Article

Thursday, September 02, 2010

One-Stop Shopping: Captain, What Does It Mean, This Term “Full Production”?

The sound company’s job is to advance the show with the artist and show up with a rig. Not so when the full production falls into your lap.

Sound companies handle “one-off” shows every day. It’s usually formulaic, and after a while, we do it by rote. 

But what happens when the client wants one-stop shopping? This is also known as “full production” or “turn key service,” and it’s quite a bit more involved than an average show. Generally months of planning and coordination are needed, as well as work with a number of subcontractors. It just can’t be done by the seat of the pants.

Normally, when a sound company is hired for a show, the client is a promoter or a venue. They provide the stage, they provide the power, and they provide the labor. The sound company’s job is to advance the show with the artist and show up with a rig. Not so when the full production falls into your lap.

Particularly for large, multi-stage festivals, hiring a single source to handle all the entertainment elements of the event is almost a necessity. The event director has too many other things to handle to have to worry about the details of his entertainment. 

Steve Rosenauer, director of the St. Mary’s University Alumni Association Fiesta Oyster Bake in San Antonio, Texas, once told me his definition of full production: “As a client, full production means working with a knowledgeable and experienced company that can produce a turn-key operation with regard to organizing, building and operating the necessary staging, sound, lights and equipment needs, with all meeting the negotiated specifications of the event as well as the bands. A company that does this can greatly enhance the quality of the event and provide a solid peace of mind to the entertainers and the event organizers.”

For the purposes of describing the process of a full production event, I will use the Fiesta Oyster Bake as my example. It’s a two-day, six-stage festival which kicks off San Antonio’s annual Fiesta Celebration every April. Fiesta has been ranked as the second largest party in the U.S. (Mardi Gras being first) by the National Meeting Planners Association. (And yes, they bake tons of oysters!) For years, our company, Sound Services, worked with this event. (Note that we recently chose to close the company for reasons completely unrelated to business.)

PREP MAKES PERFECT
In order to be ready by mid-April, we would start working in November. To be fair, we had been doing this event for nearly a decade, and had amassed a team of subcontractors with whom we were all very comfortable. Until a company gets to this point, preparations probably need to commence even sooner.

In November, we would begin talking about what our needs were going to be. Because city electrical inspectors were involved, we checked the City Code Compliance for any new electrical requirements. For example, one year (and for the first time), we were required to ground all of stages to the audio power distribution services, as well provide non-conductive covering of all power cables running in public areas. Not fun to discover things like this at the last minute!

We provided staging, sound, lights, backline, labor and all technical personnel for the festival. Because the client uses many more generators than just ours, they made those arrangements, but they used our generator provider so we were assured that power would not be a problem. The generator provider also stayed in contact on any change orders he received that might affect us.

Also by November, the client usually had more than half of the talent booked, so we got a vague idea of what to expect from headliners’ riders. By December, we started talking with our subcontractors, discussing what had changed from the previous year, giving them the firm dates, and requesting a firm price by January. 

After ringing in the new year, and still four months out, it was time to nail down the financials. Be very meticulous with this process!  Everything must be committed to paper, and math triple-checked in order to avoid any mistakes that could cost an entire profit margin.

It’s doubly vital to get this facet correct in the first year with an event, because the client will base future projections on those first year costs. Therefore, a mistake probably can’t be made up for next year.

Only after every cost is defined and listed, as well as those of the subcontractors, should the price be committed to the contract submitted to the client. Note: the one thing we found most often overlooked is the cost of a production manager. The hours and hours you spend working on this shouldn’t be done for free!

WORKING IN EARNEST
We would submit our contract on the first of February, with the understanding that requests on artists’ riders would probably cause an increase in total price. By this point, the client had all talent booked, so we could start working in earnest to learn just what those extra costs might be. My goal was to have all this information by the 15th of the month, still two months out.

There is a negotiation with contract riders and advancing the show that can - with some diplomacy - help reduce the number of additional line items for your client. Because most headliners’ riders are based on arena shows, for example, they will often concede some lighting instruments. 

On the other hand, you don’t want artist representatives to think your client is cheap, so know where and when to stop asking for concessions. It’s important to manage your client’s expectations in this regard as well. Most touring artists also understand that festivals differ from concerts, so if the stages are adequately stocked to begin with, most of the added line items will be for backline and spotlights.

Once we determined all of the additional artist-related expenses, we submitted a contract addendum. This addendum should include absolutely everything - a. client will begin to lose confidence if presented with more than one price addition. His budget is set in stone by this time, and your math errors and oversights are not his fault.

MINIMUM OF 40
Because Sound Services was responsible for the entire Oyster Bake Festival, not just the two stages we were physically covering, it was imperative that we advance the show with every artist. In this case, we’re talking a minimum of 40 bands, which made for a lot of work. But it accomplished several very important things. 

First, we got a thorough look at the requirements of every stage, and were assured that each subcontractor could adequately cover the entertainment line-up. If there was a particularly tough set change on a stage at a particular time, we could arrange to have extra help on hand at that time. 

Second, it gave each artist a feeling of confidence to know that individuals who care about their performances run the festival. Third, we established consistency in the way the artists were handled. The subcontracting sound companies all appreciated this.

And fourth, we could apprise artists of the “special quirks” of this festival. For example, it’s held on a university campus that is, itself, located in a neighborhood, not on a major thoroughfare. Getting to the venue is difficult when 80,000 other people are also trying to do the same, and there is no alternate route.

Sometimes when we told first-time performers to allow three hours to arrive, some balked, but we remained adamant. The ones who didn’t believe us were invariably late, which is a no-win for everyone. (By the way, returning artists were never late!)

Further, artists can’t drive to any stages except the main one, because they’re all positioned among campus buildings. For this reason, full backline was provided at every stage, and musicians were discouraged from bringing more gear than they absolutely had to have. To accommodate this, the university set up a team of volunteers to ferry musicians and their gear to the stages. It took several years to streamline this process.

Once all the advance work was complete, we created stage plots and input lists for every stage, and for both days. These were then dispatched to the sound companies working the festival with us.

GETTING CLOSER
A pre-production meeting with the festival committee and all stage managers was held six weeks to two months out. Each committee reported on their progress and, although we weren’t involved in things like pizza ovens and beer sales, it helped us to know what was going to be happening around us. 

Entertainment production is an important part of this meeting, and we made it a real bonding experience. Construction of “Stage 1,” for example, meant an entire campus parking lot has to be closed two days prior to the event, and thus it was critical that the timing be executed properly by the university security department. 

We also got to meet the stage managers and orient them as to what was expected of them. These folks are critical for smooth-running shows, and we let them know that. While their duties are light, the few things we needed from them are all important to the show.

Other things covered in this all-important meeting were issues of water, green rooms, use of volunteers (there are hundreds!) and getting musicians to the event and their respective stages. Over the years, and learning from our mistakes, we developed methods to efficiently accomplish these tasks, but until you’ve worked with an event for a long time, these issues are extremely important to thoroughly think through. For example, from experience we all learned that as much water as we thought we needed - double it!

At this time, we also walked the campus with the festival director, making note of things like trees that needed trimming or light poles tp temporarily remove. (Grounds and electrical departments need to be notified in advance to schedule work like this!)

WHO’S DOING WHAT
By one month out, we had a firm grip on exactly who was doing what. For example, if there was a sound company short a monitor engineer, this was the time to step in and lend a hand. Each subcontractor provided us with a list of personnel and how many vehicles (and of what type) they would be bringing on site. One aspect to double-check: be sure each contractor is providing enough people. For example, backline duties done properly for six stages requires more than two techs.

At this point, we would tally up all production people (including stagehands and spotlight operators) and provide the festival director with the number of parking passes and wristbands needed. Remember - on a multi-day festival, each person might need a fresh wristband each day. We also padded this number by a few more to replace ones that were inevitably lost.

Very key: the best technical person on staff must be in charge of production management. Even with the best preparations, all kinds of little things can go wrong, especially at multiple stages. One person not involved in production at any one stage has to be free to fight the fires, and this person should be well versed in technical knowledge as well as diplomacy. 

Our production manager for the festival spent each day traveling between stages, providing a break to a beleaguered engineer here, dealing with a power problem there, handling a recalcitrant band engineer somewhere else.  He also carried a radio for instantaneous contact. And, this person must have healthy legs – in a very crowded festival, a golf cart won’t work!

Three weeks out, we assembled packets for all of the subcontractors involved.  These included parking passes and wristbands, a map of the campus showing all stages and parking areas, a complete schedule of the event, and for the sound providers, stage plots and input lists. Load-in times were also provided.

Scheduling personnel is critical at this point. We staggered the load-in times so that we could make the best use of our stagehands. Stagehands have a four-hour minimum, and each is usually scheduled to work at more than one stage during a shift.  For load-out, we scheduled a much larger number of stagehands. This schedule was then filed with the labor company as a written work order, and note that this also included spotlight operators as well.

IT’S SHOWTIME!
Two days before the festival, we began to build the stages. The provider arrived with semi-trucks loaded with staging, and we again walked the site with the festival director, spotting the stages, front-of-house risers, spot towers and security towers.

The day prior to opening, we loaded in at our two stages, which then left us free to address the mayhem of everyone else loading in the next morning. The lighting contractor also loaded in with us in order to be out of the way, and this left the lighting directors free to work with headliners who might arrive early. On-site security was continuous at this point.

Day one of the festival would arrive, and we were free to conduct headliner soundchecks on our stages. Fortunately, the first act didn’t begin until 6 pm, so the atmosphere wasn’t too stressful.

