Thursday, October 17, 2013
Decisions, Decisions: Thinking About Upgrading Your Inventory?
There are a whole lot of variables beyond sound that come into play
Like most production company owners, I regularly take stock of my equipment inventory, evaluating the need for updates or additions to match current and future needs.
A recent audit led me to consider investing in higher powered 12- or 15-inch-loaded stage monitors.
The process of investigating what matched our audio requirements with what’s available in the marketplace emphasized that there are a whole lot of variables beyond sound that come into play regarding the acquisition of new gear.
All businesses, no matter what field of focus, face many of the same challenges when confronted with the task of updating inventory.
The biggest issue is return on investment (ROI): will the equipment pay for itself and hopefully make money for the business before becoming outdated and/or obsolete?
Because if not, unless you’re a trust-fund baby with unlimited funds to blow on gear, it’s best not to spend the money.
I’ve seen this seemingly simple concept get a lot of sound/production companies in big trouble over the years, largely because it’s not quite as simple as it seems. Here’s how I go about it, using my recent “wedge quest” as an example.
To begin the process, I evaluated current needs and inventory. My company mainly does corporate gigs requiring small-format monitors to provide speech and audio playback onstage for presenters. We also encounter a wide variety of performers at corporate events, including dancers, acrobats, singers and bands.
The inventory already includes small dual 5-inch monitors that handle speech and cueing applications, and compact 10-inch coaxial boxes that are great for theatrical performances, singers, and bands that don’t require a ton of volume out of a wedge.
For larger/louder acts, we have older 2-way cabinets with 15-inch woofers and 2-inch compression drivers on horns. They’re big and heavy, and while they pretty much still do the job sonically, age is becoming a factor.
Looking to the future, I see us continuing to serve the same types and quantity of gigs, but there’s also an opportunity to do more festivals and one-off concert shows.
Now, here’s the point where I’ve seen some folks get into trouble, because they tend of foresee a “best case” scenario that rarely materializes. I take the opposite approach, playing out the “worst case” scenario, and then go to a “realistic case” of somewhere in the middle of both extremes.
Worst case: we get no new business. Best case: we book several new gigs a month. Realistic case: we’ll probably realize some additional business if we work for and earn it, but it’s not easy to ascertain just how much.
Decision: initially rent the appropriate wedges as we need them, and if this new direction gets solid traction, invest in new wedges. Caveat: our current larger wedges are quickly becoming outdated, so perhaps adding newer models is justified anyway.
The next thing I looked at was our budget.
Figuring out how much to spend on a purchase is one of the hardest things any businessperson has to do.
There are so many variables to take into consideration – purchasing outright verses leasing, leasing with the option of buying, purchasing new or used, and financing options.
The advantages of leasing include limited or no down payments, the ability (in many cases) to get a tax deduction on lease payments, and not getting stuck with gear that may become obsolete in a few years as technology changes.
The disadvantages are that it costs more to lease, and you won’t own it when the lease is up. Leasing with the option to purchase at the end of the lease solves that last problem, but will add significantly to the cost of the item.
Purchasing works best for items that will provide long, useful service, not likely to become outdated for the foreseeable future. It’s a good idea to investigate if there’s a significant technology upgrade or innovation coming in the product category.
In my case, I was looking at stage monitors, and while new improved models hit the market regularly, they don’t invalidate existing units. (Advantage to purchasing.)
One option that can get overlooked is buying used. Well cared for used gear can be often be acquired for a fraction of the cost of new. Equipment no longer providing ROI to the current owner may be the right fit for your needs and budget.
Further, some companies like to invest in the “latest and greatest” gear, meaning that their used stuff might not be all that used and therefore provides strong value.
Having performed some strategic thinking and with a rough budget in place, here are some additional factors that I considered.
First, note that I said a “rough” budget because gear shouldn’t be chosen based solely on price.
For example, reliability is important to me. My clients don’t want excuses as to why something has gone wrong at a gig, so a product’s reliability is of paramount importance.
Look into warranty specifics and service options. Shipping a large heavy item across the country to the only authorized service center that can fix it might tie up your investment for weeks or even months. Worse would be paying for shipping.
Further, assume any unfamiliar piece of gear will need to be used well within its defined capabilities – if you foresee running it at the upper end of its operating range, consider stepping up to something bigger. This might cost a little more now, but will save money later in terms of reduced downtime and maintenance costs.
The same goes for the feature set. For example, buying a console with more inputs or a snake with more channels now may seem like overkill, but if you anticipate needing these features in the future, go for it now.
This brings us to expandability. Some products, especially software-based ones, offer the ability to upgrade and expand without having to purchase an entire newer unit.
Rider acceptability can be a big consideration. If you have to meet riders on a regular basis, this may limit your choices. But if you only deal with a few riders each year, and don’t expect that to change, then renting an item for those few instances probably makes financial sense. Do the math to be sure.
