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Tuesday, February 07, 2012
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.
In this article I’ll explain how I use mid-side (MS) processing on stereo sources for practical or creative effects.
Mid-Side?
Two channels of audio can be combined in a way that gives us control over what is the same in each signal, the middle, and what is different, the sides.
The middle is where the kick drum, snare, bass, vocals and a lot of other instruments are, the sides have any hard-panned instruments and spatial effects like reverb.
It can be pretty interesting to listen to music like this, there can be a lot hidden in the side channel.
MS is also a stereo microphone technique using a cardioid microphone facing the source and a bidirectional mic turned 90 degrees away just picking up ambience.
In this situation the two signals would need to be decoded into stereo. The side mic signal is duplicated, polarity inverted and the two side signals are then panned hard left and right. This is not a true stereo mic technique but can sound very nice. The balance of mid and side signals can be adjusted as needed by changing the level of the three tracks.
You can manually encode and decode stereo files to MS and use mono plugins to process mid or side individually. A lot more plugins have an MS mode now. Many of the modules in the T-Racks suite allow mid side processing, as does Ozone, a few compressors and equalizers and a distortion also come to mind.
You can do this for subtle or crazy effects, its a fun way to experiment with plugins and get some unique sounds.
Loud & Wide
For a recent mastering job I used a Fairchild compressor plugin in MS mode (Lat/Vert) to compress the middle and increase the level of the sides. I did this in parallel so I could blend the effect in easily. I was also using this to get a lot of extra loudness. You can call this parallel MS compression.
Compare -
The master without parallel MS compression: listen
No More Messy Verb
Someone asked ma about clearing up the middle of a mix when using a lot of reverb. Using MS compression on the reverb return can work well. Compress the middle more than the sides and increase the side volume if you want more width.
Here is an example of that on some drums - Steven Slate playing in KONTAKT. The whole kit is sent into Valhalla Room. With the Fairchild after the reverb I’m lowering the middle by 2 dB and raising the sides by 2 dB.
Here is this effect with lots of reverb on the drums: listen
And now with MS compression on just the reverb bus: listen
There is NO compression on the drums themselves, I’m only compressing the reverb return and widening it.
Wacky Effects
Here is an example of what you can do with a stereo loop and any plug-in. This is a little more complicated, and only works if there are hard panned sounds.
The loop I started with had a hi-hat that wasn’t panned very hard - I copied it to a new track, filtered out all the lows, boosted some highs and then panned it hard left. Then I recorded the combined original and panned track to a new file.
Now that I had something on the sides I could mess around with MS processing.
The first thing you have to do is convert left-right to mid and side. I use the free +matrix MS decoder from SoundHack.com. After that I used a delay plugin to add some filtered echoes just to the middle by disabling the right side input.
In the next insert I used a distortion on just the right side. This brought out a lot more of the reverb than was heard in the original loop. Lastly, second MS decoder was used to bring it back to stereo.
SoundHack + matrix MS encoder/decoder.
Here is how the loop sounds now with delay in the middle and distortion on the sides: listen
Pretty cool right!? I hope you have found these tricks useful.
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.
I’m in the process of helping one of my churches transition from an analog mixer to a digital mixer.
They were in need of more channels than their Allen & Heath 16-channel MixWiz with some outboard gear (front of house EQ, couple of compressors, effects unit) could provide.
Based on the maximum number of channels that they anticipated needing over the next five years, I recommended the PreSonus StudioLive 24.4, one of the least expensive 24-channel digital mixers on the market.
The church has two audio volunteers that are pretty much average in their knowledge of sound and sound systems so this would be a typical transition for a lot of churches in the 100-400 person attendance range. Volunteers selected more for their willingness to serve than their knowledge of audio. I know that nothing has been touched with the front of house EQ, compressors and FX since I helped them set it up about a year ago.
Some things that you need to consider in this transition is how uncomfortable the volunteers are going to be until they make the paradigm switch from the analog WYSIWYG (what you see is what you get) to the digital layers.
Depending on the digital board, layers control everything from different grouping of faders (1-8, 9-16, etc) to control over the aux sends, FX, etc. Outboard gear usually goes away and everything is now handled with the digital mixer. It’s a big transition and you shouldn’t minimize it, but treat it with care and planning and the transition will go smoothly.
Getting Started
What I recommend is that the digital mixer not be put into service immediately but be brought into a two-to-four-week training duty cycle. It requires some mics and cables as well as a couple of speakers for monitors and front of house stand-ins. If you have instruments that you can plug in that helps as well. Keep the existing analog system going as the production system until everyone has been trained and is comfortable with the digital board.
Before you start with the digital mixer, make sure everyone has reviewed the user manual. A digital board is a computer with knobs and faders and is significantly more complex than an analog mixer. While they are pretty robust, you can still mess them up and repairs can be costly.
An Investment of Protection
One thing to invest in if you haven’t is a top-line power conditioner like those from Furman. I also recommend a computer UPS (battery backup) from a company like APC or Tripp Lite. Get a decent capacity one. The reason is that because a digital mixer is a computer, when power is interrupted you can’t just switch it back on like an analog mixer. You need to boot it up and, depending on the mixer, that could take anywhere from a minute to several minutes.
Having a UPS unit, the mixer will stay powered on, so even if the rest of the system is knocked offline by the power interruption, when the power comes back on, the mixer will still be up.
Unboxing The Mixer
Once you get the mixer unboxed, check for any damage. If everything looks good bring all faders down to minimum and turn on the mixer. I like to let the mixer “burn in” for about four hours with nothing going on or plugged in just to let all the electronics warm up to full operating temperature. This will check to ensure that nothing is shorting out. Be aware of any burning electrical smell or smoke. If you detect either one shut the mixer down immediately and unplug it. Contact the vendor.
Preparing For Training
The StudioLive is close to an analog board in that all the channel faders are on one surface as opposed to layers. This makes the transition somewhat easier. All effects, aux send levels are controlled through the center “Fat Channel.” That will be where most of the confusion is going to come in so be prepared to spend a lot of time going through this area.
The StudioLive is set up pretty easy so I was able to figure 85% of the board out without looking at the manual. There are also a ton of video tutorials on the PreSonus site and YouTube that can help with anything to do with the board. But for volunteer sound techs it will be a bit of a challenge.
Building A Mini-System
Hook up a mic to channel 1 on the mixer and hook up a speaker to aux send 1 and to front of house. This will be the basic training setup.
Once you get it hooked up, bring up the gain to an appropriate level. A digital board is less forgiving about exceeding the 0 level than an analog board before going into clipping so run the level less than needed for training until you get comfortable with the way the board handles signals.
Don’t worry about EQ settings or FX yet. All you want to do is to learn the signal flow from the channel to the aux send and FOH.
Once you’ve figured out how to adjust the aux send levels for the channel and you can adjust FOH level you’ve gotten over the initial hump.
Using EQ
The next thing you’ll want to learn is how to adjust EQ’s for each channel. Depending on the digital mixer you’ll either have a screen that will have a parametric equalizer, or in the case of the StudioLive, you’ll have the knob adjustments for high, high mid, low mid and low bands. As with all digital mixers you are able to set the frequency points for all these bands as well as the Q, which is the width of the frequency adjustment. This is a lot more adjustability than what an analog mixer has and is worth spending some time practicing.
After the channel EQs get figured out you’ll want to adjust the front of house EQ. On the StudioLive it’s set the same way that the individual channel EQs are set. One nice advantage about digital mixers is that most of them have a library of preset EQs that you can start with. The StudioLive has built in a nice set of professional quality EQ presets that are good enough to leave alone and assign to each channel.
The other nice feature of digital boards is the ability to save all your settings to a scene. So you are able to set up multiple scenes for different worship teams or different instruments and recall them just by dialing up the scene and pressing the load button. So no more needing to reserve channels based on who’s playing that day.
Enter Effects
The power of digital mixers means that you can assign FX to each and every channel, both to auxes and to front of house, so you’ve got a lot of flexibility. Just remember that just because you can doesn’t mean you should. Less is more, at least in the beginning. Some boards give you more FX capabilities than others. The StudioLive offers two channels of FX, others more.
Multi-track Recording
Another advantage that digital mixers have is that they usually provide some form of multi-track recording capability. In the case of the StudioLive, it’s provided by a FireWire port into the provided Studio One software. This means you can record each channel separately into your computer, as long as it has a Firewire port.
One very cool reason for doing this for the worship team is the ability to do what’s called a Virtual Sound Check. What that means is that you don’t need the worship team there to set up the board. You can play back the individual tracks back into their respective board channels and use those tracks as the sound check.
Then, once the band gets in, sound check is very minimal. It’s also a great way for the sound team to train on the board and allows them to massage settings without needing the musicians.
Saving Scenes
Once you get everything set the way you want it remember to save your settings to a scene. I usually recommend naming the scene with the church name and 1. That way you can always recover your baseline settings.
Sound techs should create their own “sandbox” scene, which allows them to manipulate settings and save it to their own scene without affecting the master scene. Make sure that no one other than the lead sound tech saves to the master scene.
Once you’ve got the master scene saved it won’t matter what changes people make to the board during the week. Bringing back the master scene will only require a quick push of a button, and in the case of the StudioLive, resetting the gain and adjusting the faders. In other digital boards, gain settings and fader positions are saved within the scene.