The production manager was also available to address the various surprises that unfold, as they invariably will. This is where months of planning pay off and you can look really good to the client, who’s running around putting out all kinds of fires while his production people are calmly doing their jobs.

If all subcontractors are competent and well prepared, the event should run like an average one-off show. One caveat, however: it’s still a multi-day, multi-stage festival, with thousands of people swarming all over, so competent, well-informed stage managers become critical to your existence. 

They aren’t needed to get artists on and off the stage – we had already planned that out. They are most definitely needed to competently answer artist questions - “Where are our food coupons?” and “Where is our dressing room?” and the like. They also kept lots of water on ice, and plenty of ice in the ice chests.

The most important thing stage managers did, however, was manage the radios. Each stage had a radio, as did the production manager and the lead backline technician, and they were on a common channel with the event director. 

As the production staff performed its various tasks, we didn’t have time to monitor a radio, but when we had a problem or needed help, we simply asked a stage manager to contact whomever we needed. Previously we carried individual radios, but learned that this alternative approach worked so much better for everyone, plus it gave the stage managers a sense of ownership of their jobs as well. 

The best advice: “be round.” Roll with the punches and don’t get too excited by the inevitable little surprises that spring up. Make the production of entertainment as smooth as possible and don’t create tension or problems. That’s a big reason you were hired!

THE AFTERMATH
When it’s all over, the results of diligent planning and scheduling should continue to pay off. We found that handling a large number of stagehands at the end of the festival worked best if we arranged for the crew chief to assemble all of them at a pre-arranged site and make assignments from there.

Stagehands were first dispatched to the stages manned by our subcontractors, then re-routed to our stages last.  We always got this show loaded out within our four-hour labor minimum, by the way.

The production manager continued to make a circuit of the stages, being sure each stage had its allotted stagehands and collecting any left-behind belongings. We later attempted to repatriate these items with their owners.

When all the dust cleared a week or two later, we sat down and created a recap of the event, and this went into the file for next year. We also sent this recap to the festival director. Included were a summary of any issues that came up, general incidents, what worked well and what didn’t, and suggestions for improving next year’s event.

By working with the client in this fashion, we made ourselves a part of the event team, and enjoyed a multi-year contract. We also ingratiated ourselves to our subcontracting partners, who appreciated the work and reciprocated when appropriate. 

It’s just good business to develop this kind of working relationship with your clients and fellow business people, and it leaves you feeling pretty good about yourself as well.

Teri Hogan is a long-time audio professional and was co-owner of Sound Services Inc., a sound company based in Texas.

{extended}

Tech Tip Of The Day: Creating A Spacious Rhythm Guitar Sound

I tracked my rhythm guitar in mono but I think it needs to found more full. What can I do?

 
Q: I’m working on an album and I’m really having alot of trouble getting the rhytmeh guitar to sound just “right”.

What i’ve discoverd is that it needs to sound more open, with maybe a more stereo feel.

However, the problem I’m having is that when I tracked the instruments I only threw one mic on the amp.

Of course, basically everything else was recorded stereo!

So, now I’m at a loss. What can I do?

A: No doubt about it, this is one of those things that most easily could have been solved in tracking.

Back when tracks were valuable real estate (stop laughing, that time existed!), few instruments got recorded in stereo. Instead, they were recorded in mono, then panned to a position within the left-to-right sound field.

Today, nearly everyone has access to more than we ever dreamed possible, and those holds true whether you’re using a hardware or software based DAWs.

All this really just means that we have more at our disposal to create a more spacious (and more natural) stereo spread, on really most anything, as track count becomes less and less of a consideration.

Since the rhythm guitar is often the instrument around which all other tracks are built, it’s important to give it that big sound field.

Thankfully, however, there is a way to accomplish this aside from initially recording the instrument in stereo.

With an electric guitar, you can usually use a multi-effects processor to convert a mono input signal into an enhanced stereo output via a stereo chorus or panning tremolo effect or by using a delay in which the left side delay is shorter or longer than the right side delay.

Some processors allow you to assign a dry signal to one output and the delayed signal to another.

By using a very short delay, you can fool the listener’s ear into believing it’s hearing two guitars, just because the left image is offset in time a bit from the right.

As always, we welcome input from the PSW community and would love to know how you would solve this mixing issue. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

{extended}

Wednesday, September 01, 2010

Audio In Houses Of Worship

A comprehensive guide that will get you started in the world of church sound, no matter your level of familiarity.

 
Audio is an essential element in any modern-day religious service.

What is heard by the congregation is a combination of the acoustic qualities of the room and the performance of the audio system.

Some of the desirable acoustic qualities in a house of worship are:

Reverberance: When well controlled with early decay, the effect is perceived as a beautiful sound that enhances the quality of the audio. See the Rane Pro Audio Reference for a definition of “reverberation.”

Clarity: The ratio of the energy in the early sound compared to that in the reverberant sound.

Early sound is what is heard in the first 50 - 80 milliseconds after the arrival of the direct sound. It is a measure of the degree to which the individual sounds stand apart from one another.

Articulation: Determined from the direct-to-total arriving sound energy ratio. When this ratio is small, the character of consonants is obscured resulting in a loss of understanding the spoken word.

Listener envelopment: Results from the energy of the room coming from the sides of the listener. The effect is to draw the listener into the sound.

Where a conference room would be optimized for articulation and clarity, a symphony hall is optimized for reverberance and listener envelopment.

A good house of worship is optimized as a compromise between the somewhat conflicting requirements of music performance and the spoken word.

Articulation must be excellent but sufficient reverb is required to complement music performances. All reflections must be well controlled to achieve this balance and ensure the best possible listener experience.

An Example Of Good Sound

There are other possible examples but the author really likes this one. In some mosques, cathedrals and tabernacles there are wonderful domed ceilings that have marvelous natural acoustic properties.

The acoustic coupling from performers to the congregation grouped under the dome makes for a very (dare I say) “spiritual” experience. For the purpose of this article, this level of performance is a “gold standard” to which other acoustic spaces will be compared in the search for improvements and recommendations.

The U.S.A. Pavilion at Florida’s Epcot Center makes for an interesting case study. There is a dome ceiling in the pavilion. Under the dome an eight-part acappella group called the “Voices of Liberty” performs. For those under the dome listening to the group, the sound is beautiful and inspiring. Moving out from under the dome, the “magic” is gone.

This level of performance is not feasible in a typical house of worship but it does establish an icon as to what could be if there was sufficient skill (and budget) applied to the acoustic and audio system design.

And Now The Ugly World In Which We Live

Contrast this to a typical public address system squawking bad sound to the congregation.

That which was good is replaced with misery. You reach for a bottle of aspirin to calm the headache induced by a pair of blaring powered speakers.

Some of the problems encountered by audio designers/consultants include:

Excessive Reverberation—such that articulation and clarity is poor.

Echo—where a discrete sound reflection returns to a listener more then 50 milliseconds from the direct sound and is significantly louder then the reverberation sound.

Flutter echo—repeated echoes that are experienced in rapid succession that occur between two hard parallel surfaces. All echoes ruin the acoustic properties of a room and a flutter echo is particularly damaging.

Coloration due to reflections—when a reflection destructively recombines with the direct sound modifying the frequency response in the process. These are non-minimum-phase colorations as correction with equalization is not possible.

Delayed Sound—from coupled volumes (contamination from adjacent rooms storing sound energy and then returning the energy to the main room).

Psychological preconditioning—It is a common problem for the clergy and congregation to be so preconditioned by bad sound that they become resistant to change and find it difficult to (at first) recognize good sound.

This can also work in the audio consultants favor when the customers are preconditioned by good sound and are willing to invest the required resources toward good audio design.

For those of us designing audio for houses of worship with a rectangular room, flat walls and probably a vaulted ceiling, some form of sound reinforcement is required. Through attention to detail and careful design of the audio system, the experience of the congregation can be non-aspirin inducing and the system simple to use.

Common Signal Processing Blocks
Let’s begin by looking at the universal signal processing chain common to all audio systems. In the simplest systems these functions are accomplished in an audio mixer that feeds a pair of powered speakers.

More sophisticated systems include equalization, compression, limiting, automation, feedback suppression, electronic crossovers and other tools of the trade. These days it is possible to include all of these functions in a DSP (Digital Signal Processor). One example of the signal chain from the minister’s microphone to the power amplifiers is shown in Figure 1.

The signal processing flow starts at the Analog Input. A 2-band Parametric Equalizer filters out-of-band low frequencies. The microphone signals are summed together in an Automatic Mixer. An AGC (Automatic Gain Control) reduces the dynamic range and a High-Pass Filter in the side chain improves the performance of the AGC.

The Level control can be tied to a pot on the wall or a smart remote. There is a Feedback Suppressor for good measure. A 2-way Crossover supports a biamplified system. The 10-band Parametric Equalizers are utilized for both wide- and narrow-band corrections.

Generally, wide-band filters correct minimum-phase frequency response irregularities in the speaker drivers and in the room response. Narrow-band filters are useful to partially correct non-minimum-phase related problems such as energy stored in room modes (reverberant energy).

A Limiter could also have been added to protect the system from clipping if that feature is not included in the power amplifier.

Now let’s take a look at some of these signal processing blocks in greater detail.

Analog Input / Microphone Preamp

It is surprising how often even experienced audio consultants will configure an audio input incorrectly.

It is important that as much gain as possible is accomplished at the front end of the system in the Analog Gain stage.