Another factor is size and weight, as well as truck packing. Most of my company’s work is here in town, providing modest-sized systems, so it isn’t nearly as much of a concern as it is to those providing large-scale concert systems hauled hundreds of miles in a big rig.
Still, I don’t like lifting heavy pieces like monitor wedges whether it’s in the shop or on the job site, so smaller and lighter is a plus.
Cross-rentability is a further consideration. Some companies choose equipment such as loudspeakers and truss based on what their friendly competitors provide so it’s a snap to rent more when needed. Rentability can also be a consideration if you rent equipment to the general public.
As a smaller company, we look for gear that can multi-task, filling more than one role. For example, our 10-inch coaxial stage wedges have a stand mount so they can be placed on sticks for main and fill duties.
However, while multi-tasking is great, make sure that the item can do the additional job(s) adequately.
Finally, evaluate how a purchase might affect existing system infrastructure. For example, if new electronic gear doesn’t fit into an existing rack, the expense of a different, larger rack also becomes part of the equation, and then, will that new rack be more difficult and/or expensive to transport and store?
Further, will the electronic gear require different cabling and connectors? Will it incur training costs getting the crew up to speed? Will it require another crew member, or conversely, allow us to use one less crew member on certain gigs?
Every new piece of gear needs to undergo this type of scrutiny, even though some of it’s not obvious. And this leads to another question: beyond issues of ROI, will an upgrade matter to your clients?
Certainly our goal is to always do the absolute best for them, but will the added investment make a noticeable, justifiable difference in your service to them, while also serving your own bottom line?
Ultimately, I’ve decided in favor of new stage monitors, and am now in the process of investigating specific options. It really is time to retire the old wedges due to reliability, performance and aesthetic issues, and we also have decent prospects for expanding into new markets.
And did I mention I hate lugging around big, heavy, ugly wedges? Sometimes an aching back, like a squeaky wheel, gets its way.
Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.
Wednesday, October 16, 2013
Consoles & Mixers For Church Sound, In Context
Options are plentiful and growing in this dynamic marketplace
Over the past decade, we’ve seen the proliferation of digital consoles in both live and recorded sound. This is not to say that analog consoles are going away at all, particularly with respect to house of worship systems.
In fact, it’s a very safe bet that the vast majority of church systems are still headed by an analog console.
Hector La Torre, Managing Partner and producer of the national HOW-TO Sound Workshops, notes that while digital consoles are a major topic of discussion at his organization’s audio education seminars presented to more than 1,500 church sound personnel annually - largely volunteers - throughout the U.S., most churches are still hesitant to dip their toes into the digital technology stream.
“There are two primary factors with respect to the bulk of the church sound market - cost and complexity,” La Torre explains.
“Digital consoles have largely been out of the price range of all but the largest churches, and while digital mixers are not necessarily more complicated to operate, keep in mind that about 95 percent of church sound system operators are volunteers who have limited experience, and that’s who is being asked to take on a new learning curve.
“Although most churches and volunteers who take on digital consoles find that they become more efficient and proficient at their job, some still hesitate because of the initial learning curve and overall lack of knowledge of the technology.
“A board for a higher end professional application like a tour or a performing arts center is usually an upgraded version from what you’ll find being used in your average church service,” he adds.
“And in the mainstream of the professional audio marketplace, new technology often wins out over cost issues, but it’s pretty much the opposite with the majority of churches.”
“That’s why education is the key to the future of digital consoles in worship. If church folks don’t know or understand a technology, they won’t adopt it.”
Both aspects are changing, with manufacturers now increasingly introducing digital models in line with church budgetary needs while maintaining functionality and feature sets to meet all but specialized applications.
Affordable digital consoles cited by La Torre are the Yamaha LS9 Series, as well as models from Tascam and Soundcraft, and he’s also talked with a number of other manufacturers who indicate they’re quickly moving in the same direction.
On The Upswing
Some context about the church market is in order. There are an estimated 450,000-plus churches in the U.S. alone, and three-fourths (and likely more) of that number is comprised of venues offering seating for 500 or less, with the norm in this range being 250-300 seats.
While we read about sophisticated church sound systems (that often include one or more digital consoles) on a regular basis, these are often deployed at larger venues ranging in scope up to the “megachurch” realm.
The production needs at most churches are not nearly as ambitious, budgets follow that scale, and the “technical staff ” is made up of a few volunteers who might spend their weekdays selling insurance, driving a truck, teaching school and so on.
Within this context, there are countless analog consoles and mixers that are at least 15 years old still working great and meeting expectations, and when a church is seeking a new board, their mindset has still tended to analog.
In general, worship services are generally one of two categories - traditional or blended/contemporary.