Making The Switch
Once the sound techs are comfortable with the digital board then it’s time to switch out the old analog board with the new digital one. Check all your settings. Be sure any settings you change are saved to the master scene once you’re happy with how everything sounds.
Finally, when you shut things down, do NOT shut things down by just turning off the power conditioner. This WILL damage the digital mixer. Follow the shutdown procedure in the manual. It can be anything from just powering off the mixer with the mixer’s power switch to a shut-down procedure on the screen.
Summary
A digital mixer is a whole new way of doing the same old things. It’s exciting as well as terrifying for volunteers, so go slow. Take it one step at a time and ensure they are comfortable with the new system before putting it into production. You’ll achieve a seamless transition and have fun doing it!
Brian Gowing has helped over 30 churches meet their technology requirements. Brian works towards shepherding the church, analyzing their technical requirements, sourcing the equipment, installing the equipment and training the volunteer personnel. As he likes to say, “equipping the saints with technology to help spread the Good News.” Contact Brian here.
The Right Sonic Blend For An Electronic Ensemble & The New York Philharmonic
Reinforcing the live performance of a motion picture score at Avery Fisher Hall in Lincoln Center
The Philip Glass Ensemble, along with members of the New York Philharmonic Orchestra and the Collegiate Chorale symphonic choir, recently performed Glass’ powerful score for the 1982 landmark motion picture “Koyaanisqatsi: Life Out Of Balance” as the film screened at Avery Fisher Hall in Lincoln Center.
The two exclusive live performances (and screenings), held on consecutive nights for sold-out audiences at the 2,738-seat home venue of the Philharmonic, presented some sound reinforcement challenges.
The hall does not have a house system, yet the Philip Glass Ensemble, founded by composer Glass in the late 1960s to perform his experimental minimalist music, is always amplified when playing live.
As a result, Dan Dryden, long-time front-of-house engineer for the ensemble, worked with Audio Production Services of Amawalk, NY to design a reinforcement system to serve the unique needs of the event while fitting within the scope of the hall.
“With an event like this you want all of the instruments, acoustic and electronic, to sound like they belong together,” Dryden explains. “The sound system needs to be clean and consistent, in addition to being capable of covering the entire hall without impeding any stage site lines.”
He adds that, in general, he prefers the footprint of compact line arrays, and following a site review, decided that approach would work for this project as well. The choice was the compact RCF TT+ Series, with single arrays each comprised of 10 TTL31-A modules flown left and right, attached to the overhead stage grid.
A view of Avery Fischer Hall with the main RCF TTL31-A arrays flown to each side of the stage. (click to enlarge)
“When specifying systems for the ensemble I’m looking for smaller line arrays with flat frequency response,” explains Dryden. “These were perfect. The low-mid frequencies are rich and warm, and the coverage was excellent.”
The overall footprint of these arrays indeed was relatively miniscule, measuring just less than two feet wide by only about 10 feet deep. The self-powered, 2-way active line array modules are outfitted with a single-8-inch cone driver and three compression drivers feeding a horn with horizontal dispersion of 100 degrees. They proved capable of covering all four levels of seating (main and three balconies) as well as boxes.
“The arrays had no problem throwing all of the way to the back row of the top balcony without any need for delay fills. We had plenty of power for the space,” Dryden states.
The mains were joined by four RCF TTS56-A dual 21-inch subwoofers, two side-by-side on each side of the stage, and each of these sub sets hosted a single TT25 compact powered loudspeaker supplying in fill presence, particularly for higher frequencies.
The house loudspeaker complement was completed with front fill via four TT052-A low-profile 2-way loudspeakers deployed evenly along the front lip of the stage.
The ensemble, positioned centrally on stage, was comprised of eight players, including three on keyboards, three more on woodwinds, one soprano vocalist, and for this show, a bass vocalist. The orchestra’s 30-piece string section and 19-piece brass section, as well as the 40-member choir, resided in a semi-circle around them.
Each string instrument – violas, cellos and double bass – was outfitted with a DPA 4061 omnidirectional miniature clip-on microphone, while Sennheiser MD 421 II dynamic mics were stand-mounted for each trumpet, trombone, French horn, bass trombone and tuba in the brass section. Each two vocalists of the choir shared a Shure SM58 mic, also stand-mounted.
A closer look at one of the compact arrays that provided the advantage of a minimal footprint. (click to enlarge)
The ensemble feeds went directly to both front-of-house and monitor consoles, with Dryden manning a Yamaha PM5D board for house and Stephen Erb on another PM5D for monitors.
All of the orchestra and choir feeds (more than 80), meanwhile, routed to a DiGiCo D1 Live console. There, Dan Bora did a mix of the individual stems that were then supplied to the house and monitor consoles.
“One big challenge for a performance of this scale is the number of inputs,” Dryden notes. “In this case we decided to utilize a sub mix, which ended up being a very big job. Not only did Dan Bora have to make sure signal integrity and placement of each of the microphones were good, but the mixes provided to house and monitors were key to the sonic performance.”
All effects were supplied via the PM5D consoles with the exception of a Lexicon 300 reverb at front-of-house that Dryden likes to apply to certain passages or sections.
The ensemble on stage surrounded by the orchestra and choir during one of the performances of “Koyaanisqatsi.” (click to enlarge)
“The Lexicon algorithms are excellent,” he says. “I’ve used Lexicons forever – for me they’re the smoothest, best-sounding digital reverbs.”
Monitor engineer Erb fed mixes to 12 dBTechnologies DVX D12 powered 2-way loudspeakers that acted as stage monitors for the ensemble - keyboards, woodwinds, soprano vocal and bass vocal.
The strings, brass and chorus sections were served monitor mixes with stand-mounted dBTechnologies K70 multipurpose ultra-compact loudspeakers (also powered).
Dryden reports that the project produced the results he was seeking. “I think it’s always important to remember that you need to work with a room rather than try to impose your will upon it,” he concludes. “In this case, it’s a terrific room and, when equipped with the right system, it sounded fantastic. The musicians in the symphony and the chorus added so much to the ensemble’s performance. It all added up to a lot of fun.”
Julie McLean Clark is a writer and marketing consultant working who has worked in the pro audio industry for more than 15 years.
Haven’t we all had stories of misheard words? It could have been a song lyric or you misheard your spouse? Maybe they mumbled a word or it just wasn’t clear what was said. This has been the cause for a few hilarious moments at our dinner table.
The problem is unclear words are a distraction from the message.
In the church environment, the pastor’s words must be clear. We can ensure this maximum intelligibility through proper speech EQ.
There are four topics to consider when it comes to the EQ’ing needs for the spoken word.
1. Microphone location. We are fortunate in that most pastors now use wireless microphones. This means that the distance between the mic and their mouth is pretty consistent. In the case of the headpiece, this is especially true.
In the case of the lapel mic, remember they should drop their chin to their chest and put the mic directly below that point. Long ago, I was taught “a fist away from the chin.” The point here is that we want the best sound isolation we can possibly get while having a good gain structure in place.
Remember, the closer to the source, the more the proximity effect comes into the equation and you’ll need to EQ out some of that added bassiness.
2. The speaker’s natural voice. Just as every guitar has a unique sound, so does every person. You want to bring out the best qualities of their voice. You don’t want them to sound like a different person. Their vocal characteristics are also “what you have to work with.”
This means you’ll need to know how to deal with quiet speakers, bassy talkers, and nasally preachers, just to list a few. Not everyone has a great radio voice.
3. Presence of background music. Depending on your church, your pastor might talk with a running soundtrack. There is definitely an art to being able to play the right music for this.
However, any type of music bed means you now have to make a space for the voice amidst the instrumentals. Instrumentals can easily blur the spoken word so you’ll have to plan on tweaking the EQ for the musicians as well.
4. The environment. Just because a vocal boost at 400 Hz sounds good in one room doesn’t mean it will sound good in another room. One of myreaders runs audio outside…in Egypt. Any EQ work must take the environment into account. The settings for a “quiet room” won’t be the same for an echo-y room or an outdoor venue.
Now that we’ve got those out of the way, let’s turn to…
The Frequency Make-Up Of Speech
Our speaking voice has three frequency ranges that need to be understood: 1. Fundamentals. The fundamental frequencies of speech occur roughly between 85 Hz and 250 Hz. 2. Vowels. Vowels sounds contain the maximum energy and power of the voice, occurring between 350 Hz and 2 kHz. 3. Consonants. Consonants occur between 1.5 kHz and 4 kHz. They contain little energy but are essential to intelligibility.
In short, this means that the “power” of the voice does not equate to the intelligibility of the voice. Think of it like this…just because a person has a booming voice doesn’t mean they are easy to understand.
Now that you understand the audio dynamics (fundamentals, etc) in a voice and the environmental concerns (background music), let’s turn to…
What You Can Do To Provide The Maximum Speech Intelligibility For Your Pastor
There are three things you can do for tackling the EQ’ing process for the spoken word:
1. Make room for the voice. As I mentioned above, the environment makes a difference in how you EQ the spoken word. We can only control what is coming into the mixing board, so wind and rain aside, let’s talk about music.
Mixing a large band means making space in the sonic spectrum where each instrument/vocal can sit and sound unique; and of course then blending these sounds together into a tight mix.