Any additional gain from Digital Trim after the input stage degrades optimum signal-to-noise performance.

As an example, let’s set the input gain to a value of +40 dB.

One way is where the analog gain is set to a value of +45 dB and the digital trim is set to -5 dB (as in Figure 2), the measured input referred noise is -127 dBu.

A common (but incorrect) way would have the analog gain set to a value of +30 dB and the digital trim set to +10 dB (the author has seen this repeatedly), to give the same Mic gain of 40 dB—but now the input-referred noise is degraded to -114 dBu.

That is an increase of 13 dB for the noise floor, or a change (in the bad direction) of 8 dB in the maximum SNR (Signal to Noise Ratio). Your exercise is to determine why the SNR was only degraded by 8 dB rather then the intuitively obvious value of 13 dB.

Answer: The noise floor does drop by 13 dB, but this combination of settings causes the analog input stage to clip at an input level that is 5 dB lower. Hence, the change in system SNR is 8 dB.

Applying attenuation after the input stage (rather then gain) reduces overload performance and so should be used with skill and discretion. It is the proper technique to maximize noise performance.

Input Low-Cut Filter

A very good idea is to add a low-cut filter set to ~80 Hz after the input stage to minimize the effects of undesirable low-frequency noises such as bumps and thumps that come from handling the mic and also wind blasts and pops from speaking too closely into the microphone.

In Figure 3, both 2nd-order filters are set to the same frequency to produce a 4th-order filter.

There should also be a low-cut filter in line with the SC (Side Chain) input of the AGC (Automatic Gain Control).

This filter can be set to a higher corner frequency (such as 120 Hz in Figure 4) to improve the performance of the AGC by rejecting the effects of low frequency noises.

The Auto Mixer—A Little Automation Buddy

An Auto Mixer (shown in Figure 5) is a good idea when there is more then a single open microphone.

Auto Mixers combine the signals from multiple microphones and automatically correct for the changing gain requirements as the NOM (Number of Open Microphones) changes.

Threshold with Last On is a useful setting for all microphones used in a worship service (Figure 6).

Unused microphones (input levels are below threshold) are gated. When the input of a microphone is above threshold then other inputs with a lower assigned priority level are ducked.

Automatic Gain Control

A Compressor is the correct processing block in this link of the audio chain. Something is needed here to prevent exuberant preaching from melting down the congregation.

Surprisingly, an AGC can be very useful in this position but configured to behave more like a specialized compressor by using the settings shown in Figure 7.

The value of “Threshold re: Target” is set to have an offset of 0 dBr so that “Threshold” has the same value as the “Target.” “Maximum Gain” becomes 0 dB and the gain curve starts to look like a compressor but there are additional controls in an AGC for Hold and Release that are useful when the input level is below threshold.

These settings avoid the problems of compressor “pumping” when that exuberant speaker is at the microphone as attenuation levels are held between spoken phrases.

Then, when transitioning to a more reserved speaker, the hold time (below threshold) is short enough to expire so that the gain returns to a normal level.

An Exciting Labor-Saving Tip—Put a Control On the Wall

Here is an exciting tip. A level control can provide attenuation as needed under the control of a pot on the wall or a smart remote.

This is handy in systems where a minister needs to run a system alone without the assistance of an audio specialist who is running a mixing board. The remote can be located on or close to a pulpit which places control of the audio system at the fingertips of the minister. The DSP control is shown in Figure 8.

Feedback Suppression—A Gift From Above?

The next item in this processing chain is somewhat controversial. It is a Feedback Suppressor.

To some audio consultants a Feedback Suppressor is heresy! The argument is that a properly calibrated system has no need of such a Band-Aid.

This is generally true, but there is one case when it is wise for an audio consultant to suffer the ignominy of using a Feedback Suppressor—a lay clergy where the person speaking is untrained and/or unfamiliar with proper use of a microphone.

The author has witnessed such a person cup their hands (in the attitude of prayer) directly around the microphone capsule. The hands form a resonant chamber that results in squealing feedback.

A good Feedback Suppressor would have locked on to the offending tone and notched it out posthaste.

Parametric Equalization: Now We’re Having Real Fun

Parametric equalizers are used for both wide and narrow band corrections.

Generally, wide-band and shelf filters can correct for minimum-phase frequency response irregularities.

One interesting detail of Figure 10 is Hi-Shelf Filter 1. This filter was added after achieving flat in-room response.

Since the system was calibrated in an empty room, this extra high-frequency energy is intended to compensate for the high-frequency absorption of the congregation when the room is full of people.

There is also a noise-masking effect in some congregations that will tend to obscure the intelligibility of the spoken word. In practice this approach of adding a bit of extra high-frequency energy into the room works well.

Narrow-band filters (see Figure 11) are useful to partially correct non-minimum-phase related problems such as energy stored in room modes.

At low frequencies this energy causes bass to sound indistinct, and in midrange to lower treble this energy is perceived as reverberation.

These filters attenuate the frequencies that bounce about the room. In an acoustically live room, room resonances can propagate for a surprisingly long time causing these frequencies to “build up.”

Narrow-band filters are just a partial solution. Greatest effectiveness is achieved when filters are used in conjunction with acoustic room treatments such as diffusers, high/mid frequency absorbers and bass traps.

Specific Examples

Example #1: A Small Church

Description:
The ceiling is low suspended acoustic tile over an open space covered with thin carpet. The RT60 (the time it takes the reverberant sound to decrease by 60 dB) is short, so controlling reverberation is not a problem for audio clarity.

In fact, the room is a touch “dry” for music, and content of the worship service includes live musical performances.

The sources of audio are the minister with a wireless microphone and the band.

Additional sources are DVD/CD players and other devices as needed. Control is via a 24-channel mixer with all inputs used.

Output is to a pair of powered speakers mounted high in the corners of the room in a stereo configuration. This installation was done by members of the congregation without consultation with an audio professional.

Next, let’s look at some specific examples to bette illustrate these points.

Problems:
• The quality of the audio is poor with numerous problems including uneven frequency response.

• An experienced sound person is required to run the mixer for all audio system use.

• There is poor coverage of the congregation from the stereo speaker pair. People sitting in the hot spots just in front of the speakers are blasted with excessive level, and the rest of the congregation is exposed to a strong interference pattern between the two speakers.

The system is uncompensated for room modes, room response and speaker response irregularities.

There is a small “sweet spot” in the center of the room where the two speakers combine coherently but there is an isle down the center of the seats. Since there are no chairs, no one is seated in the “sweet spot”.

So does this audio system work the way it is? Yes, but even the pastor knows the congregation may not be receiving the best possible audio experience. This example is rich in possibilities.
Recommendations

Improvements to this system are accomplished in a number of ways. A DSP can be used for equalization, other processing and to add automation to the minister’s microphone.

The entire worship band could be run through a mixer with each individual input processed by an AGC.

There are admittedly downsides to automating the audio mixing of a large group, as the automation is not as intelligent as an experienced sound person, but is possible in some cases.

The speaker system is examined to look at options that provide more even coverage of the congregation. Improvements to this audio system can be introduced in phases.

Phase 1:
Add a DSP box between the output of the mixer and the feeds to the main speakers and on-stage monitors. Features added could be:

• Parametric Wide-Band Equalization. This alone would greatly improve this system.

• Parametric Narrow-Band Equalization. A short RT60 makes this unnecessary at this time. However, remodeling could increase RT60 to where narrow-band equalization would be needed. (This room could use bass absorbers).

• High-Pass Filtering. If not in the 24-channel mixer already.

• Compression. Always a good idea with microphones because of the inverse square law relationship between the preacher’s mouth and the location of the microphone.

• Feedback Suppression. If needed.

Phase 2:
Automation is incorporated with automixers and remote controls. There are many exciting ways to add these features depending on the needs of individual congregations.

The most obvious upgrade would be to add the ability for a minister to turn on and control the main microphones from a simple control panel located in easy reach at the front of the room.
Phase 3

The very uneven coverage of the congregation by the stereo speaker pair needs to be addressed, as shown in Figure 12. The seats directly in front of the speakers have enough level to kill small animals.

If the audio system were perfect then each seat in the congregation would have the same audio level. In the author’s experience, similar rooms have been controlled within a couple of dB.

In this example, the seat closest to each loudspeaker is about 15 dB louder then the worst seat on the floor, and interference between the two speakers adds to a very lumpy and unpleasant frequency response.

Another problem is that the FOH (Front Of House) Mixer is placed in a location for good sound, causing the levels at the ends of the front rows to be way too loud.

Line Array Speakers

One improvement is to remove the stereo pair of point-source loudspeakers, and install a floor-to-ceiling line array located in the center of the back wall as shown in Figure 13. Coverage of the congregation is more even, and the level at the FOH Mixer location is very similar to the coverage level over the whole floor of the congregation.

The level of the stage monitors is greatly reduced and some of the stage monitors may no longer be needed depending on the individual needs of the musicians.

Within the near field of the line array there is a range were the audio level will decrease by only 3 dB for each doubling of distance which greatly helps even the coverage across the entire floor.

One other characteristic of this application is that the audio is distributed across the whole line so that even if a microphone is right next to the line there is little tendency to feedback.

In this example, there is a low suspended-acoustic-tile ceiling that shortens the length of a line array speaker. This limits some of the good qualities of a line array so this might not be the best solution.

If the room were remodeled so there was a high ceiling, then a line array would make more sense because a longer line array would fit. This is especially true if the newly remodeled ceiling was acoustically reflective causing the RT60 of the room to be much greater.