Let’s look at traditional first. Often there will be a pulpit microphone, an altar mic, a lectern mic, a wireless mic on the pastor, and perhaps a feed from a digital piano.
Chances are this type of system has been in place for quite a while, and the mixer is sometimes mounted in the rack and offers only volume control for the various inputs. The rack is stuck out of the way in a closet somewhere - “set it and forget it.”
If this mixer needs to be replaced, a similar analog rack-mount mixer with level controls only and a good equalizer (properly tuned) can create a reasonably acceptable result.
The downside is that there is no individual EQ for the various input devices, so all channels will have to compromise with the EQ needs of the other channels.
For those seeking more capability but wishing to stay in the rack realm, a digital rack-mount mixer is an option.
Once installed, the system contractor hooks up a laptop to the sound system, and via system software, uses the digital output equalization to give the sanctuary as close to a flat response as the loudspeakers and the room itself will allow.
After that, the contractor can set the digital EQ for each individual input device to get the most pleasing possible result for that specific channel. No compromises are necessary.
Further, a simple touch-panel can be provided at a remote location, allowing for several types of adjustments by the user without the possibility of damaging the loudspeakers or causing feedback.
Of course, another choice is to replace the rack-mount mixer with a small console/mixer that offers EQ on all individual input channels.
This can also present accessibility advantages, as well as locating the operator in the sound coverage field to make adjustments.
And, most of these models usually offer more than eight channels at attractive price points, making them well-suited to meet future system expansion.
Some traditional services - up through virtually all blended/contemporary services - need a true mixing console, hopefully located in the primary listening area and manned by a competent system operator.
The size of the console is determined by the number of channels needed plus future growth considerations, and even a few more for good measure.
Both analog and digital consoles/mixers definitely track with the rule of “you get what you pay for” - particularly in terms of analog, there are some inexpensive models that are surprisingly decent, and there are some absurdly inexpensive models that should be absolutely avoided.
The system operator is another key to the selection process. A skilled, experienced operator can make beneficial use of a more sophisticated, feature-laden console.
The control surfaces of early digital consoles/mixers were not very intuitive for the majority of church operators.
As noted, that’s changing rapidly, with most modern models offering user-friendly interfaces that track well with analog intuitions.
As a result, any operator with a decent amount of analog experience can now successfully perform basic digital mixing. This is probably even more true for younger operators, who have practically grown up using touch screens to scroll through menus and so on.
Digital console platforms offer several advantages for church sound applications.
The dedicated offering of EQ and dynamics to every single channel (in addition to scads of other operations) mean more highly tailored and adaptive sound, as well as saving the space and cost required for outboard gear providing the same capabilities.
Further, there’s a lot of value in the ability to save preset mix “scenes” for instant recall so that specific settings may be accessed instantly at the touch of a button.
This is quite useful for contemporary services that can feature dozens of performers and talkers appearing in rapid succession, as well as at churches where the first service is traditional, the second service is blended and the third service is full-blown contemporary.
It can also be a significant benefit to the pastor who needs to conduct a funeral service or a simple wedding ceremony mid-week without benefit of a system operator.
Times they are a-changin’ with mixing consoles, and it’s all good for churches.
Digital is coming on for very good reasons, but there’s still a plethora of options for those who prefer an analog workflow and layout.
Jon Baumgartner is a veteran system designer for Sound Solutions in Eastern Iowa, and Keith Clark is editor-in-chief of ProSoundWeb and Live Sound International.
Friday, October 11, 2013
The Old Soundman: Earplug Wearin’ Musicians & An Insolent Newbie
Reaching the boiling point with those who wear earplugs while demanding insane monitor levels
Dear Old Soundman—
Yes, SPA1! Were your parents named SPA.09?
I’ve enjoyed reading your comments for some time now.
See, even though you have named yourself after a decadent outdoor love tub, you’re okay!
I wonder if you might give us your insight regarding musicians who use earplugs (sometimes very sophisticated earplugs,) but then require ridiculous monitor levels on stage.
There was a lack of love in these people’s lives when they were children.
I run into more problems with drummers that wear earplugs than I’d like to.
I’ll bet you do! Back in the seventies, rock musicians weren’t a bunch of soy latte sipping sissies! They turned up their big old amps and beat the heck out of their clear plastic drums, and we gave them as much side fill as we could and then they did a show, dangit!
You, like me, are probably sick of these shaven-headed suburbanites, these tame domesticated animals with their silly “tribal” tattoos, their click tracks and their carefully assembled multi-layer loops and their $10,000 kits with the 15 cymbals. Don’t even get me started on the dyed black hair!
Kick, snare, hat, Little Lord Fauntleroy! A rack, a floor, a crash and a ride, Trust Fund Jimmy!
If these kids played music that made any sense, they wouldn’t have so much trouble following each other.