The spoken word needs the same treatment when music is played underneath it. This can happen in two ways—
—A. Adjust volume. This can be done using compression or simple volume adjustments. The general rule-of-thumb is the music is there to support the spoken word – to sit underneath it. Therefore, look to cut volume levels of instruments before you boost the volume of the speaker. You can also use compression to bring volume levels up and down as you wish. —B. Adjust the mix. Cut the frequencies of the instruments where they are the same as that of the speaker. Boost the spoken word EQ in those areas a little if needed to present the music and the voice as two distinct sounds.
2. Know sibilance and how to avoid it. Sssssssibilance in vocals is when the sound of the letter “S” sounds more like a hissing snake. You can accentuate vowel sounds/add presence by increasing the EQ in the 4.5 kHz to 6 kHz.
However, the “S” sound lives between 5 kHz and 7 kHz. Therefore, be careful when adding presence because you can easily go from a great sound to a hissy sound.
3. Focus on vocal quality. There is no simple 1-2-3 process to EQ’ing the spoken word. Therefore, take these points into consideration:
—Roll off the low frequencies if the proximity effect is causing unusual bassiness. —Don’t roll off so much low end as the voice loses some of its umph. Yes, I’m using “umph” as a technical word. —Boost in the 1 kHz to 5 kHz range for improving intelligibility and clarity. —Boost in the 3 kHz to 6 kHz range to add brightness. This can help with speakers with poor intonation. —Boost in the 4.5 kHz to 6 kHz range to add presence. Note that too much boosting in this area can produce a thin lifeless sound. —Boost in the 100 Hz to 250 Hz for a boomy effect.
In Case Your Head Is About To Explode From Information Overload, Remember:
—The above points can contradict each other. There is no hard and fast rule. Mixing is as much an art as a science. Trust your ears over everything else. —It’s possible that once you EQ the vocal channel that it’s a little lacking in the low end. Boost it a bit give it that full sound. Again, trust your ears. Close your eyes and ask yourself if it a) sounds natural and b) sounds clear.
Finally
EQ’ing the spoken word is about improving the quality of the sound so it sounds clear, is easy to understand, and sounds natural.
So much of our mix time goes towards the band. Make sure you spend those few crucial minutes working on the pastor’s vocal as well.
Church was about the sermon long before music, skits, and cool videos rolled onto the scene.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
Ells has long been recognized as one of the finest engineers working today and has a shelf full of industry awards (five Grammys, four Surround Music Awards, Surround Pioneer Award, Tech Awards Hall Of Fame and too many total award nominations to count) from his work with The Eagles, Beck, Steely Dan, Fleetwood Mac, Sting, John Fogerty, Van Morrison, Toto, Queen, Faith Hill, Lenny Kravitz, Natalie Cole, Doobie Brothers, Aerosmith, Phil Collins, Aretha Franklin, Barbra Streisand and many, many others to prove it. He’s also one of the nicest guys in the business.
In this interview, Elliot talks not only about his approach to mixing but about some of his projects as well. ——————————————————————————————————- Do you have a philosophy about mixing?
Elliot Scheiner: I’ve always believed that if someone has recorded all this information, then they want it to be heard, so my philosophy is to be able to hear everything that was recorded.
It’s not about burying everything in there and getting a wall of sound. I’ve never been into that whole concept. It was more about whatever part was played, if it was the subtleties of a drummer playing off beats on the snare drum next to the backbeat, obviously he wants that heard. So I always want to make sure that everything that’s in that record gets heard.
If you were able to accomplish hearing every single instrument in the mix, that was a huge achievement. Granted, maybe there wasn’t as much information when I started as there is now. I myself have come across files that have been a hundred and some odd tracks, so it’s not as easy to do that today.
I have to admit that the way some people record things today is a bit peculiar. All of a sudden you’ll be dealing with 7 or 8 different mics on the same instrument. Like, for example, an acoustic guitar will all of a sudden have 7 different viewpoints of where this guitar’s being recorded.
It’s mind boggling that you have to go and make a determination and listen to every single channel to decide which one you want to use. And if you pick the wrong ones they come back at you and say, “Oh, we had a different combination” or “It doesn’t sound quite right to us”, but they don’t tell you what they did! So granted, it is a little more difficult to deal with those issues today, but I still take the same approach with every mix. If you have a hundred tracks, will you try to have them all heard? Or do you go in and do some subtractive mixing?
Elliot Scheiner: Well, it depends if that’s necessary. I don’t usually get those kind of calls where they say “Here’s a hundred tracks. Delete what you want.” It’s usually not about that. And I have to say that I’ll usually get between 24 and 48 tracks in most cases and hardly ever am I given the liberty to take some of them out.
I mean if something is glaringly bad I’ll do that, but to make a judgment call as to whether background vocals should be in here or there, I generally don’t do that. I just assume that whatever an artist and producer sends me is kind of written in stone. They’ve recorded it, and unless they tell me otherwise, I usually don’t do subtractive mixing. How long does it take you to do a mix on average?
Elliot Scheiner: Depending on how complicated it is, it usually takes anywhere from 3 hours to a day.
3 hours is really fast!
Elliot Scheiner: Yeah, well a lot of time you just get a vibe and a feel for something and it just comes together. Then you look at it and say “How much am I actually going to improve this mix.” I mean if it feels great and sounds great I’m a little reluctant to beat it into the ground.
For me it’s still about a vibe and if I can get things to sound good and have a vibe, that’s all I really care about. I still put Al Schmitt on a pedestal. Look at how quickly he gets things done. He can do three songs in a day and they’ll be perfect and amazing sounding and have the right vibe. So it’s not like it can’t be done. Some people say that you can’t get a mix in a short time and that’s just not true and Al’s my proof. Where do you usually start your mix from?
Elliot Scheiner: Out of force of habit, if there’s a rhythm section I’ll usually start with the drums and then move to the bass and just work it up. Once the rhythm section is set I’ll move on to everything else and end with vocals.
How much EQ do you use?
Elliot Scheiner: I can’t say that there are any rules for that. I can’t say that I’ve ever mixed anything that Al has recorded, but if I did I probably wouldn’t have any on it. With some of the stuff done by some of the younger kids, I get it and go, “What were they listening to when they recorded this.”
So in some cases I use drastic amounts where I’ll be double compressing and double EQing; all kinds of stuff in order to get something to sound good. I never did that until maybe the last 5 years. Obviously those mixes are the ones that take a day or more.
When you’re setting up a mix, do you always have a certain set of outboard gear, like a couple of reverbs and delays, ready to use or do you patch it as you go?
Elliot Scheiner: Usually I don’t start out with any reverbs. I’m not one for processing. I’d like to believe that music can survive without reverbs and without delays and without effects. Obviously when it’s called for I’ll use it, but the stuff I do is pretty dry. The 70’s were a pretty dry time and then the 80’s effects became overused. There was just tons of reverb on everything.
Most of your Steely Dan stuff is pretty dry, isn’t it?
Elliot Scheiner: It’s pretty much dry. What we used were plates usually.
Real short ones?
Elliot Scheiner: Not necessarily. In the days when I was working at A&R [studios in New York city] we had no remotes on any of our plates there. Phil [Ramone - producer and owner of A&R] wanted to make changing them difficult because he tuned them himself and he really didn’t want anybody to screw with them.
There would be at least 4 plates in every room. Some of them might be a little shorter than another but generally they were in the 2 to 2 1/2 second area. There was always an analog tape pre-delay, usually at 15 ips, going into the plates. The plates were tuned so brilliantly that it didn’t become a noticeable effect. It was just a part of the instrument or part of the music. You could actually have a fair amount on an instrument and you just wouldn’t notice it.
Is the sound of the A&R plates something that you try to get today?
Elliot Scheiner: Oh, I’m always trying to get that reverb sound If I’m using plates either at Right Track or Capital, I’ll still use an analog tape delay going into it.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.
From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature provides an interesting look back at quadraphonic recording. This article dates back to September of 1970. (Volume 1, Number 3). The text is presented unaltered, along with all original graphics.
As a complete oversimplification, a microphone is an instrument which measures differences in air pressure.
It is not surprising that somebody would, in light of the interest in Quadraphonic sound, experiment and perfect an instrument which would measure and transduce the differences in air pressure around a full 360 degrees - to effectively create a quadraphonic microphone.
Figure 1 (click to enlarge)
Such a truly Quadraphonic device, developed by engineer Carl Countryman and producer Brad Miller, is in external appearance no different than the several models of standard microphones (Figure 1).
This Quadraphonic microphone has been designed and built using the case and chassis of a Neumann SM-2, into which four independent microphone heads have been built to provide full 360-degree pick-up.
The pick-up patterns (Figure 2) are cardioid, front and back, and figure-8 at the sides.
Although the obviously complicated matrixing data are proprietary, and unavailable for publication, the discussion of pickup patterns, generally, yields an understanding of how the design provides excellent separation and naturality of sounds.
Cardioid, also sometimes called unidirectional, is a heart-shaped response. It is resultant of an omnidirectional and figure-8 pickup.
The signals are superimposed on each other; at the very rear they are anti-phase, and so cancel out.
At the front they are in phase, hence the tapering hear-shaped response toward the rear.
Figure 2 (click to enlarge)
Figure-8, or bi-directional pickup-patterns, are the result of two directional pickup patterns, one in phase and the other anti-phase.
The output at the front and the back are equal, although opposite,.
As the input signal moves to the side, the output is gradually reduced until at 90 degrees, the two patterns have, for all intent and purpose, canceled each other out.
Figure 3 shows microphone capsules as they are arranged in the microphone head.
