plate and table box products through all levels of distribution.
Along with his Kramer team, Lewis is confident that he can uncover new opportunities, not only directly related to cable products but opportunities within the A/V marketplace as a whole, ensuring the continuation of Kramer’s success.
Dave Bright, President of Kramer Electronics USA, states that “Mike is the perfect choice with his many years of cable knowledge and experience prior to joining the Kramer family.”
“Furthermore, I am very proud to have Mike join our Kramer sales management team. He is one of the purest salesmen I have ever had the pleasure of working with. At Kramer, much of our success is the result of relationship selling…and Mike is the master at that.”
“His great work ethic and the always positive attitude he brings to his job every day is what makes him so successful. I am sure Mike will continue to excel in his new position.”
Do you know what to do if the drums in the sanctuary are too loud?
Any time a band has some of its sound coming through the main PA system (usually vocals and electronic instruments) and some of the sound coming from the stage acoustically (most notably the drums) you have problems.
The drummer must play loud enough to keep up with the sound system, which he cannot hear.
However, playing loudly enough for the back row of listeners means that the drums are often too loud for the first several rows. It’s even louder on stage, which requires the rest of the band to play louder and turn up the stage monitors.
The result is a stage volume that is overwhelming – too loud for the room, and often louder than the main sound system in the room, and still unclear.
People get frustrated and irritable, and some leave to find another church where they can understand the music.
A major part of the solution for this problem is to control the sound of the on-stage instruments, beginning with the drums. There are three steps in controlling drums in church:
1. Contain the acoustic energy from the drums, 2. Absorb the acoustic energy from the drums, and 3. Reinforce the sound that you want from the drums and provide monitoring back to the drummer.
Containing the acoustic energy from the drums is the easiest part. The sound of the drums travels from the drum head to the ears of the people hearing it.
The strongest part of that sound is generally direct line-of-sight.Many churches have installed plexiglass drum shields around the drums for this purpose. It’s cost-effective and it’s a reasonably effective starting point.
The plexiglass reflects most sound, preventing the direct line-of-sight sound from reaching the people in the congregation.
This solves one problem and introduces a couple of new ones.
Click to enlarge.
Plexiglass does not absorb sound; virtually all of the sound created by the drums is reflected; that means that the sound is still in the room, it’s just not traveling to the listeners in a direct route.
The drummer often feels more confident now that he’s behind the plexiglass, and often times he plays harder, creating even more sound than before. Now that sound is bouncing around the room as reflected sound.
Reflected sound is, by definition, noise: it has the same amount of energy as direct sound, but because it is reflected, it has become “incoherent.” Now instead of hearing the clear “slap” of the snare from a single source, we hear reflections of that slap from various reflective surfaces around the room.
The clarity is decreased, but the energy of the snare is still there, rattling around the room, muddying up the rest of the sound.
The second problem with a plexiglass drum cage is that the first reflection of the sound is concentrated back at the drummer’s ears.
The potential for hearing damage is greatly increased. That’s one reason some drummers want to play loudly – they can’t hear the sound as well as they used to, so they feel the need to play louder.
The result of plexiglass by itself is that the total energy of sound is not decreased. Instead, it’s just bouncing around the room, making the rest of the sound muddy, and damaging the drummer’s hearing.
After we block the direct sound of the drums with a plexiglass drum shield, the next step is to absorb a good portion of the sound, to keep it from filling the room with incoherent echoes. This is generally accomplished with sound-absorptive foam.
Generally, the foam is installed in three locations: on the wall behind the drummer, on the plexiglass itself, and as sound-absorptive “lid” over the top of the drummer.
How much absorption to install is governed by several factors, some practical and some aesthetic. Since the drummer needs to be able to see the rest of the band, it’s best to not block all of the plexiglass.
Click to enlarge.
Rather, install foam along the bottom and sides of the plexiglass. Generally, the foam is not installed above the height of the drumheads themselves, and often only to the top of the kick drum. On the sides, install the foam higher, particularly on the side with the snare and hihat, as these are the greatest sources of sound.
Install a greater amount of sound absorption on the wall behind the drummer. In fact, complete coverage of this wall is often appropriate, up to the height of the plexiglass drum shield.
Since the sound from the drums is omnidirectional it will either strike the wall first or it will reflect off of the plexiglass and then strike the wall. Absorption on the wall behind the drummer will be a big help in keeping the reflections around the room under control.
Some of the sound from the drums, of course, goes straight up, where it will bounce off of the ceiling before eventually making its way to peoples’ ears.
If you have done an effective job of absorbing the sound inside the drum cage this reduced amount of reflected sound may be acceptable, or even desirable.
It may still be too much sound, especially in a low-ceilinged room, or with a large drum kit, or with a particularly physical drummer. In this case, it may be necessary to add a sound absorptive ceiling over the top of the drum kit.