The devil was working overtime when he came up with his masterpiece, the drum machine! Bands that paid their dues in the old days, that played in bars night after stinking night, they understood how to strip it down:
“She’s got legs!”
“Round and round!”
“All she wants to do is dance!”
Every one of Sheryl Crow’s big hits! Boom bap! Boom boom bap! Huey Lewis, you know what I’m saying, you were just workin’ for a livin’, and you wanted a new drug!
Stevie Ray Vaughn, much love to Stevie Ray!
But noooo! These loco children of today are forced to go along with their idiotically effete keyboardist’s asymmetrical programming. It burns me up, SPA1!
Speaking of which, check out the nerve of this little creep Rik. He really knows how to push my buttons!
Yo there old guy,
You got some kind of problem? Why I oughta…
How did you get into the industry, and how did you get so disillusioned? (beats me)
I am a young budding 19-year-old trying to do a physics degree, (hahaha) and I would love to do live sound (no kid, get a good job).
I believe in moderation in life, as in not busting the guts out of some speaker (ooh goody) if we may savour a little respect for a love of music (so young, so innocent).
I have had a fair bit of experience, but I can never find anything past offering my services to people I know (what’s new?).
I love your humor, and have provided anticipated responses in () parentheses.
Rik, let’s explore the inner workings of your mind (if there’s anything there).
See? I can use parentheses, too.
I deduce by your use of some anglicized spellings that you are either across the pond in merrie olde redcoat land, playing your penny-whistle, or else down in Qz, happily making crazed guttural noises into your culturally appropriated (stolen) didgeridoo.
You seem to think that the “me” that exists in your brain is actually who I am. You think your perceptions of me, that virtual voodoo doll that you call the Old Soundman, is the entirety of my existence.
On top of that, you have defied my repeated admonishments to you pilgrims—that I handle the funny stuff. Your role is to just politely ask the audio questions and then shut up and step back so you don’t get any on ya.
I spray humor like a tribe called quest spits rhymes!
More people read me than read the New York Times!
Everybody applauds when my bell chimes!
I’m a master perpetrator who never gets punished for my crimes!
You’ve never had to suffer any really hard times!
If you don’t shape up, I’ll bust your grill with a roll of dimes!
And these are the breaks, Rik!
The Old Rappin’ Soundman
There’s simply no denying the love from The Old Soundman. Check out more from OSM here.
Church Sound: When System Failure Is Not An Option….
Keeping cables in working order under the rigors of heavy use
Noisy mic cables can happen to anyone almost anytime. No matter how careful you are XLR cables are prone to being stepped on, run over and pulled too hard by musicians, singers and, well, you. The result of all this abuse can be intermittent shorts, open circuits and noise issues.
Of course, your cable problems will often turn up in the most audible and important signal path, such as your Minister’s microphone or signal feed to your radio station, so here’s how to find and fix problems before they get out of hand.
First, identify the source of the noise. If you hear a crackling sound during your service, grab your headphones and start soloing individual microphones and instruments until you hear the noise in your own ears. You can now mute that channel — if you can get away with it for a song — or perhaps get your preacher to step over to his backup microphone.
At this point you’ve identified the signal path with the noise, but not the particular cable. So mark every cable in this signal path with a piece of gaff tape and pull them out of the sound system after the service for later testing and repair.
Next, test the cables! If you don’t have a cable tester, buy one now. For example, the Swizz Army Tester from Ebtech costs less than $100 and is a great option.
This versatile tester will check any combination of XLR, Phone, RCA and MIDI cables for shorts, opens, cross-circuits and grounded shields. It also checks for intermittent open circuits with a “Reset” button function.
After you plug in the cable, momentarily press the Reset button you’ll see the “Intermittent” lights go out. Then when you wiggle the cable around, any momentary break in the connection will cause the appropriate light to lock to the “on” position. Sure beats trying to watch for a light to blink off while you’re attempting to make a cable fail.
In our shop, we test every single XLR cable that’s going out on a gig, especially if it was used by someone else on another system. That way we’re not surprised by a bad cable at the worst possible time. If your cables stay “home” then at least a yearly verification of every cable is a good idea.
For cables that get moved around a lot, say if you’re a mobile ministry, once a month testing is indicated. However, not testing your cables regularly is just a failure waiting to happen during your worship service.
How many of you have tested your mic cables in the last month? Let me see a raise of hands… Hmmmm…
Mike Sokol is the chief instructor of the HOW-TO Church Sound Workshops. He has 40 years of experience as a sound engineer, musician and author. Mike works with HOW-TO Sound Workshop Managing Partner Hector La Torre on the national, annual HOW-TO Church Sound Workshop tour. Find out more here.
Tuesday, October 08, 2013
Church Sound: Finally Mixing on a Digital Board? Escape These Three Traps
With great mixing power comes great mixing responsibility
You hear talk of people transitioning into the wonderful world of digital mixing, but you never hear of what happens after they make the transition.