“Front to Back” and “Left to Right” are one above the other at 90 degrees to each other.
Three demonstrations, on very spontaneous, served to convince that development of the unit is very nearly complete.
Figure 3 (click to enlarge)
The microphone was hung in Miller’s back yard garden, surrounded by about 200 degrees of sound source emanating from a waterfall with various small tributary streams flowing from it. It presented an excellent opportunity to “hear” the complete environment; the waterfall in stereo on the two speakers in “front,” and from behind, the beautiful ambiance of the total environment and the reflected sound.
Several minutes into the demonstration, on the Southern Pacific tracks bordering on the rear of the Miller garden, a slow-moving freight train ambled by. The completeness of the sound, the way it engulfed the listening room, is difficult to describe. It was totally complete… almost frighteningly so.
Figure 4 (click to enlarge)
Miller completed the demonstration by playing a 4-track tape of his “Mystic Moods Orchestra” on an especially adapted Sony. The machine (Figure 4) has been adapted for 4-track, in and out, and will be able to accommodate 10-inch reels of 2-inch tape.
The machine is the forerunner of a new design from the Countryman/Miller collaboration which will weigh in the vicinity of 20 pounds.
The “Mystic Moods” piece only served to further impress that Quad or Multi is certainly on the way… with an endless spectrum of sound combinations and tonal effects.
Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.
Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970
Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.
Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).
John Penn Of SSE On The Current (And Future) State Of The U.K. Live Sound Market
“What really worries me is that artists don’t achieve the same mega-sales status that they used to."
John Penn has seen a lot in his 35-plus years in the pro audio business, which officially began with the founding of Disco Sigma Sound Enterprises in 1975 at the age of 19.
Now the managing director of that fledgling start-up - SSE Audio Group, the largest PA firm in the United Kingdom – John recently sat down with me and shared his thoughts on the sound hire market in his part of the world, as well as some of the long-term goals for the company.
Paul Watson: How is business in general, and which areas in particular have you seen the most change?
John Penn: Well, as usual, we had a very busy festival season, as we supply to more in the U.K. than any other rental company, but what’s impacted us most is that touring is way down. There are far less shows being booked, and we’re also seeing a double-whammy, where the Academy market is now fully installed, so we’re no longer supplying to those venues either.
It’s partly down to the recession in that people have got less money in their pocket, but people are now being far more careful generally about where they spend their money.
PW: And what about trends?
JP: Well, for years now, we’ve seen the trend of the market expanding, but now, there just isn’t a next generation of concert-goers to work with, so the market isn’t expanding. That’s a big change. Youngsters have all of these other things to do as social activities, such as gaming, so perhaps making sure that they see going to concerts as a sort of mission-critical thing would be a good idea.
There is still that cultural phenomenon of going to a festival and experiencing the vibe, which I understand fully, as it’s something I engaged in myself when I was 19 years old - that’s really strong. The problem is that it isn’t translating to people just wanting to go to a show as much as we would like it to.
PW: Do you think it’s due to cost of live shows?
JP: Well, even the Academy venues are moderately more expensive compared to more of the small clubs you used to be able to go to, so that is a factor, yes. And this fall, there were far less shows generally as promoters couldn’t afford to keep losing megabucks on promoting shows that don’t sell, and that’s not good for anyone.
Editor’s Note: Academy Music Group (AMG) is the U.K.’s leading owner and operator of live music and club venues.
PW: U.K. bands don’t seem to be touring anymore, opting to play festivals instead, and I guess it makes sense in a way: less outlay on production costs, the audience is already there, so artists don’t have to worry about selling tickets, and when the U.K. festival season ends, bands can go and do the same thing in Europe, or even Australia…
JP: This is exactly right – there just aren’t many tours going out, which means there’s a lot less work around. And at the same time, the work that is out is on very tight budgets. It used to be that the money came from the record companies, but that’s no longer the case; they have to make enough out of the shows not just to sustain them when the shows are out, but to sustain them until they go out again.
PW: So there’s a lot more pressure on achieving a good return on investment then?
JP: Oh, a huge amount more. Budgets are being squeezed, the artists have to be confident that they can sell the shows, and the promoters are less confident than they were, so are offering smaller guarantees. This squeezes pricing even more.
PW:With more venues installing their own systems, how do you as a rental company compensate that effective loss in business?
JP: The difference between SSE and a lot of our competitors is that we’re a much more diversified business. Our installation department and sales department are very big parts of the business, and they’re compensating for some of the loss of touring work.
We have this problem that through the summer months, we are flat out (busy), but it’s difficult when you have such a feast and famine situation where you do so much business in those three months in the summer compared with what you do between September and May.
The diversification has certainly enabled us to hold our own, but it’s not enough. We have a key problem that with the amount of investment that’s required, we can’t possibly make the money back to work at the level we are with a three-month season each year.
We have significant issues to deal with as we’re pretty much geared up in terms of staffing levels to be able to deal with working at that volume all year round, so it’s even harder for us to make money in those down times.
PW: So how much of the business is install at the moment?
JP: The installation side is a very big variable, because you win a job and you’re suddenly really busy, then for whatever reason, you don’t win any more, maybe because you’re busy doing those installs! [Laughs] The installs themselves are a bit seasonal - companies realize that there are key times of the year to get a re-fit done that don’t disrupt their businesses; thankfully, those key times are January and August, and they’re our quietest months.
An SSE system for 75,000 to hear a public mass by Pope Benedict XVI at Bellahouston Park in Glasgow, Scotland in late 2010. (click to enlarge)
PW: How will next year’s [London] Olympics affect business?
JP: Actually, we’ve already got some extra work off of the back of the Olympics. Live Nation has an event in Victoria Park that we’re doing, which will be running throughout the Olympics – and that’s a big job. I also think that there will be a bunch of other work to pick up, which I think we’re well placed to deal with.
PW: And how do you see business changing in, say, five years?
JP: Well, it’s unlikely I’ll be doing this job, because I’m 57 next year. You can’t work at 57 in the same way you can when you were 40, and I feel that difference already. People who put themselves under too much stress and pressure are never able to enjoy their retirement at the end of it.
While I’m not about to hang up my spurs and say ‘well that’s it’, like my Dad did, I plan to wind down gradually; but you have to have a vehicle to enable you to not be here, and to that end we’ve been building a management team at SSE with a global vision, because the industry’s definitely going to change.
PW: How will it change, exactly?
JP: Well, what really worries me is that artists don’t achieve the same mega-sales status that they used to.
Two of the biggest on the circuit are Springsteen and U2 – and they still sell a phenomenal amount of records, too; they’re still able to deliver relevant albums, which isn’t true of some of all the major players.
Look at the proportion of the top selling artists that are over 60: The Rolling Stones, Elton John, Bruce Springsteen, Sir Paul McCartney - all of the top grossing acts are in that league.
PW: And the likelihood is that they won’t be touring to the same degree in five years time either…
JP: Exactly, and there’s no one to replace them! That’s my great fear about The X Factor: everyone gets their 15 minutes of fame, but how many are able to build on that? Take That is a remarkable phenomenon – a British boy band that started 15 years ago, then came back bigger and better, but they’re the exception rather than the rule.
We need raw, groundbreaking talent. Adele is a phenomenal talent; O.K., she has a health problem at the moment, but hopefully she will get over that and go on and be a mega-seller, but it’s whether we have enough of them. That’s the big question.
The SSE warehouse in Redditch, England. The company also has offices and warehouses in London and in Bradford as well as in France. (click to enlarge)
PW: Staying on long-term, with more and more European rental companies providing all-in-one touring packages, do you think the U.K. market will eventually have to say, “Right, I’m going to have to buy a lighting company’?
JP: We’ve been discussing this concept for years, and first of all, the U.K. market is too fragmented, even as sound companies. Our major-player sound companies are all independents, and although the Europeans always did a bit more of the whole production than the audio anyway, you’re right, they now do the trucks, the buses, the whole nine yards.
It’s an obvious way of saving money on production, and it’s a winner-takes-all situation. We feel that’s the obvious way to go in this country, but only up to a certain point, because you’ve still got quite a few big independent lighting and video businesses. As long as you’ve got Neg Earth and XL Video as independents, it will continue as it is.
The problem is, if we’re going to maintain ourselves as a sound company, we’ve got to be a much bigger business, so that we’re reducing the overhead and cost of delivering what we’re doing; and with regard to providing 360-degree production, the other problem is that the big three lighting companies, Neg Earth, PRG, and HSL, are all turning over big money, so we can hardly say: “let’s get a lighting division and find a company to buy,” because, well, who is there?
PW: Good point. But in theory, at least, you think it makes sense?
JP: Absolutely. The one-stop shop is something that will eventually be inevitable. It’ll just be a case of finding a partner so we can make it happen.
Paul Watson is the editor for Europe for Live Sound International and ProSoundWeb.
Church Sound Files: The Reason For “Bad Sound” May Not Be The Sound System
Three factors, roughly equal importance, play the key role in good sound - and “two out of three” isn’t good enough
Many things around us are getting better. Computers are faster, televisions have more resolution, and dishwashers are quieter and more powerful than ever.
But with all of our digital wiz-bang processors, technology has been unable to eradicate “bad sound.” Why is this so? This short piece is an attempt to shed some light on three possible causes, two of which have been completely unaffected by the technological revolution.