The high directivity of a long line array greatly helps to project the audio out to the floor rather then have the audio directed toward the ceiling where it contributes to the reverberant energy and slap echoes in the room.

Supplemental Distributed Array Speakers

Because of the dropped ceiling, another option would be a distributed array of supplemental ceiling speakers in the back of the room as shown in Figure 14. The loudness level of the main stereo pair could be reduced by at least 12 dB.

This would greatly diminish the effects of the hot spots in the front of the room but would leave the level at the back of the room way too low. Ceiling speakers can be added in the locations shown to fill in the audio in the back of the room.

It would be very important to include a speaker over the mixer location so the audio at that location matches the level in the congregation to aid in achieving an accurate mix.

Why The Delay?

The ceiling loudspeaker signals should be delayed in time so their output combines coherently with the output from the point-source pair in the front of the room.

If the rear loudspeakers are not correctly delayed then the loudspeakers in the room will not combine correctly.

This room is too small for audio from the front of the room to be perceived as a distinct echo.

Applying a proper delay to the ceiling speakers can minimize the problem of localization confusion that occurs if the first arrival sound is coming from the overhead loudspeakers and not the front of the room.

Example #2: A Mid-Sized Contemporary House of Worship

Description:
This second example is a medium sized house of worship. The vaulted ceiling is high and the floor in the congregational seating area is covered with hard-industrial vinyl.

The RT60 is longer then the first example at approximately 1.5 seconds so reverberation is a problem in an empty room. The sources of audio are again ministers on a microphone and a worship band.

Control is via a 32-channel mixer. The speaker system is an array of three large boxes mounted as a central cluster high in the peak of the ceiling. A professional audio company did the installation and calibration of the audio system.

The quality of the audio in this church is much better than in the first example. An interesting question is: how good is “good enough”? When interviewed, members of this congregation can usually hear. Rarely is the audio painful to listen to so some say that the audio quality is fully acceptable.

This is a good time to reflect back on the example in the introduction where domed ceilings were held up as an icon of natural acoustic wonderfulness. Let’s examine each individual audio characteristic previously discussed and see how this audio system installation stacks up.

Problems

• Reverberance is not well controlled and is dependent on the configuration and occupancy of the room. Low-mid frequencies are a particular problem as the energy builds up and is never trapped or controlled.

• Clarity is fairly good and meets a minimum standard.

• Articulation is acceptable but not outstanding. The ALCONs (Articulation Loss of Consonants) rating of this room is fairly low but in the acceptable range. However, there is room for improvement.

• Listener envelopment is nonexistent and completely pales in comparison to the example of a domed ceiling.

• Again, as in the first example, an experienced sound person is required to run the mixer for any use of the audio system, as there is no automation in the audio system.

• There is good coverage of the congregation from the central cluster, but people sitting in the area where the coverage patterns between two of the speakers overlap experience uneven frequency response due to the comb filtering caused by the interference between these two speakers.

• Bass response is particularly poor. The poor bass response leads to the impression that the system lacks sufficient power.

Recommendations

A DSP is already in the system and can be used for additional equalization and other tasks. The same recommendation applies to add enough automation so that a simple service can be done without bringing in a sound person.

The speaker system may already be fully adequate.

The first temptation may be to add a subwoofer to add bass power, but after a quick survey it is probable that the buildup of mid-bass energy in this room makes the quality of the bass so poor that adding more bass will only make matters worse.

To fix the room, the ceiling and walls could be completely covered in bass absorptive panels, but this is not really practical so a compromise is to add bass traps to the corners of the room and the ridge of the ceiling.

If it is not possible to tame the room with traps, then narrow-band filtering techniques could be employed.

This is where the room is evaluated for the natural modes that build up energy in the room and these frequencies are notched out with a very narrow filter. A combination of some absorptive panels and narrow-band filters might be the best compromise.

There are regions (as shown in Figure 15) where the coverage from the individual speakers in the cluster interfere with each other rather than combine cooperatively. This interference is frequency-dependent.

The solution is to reduce the contribution of some of the speakers of those problem frequencies so that interference is minimized.

The system would then require re-calibration to complement the above changes. That should do it.

 
A the time of publication Michaël Rollins was a senior digital design engineer for Rane Corporation.

 
Download a copy of this article. (pdf)

Editors Note: This and other educational articles are available in the RaneNote Library, a subset of the Rane ProAudio Reference.

{extended}

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

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

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. I

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.

THE MIGRATION
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.

INNOCENTLY ENOUGH
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.

COMBINATION OF FORCES
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.

A LONG TIME
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.

{extended}

Tech Tip Of The Day: Word Clock Confusion

Is there any specific way my word clock should be run throughout my studio?

 
Q: I run a small recording studio, where I have a master mixdown deck that has a separate word clock input.

I usually go from my digital console through an outboard digital processor, and then on to my master mixdown deck.

Not terribly long ago, someone mentioned it would be a good idea to run a separate word clock cable to this deck for better performance.

Is this really important?

A: Interesting question! Arguably, it is of some importance to hook your master deck into the word clock, but the biggest benefit may come from better distribution of the clock itself.

There are a variety of subtleties that are beyond the scope of what can realistically be covered here. However, using separate word clock cables to connect your mixer to the processor, and then the processor to the recorder probably wouldn’t make much of a difference.

The word signal that’s included in the digital audio data will generally serve this function fine. However, running a separate cable from the mixer directly to the recorder and the Finalizer separately could make a significant difference.

There are so many other variables it’s hard to say for sure what your results will be, but the quality and stability of a word clock signal can (and usually is) degraded as it passes through multiple pieces.

In fact, an ideal setup for you would include a dedicated house sync generator combined with a distribution system (often the same device) that will deliver a high quality word clock signal directly to each digital device (mixer, recorders, etc.), without it having to pass through other devices along the way.

Not only will this help maximize your system’s ability to sound its best, but it can also help things be more stable..

As always, we welcome input from the PSW community and would love to know your feelings on word clock. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

{extended}

L-Acoustics Monitor Package Chosen By On Stage Audio

The new LA-RAK touring racks and coaxial 115XT HiQ wedges made their initial debut at Chicago’s popular Ravinia Festival.

On Stage Audio (OSA) recently made its inaugural L-Acoustics purchase with the addition of a new monitor package comprised of three LA8-equipped LA-RAK touring racks and 16 coaxial 115XT HiQ wedges.

OSA purchased the system at the request of Chicago’s Ravinia Festival, which used the gear for three months this summer in its 3,200-seat, open-air, covered pavilion.

Hosting the summer residency of the Chicago Symphony Orchestra since 1936, Ravinia also featured a diverse array of non-classical artists who used the monitors.

Following the festival’s summer season on September 7, the L-Acoustics racks and wedges will be available as a rental system package for tours and other productions.

“The decision to buy this system was initially driven by our client, but we’ve all come across these wedges on various tours and have always really liked them,” said OSA Senior Staff Engineer Carmen Educate.

“So when Ravinia told us they wanted to use 115XT HiQs in their Pavilion this summer, we jumped at the chance to add them to our rental inventory. I’ve come to love the clean SPL that the wedge delivers as well as its extremely linear response when boosting level.”

“It’s a very smooth and tight-sounding little speaker.”

Educate said that the festival’s management, crew and performing artists have all been extremely satisfied with the monitor package’s performance. “Everyone there loves the rig.”

““They’ve been more than happy with it, and so have we. Although this was officially OSA’s ‘maiden voyage’ with the brand, we really like the product and are hoping to move further into L-Acoustics’ larger systems.”

According to Ravinia Festival Master Audio Technician Sam Amodeo, “Our summer festival schedule is extremely full, so every second is critical.”

“The quality and fidelity of the 115XT HiQs and LA8s have been great and actually enabled us to save time on sound checks, so they’ve been a prized addition this year.”

“This is the first season for On Stage Audio at Ravinia and we could not be happier with the condition and performance of their audio package,” said Ravinia Festival Technical Director Mike Robinson.

“Having OSA as a vendor and L-Acoustics as a brand has pretty much removed all reliability concerns.”

Founded in 1904, Ravinia Festival is the oldest outdoor music festival in North America and attracts approximately 600,000 people to as many as 150 diverse performances each year.

Over the past century, the festival has hosted such luminaries as Louis Armstrong, Leonard Bernstein, Duke Ellington, Ella Fitzgerald, George Gershwin, Janis Joplin, Yo-Yo Ma, Luciano Pavarotti, Itzhak Perlman, Stephen Sondheim, Isaac Stern and Frank Zappa.

More information on the festival is available on their website.

 
L-Acoustics Website

{extended}

Tuesday, August 31, 2010

The Most Important Thing In Audio Is…? The Conundrum Of “Ears Versus Education”

For the best results in audio mixing, context is vital. But can it be taught?

I’ve been thinking quite a bit about the role musical education plays in audio mixing. There have been numerous threads about the subject in several on-line forums, and the responses seem evenly divided between “not needed but it doesn’t hurt” and “it’s actually a hindrance” and “it certainly helps.”

Because I earned a degree in music performance, I’m biased on the subject, with my opinion leaning toward the “it helps” camp. Still, I can’t help but wonder if it really does…

When evaluating the handiwork of mix engineers, there are plenty of guys and gals that indeed do not have formal musical training. An obvious example is Al Schmitt, who’s earned a stockpile of Grammy Awards for efforts with artists such as Frank Sinatra, Toto, Diana Krall and numerous others.