Old habits must be broken, a new way of thinking about workflow has to occur, and digital mixing doesn’t mean you can finally perfect a vocalist’s mix…at least not for two weekends in a row.
The Three Traps Of Transitioning From Analog To Digital Mixing
1. Don’t assume last week’s settings are perfect for this week.
Digital mixers give you a massive amount of EQ control over each input. While I’m grateful for graphical EQs, it’s easy to set them “perfectly” for each musician one weekend and think the next week those settings will still be “perfect.”
Week-to-week, a lot of factors change. Guitarists use different guitars, different pedals, and different effects, all according to the song arrangement. Oh yeah, and then what works for one arrangement doesn’t work for another.
I’m not against using the previous week’s EQ settings as a basis for the mix, but don’t assume it doesn’t have to change.
2. Before setting your gains and faders, check your group levels.
This one still gets me from time-to-time. Coming from an analog work, it’s easy to look at your board and know exactly where your group volumes are set; these could be your groups, DCAs, “SUBs”, whatever your board uses and calls them.
In the digital world, where some mixers work as a “surface” where the faders represent whatever channels you have selected, you might not see your group level fader settings unless you select the mixer’s surface to show them. Yamaha M7CL users know what I mean with the DCA button.
3. What you see is not always what you get.
In the analog work, you can look down at the mixer and see all of your settings (rack unit settings excluded.) In the digital work, what you see in front of you is only a small representation of what’s actually set for a channel. And in some cases, the digital screen before you might not be the same as the channel which you are focused.
For instance, on the M7CL, there is a bank of faders and controls on the mixer which are tied to the bank of channels displayed on the screen. If you change the view to a different bank, say channels 1-8, but think you are on channels 10-16…you are changing the wrong channel. I saw this happen to a tech when he thought he had un-muted a microphone but was working with a different bank of channels.
Additionally, when a problem occurs during a service, such as with a microphone channel, using an analog board, you can scan over the whole board and spot an incorrect knob setting. With digital mixers, you don’t get that ability. Look to the channel you believe to be the problem and make sure you select that channel so your display settings are for that channel.
The Take Away
With great mixing power comes great mixing responsibility. Digital mixers give you a lot of control but to be used effectively, you must know how to use them…and never assume that what worked last week will work this week.
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, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
Wednesday, September 18, 2013
Properly Cleaning Mixing Console Faders
Cleaning a fader is not brain surgery, but it takes practice and a lot of care. Here's how to go about it - successfully.
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
I’ve cleaned a lot of faders over the years and suppose I’ve gotten a little bold when it comes to tearing a fader apart and giving it a good bath.
And I’ve learned the hard way just how much punishment a fader can take before it breaks.
In some cases, a certain amount of brute force is required to crack open a fader, but then a certain amount of gentle finesse is needed to clean its individual parts.
I recommend practicing with old junk faders – without experience, it’s all too easy to ruin a good one.
Cleaning a fader is not brain surgery, but it takes practice and a lot of care.
Before we getting into a total fader rebuild, let’s talk about quick cleaning. Much of the time, if the gear has been well cared for and the faders are not too dirty, then a little routine maintenance is all that is likely required.
Besides, keeping faders clean is always a good idea, preventing dirt from becoming embedded deeper inside where it can cause more wear and tear.
Keep in mind: with relatively new faders in particular, do as little as possible in order not to undo the original lubrication. And overall, don’t go any further with this process than you feel you need to.
The first step is to use compressed air to blow as much dirt as possible out of the fader.
Figure 1: Start by blowing one end, and then the other. (All photos by Alex Welti)
There is usually “dust bunnies” in the fader that will come out easily, and this might be all that needs to be achieved in terms of cleaning.
Move the fader carriage to one end and blow air into the slot aiming away from the carriage so that dust can escape through the slot. Then move the carriage to the opposite end and blow air aiming the opposite way.
Skip this step and compound the laziness by spraying some off-the-shelf cleaner-lube into the fader, and it’s likely that the dust bunnies will be matted down and stick in the corners.
Laziness can lead to temporary improvement but later, the dreaded “dust bunnies” in the corner syndrome.
A fader might seem to work better for a while, but this won’t last and might lead to the need for a more substantial (and time consuming) cleaning effort.
Note that the compressed air must be clean and dry.
I do a lot of cleaning, so I’ve invested in a $100 air compressor and then added an air filter / dryer unit for about $40. To this I’ve added a dryer cartridge that contains silica beads for about $5.
If an air compressor isn’t available, cans of aero-duster will work, but they don’t last long.
If the plan is to clean a couple/few consoles, an air compressor is a worthwhile investment, and it helps do the job right because you don’t need to be worried about running out of air.
In addition, the compressor will offer higher pressure.