The goal of most sound reinforcement systems is to deliver high quality sound reproduction to the listener. While we would like to think that a high quality sound system guarantees this, it does not.
The quality of the reproduced sound will only be as good as the weakest link in the reproduction chain. Let’s examine some of the major “links” individually.
The Room
The room is a major factor in the reproduction chain. Most large spaces are hostile environments for sound systems, unless they have received special attention from a professional and a considerable financial investment from their owner. Good acoustics doesn’t just “happen.” It is the by-product of careful planning.
A quality sound system may radiate an exceptionally high-fidelity sound field into the room. Unfortunately, most of the radiated energy will create acoustic events that detract from the listening experience. While small rooms have their share of acoustic problems, these problems pale next to the late reflections, reverberation, and energy build-ups encountered in large spaces.
If your sound system doesn’t sound good, ask yourself the question “What have I done to provide a good acoustic environment?” If the answer is “nothing,” then you got what you paid for.
The Sound System
Of course, a good sound system is a vital link in the reproduction chain. But this doesn’t just mean expensive equipment. It means that equipment that is suitable for the environment has been selected and implemented by someone who understands the compromises involved in large room reinforcement systems. Money can be wasted on “features” that offer no real benefit for the large room environment.
The vast majority of auditoriums that I have visited are not suitable for multi-channel formats such as stereo, surround sound, etc. since each channel must be delivered to all listener seats. Loudspeaker placements that are optimal for stereo reproduction are horrible choices for single-channel systems.
Even with monaural systems, “first choice” loudspeaker placements often create problems with sight lines and aesthetics, and are therefore ruled out by venue owners. Multiple loudspeakers must overlap somewhere, and there will be sound problems in these areas.
A properly designed system will often sound bad in the aisles – the very place where casual onlookers will stand to evaluate it. We all have good sound at home, but the rules change as the listening space grows. Intuition that is not filtered through the proper large-room principles leads to errors.
Sound system designers are often forced to compromise away the performance of the system to make it fit aesthetic concerns, budget limitations, and fashion trends within the industry.
The Operator
I’ve intentionally saved this one until last. The most overlooked link in the chain is the end user of the system. This includes the mixer operator and any supporting staff, such as those who run the monitors and place microphones.
A monitor system that is too loud will dump excessive energy (usually low/mid frequency) into the audience area. This excess energy will upset the spectral balance of house sound system, tempting the front-of-house operator to compensate by over equalizing (usually in the form of high frequency boost). This results in a reduction in gain-before-feedback and an unnatural sounding system. Microphone placement is equally critical, as is an understanding of the shortcomings of various miking techniques.
If a lapel mic could sound like a hand-held, then no one would use hand-helds. The overhead drum mic that captures the cymbals also captures the stage monitors and “spill” from other instruments, as does the vocal mic used at arm’s length. And that “mellow” bass guitar sound that the musician likes in the practice hall turns to “mush” in a large space, where increased definition provided by the use of a pick and brighter strings may be required.
These factors and many more “eat away” at the sound quality of the system as a whole. A good mixer operator will evaluate and optimize the sound of the instruments individually before allowing the band to perform as an ensemble. There’s no room for democracy here – effective mixer operators learn to say “no” and “be quiet.”
A question that I recommend for an interview of prospective mix personnel would be “What will you do if something starts to squeal?” If the answer is anything other than “Turn the offending channel down slightly until I figure out what the problem is” move on to your next applicant. Filters implemented in desperation do nothing to preserve sound quality.
Modern mixing consoles pack a considerable “wow factor.” It’s fashionable to sit behind a large one and move knobs all of the time. But doing so doesn’t make one an engineer. Completing an accredited academic program or piloting a locomotive does. The decision as to which console to purchase is often made with no consideration as to whether anyone at the facility will be able to operate it. The result? Bad sound.
I have personally witnessed the performance of many good sound systems ruined by bad rooms and incompetent operators. I have also seen skilled operators “salvage” the sound reproduction in situations where the room and system were less than optimal.
The performance of a sound system is only as good as its weakest link. Unfortunately, all of the links that I have mentioned are of roughly equal importance, meaning that “two out of three” isn’t good enough. Good sound requires all three.
Experienced, well-trained audio people realize this and are there to help you find your weakest link. Pay for their advice and follow it.
Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. Synergetic Audio Concepts (SynAudCon) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, SynAudCon is dedicated to teaching the basics of audio and acoustics. For more information, go to http://www.synaudcon.com
A reader is puzzled by stereo (2-mic) acoustic guitar recording:
I recently got into mixing acoustic guitar with 2 mics. The problem is that I do not know how to create as much ‘space’ as some tracks I know of. I’ve tried XY, ORTF, and spaced pair.
XY and ORTF are too narrow. Spaced pair seems reasonable (following the 3:1 Rule), but the mic pointed closest to the body becomes overly ‘bassy.’
How can I balance the stereo image? EQ can control the problem but not by much. How would you go about on fixing this problem?
I know mic position has to do with it but I don’t know where to start. Just wondering if you had to overcome this type of problem before.
As much acoustic guitar recording and mixing as I do, I’ve dealt with problems like this a LOT.
(And this applies to ANY instrument, not just acoustic guitar.)
First things first…
Mic Placement Is Everything
I’ve played the “Hey, I’m Just Going to Throw a Couple of Mics in Front of the Guitar and Hope it Works” game.
It’s not a very fun game, trust me. You always end up losing.
Whenever you’re recording an acoustic instrument, always plan to give yourself at least a few minutes to try a few different mic techniques. I
s one mic (mono) appropriate? Does it need the wider sound of a stereo (2-mic) technique? If so, which technique is best?
There are a lot of options, and it would behoove you to try at least a couple of them before committing the recording to tape. (Tape…who says tape anymore?)
I feel like I’ve come full circle when it comes to stereo recording. I used to love a nice, wide acoustic guitar sound. But the last year or so I’ve simplified a lot.
Nowadays I’ll either go with a single mic or use two mics in an XY configuration. Why? Because having a really “wide” recording isn’t all it’s cracked up to be.
Even though XY doesn’t give the widest stereo image, it doesn’t lend itself to phase issues and lopsided recordings.
Speaking of lopsided, let’s talk about that stereo recording that has too much bass in one mic versus the other.
Even if you do your job on the front end with mic placement, sometimes one mic (the one pointed at the sound hole) picks up more bass than the other.
Here’s how I deal with it:
—Place an EQ plug-in on the bass-heavy track ONLY. —Use the EQ to remove some of the excess low end, until the two tracks are more balanced. —Bus the two tracks to a stereo aux track. —Put any additional EQ and compression plugins on the stereo aux track.
The first EQ is simply corrective. It lets you balance out the sound. (No more lopsided guitar.)
The second EQ (on the stereo aux) is what you’ll use to carve the entire sound of the entire stereo recording to fit it in the mix.
As you may have guessed, mic placement and technique play a HUGE roll in how awesome your acoustic guitar recordings (and mixes) are going to sound.
Make that a priority on your next session.
If you’re interested in diving in deeper, I created a 4-week class on getting consistently awesome acoustic guitar recordings. You can join any time here.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
This article, #InstallationFail, has been inspired by the many, many instances of bad installation practices I’ve seen throughout the years.
I’ve been taking pictures and cataloging these for quite some time, and I figured it’s time to share them with the world.
Now, I should point out I’m not publishing these to make anyone feel bad. Though some of you should feel bad for some of these installations. But that’s not the point.
The point of this article is to show you how not to do things. My thinking is that many bad installation practices are repeated because someone saw it done that way before and no one told said person it was wrong.
Surely, if you knew it was wrong, you wouldn’t do it this way… right?
In that spirit, here we go. Our first installment is a run of Cat-5 cable. The cable comes up from a lower floor into a pull box; and it comes up in a conduit.
That’s where things get weird. I’m not sure if they found pulling the cable through the conduit just too dang much work or what, but instead of continuing on out through the conduit at the top of the box, they punched through the side of the box and tie-wrapped to the conduit. I’m not kidding.
Easier than running through the conduit? Perhaps. Correct? Uh, no.
Now, there is a proper way to bring cable in and out of a box without using conduit. Cable clamps and bushings are two such options. Apparently, neither were handy when this cable run was done.
And in case you’re wondering what might be wrong with pushing cable through a box in this manner, take a closer look from inside the box.
See those nicks in the insulation? They’ll get worse with time as the building vibrates.
Those sharp edges on the box will gradually cut through the cable as the building vibrates. It may take a while, but the wire will eventually be compromised.
They also pulled rather tight, which puts pressure on the cable, another no-no.
Finally, after continuing up the outside of the conduit, the wire goes through a fire-rated wall; a big no-no. This is the kind of stuff that will get you shut down if you have a fire inspector with a limited sense of humor.
There are proper ways to go through a fire-rated wall. This is not one of them.
This #InstallationFail has a lot going for it (perhaps more correctly, not going for it). The sad thing is that it was installed by a company that does cable installation. Meaning, this was not the work of some well-meaning but uninformed volunteers.
Nope, this was a “professional” job. And frankly, that irritates me.
Now, let’s move long to some great reader finds. I won’t give credit, largely to protect the guilty, err… innocent. But you know who you are.
Actually, these were all found by people who were just as amused and disgusted at the same time as we all are.
Why try to cram too much into a work box? (click to enlarge)
This is a classic case of “Why work hard if you don’t have to!.” I’m not exactly sure how the bolt is connected to anything structural, but I’m pretty sure that little metal dome is not rated for holding a Parnel.