Even though he’s a studio engineer, I think his example can still be applied to sound reinforcement. One thing’s for sure – Mr. Schmitt has never been called “unmusical,” or at least I’ve never heard it said.

My hunch is although he doesn’t have “formal” musical training, he still has listening skills quite sensitive to musical aesthetics, an amazing sense not only for the technical but also for how all of the sounds relate to one another in context.

This leads us to a key point: for the best results in audio mixing, context is vital. But can it be taught?

MATTER OF STYLE
With any art form, there are those who specialize in a particular style and then those who seem to be able to transcend their particular era and become “timeless.” Relating this to audio, I’ve heard mix engineers who seem to meld their style of mixing to the music itself, while others try to force the music into their mixing style.

Back when I was touring as a mix engineer for the Airmen of Note (U.S. Air Force Jazz Band), I found it was important to spend time with the band in rehearsal to get a sense of the issues at hand: arrangement, internal balance within sections and between sections, and the general “feel” produced by the music.

In the process, I came to the conclusion that the drums, along with the bass, generate a certain rhythmic element that actually drove the way the horn players stayed “in the groove.” It was an actual physical thing, where the acoustic wave from the kick drum had an impact on the diaphragms of the horn players. Stand close enough to this type of group while they’re playing, and you can pick up this sensation.

So I set about trying to bring some of that feel to the audience while I mixed, but without making it too overpowering or “rock ‘n’ roll” – which I felt would not be representative of the big band style. The approach involved how I mic’d the drums (three mics – kick, and two overheads), use of EQ (not much, except to bring out certain things and make sure other elements didn’t become overbearing) and setting the drum levels relative to the rest of the mix (supporting the sound).

I felt that the result was a convincing live portrayal of the band, bringing out the dynamics and impact they worked so hard to do attain, but without too much power from the rhythm section. But did my music education help me attain this, or was it some innate musical sense that can’t be taught?

THE INNER VOICES
Another aspect of mixing, and it was clearly important in big band work, is the inner voices. No, I don’t mean the little voices in my head saying, “check out that woman in the third row.” Rather, I’m referring to the relationships of all the instruments between the bass and cymbals.

Any arrangement - rock, jazz, classical, or whatever - relies on specific voicings. I’m talking about the order of notes from the lowest to the highest within a chord. As a mixer, if you’re not aware of this, then you likely don’t realize that the third of a chord determines whether it’s major or minor, that the fifth along with the root make up the “frame” of the chord, and that everything above the fifth is harmonic embellishment but nevertheless important in terms of leading notes, harmony, and what kinds of scales might be used for melodic material.

And perhaps the mixer might miss (or not know) that inversions (chords where the root, third, etc. are stacked out of order) are extremely important to musical harmony, and thus are a critical element of a musical style like jazz. An example is the horn section for a swing band (think Brian Setzer’s Dirty Boogie), where if one of the horn mics is turned up too “hot,” then the wrong note in some chords may be emphasized. The difference might be subtle, but it may also throw a certain amount of “aural sand” into the musical experience for at least a portion of the audience. And let’s face it – it’s just not right.

But these are “rules of thumb” taught by the educational process. Another way to figure out “who’s playing what” might be to listen and think, without cluttering up the works with confusing terminology. In other words, how do you think it sounds?

THE NEW RESPONSE
“The most important tool in audio is… ?” I ask this question often when giving presentations. It used to be that the answer I wanted to hear was “our ears.” Recently, however, I’ve preferred the response of “our brains.”

Of course, good ears are a critical component in mixing, and without them, there wouldn’t be much of a purpose for audio systems. (Although I’m sure that marketing departments would find a way to put a spin on that!)

But my thinking began to change as I realized that without the brain, what the ears are telling us can’t be interpreted and no plan of action can be developed. In other words, we may hear a problem, but if we can’t produce a solution, then what’s the point?

For example, if there’s a buzz in the system, is it at 60 Hz? 120 Hz? 180 Hz? And if it’s indeed at 60 Hz, where to start in looking for a solution?

On the flip side, those without the sense to apply their knowledge in order to generate an aesthetically pleasing mix lead me to question the value of any understanding of things like gain structure and signal flow, let alone voicing and spatial relationships. In other words, it may be technically “right” but does it sound good?

Perhaps their mixes are “good enough,” and certainly any situation involving art and technology must by nature be a form of compromise. However, if you knew of a way to improve your mixes, wouldn’t you want to employ it?

EQUAL VALUE
My resolution to these conundrums has been to settle on the theory that both musical ears and musical education have relatively equal value, and therefore, for better mixes, the focus should be on both. My theory guidelines track along these lines:

- If considering attending an audio school, see if the curriculum includes courses in musical training (ear training, theory, etc.). Purely technical audio training can result in a set of skills, but musical training allows you to “speak the language” with musicians and within your own mind.

- Spend a lot of time listening to a wide variety of music, and try to determine the common elements between them as well as those things that distinguish between different styles. It’s also vital to listen to acoustic music as much as possible – if you don’t know what instruments sound like un-amplified, where is your frame of reference?

- Come to terms with your own mix style and types of music. There are even differences between punk music from New York and L.A., right? (I suppose I’m showing my age with that one.)

- If the music you’re mixing was developed before amplification (classical, big band jazz, etc.), understand the context, both musically and in terms of acoustics. For example, what types of rooms originally hosted these types of performances? In other words, why put major amounts of reverb on a baritone sax solo in a big band performance? It just doesn’t fit. Not only that, but the players and the audience will expect to hear it as it is supposed to sound.

The track record of many successful folks working as mixers in pro audio without a formal musical education makes a persuasive argument that such an education may be largely irrelevant towards enhancing mix skills. Perhaps their abilities and success are a matter of an innate, natural musical sense, along with great ears and a lot of real-world experience.

Yet it also begs the question: would they be even better at what they do with further learning? Aren’t we all usually better for having learned more?

Karl Winkler is director of business development for Lectrosonics and has worked in professional audio for more than 15 years.

{extended}

How to Record Violin (Or Any New Instrument)

Have an upcoming session but recording an instrument for the very first time? This tutorial has got you covered.

 
At various points throughout your recording career, you’ll have chances to record certain instruments for the first time.

Maybe you’ve never recorded an actual piano, or a drum kit.

Or maybe you’ll need to record something truly esoteric like an accordion, or a hammer dulcimer, or bagpipes.

Having never done it before, where do you start? I had the pleasure of recording a violinist in my studio last week.

I’ve recorded violin before, but never in my home studio. It’s a lot of fun to mic up a new instrument, but sometimes it’s hard to know where to begin.

You Don’t Have To Re-Invent The Wheel
Chances are someone else has recorded the instrument you’re about to record. Before your session, take a few minutes to find out as much as you can about recording that particular instrument.

Don’t make this a 4-hour research session, though. That’s how people end up researching for years and never actually finishing anything.

Ask a friend. Better still, check with your Twitter/Facebook friends. Post something in a forum, like PSW’s REP. All you’re looking for here is some quick feedback for where to start.

I’d also recommend simply doing a Google image search. You can learn a lot from pictures of recording sessions.

For instance, I just googled “violin recording session,” and in about 30 seconds of perusing the image results I had 2 new ideas for recording violin.

Don’t get me wrong, it’s a great idea to try new things. There are no rules, but just because everyone uses an SM57 on a guitar amp doesn’t mean you can’t use it, too.

Keep It Simple
For my violin session, I used a single mic (large-diaphragm condenser) through my Presonus Eureka preamp.

I set the mic up about two feet above the violin and about two feet in front of the violinist.

Could I have used 2 mics? Absolutely, however if the violinist moves around (most do), it would have caused all manner of weird phase issues.

Besides, the violin is a small instrument.

One microphone tends to capture the sound just fine. I didn’t mic the violin too closely, because violins don’t sound particularly wonderful when they’re close mic’d.

They tend to sound much better from a distance. That’s why I moved the mic back a few feet from the instrument itself. This is where having good acoustic treatment in your room really helps when you don’t want to close-mic something.

When the violinist was warming up, I moved the mic around just a little bit to find a spot that sounded good. I recorded a quick practice take and played it back.

It sounded quite good to me, so we jumped in and started recording. I didn’t waste unnecessary time trying out different techniques. Rather, I guessed at one setup, it worked, so we moved forward.

Some Quick Tips
If you’re recording an unfamiliar instrument, here are some things to keep in mind:

• If you’re using a directional mic, you can adjust how bass-heavy the signal is by adjusting the distance between the mic and the instrument. The closer the mic is, the more bass it will record (proximity effect).

• If the instrument is fairly large (piano, upright bass, orchestra), you’ll want to think about using multiple microphones. Keep in mind that the more mics you use, the more mindful you’ll want to be of phase issues. (See 3:1 Rule.)

• Use your ear first. Stand in front of the instrument and find the spot where it sounds best to your ear, then put the microphone there.

• If things aren’t sounding great, try using a different microphone. Changing from a condenser to a dynamic or ribbon can provide a whole new range of tonal options.

Have you recently faced the challenge of recording an unfamiliar instrument? Let me know in the comments below.

 
Joe Gilder is a Nashville based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.

{extended}

Church Soundguy: A Monitor Option That Kicks Butt

Once your praise band moves to in ears, do you have a plan for ensuring that everyone is truly hearing the best mix possible?

I am an audio engineer, a bass player, and a skeptic.

So, when I first saw the Guitammer Buttkicker system, I’m afraid it was the skeptic that spoke up sarcastically: “Oh boy. Another gimmick! Yet another electronic toy to pay for!”