Most canned air provides about 60 psi, with this dropping as the can is used.
With the compressor, I’m able to set pressure at a consistent 80 psi, which works very well. (And I found out the hard way that 100 psi will blow some faders and switches apart!)
If the initial “blowing out” process didn’t offer the desired results, it’s time to move on to use of chemical contact cleaner.
Figure 2: Contact cleaner outfitted with a nozzle that adds precision and cuts waste.
Some faders have lubricating grease applied by the manufacturer, while others employ a self-lubricating Teflon-type of plastic.
If used sparingly, chemical contact cleaner shouldn’t impact the self-lubricating type, but it will invariably wash away lubricating grease.
The goal is to avoid adding any more lubrication than is absolutely necessary - dust tends to fall away from dry surfaces, but it sticks to oily surfaces.
Figure 3: “Snap together, snap apart.”
After spraying contact cleaner, exercise the fader and then quickly blow out the excess cleaner.
This helps to spread the cleaner over the entire fader surface, while the excess cleaner carries away additional loosened dirt.
I’ve tried several types of contact cleaner since canned Freon was banned from the market.
There are a lot of good choices – my preference is Contact Cleaner II made by Techspray. It’s about $30 per can and worth the price. Note that I also invested another $30 for a screw-on trigger nozzle so that I can be precise and cut waste.
The fader is still feeling a little rough? Time to try a little lubrication. The key word is “little” – use as little as possible.
Did I mention not to use too much lubrication? Third time’s the charm – lubrication collects dust, so don’t overdo it!
Depending on the type of fader, I use a precision dropper to place just a few drops of lubrication in the fader, or give it just a quick squirt.
Exercise the fader and then blow away the excess with compressed air. Again with the compressed air?
Figure 4: The basic parts of a typical fader, and where they’re located.
Seriously, this helps spread the lubrication into a thin film and gets rid of any excess.
I’ve had good results with a spray lubrication called Tefrawn, made by Rawn. It’s Teflon-based and beneficial to the self-lubricating type plastics noted earlier.
Also, it smells like bananas, not that it matters!) Caig also offers products of this type.
Lesson learned the hard way: some oils react with plastic, causing it to break down. If there’s any doubt, test it out on a spare fader first before applying.
Figure 5: Be careful not to damage the wiper, which can ruin the fader.
Also, certain faders use thicker grease that results in a “smoother” feel, and these may actually feel too loose after lubrication.
If this proves bothersome, use silicon or petroleum grease (but not bacon grease!). I’ve found this step to be more trouble than it’s worth - if “feel” is that important, buy new faders.
Time to reiterate: “Air > Cleaner > Air > Lubrication > Air” About 30 seconds of effort for each fader.
Figure 6: Under and around the rails, but don’t touch the carriage.
Some of the more expensive faders are designed to be easily taken apart for cleaning.
If less expensive faders can’t be cleaned using the steps already outlined, it may not be cost-effective to go any further.
Consider replacement, but if it’s an emergency, keep in mind that you’ll be dealing with tiny parts that are easy to break and lose.
A total fader rebuild should take only about 5 to 10 minutes, after going to the trouble of taking apart the console to get to the fader.
If I take a module out for repair, I go ahead and clean its fader at the same time. Otherwise, I do fader rebuilding as part of a larger console-cleaning project.
There are several different types of fader construction. Higher-cost faders are literally a “snap” to take apart; that is, they have a “snap together” design.
A much more pleasant use of a dental pick than usual.
The main parts of a typical fader include the element that carries audio on conductive tracks, the carriage that holds wipers against the tracks, and the rails that guide the carriage.
Be extremely careful with the wipers - they’re easy to damage, and once bent, the fader is toast.
After opening the fader, first blow away the loose dust. There might be dirt wedged in at the point where the carriage and rails meet, so use a dental pick to loosen this up, and then blow it out.
Again, blow the loose dirt out.
Use a strip of clean cloth dipped in isopropyl alcohol to clean the rails, pass the strip under and around each rail.
Gently wipe the surface of the conductive element with a clean cloth dipped in alcohol or contact cleaner. Be gentle, and do NOT go under the carriage with the cloth. This can damage the wipers!
Top it off with just a dab of lubricant. Caution: a little goes a long way!
Apply just a few drops of lubrication to the rails and exercise the fader. Blow away any excess lubrication with and reassemble the fader
And that’s it. With a little practice and patience, anyone can make old faders feel like new again!
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
Alex Welti is vice president of research for Creation Audio Labs, a service facility in the southeastern U.S. He served for a decade as service manager of Soundcraft, and prior to that, worked as a technical supervisor for Westlake Audio.
Wednesday, September 11, 2013
API 1608 Graduates To Nuremberg University Of Music
Hochschule für Musik Nürnberg adds API 1608 to Studio 214 on campus.