And really, trying to stuff all the wires inside the box? Waaaayyyy too much work. Let them all hang out. Keeps ‘em cooler, anyway. Nice work, to be sure.
You can use wire nuts for almost anything. I love the art of this piece.
The interplay of the yellow and blue wire nuts is stunning. The fact that someone took the time to do it is impressive. I’m not sure what exactly the little stub of a RJ-11 may have been plugged into, or how they managed to plug anything in while it was mounted in the wall, but it’s got creativity written all over it.
I can’t say for sure, but I’m guessing there was a RJ-11 coupler somewhere in the mix here…
The safety of this device is unquestionable.
Unquestionably bad, that is. I can’t decide which I like more…the duplex outlet with only one leg attached, or the plug end with no backing housing, leaving the hot terminals right out there in the open for all to experience.
I’ve seen a lot of sketchy electrical wiring in my day, but this one takes the prize for most sketchiest. I’m not sure what in parallel universe this may be considered safe, but it’s not this one.
They could have at least used gaff tape to cover up the hot leads…
Moving along, consider this a primer on how not to install cables.
The good news is there’s plenty of cable should the equipment need to be re-located.
At least I can’t chalk the above mess to a professional installer. The next photo however…well, it’s just the way the installer left it.
Bushings? We don’t need no stinking bushings. Or the cover for that matter.
Man it was nice of those plumbers to put those pipes there for the cables to run over.
And just to prove the A/V guys didn’t favor plumbers over HVAC guys…
We’ve got loudspeaker cables going over HVAC and electrical! Score!!
Yes, we’ve seen all sorts of creative installs. This is one of my favorites. Props for using conduit. But feeding an extension cord through it to plug into a dimmer? Hmmm…
Hey, at least they put the extension cord in the conduit…
Sometimes, however, conduit seems hardly necessary.
This isn’t so much an “installation” as a lack thereof…
I really have nothing more to say about that one.
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Enter The PSW Sweepstakes To Win An Audio-Technica Microphone Or Headphones
Enter to win an Audio-Technica microphone or headphones in the first PSW Sweepstakes of 2012.
ProSoundWeb is giving away three Audio-Technica 50th Anniversary Limited Edition products each month in January, February and March.
Specifically, for each drawing, we’re giving away:
1st prize - AT4050/LE Multi-Pattern Condenser Microphone —Special 50th anniversary edition in silver-colored metallic finish with etched-on serial number and carefully crafted wooden carrying case —Transparent uppers/mids balanced by rich low-end qualities combine with advanced acoustic engineering for extensive performance capabilities and highest quality —Dual-diaphragm capsule design maintains precise polar pattern definition across the full frequency range of the microphone —The 2-micron-thick, vapor-deposited gold diaphragms undergo a five-step aging process so that the optimum characteristics achieved remain constant over years of use —Three switchable polar patterns: omni, cardioid, figure-of-eight —Transformerless circuitry virtually eliminates low-frequency distortion and provides superior correlation of high-speed transients —State-of-the-art surface-mount electronics ensure compliance with A-T’s stringent consistency and reliability standards —Switchable 80 Hz hi-pass filter and 10 dB pad —Custom shock mount provides superior isolation —Valued at $995.
2nd prize - ATM25/LE Hypercardioid Dynamic Instrument Microphone —Exclusive 50th anniversary edition in silver-colored metallic finish with serial number etched on the surface —Ideal for kick drum, toms, and other highly dynamic instruments —Handles very high SPL at close range —Big, warm low-frequency response with excellent presence —Multi-level grille and rugged construction —Offers very full sound on close-up vocals and dialogue —Corrosion-resistant contacts from gold-plated XLRM-type connector —Rugged, all-metal design and construction for years of trouble-free use —Valued at $489
3rd prize - ATH-M50s/LE Professional Studio Monitor Headphones —Special 50th anniversary edition in silver-colored metallic finish —Exceptional audio quality for professional monitoring and mixing —Collapsible design ideal for easy portability and convenient storage —Proprietary 45 mm large-aperture drivers with neodymium magnet systems —Closed-back cushioned earcup design creates an outstanding seal for maximum isolation —Adjustable padded headband for comfort during long mixing/recording sessions —Single-sided straight cable terminates to gold-plated mini-plug with screw-on 1/4-inch adapter —Valued at $209
Go here to enter the latest PSW Sweepstakes. Note that entrants are asked to register to receive the ProSoundWeb Daily e-newsletter.
The real art of audio, or, I mean the real science of audio, is...
One of my favorite sayings: “Audio is an art that everyone thinks is a science, and audio is a science that everyone thinks is an art.”
There’s no doubt that delivering an accurate (not to mention good-sounding) mix without missed cues is the right blend of both art and science.
Knowing the science helps in setting up the mix and making sure that everything is routed properly and the right things plugged in to the right parts of the system.
Knowing the art helps to creatively bring all of the various sounds from the instruments and singers together to deliver a pleasing sound without any distractions.
If that title were true, I would not be here writing this, and the unfortunate thing is that I would be dead from self-inflicted wounds! Over the years I’ve found that I can usually attribute the reason for the bad sound that I’ve mixed to one word: anticipation.
On the science side, anticipation means:
1) Being generally prepared, having the right tools, and being aware of what is going on at the event.
2) Check over the system to make sure everything is working.
3) Check all the inputs to make sure they are working and patched correctly.
4) Visually reviewing the board, making sure things are routed were they are supposed to be, the channel EQs are on and aren’t set too crazy, etc.
5) Having a backup emergency microphone on stage that everyone knows to go to if his/her particular mic fails.
And on the art side of things:
1) Thinking ahead, planning to boost the levels for solos.
2) Keeping my eyes on the stage to make sure mics are turned on ahead of people speaking.
3) Having my headphones handy so I can pfl channels to check anything, and quickly.
4) Being in tune with that is going on so I can react quickly to any changes that occur.
5) Having my cue sheet or order of service right next to me and then read ahead and mentally prepare for the next event on the sheet.
6) Listening to the worship songs ahead of time to hear what the original recordings sound like.
7) Knowing where the backup emergency mic is patched and being prepared to use it for any surprise events (unplanned testimony) or mic failures.
Obviously anticipation alone doesn’t guarantee a great mix - you still need to have the fundamentals down. But it does greatly increase the potential of having an error-free service or event.
So there you have it. The real art of audio, or, I mean the real science of audio, is… well, in both cases, it’s anticipation.
Cue the Carly Simon…
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.
During a long discussion about vintage instruments in the studio this week, it prompted me to think about this excerpt from The Ultimate Guitar Tone Handbook (written with writer, composer and good buddy Rich Tozzoli) that describes some of the intangible factors that went into manufacturing Gibson humbucking pickups in the 50’s and 60’s.
As you’ll see, there are a lot of external factors that went into making a pickup back then, and those factors can pretty much be applied to all instruments in one way or another.
————————————-
“As if the known factors in building a pickup weren’t enough, consider the many intangible factors as well. For instance, most pickups loose their magnetic strength over time because of environment and electrical interference. Pickups can become weakened or demagnetized completely by leaning your guitar against an amplifier with large transformers, or even from taking your guitar too close to the train motor of a subway (as happened with Andy Summers of The Police).
Another intangible is the fact that tolerances of just about every component were much looser until the 90’s. While the difference was indeed subtle, add enough components at the edge of their tolerances together and you suddenly get a pickup that sounds different even though it’s made the same.
Manufacturing intangibles are a whole other story and for that we’re going to go a bit into the history of the Gibson humbucker.
The Changes In The Humbucker
The first humbucking pickups on the 1957 models of Gibson guitars had a sticker on them saying “Patent Applied For” as the design was in the review cue before being granted a patent (see Figure 3.27). These became known as PAF pickups (“Patent Applied For”) and have become highly sought after today for their great sound.
The problem is that most PAFs sound different from one another due to manufacturing process of the time.
Figure 3.27 A Gibson PAF Humbucker. (click to enlarge)
Until 1961 when Gibson standardized the selection process, they randomly used different strength magnets (grade 2 through 5) in their pickups, which accounts for some of the reasons for the different sounds. To make matters worse, sometimes a shorter magnet was selected (mostly seen in gold-plated guitars for some reason), which decreased the power of the magnet as well.
In July of 1961, Gibson consistently began to use all short Alnico 5 magnets, although occasionally a few Alnico 2’s showed up. In 1965, Alnico 5’s became standard in all pickups, which finally brought about a bit of consistency to the process and the sound.
If that weren’t enough, the number of windings on the pickup varied enormously as well, especially in PAFs. The early coil winding machines didn’t have an auto shut-off so the workers would shut off the machine when the bobbin looked full, which was at about 5000 turns. As a result, no two pickups were ever the same.
Even when Gibson bought a winder with an auto-stop, there continued to be problems even though the pickups became more consistent. The stop mechanism was controlled by a fiber wheel which would wear out and break, at which point the workers would approximate the number of winds by timing the wind, which resulted in more inaccuracies.
Since the humbucker is made up of two coils, sometimes the windings of each coil were different even though the total number of turns were correct. This would cause certain mid-range frequencies to stand out and give it more bite.