But it was a cool idea, so I still went right over to look at it.

I sat down on the drum throne (why is it that drummers need a “throne” and everybody else gets “chair”?) and it was immediately evident why they’d named the product the “Buttkicker.”

They were playing some tracks through the system, and I was feeling it in my hindquarters.

Mostly, I was feeling the low frequencies: the bass guitar and the kick drum, as a drum throne and the flesh sitting upon it are not very efficient high frequency transducers.

However, being a bass player, those were the frequencies I was interested in anyway.

As, a bass player, there’s no such thing as too much kick drum.

If the band is going to be tight, then it starts with the bass and the kick drum being tight.

Since I am a “sound tech who plays bass” rather than a “gifted musician”, staying tight with the kick drum has been a slight challenge for me on more than one occasion.

A friend of mine was showing off the Buttkicker system, and he knew that I played bass, so he drew me to the bass player’s position on the stage and had me stand on a funny little platform.

It vibrated my feet, but they vibrated in time with the kick drum and the bass guitar: same principle, but we bass players take it standing up!

Aside: Since then, I’ve played bass sitting a wooden stool on the bass player’s platform and benefited significantly from this very Buttkicker effect . . .

“Now listen to this!” my friend said, and he pulled me off of the platform, clamped some good headphones on me and pressed play on a multitrack recorder.

It was through an Aviom personal monitor rig, so I could adjust the various instruments individually in my ears; I’ve always been a fan of personal monitors.

I was listening to a worship song that I knew, one I’d played with my church’s band many times, and it sounded pretty good.

I listened for a while, adjusted the bass and the kick drum so I could hear them better, then we talked about it.

Then he got this funny grin on his face, and pushed me back on the platform, said, “Now listen again!” and pressed play.

I put the headphones and listened, and immediately tore the headphones off and accused him of playing a different track.

“Nope. Same track.”

I put the headphones back on: the bass guitar and the drum were clearly out of sync; the difference was night-and-day! I stepped off the platform: it was hard to hear the problem.

I stepped back on, it was obvious: the bass player was a little bit off from the rest of the band, and without being able to “feel” the low frequencies, I could hardly tell.

As an audio engineer, I can’t tell you how many times I’ve asked the bass player to turn his amp down, and been met with “But I need to feel the bass!” And I have to admit, when I’ve been the bass player, the audio engineer has sometimes asked me to turn my amp down, and I’ve wanted to say the same to him.

Low frequencies – the kind of lows that are made by a kick drum and a bass guitar – are large waves: a low E on the bass is 27’ long; the low B on a 5-string bass is 36’ long.

A bass drum’s fundamental note can be 40’ long. These frequencies are indeed difficult to hear up close, and so the bass guitarist turns up his amplifier and the drummer plays extra loud so that they can feel the low frequencies and stay in synch with each other, and with the rest of the band.

By the time I stepped off of that funny little Buttkicker platform, I was completely convinced: a Buttkicker can be a wonderful tool for bass players and for drummers. It can allow them to play tightly in synch, creating a “groove” that the rest of the worship band can ride on.

It can allow them to play with less volume, so the rest of the band doesn’t need their monitors cranked up so loud.

It truly does make all the difference in the world for bass players and drummers when the band moves to in-ear monitors: there isn’t a headphone in the world that can replace a bass guitar cabinet, or reproduce the kick in the chest from standing or sitting near the bass drum.

After listening to the tracks with the bass and drums slightly off, my friend played another track. This time, the bass and drums were tight, completely spot on.

Standing on the Buttkicker made it obvious what the difference was, but even when I was not on the platform, the difference was significant: the whole band, the whole mix sounded “tighter” when the bass and drums finally got together. Non-musicians that I quizzed agreed that it sounded better, however, they couldn’t tell why.

Certainly, adding Buttkickers to a monitor rig would allow the bass player to turn his amp down (or remove it from the stage) and the drummer to play lighter.

And, as strange as this sounds, I have come to expect that if the bass player and drummer get Buttkickers, then the whole band will sound better.

Check out more from David McLain at the Church Soundguy blog.

{extended}

Tech Tip Of The Day: Click Bleed

Is there any way to satisfy a musicians monitoring needs while eliminating click track bleed?

 
Q: So, first of all, a big thank you to PSW for all the monitoring articles lately.

They’ve really helped me deal with the monitoring needs of some very demanding musicians.

However, I do have a problem I’m hoping you can help me with.

Despite my best efforts, the vocalist still needs their click track turned up ridiculously loud, which is constantly causing bleed.

I’ve tried riding the track, to the point where I’m not mixing the song but rather their click track! Is there anything I can do?

A: There most certainly is hope, and I’m glad we’ve been able to help you so much thus far!

To almost instantly reduce the amount of bleed you’re experiencing from that obnoxiously loud click track, there’s only one step.

Change the click sound!

How?

Well there’s more than one way to accomplish this, however, depending on the plugin you’re using to generate the click, it may be as simple as choosing a new sample. In lieu of the traditional (and harsh metallic) metronome sound, try something like snare or woodblock.

Make sure to give the musician a bit of input on the final sound, but any of these options should be equally workable for them, and would cause near the bleed in the microphone.

If your plug-in is more basic (mine is), you still have a way to solve the issue. Simply run the click through an EQ and pull out the high frequencies. The duller, EQ’d click won’t penetrate the phones as well and, again, won’t be as audible in any nearby open microphones.

As always, we welcome input from the PSW community and would love to know how you handle click track bleed. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

{extended}

Monday, August 30, 2010

Tour De Absurd: Unbound By The Fundamental Rules Of Reality

Everybody's dealt with horrible vendors from time to time, and Sully's got some tales from the road to which we all can relate.

I was just bitten by a dog.

Truthfully, not 20 minutes ago when I went to pick up a piece of gear at somebody’s house.

When I pulled into the driveway, an unholy spawn of a late night dalliance between Benji and the Geico gecko waddled over to me, growled, then bit me on the f***ing ankle.

I screamed like a five year old, which somehow triggered the garage door to open and spew a teenage girl carrying the gear I was there for.

“This yours?” she piped. “Your dog just bit me on the f***ing ankle,” I squeaked. “”Really? Sorry…” She froze with a look on her face that indicated she was now invisible and I should leave wondering where’d that girl go?

Needing more satisfaction I called the owner of the house.

“Hello?”

“Your dog just bit me on the f***ing ankle.”

“That dog’s 13 years old, he’s never bitten anyone.”

“Oh. Cool. Never mind then.”

“You sure?”

“Hang on a tic, let me make sure my portable morphine drip isn’t on high. Nope, machine’s good… the f***er definitely bit me”

The vicious dog attack left me sulking about the hound’s total lack of fear and respect for me. Then I got mad at myself for sulking about not being feared by an arthritic Chihuahua. Skillfully, I managed to cram in a 30-minute session of bi-polar self-loathing and admonishment in the time it took to drive from the scene of the assault to our bus.

A small, clarifying sidebar: Against my will, I’ve been educated about the straightforward hierarchy that exists in dog-dom by a friend who bites his new puppies. He favors the short snout breeds like Rottweillers. In order to school a puppy about who’s the “Alpha” in the hutch, he rolls the thing on it’s back and bites it on the neck.

Having come upon the disturbing scene of him hunched over, clamped to a dog and growling… I’ve gotta say it’s not something that prompts a salient comment right away.

Judging by the expression on the puppy’s face, I’m pretty sure he had some questions too. The relevant point is that once one animal bites another, dominance is established and then both go about their business.

As usual, I digress. Let’s reel this back in and connect pocket poodles with artfully deployed line arrays. The truth of the matter is a canine chomping me on the ankle merely represents a random act of nature, nothing more. It just happened to occur during a personal period of significant disillusionment with the human race.

It had suddenly occurred to me, as I stared down at the dog sticking out of my jeans, that this was a fitting coda to the four-week tour de absurd that I and the rest of my crew had just endured. During the preceding month, 90 percent of the production vendors we had met had attempted to convince us they alone were not bound by the fundamental rules of reality.

To prove this point, they had taken our advance phone calls, listened carefully to our requests, sagely reassured us all would be well… then rolled us over and tried to bite us on the neck when we showed up. Same deal as the dog. They looked us up and down and figured they could take us.

Act 1
Me: “Hey, how wide is this box?”

PA prestidigitator: “205 degrees for the long throw, 365 degrees for the downfill.”

Me: (Knowing it’s general admission) “OK.”

 
Act 2
The setting: a large field with bands of disgruntled raisins milling arrogantly about.

A2: “We’re ready, my lord. We are prepared for you to communicate with the magic box and give us the array angles for the sound system.”

Steak sauce: “I have spoken with the machine. It gives no advice today. You must have done something to anger it. Go now, butcher the factory program and burn the fatted DSP as an offering. Leave me.”

A2: “My liege, the troubadours will be upon us soon… Can you offer no wisdom for us to assuage their FOH knight?”

Steak Sauce: “Tell him…tell him… the sound will emerge crooked if you angle the speakers. Tell him flat… Yes, flat is best. Threaten to rub petroleum jelly on him and burn him as a witch if he questions you.”

A2: “You are indeed the wisest in the land.”

 
Act 3
Production manager motioning to four speakers hanging from swing chain flown with the aid of two winches off the front of a quad runner.

PM: “What da hell is that?”

Local vendor: “EV X-Array.”

PM: “No it’s not.”

Local vendor: “Yes it is.”

PM: “No it’s not.”

Local vendor: “Yes it is.”

PM: “No it’s not.”