The API 1608 serves as the perfect teaching tool in many interesting places in the world, and now, one can be found in Studio 214 at the Hochschule für Musik Nürnberg (Nuremberg University of Music).
Creativity is the focus and hands-on experience is the goal of the program.
The equipment available to the students offers unequaled opportunities with both vintage and modern effects, all in a school that dates back to 1821.
The decision to purchase the 1608 came to fruition with the help of API’s German distributor Erwin Strich.
Toni Hinterholzinger, head of the recording department, believed that the punchy and clear sound of the API preamps would make this the ideal learning tool for teaching classic recording techniques.
Most of the recording projects are student-based, but on occasion, there are some commercial projects where students assist and often take part as musicians.
Some who have recorded on the 1608 include Wolfgang Buck, Nevio, Johannes Ludwig, Tilmann Herpichböhm, Steffen Schorn, and Olivia Solner.
“This is, for sure, one of the best equipped rooms in Europe – a place where audio magic actually happens and sonic dreams come true!” said Hinterholzinger.
Posted by Julie Clark on 09/11 at 12:57 PM
Monday, September 09, 2013
In The Studio: Watch Out For The Tall Tale—“We’ll Fix That During Mastering”
The downsides to the "fix it in the mix" approach
Over the last few decades, before the average musician even knew of the mastering process, the most common lie we heard was “we’ll fix that during the mix”.
While we all know that although this statement can be true, it’s often used as an excuse to keep the talent moving along or for the engineer to avoid recording yet another take.
Of course with the invention of digital editing “we’ll fix that during the mix” took on a whole new life, being as that one could easily chop, slice, dice and mutilate a “less than stellar” performance until it became usable, thus making the statement somewhat true. However this was only after much effort that could have simply gone into recording another round of takes.
But, I digress. This isn’t about yesterday’s misguided catch phrase, but about the plague that is upon us today, FDM syndrome (fix during mastering syndrome). Let’s start off with a brief description of what mastering can include.
Mastering is where the final dimension of sound quality is brought to a recording. Experienced mastering engineers with specialized audio tools work at bringing each project to its maximum sonic potential.
Mastering is the step between mixdown and manufacturing. In the mastering studio each project is critically evaluated on a high resolution monitoring system. Any deficiency in the sound is then addressed and careful processing is applied to make the project sound bigger, warmer, clearer, punchier, louder, more three-dimensional, more natural - or whatever may be appropriate to that particular recording.
Some processes that might be applied at the mastering stage include:
Stereo width expansion
Additionally, mastering is where the final assembly of the album occurs. During this process, songs are placed in their proper order, gaps between songs adjusted, fades performed, noises and glitches removed, and so forth.
Most importantly, all changes and enhancements in the mastering process are done in close consultation with the client to make sure that the end result that fits the vision of the project.
I will attempt to lay out what I have continually seen as common misconceptions of what mastering is, and how I feel we can put and end to FDM syndrome once and for all.
Misconception 1: Many mix problems can be fixed during the mastering process.
This is the biggest lie being spoken these days in regards to mastering. Although certain overall mix issues can be helped or controlled during mastering, the only way to truly fix a bad mix is to re-mix or re-record the problematic track.
A bad mix is a bad mix, period. You can’t expect your house painter to fix foundation and framing problems. You can, however, expect him to make your house more presentable.
Misconception 2: Professional mastering will make my home studio project sound professional.
This is very common and probably the most absurd misconception of the lot! A professional-sounding recording will contain many things over and above a great mastering job. Solid material, well executed performances, great mics, pre-amps and recording gear, and a stellar mix are all more important than the mastering process.
Don’t get me wrong, mastering is vitally important, but if these other things are not in place, no mastering job, no matter how brilliant, will amount to a hill of beans. Of course, this also doesn’t discount a quality home recording. Just do it right.
Misconception 3: All recording engineers have the tools and knowledge to master a project.
As the description above states, mastering engineers possess very specific skills and gear. Most reputable mastering engineers do nothing but master, and I, for one, would never dream of having my project mastered by anyone who did not specialize in mastering only.
As far as gear goes, I’ve been in several mastering facilities where I haven’t recognized one single piece of gear or found any of the tools common to just about any recording studio.
I once read a post on an audio engineering board that stated that the BBE Sonic Maximizer was the “secret weapon” of mastering houses.
Now, I’m not saying that particular poster did not see that box in what he was told was a mastering house, but in the professional facilities I’ve seen, the only function it would serve is as a drink coaster.
Misconception 4: There is software on the market that can accomplish what a mastering engineer in a mastering facility can do.
Nope! That’s like saying that everyone with a DAW can produce commercial recordings.
Now, can you buy software that enhances you project studio’s recordings? Of course. Should you experiment with all sorts of hardware and software tools to get the best out of your rig—absolutely! Can you get $50 of software to do what years of experience and custom hardware in the hands of a pro can do—not without skill and talent.