Figure 3.28: A Gibson Patent Number Pickupr. (click to enlarge)
By mid-1962, the patent for the humbucker was granted and Gibson changed the sticker to read “PATENT NO 2,737,842” which was actually the patent number for Les Paul’s trapeze tailpiece. No one knows for sure if printing the wrong number was merely a mistake or a way to throw off the competition.
From 1963 to 1975, these “Patent number” pickups are very consistent, as are the ones thereafter when new, more precise winding machines were used (see Figure 3.28).
In the 1990’s, Gibson further refined their manufacturing and began to manufacture pickups based on the original PAF design. Thanks to precision modern manufacturing techniques, these pickups are remarkably consistent, which also means that a “magic” pickup made as a result of loose tolerances is no longer possible to get.
That being said, most experts agree that you can now get 90 percent of the way there sound-wise for 10 percent of the cost of a vintage PAF.”
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.
Everything You Wanted To Know About Sound Level Meters (SLMs)
The primer: what, how, why, what's available, techniques, applications and more
A sound level meter (SLM) is a device used to make frequency-weighted sound pressure level measurements displayed in dB-SPL.
0.0 dB-SPL is the threshold of hearing, and is equal to 20uPa (microPascals). This correlates to what one would aurally perceive when in a deep cave or in a large anechoic chamber.
Packaged as a single-function handheld test device, SLMs are intended to be held at arm’s length during measurements (to reduce the effects of the body on the measurements) or secured to a tripod stand for more stability.
All SLMs feature an omnidirectional measurement quality condenser microphone, a mic preamp, frequency weighting networks, an RMS detector circuit, averaging circuits, the meter display, AC and DC outputs used to feed other measurement devices or for recording (see Figure 1, below).
Most SLMs have the same set of user adjustments, including SPL range selection, A and C weighting filters, slow and fast detector response, and minimum or maximum SPL.
The SPL range switch provides a balance between minimizing the preamp noise level and measuring a wide range of sound pressure levels.
Most of the commonly available SLMs measure from about 30 to 130 dB-SPL and do this in 3-4 ranges.
Figure 1: Functional drawing of a basic SLM (click to enlarge)
The more advanced and expensive SLMs feature removable microphones, 1-octave and/or 1/3-octave filter sets, additional weighting filters including B, D and Flat or Linear (no filter), additional detector response options (Impulse and Peak), averaging (over time) and data logging or storage (either on-board, as computer files or both).
Almost all SLMs are designed and specified to perform to one of four internationally standardized levels of accuracy:
Table 1: Permitted tolerances as defined by the IEC 60651 and ANSI S1.4-1983.
Note that these tolerances are at 1 kHz, the standard calibration frequency for SPL measurement. In order to ensure the flatness of the SLM there are additional tolerances specified for various frequency bands and microphone classes as well.
Class-0 SLMs are employed primarily to calibrate other SLMs and may be used for very high precision noise measurement in controlled spaces and/or for academic research.
Class-1 and Class-2 SLMs are most widely used by acousticians, sound system professionals, industrial designers/ manufacturers and researchers in academia and government. Measurements made with these levels of accuracy are generally acceptable as evidence in the resolution of legal disputes.
Class-3 SLMs are restricted to noise survey meters and dosimeters.
Microphone Sizes
Most general purpose SLMs are provided with a 1/2-inch free field microphone, permanently attached to the SLM body.
Higher quality SLMs with removable microphones may be outfitted with capsule sizes ranging from 1/8-in to 1-in.
The smaller microphone capsules (1/8-inch and 1/4-inch) have three primary advantages: —They are capable of higher SPL measurements; —They remain omnidirectional up to higher frequencies; —And they present less disturbance of the air due to their reduced physical size (the mic itself has less impact on the measurement).
Larger 1-inch capsules exhibit less self-noise and are therefore best suited to measurement in very quiet spaces, of very low-level noise sources, and of low-frequency energy.
Microphone Classes
One can also choose between three different classes of measurement microphones: Random Incidence, Free Field, and Pressure.
In reality, these three classes of microphone are all fundamentally “pressure response” microphones and are designed to sense pressure - not pressure gradients or particle velocity.
The difference is primarily in how they pick up sound at high frequencies, and this relates to how they are positioned relative to the sound source.
At higher frequencies (where wavelengths become so small as to equal, or be smaller than, the size of the microphone diaphragm) the direction in which the microphone is aimed - relative to the direction of the sound being measured - will have an effect on the accuracy of the measurement.
Random Incidence microphones are intended for use in a diffuse sound field where the sound is arriving from all directions due to a high level of reflections.
This type of microphone must be angled approximately 70 degrees to the sound source in order to yield accurate measurements.
Free Field microphones are designed for use in an open space (free of reflections) and should be held at 0 degrees to (pointed directly at) the sound source.
Pressure microphones are designed specifically for measurements in couplers and closed cavities such as plane wave tubes, ear cavities in dummy heads and for noise measurements in air ducts.
Some SLMs provide a switched correction filter allowing one type of microphone to be used in place of the other.
Note: In general, a “measurement quality” microphone differs from a recording or sound reinforcement microphone in its flat response, the absence of an onboard preamp, and its less rugged construction.
Detector Averaging Time
The standard averaging time choices provided by the majority of SLMs are Fast and Slow.
The Fast setting has a 125milliseconds (mS) averaging time (developed in 1950 to match the ballistics of the moving coil meters used at that time) and is useful for steady state (not dynamic) sound.
For measurement of more dynamic sound (when the Fast setting results in fluctuations of 4 dB or more), the Slow setting provides a 1 second averaging time resulting in much more consistent and stable readings.
It is this setting (along with an A-weighting curve) that is most frequently used for measurement of pop music SPLs.
Some higher priced SLMs also provide Impulse response time (four times faster than the Fast setting) used for extremely dynamic and short term (impulse) sound and peak response time, which accurately responds to signal pulses as short as 50 microseconds.
Frequency Weighting Filters
Because we humans do not hear frequencies in a linear manner (as illustrated in the often referred to Fletcher-Munson curves, aka: equal loudness contours), sound level measurements made with a flat response do not accurately reflect how we perceive sound.
When measuring pop music sound reinforcement systems (with the program signal containing generous amounts of bass energy), broadband readings would be artificially high without the frequency shaping provided by the A-weighting filter.
C-weighting, with its more gradual low-and high shelving filters, is used for very high SPL sound system and noise measurements.
On SLMs without a Flat setting, the C-weighted setting may be used to feed a spectrum analyzer such as Rational Acoustics Smaart for frequency analysis of the broadband sound.
Table 2 below, and then Figure 2 below that, show the frequency response for A and C weighting filters, plus two other weighting filters employed for more exacting and specific noise measurements.
Table 2
All weighting curves are equal in level at 1,000 Hz, the standard reference frequency that SLMs are calibrated to using acoustic calibrators.
Note On SPL Annotation: When communicating or recording (speaking or writing) SPL values, it is necessary to annotate what - if any - weighting curve has been employed. A-weighted SPL is written or stated as “XX.X dBA” and C-weighted measurements as “XX.X dBC”. Such annotation eliminates the need to provide the “SPL” suffix.
The absence of weighting notation implies that a Flat setting (no weighting filter) was used and in this case the suffix “SPL” must be used to differentiate from other, non-acoustic decibel values.
Historical Note: The A, C and B frequency weighting curves were first developed for specific sound levels based on equal-loudness contours.
Figure 2: Standard A, B, C and D weighting curves (click to enlarge)
Because equal-loudness contours were developed from research utilizing pure tones, it later became evident that the weighting curves chosen for SLM measurements at certain levels do not correlate very well to how we actually hear complex sounds.
More recently it has been found that the A-weighting curve provides a realistic approximation of how we hear over a fairly wide range of sound level measurements.
Display
With very few exceptions, modern SLM’s are provided with LCD displays with 3- or 4-digit readouts. Most of these provide 1/10-dB resolution except in the case of 3-digit displays when they go over 100 dB-SPL.
Prior to LCD technology, analog meters were employed and these suffered from all the inherent shortcomings of this method of display: slow rise and settle time, susceptibility to damage from pinging (over loading) and overall mechanical fragility.
Another shortcoming of analog meters is the difficulty in establishing stable readings due to the range selecting that is common in most SLMs. One must take the range that is selected and then add or subtract from the meter position accordingly.
Measurements made when the sound pressure level has changed and is out of range are prone to be inaccurate, as well. Some feel that fast impulse noise measurements are easier to read with an analog SLM than with a “digital” SLM.
Both styles of display are difficult to read in dim lighting but some SLM’s feature backlit LCD’s for this condition. And either type of display may be quite difficult to read while at arm’s length if the display size is too small.
Equivalent Continuous Sound Level
Although we live sound folk generally believe that we are able to accurately determine the overall sound pressure level when we view the displayed real time values on our SLM’s, these readings are actually no more than a glimpse, or snapshot, of the varying sound pressure levels that we are exposed to.
Due to the dynamic content of the sounds we normally work with (this applies to both speech and music) and our own human shortcomings, we are unable to mentally integrate the frequently changing SPL’s and determine an accurate mean SPL value.
Research into hearing safety and annoyance and the standards that have been developed from this research are based on the accumulative exposure to sound energy over periods of time.
To this end, integrating SLMs have been developed to provide measurements that are conducted over a selected time period and averaged so that they provide an accurate equivalent energy value, designated as Leq (LAeq or LCeq when weighting is employed).