Local vendor: “O.K., no it’s not. I bought an X-Array box and copied it.”

PM: “You mean EV X-Line. You copied X-Line.”

Local vendor: “Yeah, the big EV box. X-Array-Line.”

PM: “O.K., just so we’re clear…you pirated something from EV and call it X-Array.”

Local vendor: “Yeah. Sounds great too. Wanna see our V-Disc wedges?”

Act 4
The three principal characters enter upstage center and proceed downstage in slow motion, their movements reminiscent of Apollo astronauts bravely approaching an ill-fated capsule.

Bonded by an invisible energy, their gaze begins tracking the seventy-five degree seating angle until at last their eyes settle upon the top seat, 600 feet aloft. One holds a laser range finder and whistles quietly at the data it yields.

Their attention is suddenly diverted to the single horizontal row of two EAW KF750’s stacked neatly on the stage deck. A small man rapidly approaches the group.

He is equipped with a large black belt dubiously supporting a bricklike Tandy walkie-talkie with a solid three-foot antenna fully extended.

The effect is not unlike a remotely controlled Hobbit. A roll of gray tape used to seal air conditioning vents dangles from his meaty wrist, and he is thrusting an irate digit at the tiny speaker array.

Small Man With Big Belt: “I don’t want to hear it! Them speakers cover front row to top row perfect. They’re 70 degrees up and down so we don’t even need to tilt them. Sounds exactly up there like it do down here. I don’t want any of your smart-alecky talk about math. We done it this way for 10 years and it sounds great. Now, welcome and go away, I mix the opener tonight and I gotta make sure they’re happy”.

 
Act 5
A man stands beaten, his feet loosely clutching the prefabricated stage. His attention is captivated by the scene unfolding before his weary blue intelligent eyes…Men of ill-advised employment are hoisting a large-format console by attaching a 1/4-ton drape motor to its top-riveted session handles.

They stand under it, marveling at the graceful way it swings in the cool breeze. Our hero calculates that when the first handle lets go, the desk will swing low, hijack a stagehand at it’s nadir and force him to ride it bareback halfway to the rafters.

As the console reaches it’s apex and the second handle shears away, the desk will immediately divest itself of it’s passenger and enter a vertical spin, 25 feet off the ground, shortly proving wrong the load-out adage, “gravity is your friend”.

Quickly, without remorse, the sad man dispatches an intern to the balcony with a bin of economy popcorn and two video cameras. Word must reach the outside world of the transgressions that have transpired here…

 
Act 6
Me: “What version of the prediction software are you using?”

Them: “Ashly crossovers. They’re out front.”

I own a cat that hides behind the drapes when in trouble. It sits perfectly still, avoiding all eye contact, staring straight ahead looking like a paisley tumor respirating below the front window.

She is so convinced of her sudden undetectability that I have no choice but to accept the fact that the curtains have spontaneously evolved a tail and I should look elsewhere for her.

I marvel at her ability to gaze directly into the face of truth and maintain plausible deniability. Like the vicious miniature wolfhound noted earlier, the cat has eyed me up and come to the conclusion that she’s got my number.

I’d start dutifully working on a complex about my lack of respectability within the various animal phyla, but I know from experience, it’s not just me.

Many of the band guys I run into step off of the bus in the morning with dingoes latched to their ankles. They all have stories that somehow involve PA and lighting vendors avoiding eye contact and hiding behind backdrops with only their five D-MAG lights sticking out.

Sometimes I’ll look into their eyes, pat their dogs and smile with them, offering these words of solace: “get your sun block out boys, we’re goin’ to Hell.”

 
Finale
Me: “Two horns are popping red and two are green. Which is correct?”

System provider: “Which is better?”

 
Sully is a veteran live sound engineer and really has no clever off-hand remarks for this space at this time.

{extended}

On Three, You Lift: Methods Behind The “Check One, Two” Madness

Sometimes there actually is a little method beyond the madness we all endure...

“Check one, two… one, two…” As a soundperson, I’ve had to repeat this phrase ad nauseam - at times I even hear it in my dreams.

Everyone wants to know: “Why can’t you count to three?” To which the punch line follows: “Because on three, you have to lift.” Ugh, like I haven’t heard that a thousand times before. (In addition to “when will this be over” and “is that really necessary?”)

Years of enduring these tired refrains finally prompted me to give the matter some serious thought. What is it about those little words - “check one, two” - that help in quickly setting up and tuning a system and room? I soon re-discovered a few things that had become instinctive and subconscious over the years.

After getting a system up and running, I make sure to flatten the channel and house equalizers, but this then begs the question: What’s the point of having a 1/3-octave EQ for some of these smaller shows anyway?

It’s a question I’ve actually had to answer a few times, and sometimes is followed by quite a fight to get even one EQ unit actually included for some of my systems. It’s amazing how many folks ignore the importance of EQ.

Yet two of the biggest factors in my area of specialty, which is corporate sound reproduction, are feedback reduction and tonal balance, both of which can be optimized with a decent graphic EQ. Therefore, it’s invaluable.

Anyway, continuing the process. After returning the EQ and console to “normal” positions, I’ll grab a microphone to begin testing. For this portion, I employ the same mic model in use on the lectern, and stand-mount it to keep my hands free for adjustments.

Then, “check.” The first thing I look for are any frequency imbalances in the system. “Check” can cover anything from 2 kHz down to 630 Hz, depending on the voice. The “ch” sound can be hit harder for the upper mids, and separating the “eh” sound can help in identifying problems in the mids. Try dipping each one of these bands while saying the word in order to filter out any harshness without sacrificing clarity.

LIKE A WAH-WAH
Once this range is addressed I move on to “one,” altering pronunciation - “wahn, wawn, won” - slowly opening my mouth to shape these sounds. Just like a wah-wah pedal for a guitar, or a filter sweep on a synth, the idea is to slowly sweep through a range of frequencies to explore the vowel shapes.

These sounds usually cover anywhere from about 1 kHz down to 250 Hz, again depending on the voice. Look for a boxy, wonky sound with the “wah,” pushing the mic more toward the nose for the 800 Hz range, and then more toward the mouth for the 400 Hz to 500Hz range.

Finally, the word “two” goes through a few variations - “tieuh, tyou, too.” Hitting the “t” extra hard can give a feel for the highs around 4 kHz, while dropping the “ooh” sound smoothly through the vocal range can provide a look at the lows. Everything from 4 kHz and up can be explored with a sharp hissing “s” sound, made by tighten up facial muscles and raising eyebrows to get into the upper high ranges.

In working this process, your own voice is going to be the sound measuring tool that you should be most familiar with. Upon learning where the tones of these three little words fall in your own vocal range (everyone is going to be a little different), tuning can be fairly quick and easy. As long as “the peanut gallery” stays quiet!

After getting a decent tonal balance, it’s time to start searching for feedback. Many use the technique of turning up the level until the point of first feedback, finding that frequency, and then dipping the graphic slider. Generally, this is repeated about three times.

However, my own preference is to go “fishing for feedback.” I’ll bring the system up to the first ringing tone and back it off slightly, then go through the entire 1/3 octave, tiny fader by tiny fader, looking for feedback on each band.

Some won’t ring at all so I’ll leave them flat. The others will range from immediate ringing to nothing at all until full boost. The amount of cut I use is inversely proportional to the amount of boost needed to achieve feedback.

There’s no exact formula to it - if 2 kHz starts ringing with only a nudge of boost, I’ll take a pretty good chunk out of that range. If nothing happens until the fader is nearly completely boosted, I’ll only dip it a slight amount. Remember to continue checking “one, two, one, two” throughout this process because feedback normally needs some initial sound to occur.

The bottom line is how the system sounds with a human voice. If the feedback is obliterated but the system sounds like a telephone line, it’s probably time to sacrifice some volume to get a good tonal balance. And once you find a tone that’s ringing, don’t be embarrassed to hum or sing that note until you really lock in on it. Don’t turn this into karaoke or “Showtime At The Apollo,” but singing the pitch of a ringing tone can make it a lot easier to find when fishing.

BRING IT TOGETHER
After I’m satisfied with the overall sound of the rig, I bring in sources that will be used onstage and slowly add them in together. Again, in my world, this can mean a lectern mic, a series of table mics for a panel discussion, wireless lavaliers as well as wireless handheld mics for “Q & A” from the house.

I adjust each individual channel for feedback and tone using the same fishing techniques used on the 1/3-octave EQ. Bring the fader up to the “0” position, and slowly bring up the input or trim pot at the top of the fader until there’s a good level.

For a simple panel discussion, in general I’ve been able to set the mics at 75 percent open and then ride them up and down as the individuals make comments on mic. Having every mic 100 percent open can add noise to a system and create additional feedback problems.

If faced with wireless lavalier mics, there may be need for some pretty severe adjustments to the 1/3 octave curve. Make notes of these adjustments and compensate for them on each of the other non-lavalier sources, with individual channel EQ to make up the difference.

For example, if you’ve just rung out the room and start working with a series of lavs that need a huge dip in the 400 Hz range, compensate on the other channels by bringing that range up a bit. If there are wireless handheld mics in the house, have someone stand out front and “bless” the main loudspeakers by making a papal, sweeping motion in front of them with the mic. The last thing anyone wants is for someone to walk in front of those loudspeakers with a hot mic that hasn’t been tested.

See, sometimes there actually is a little method beyond the madness we all endure. After a little thought and experimentation, it can be as easy as one, two…

You didn’t think I’d actually say three?