All this absorption sounds expensive, but it is possible to cover all three sections - on the plexiglass, the wall behind the drummer, and the lid - for about the cost of the plexiglass drum shield itself.
To this point, we have been reducing the overall volume of the drums. The stage volume is under control, so the musicians can hear themselves, and the sound from the stage doesn’t overwhelm the main speakers. The front several rows of the congregation are no longer being overwhelmed by sound. But now the back part of the sanctuary isn’t being reached.
The third step of controlling the drum sound is to put the drums into the sound system. At the very least, you’ll need to mic the kick drum, the snare drum, and the hihat. With careful placement, a single mic can pick up both the snare and the hihat, for a two-mic minimum.
As far as mic selection goes, my preference is to use a large diaphragm mic on the kick drum – either a dynamic mic like the Shure Beta 52 or the Sennheiser E602II, or a large condenser mic like the CAD E100.
Dynamic mics tend to capture the “boom” of a kick drum well, and condensers can capture the “snap” of the sound.
Audio Technica makes a mic (AE2500) that has both a condenser capsule and a dynamic capsule in it. Be sure that the mic can handle the high sound pressure levels of a kick drum closely miked.
Your first choice for a snare mic is a simple dynamic microphone, with the ubiquitous Shure SM57 being the most popular. It’ll take a number of accidental whacks from overly-enthusiastic drumsticks and keep working well.
Dynamic mics can also be used on the toms, but there are several very nice tiny condenser mics that have become popular, like the AKG C418 or the Audio Technica PRO 35.
Sennheiser makes a small dynamic mic for this purpose, the E604. These small, specialized mics generally come with their own mic clips which attach directly to the drum itself, reducing the number of stands and cables sticking out of the drum kit, and allowing the plexiglass drum shield to be brought in nice and tight.
In a large room, you’ll want to add a pair of overhead mics, to capture the overhead cymbals and the overall ambience of the drum kit. Small condenser mics like the entry level AT Pro37R or the higher priced Sennheiser E914 are common choices.
Recently, the trend has been moving towards large-diaphragm mics overhead, including the inexpensive CAD GXL2200 or the versatile AKG C3000B. Be sure to experiment with mic placement, listening closely to the sound of each mic, to determine best placement on your drum kit.
Click to enlarge.
Once you route the new mics to your mixing console, you’ll need to consider monitoring for the drummer.
The simplest way to give your drummer the ability to hear what he needs to hear is to use an unused Auxiliary Send from your mixing console. Send that aux to a headphone amp (there are many entry-level manufacturers including Samson, Behringer, Rolls, and Carvin).
Headphones with significant isolation help ensure the drummer will get the reinforced sound and not just bleed from around the ear muffs, and some drummers like headphones that emphasize low frequencies. In-Ear Monitor models which produce extended low frequencies (dual driver models) are also an option.
Beyond this simple setup there are many other more advanced monitoring solutions such as a separate monitor console or personal mixing devices from folks like Aviom, Hear Technologies, Furman, and MyMix.
If you have the room, I prefer using a compressor on both the kick drum and the snare, and an ideal world would call for gates on the toms, the snare and the hihat, to tighten up the sound, but most churches will stop before that point.
The main goal is to prevent the acoustic sound of the drums from either overpowering the rest of the band, or reverberating around the room, by bringing the drums into the sound system with the rest of the band. You’ll be surprised how much cleaner your band sounds, and how much easier it is to keep the volume under control.
DPA Miniature’s Reinforce Concert At Hong Kong’s Polytechnic University
Jacky Cheung’s Private Corner made extensive use of DPA miniature microphones to amplify instrumental accompaniment.
A range of DPA miniature microphones were used at Jacky Cheung’s recent Private Corner concert in the Jockey Auditorium, Hong Kong Polytechnic University.
The DPA 4099S and 4099T instrument clip mics were used at the concert for trombone, trumpet and saxophones, while a grand piano was miked with 4061 miniature omnidirectional microphones.
Mike Wong, Cheung’s monitor engineer, chose the cardioid 4099s for their excellent focus on the instrument as well as isolation and rejection from adjacent musicians.
“In this concert, we were striving for a natural and uncoloured sound and therefore really appreciated the musical accuracy of the 4099s,” he says.
Added front of house engineer Daniel Kwan, “Besides the superior sound quality, we like DPA microphones for their mounting and positioning possibilities. The 4099 series has mounts and clips for a wide range of string, brass and woodwind instruments.
“We also appreciate DPA’s uncompromised attitude towards their products and the service provided. The local DPA distributor, DMT, has done a good job supporting us, too.”
Quote meon an estimate et non interruptus stadium. Sic tempus fugit esperanto hiccup estrogen. Glorious baklava ex librus hup hey ad infinitum. Non sequitur condominium facile et geranium incognito.