Integrating SLMs also provide Lmax and Lmin values, and the user can choose the time period that the Leq averaging takes place (normally from several seconds to 24 hours).
More and more organizations and governments are adopting this as a more accurate means for establishing risk of hearing damage, community noise standards, annoyance issues, etc.
Although stand-alone integrating SLMs are considerably more expensive than basic SLM’s, some software is available to provide this statistical averaging from the AC output of any SLM. and both the TerraSonde Audio Toolbox and Neutrik Minilyzer provide this function along with other SPL measurements and other useful audio system measurements.
Relative Vs Absolute Measurement
Absolute measurements are those made with calibrated SLMs that provide actual sound pressure levels referenced to 20uPa (0 dB-SPL).
Such absolute SPL measurements may be used with and accurately compared to those provided by other calibrated and equal quality SLMs.
Conversely, measurements made with un-calibrated SLMs or those with of low quality (including less then linear frequency response) do not accurately indicate actual sound pressure levels.
SLMs that have been incorrectly user-calibrated or have simply lost their calibration over time also do not provide accurate absolute SPL values.
Relative measurements are simply comparisons between sound pressure levels and do not require absolute calibration nor other “precision” performance characteristics such as a flat frequency response.
Non-calibrated and lower-quality SLMs do provide meaningful comparative measurements as long as the same SLM is employed for these measurements and the measurements are made in the same manner.
Example: A Radio Shack SLM, which is specified as providing +/-2.0 dB-SPL accuracy and which has a less-than linear frequency response, is still able to provide realistic indication of the changes in relative sound pressure levels even though it does not provide accurate measurement of absolute sound pressure levels.
Likewise, the SLM function contained in Smaart may be used “as is” (without calibration) for accurate relative SPL measurement.
One relevant and valid relative SPL measurement that can be made in sound reinforcement work is determining the even-ness of coverage provided by a loudspeaker system.
In this case virtually any SLM is able to indicate any deviations that may exist in the sound pressure levels as the SLM is moved about the coverage area of the loudspeaker system.
In a similar manner we can observe changes in relative SPL as the gain of a sound system is brought up and down and we can accurately determine changes in SPL as a noise source (machinery, for example) is turned on and off.
Available SLMs
As stated previously, SPL measurements made with Class-2 and Class-3 SLM’s provide sufficiently accurate (within +/- 1.0 dB) readings for what we require in sound reinforcement work.
Until recently, SLM devices that met this level of accuracy were prohibitively expensive ($1,200 and up).
Most of the old classic SLM’s are cumbersome to set up and, due to their inherently fragile construction, are seldom used for touring sound or on-site installation work.
Without question the most common SLMs used by live sound folks have been those made by Radio Shack. These appear to represent a very good value, are easy to use and are readily available almost anywhere in North America.
In researching for this article I have found that, along with all of the sound system operators and contractors who own a Radio Shack SLM (or two), there are large numbers of other users including university physics/acoustics and biology departments, municipalities, hi-fi enthusiasts, and theater system installers and property management companies.
And there are at least a few websites devoted to using the Radio Shack SLM for tweaking hi-fi, project studio and sound reinforcement loudspeaker systems and for simple noise measurement.
The original Radio Shack model 33-2050 utilizes an analog meter display and has been in production for the past few decades, and several years ago, the model 33-2055 was introduced, providing a “digital” (alphanumeric) LCD display, recessed and better-protected switches and a maximum/minimum measurement function.
Figure 3: Radio Shack 33-2050 (left) and 33-2055 SLMs (click to enlarge)
Otherwise, these two models employ the same microphone, base circuitry and battery complement.
Alternative SPL Measurement Tools
One alternative to the stand-alone SLM’s described in this article is the SLM function included with some of the better portable RTA’s (real time analyzers) that are often employed for live sound and acoustic measurement.
Models from Ivie, Audio Control, and GoldLine all provide accurate broadband sound level measurement (Class-2 or Class-1, depending on the microphone used) plus octave and 1/3-octave spectrum analysis.
Figure 4: A-weighted frequency response for Ivie IE30 (red), Smaart with MK-10 microphone (green), Terrasonde Audio Toolbox (violet) and Radio Shack 33-2055 (blue). (click to enlarge)
This combination of measurements is very useful for determining both the SPL and the frequency components of the sound/noise under con sideration.
Most, if not all, of the current crop of computer-based electroacoustic measurement system provide an SPL function along with their frequency and time-related measurements. Of these, SIM from Meyer Sound and TEF from Goldline are sold with microphones and are calibrated at the factory.
Smaart (any version), SpectraFoo, SysSid, PC-RTA and other similar systems utilize third-party microphones and preamps that must be calibrated by the end-user, using an acoustic calibrator or piston.
Once calibrated, these systems provide very good accuracy and feature a number of variables (weighting filters, etc).
Unfortunately, they are not really “handheld” nor nearly as portable as stand-alone SLMs. And (most importantly) they do not retain their calibration from site to site and therefore require an acoustic calibrator for continued use in measuring absolute SPL’s.
Another product that is a variation of a SLM deserves mention here. SPL “management” systems have been available for perhaps a decade and are directly related to the issues of sound system volume levels.
These simply employ microphones and a software-based SLM system that provides either an obnoxious visual warning or more extreme reactive functions (such as muting of the sound system signal!) when the predetermined SPL limit has been reached or surpassed.
Calibration
In situations where the accuracy of sound level measurements must be recorded and/or verified, it is necessary to calibrate the measurement system both before and after the measurements are made.
Software-based electroacoustic measurement systems that feature an SLM function also need to be calibrated if used for absolute SPL measurements.
An acoustic calibrator provides the standard 94 dB or 114 dB (or both) 1 kHz test tone and should also include the proper size opening or adaptor rings for the microphone(s) you intend to employ.
As is the case with SLMs, calibration devices are manufactured to several levels of tolerance (see table near beginning of article) and they have historically been prohibitively expensive.
Recently there have been several acoustic calibrators introduced that provide very good accuracy for sound system and acoustics measurement.
SLMs may also be calibrated with a pistonphone, but the cost of this device is prohibitively high and the high level of accuracy it provides is not practical for most general sound system related measurements.
Why We Use Them
The simple and snappy answer as to why we use SLMs: so we can know what’s going on.
The need for accurate and repeatable sound level measurement is due simply to the tendency for humans to aurally perceive sound in a very subjective manner. This applies to all aspects of sound perception in the audio field: frequency content, mix “correctness”, system EQ, dynamics, etc.
At some point or another we will disagree on the relative loudness being produced plus these other subjective sound qualities.
Whereas other electroacoustic measurement tools remain very subjective in how we interpret the data they provide, a reasonably accurate SLM provides an irrefutable means to objectively verify how loud the sound is (or is not).
SLM measurements may also be used to verify exactly how much the sound level has been changed (as noted in the section on relative versus absolute measurement).
Anyone involved in live sound system work is aware that there is an ongoing and increasing problem with the issue of volume levels produced in public performance, and specifically, in pop music reinforcement.
It’s wise for anyone involved in this field to become familiar with the actual measured numbers, not to mention the need to continually monitor what we are doing. Issues of personal health and legal liability are an ongoing and increasing concern.
Measurement of unintended noise levels and/or the effectiveness of acoustic treatment are additional exercises that most live sound folks eventually will become involved in.
How We Use Them
For basic every day sound system level measurement, using an SLM is fairly straightforward.
The standard SLM employs a free field microphone and is intended to be held at arm’s length and tipped up slightly towards the sound source.
As you move away from the source and into the reverberant or diffuse field, or when measuring in a multiple source sound field such as on stage, the angle of incidence of the microphone becomes much less of a concern.
When measuring noise levels at installation sites, some care should be taken. If measuring the noise level of a rack room, for example, be sure that the SLM is positioned away from walls, floors and other large boundaries.
Signal-to-noise levels are seldom an issue for loudspeaker system measurement because we can simply turn the system up, more than far enough above the noise floor.
But when measuring the noise from a loudspeaker system (hum, hiss and/or the fan noise that may exist in self-powered systems), or from a mixing console and its power supply, etc., or from other common noises such as HVAC systems, vacuum cleaners, power tools, etc., the unintended noise must be at least 10 dB-SPL lower than the noise being measured.
When measuring and logging levels to help resolve a dispute, both A and C weighted measurements should be provided, as should the distance from the source and any other detailed notes.
Also don’t forget that a non-annotated measurement (“00.0 dB-SPL”) will be interpreted to be flat (unweighted).
General Measurement Guidelines
When conducting sound pressure level measurements there are a few basic rules that apply:
• Keep the SLM at arm’s length, aimed in the direction of the sound source(s) and positioned away from room boundaries. By the way, the mixing surface on a large (40-plus channel) mixing console will effect SPL measurement when the SLM is laid down on the mixing surface or angled against the meter bridge.
• For almost all sound system measurements, use the A-weighting filter and Slow response setting.
• Be sure that you’re measuring within the range that you have selected. SLMs with multiple range positions and those with analog displays, in particular, may provide incorrect readings when the SPL is over/under the selected range.
• Prior to making SPL measurements, ensure that the SLM has been turned on for a few minutes and has reached room temperature.
• Wind and air-blowers will effect SPL measurements.
• Be aware of where you are taking your measurements. For example: if you measure from within the area where the bass energy from the sound system has combined (aka “power alley”) or within a standing wave null, the readings will be non-representative of the SPLs for the majority of the audience.
Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.
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