Paul LaPlaca has been working with sound for more than 25 years and heads up Stentor Productions in New York, specializing in the corporate audio market.

{extended}

Top Ways To Help Musicians Hear Themselves

The key to a great mix is getting your musicians working together creatively, which starts with a great monitor mix.

 
I’m often heard saying that the recording engineer’s job is to create an environment conducive to musical creativity and then capture that creativity.

Headphones are usually the only way that a musician will be able to hear themselves and (more importantly) how what they are playing works with the rest of the band.

Every musician will (eventually) ask to hear themselves much louder than everyone else. This makes sense as it will allow them to play the nuances of their instrument.

However, if they are only listening to themselves (or the click track) and not the everything else then what they play may be wonderful by itself but terrible within the mix. They may even compromise the art of their own playing as a result of a poor headphone mix.

For Example:
• Guitar players who hear themselves too loudly will not “bear down” with the pick as much as they may need to.
• Piano players who hear themselves too quietly may not play with the full dynamic range of the piano if they cannot hear themselves play softly.
• And finally, any musician that cannot hear the full rhythm will cause a combined pushing and pulling of all the instruments, and no one will be together or “in the pocket”, even if they are overdubbing alone.

Remember, you must make the musicians feel like they are playing together in a room without headphones (in fact I prefer to record bands that way). They have to be able to hear and feel each other clearly.

Sometimes you may have the luxury of multiple headphone feeds, which will allow you to tailor different mixes for the players that require them. Even given the advanced personal mixer technology available today, always be wary of letting musicians mix their own headphones completely by themselves, as they will tend to want to hear only themselves.

A Few Pointers:

1. No matter how loud the drums may be in the room, everyone needs some kick, snare, hat and other drum microphones. The timing and feel of the drum mics will sound different from the drum sound in the room.

2. Panning can be your friend. Sometimes moving some instruments just slightly off center will make it easier to the players to hear themselves without increasing volume or resorting to making the moniutor mix a solo mix for certain indivitduals.

3. You can always change the sound musicians hear in their headphones without compromising the sounds you record.

Once, I was recording a large horn section that was used to a compressed edgy sound. I wanted to go for something full, so I recorded them using a combination of ribbon and condenser mics going flat from Neve mic pre’s straight into the tape machine.

The section was not happy and complained that the sound was not what they were used to. I did not want to lose the fullness the mics were giving me, so I EQ’ed and compressed the monitor channels coming off the tape machine. Suddenly they were all happy and played well.

When I mixed, I was able to use all of the sounds with absolutels no EQ or compression (until those effects were called for) and was very pleased with the results. If I had changed the sound I was capturing to match what the musicians were used to hearing in their headphones, the final sound of the section would have suffered.

4. Make sure the musicians hear enough of the band and even the beat that they can perform to the song rather than just lay down their parts. Musicians will (and should) be concerned with their performances, but do not let them lose sight of the fact that they are playing within a song along with other musicians.

If they do not hear the others they will not be able to interact with them, even if it is only on a subconscious level.

5. Some drummers will ask for loud click tracks in their headphones. If you have only one headphone feed and the drummer needs to share the cue with other performers, it may be tricky for you to keep everyone happy. You may need to ride the click. And, speaking of riding the click....

6. Be prepared to ride the click track down in softer sections of a song, especially at the end.

There is nothing worse than trying to mix the very end of a song and having to fade out too quickly to keep the click from the drummer’s headphones from being heard.

Bruce A. Miller is an acclaimed recording engineer who operates an independent recording studio and the BAM Audio School website.

{extended}

Worship Basics: Quick Tips For Maximizing Your Monitoring

Excerpted from recommendations to a client following a worship visit, these are critical aspects of monitoring often overlooked by the worship team.

Often times, it’s only through the critique of others that we are able to see the deficiencies in our systems, which is why I’m often called to visit clients services “in action.”

In the hopes that you may find some of follows below is an excerpt from the recommendations I made to a church client following a visit to their Sunday morning service.

The most prevalent issue throughout the soundcheck was that while the band sounded great, it took them a great deal of time to get the monitors right.

My Advice
1: There is always this tension that exists between tech and talent about stage volume. Thankfully with the entire band (except, well, we’ll call him Jimmy, who was also singing) using “ears” the stage volume was really at a minimum.

2: Structured monitor checking. When using ears it is very difficult for the engineer to know what the musicians are hearing.

My suggestion would be to have the engineer (or someone who can assist him) ask each musician what they want in their mix before the rehearsal starts. 

After the engineer has roughed in those levels on the soundboard the band should play through one song (unless it is a total disaster, don’t stop).  After the song each musician in an orderly fashion should give the engineer direction as to how they would like their mix changed. 

The band should then play through two-three more songs before the musicians can make any additional requests.  This forces the musicians to be precise in their monitor requests and also gives the engineer an opportunity to work on the house mix. 

After the two-three songs, each musician can once again in an orderly fashion ask for adjustments in their monitor mix.  Once this adjustment has been made, the musicians will have to live with the mix allowing the engineer to forget about monitors and only be concerned with the house mix. 

Note: The engineer should not adjust the master gain on any channel (except for an emergency) after the completion of the first song. 
 
All adjustments need to be made using the faders (for house sound) and the aux sends (for monitors). Assuming a pre-fade auxiliary send the levels of the monitors will then not change when the channel faders are moved.

3: Shield or no shield?  As you may or may not be aware, I’m not a big fan of drum shields (yes, sometimes they are necessary) as it has been my experience that when the shield is removed the drummer begins to play with more finesse and stage levels begin to decrease.

In a room the size of many worship spaces (including this one) I would say that a shield is not necessary.  In the drum mics, I find that around 90% of what is picked when using a cage is cymbals. 

I didn’t have much of an opportunity to look at mic placement, and there’s often some room for improvement, however overall I would say take the shield away.

While these recommendations were made after hearing a specific clients service, they’re the same recommendations I find myself giving more often that one would imagine.

So, take note, and remember that most of what an engineer does and thus how he is perceived comes from his attitude toward the musicians.

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

{extended}

Tech Tip Of The Day: Different Types Of Shielding

Is a cable that's shielded one specific way any better than another?
 
Q: I’ve been working with audio for quite a while now, and recently, I’ve been repairing a lot of cable.

One thing I’ve noticed is that there seem to be lots of different ways to shield a cable.

Is any one way better than another? Does it even matter, or is it just the choice of the manufacturer.

A: A very interesting question!

Each cable is always shielded in a way that provides the maximum amount of RF-shielding possible given the circumstances in which it will be operating.

What does that mean? Well, some shields are super-flexible and good for in the field, while others not so, and better for installation.

The first step to answering this question is understanding the different types of shielding. Let’s get started!

Braided Shield: A cable shield that is applied by braiding bunches of copper strands called picks around the insulated, electrostatically shielded center conductor. The braided shield offers a number of advantages.

Its coverage can be varied from less than 50% to nearly 97% by changing the angle, the number of picks and the rate at which they are applied. It is very consistent in its coverage, and remains so as the cable is flexed and bent, unlike Serve Shields.

This can be crucial in shielding the signal from RFI, where there are very short wavelengths that can enter very small “holes” in the shield. This RF-shielding superiority is further enhanced by very low inductance, causing the braid to present a low transfer impedance to high frequencies.

This is very important when the shield is supposed to be conducting interference harmlessly to ground.

Drawbacks of the braid shield include restricted flexibility, high manufacturing costs because of the relatively slow speed at which the shield-braiding machinery works, and the laborious “picking and pigtailing” operations required to solder them to connectors or circuit parts.

Serve Shield: Also known as a Spiral-Wrapped shield, is applied by wrapping a flat layer of copper strands around the center in a single direction (either clockwise or counter-clockwise).

The serve shield is very flexible, providing very little restriction to the “bendability” of the cable. Although its tensile strength is much less than that of braided shields, the serve’s superior flexibility often makes it more reliable in “real-world” instrument applications.

Tightly braided shields can be literally shredded by being kinked and pulled, as often happens in performance situations, while a spiral-wrapped serve shield will simply stretch without breaking down. Of course, such treatment opens up gaps in the shield that can allow interference to enter.

The inductance of the serve shield is also a liability when RFI is a problem; because it literally is a coil of wire, it has a transfer impedance that rises with frequency and is not as effective in shunting interference to ground as a braid.

The serve shield is most effective at frequencies below 100 kHz. From a cost viewpoint, the serve requires less copper, is much faster and hence cheaper to manufacture, and is quicker and easier to terminate than a braided shield. It also allows a smaller overall cable diameter, as it is only composed of a single layer of very small (typically 36 AWG) strands. These characteristics make copper serve a very common choice for audio cables.

Foil Shield: A type of cable shield composed of a thin layer of mylar-backed aluminum foil in contact with a copper drain wire used to terminate it.

The foil shield/drain wire combination is very cheap, but it severely limits flexibility and indeed breaks down under repeated flexing.

It’s the only commonly available shield type that offers true 100% coverage, however, this advantage is somewhat compromised by its high transfer impedance (aluminum being a poorer conductor of electricity than copper), especially at low frequencies.

It does allow for very small cables and is often a preferred shield in permanent installations where space is much more of a concern than durability.

Conclusion: Now you have an understanding as to what specific uses each type of shield is intended, so which one is “better?”

The answer, as you’ve no doubt realized, is that it depends completely upon the needs of your specific project. Your best bet is to evaluate the needs of your project, and make a decision from there.

As always, we welcome input from the PSW community and would love to know your thoughts on shielding. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

{extended}