Misconception 5: All mastering facilities/mastering engineers are created equally.
Nope. Go to 10 different mix engineers, get 10 different mixes, and the same goes for mastering engineers and facilities. Just because some people have had bad mastering experiences, don’t assume that you’d be better off doing it yourself.
The same also goes with good mastering experiences. Every project is different and has it’s own set of challenges.
Misconception 6: All recording projects must be mastered.
Simply put, this is not true. It all depends upon the purpose of the recording. If your final (un-mastered) mixes sound good to you, and you want to release them, then more power to you!
I’ve produced and heard plenty of songwriter demos, indie EPs, short run releases, and project recordings that would not have been cost effective for the artists to have them professionally mastered, and would have served little purpose for the intended use of the recordings.
Misconception 7: Louder = Better
Read this great article by Rip Rowan about the atrocities of the loudness war.
I will summarize by saying that a recording can be ruined by a substandard mastering job, and that record labels and artists (and even some mastering engineers) seem to think that louder is somehow better. And it can ruin the dynamic aspects of otherwise great recordings.
Now, go record some great music!
Thursday, September 05, 2013
Manley Labs Announces HHB Communications Canada As New Canadian Distributor
Manley Labs announced that they have signed HHB Communications Canada as their exclusive Canadian distributor.
Manley Labs announced that they have signed HHB Communications Canada as their exclusive Canadian distributor.
HHB Communications Canada Ltd is a major international manufacturer and distributor of recording equipment to the recording, MI, DJ and production sectors of the professional audio industry. HHB and HHB distributed products are sold through all major MI, production, pro-audio and DJ dealers across Canada.
Manley Labs, a privately owned company that designs, manufactures and distributes high-end analog audio equipment, is thrilled about this expansion.
“Canada’s size presents a big challenge for distribution but it also offers vast opportunities, “ said Rick McClendon, Vice President of Sales and Marketing at Manley Labs. “We are confident that the combination of HHB’s resources and their excellent staff will guide Manley Labs growth in Canada.”
“HHB Canada is very pleased to be selected as the new distributor for Manley Labs in Canada. We have always had the greatest respect for the sound and quality of the products. We look forward to increasing the visibility of the brand in the Canadian marketplace, ” commented HHB Communications Canada President Dave Dysart.
“I have always admired the look, feel and sound of Manley products. Our goal at HHB Canada is to increase the sales of Manley products through our network of dedicated dealers and sales representatives across Canada. Hanging out with the fun crew at Manley will be an added bonus!”
Tuesday, September 03, 2013
Compact Console Connectivity: I/O, I/O, It’s Off To Patch We Go…
Taking stock of onboard and expansion connections
Back in the analog (only) days, hooking up to a console was a rather simple process. Stage inputs ran through a copper wire snake that plugged into the front of house console’s inputs. Console outputs were run back to the stage through the same snake line’s returns or they ran through a separate “drive” snake that sent the line level output signals to the amp racks.
If you needed to insert a processor to a channel or group, you simply grabbed a TRS to dual 1/4-inch cable and hooked up the unit to a console’s insert jack. If you wanted to use a board microphone or an audio playback deck, you simply reached around the back of the console, parted the rat’s nest of cables and found an unused input to plug in to. Even if the system used multi-pin connectors, setting up and patching FOH took a bit of time as everything was point-to-point.
Another challenging aspect was patching the stage on shows that had multiple bands. Most of us old sound folks paid our dues playing “patch monkey” on shows that required us to swap out stage snakes and stage lines into the main snake head, while trying to get every signal into the correct snake channel. Trying to do it all in the short turnover times the promoters wanted is why many of us became grumpy old sound folks.
While it was – and still is – relatively simple to interface analog gear, digital consoles have been a boon. Many digital consoles have a way to connect to a digital snake system, saving our backs from having to deal with heavy multicore copper cables.
Another obvious benefit is the need for far less FOH racks, as well as not needing to patch all of that outboard gear into to the console. With built-in processing, digital consoles have lightened our trucks and shrunk the area required for the mix and control position, allowing promoters to sell more seats.
From a connection standpoint, the biggest caveat (at least to me) is that many digital consoles allow patching and routing of signals directly in the software. No more reaching into a pile of cable spaghetti to find an open jack while also trying to to accidentally unplug something. With many digital boards, you can open a set-up/patch menu and assign inputs and outputs to wherever you need. Routing effects, processing and audio mixes has never been easier.
Another great feature is the ability of many digital systems to record either directly to a USB connected drive, to a DAW, or to a stand-alone multi-track deck. Being able to archive shows and do “virtual” sound checks (where you play back feeds directly recorded from the band and use these signals to help set up and tune the PA at the next gig before the band shows up) has been a real blessing to many who work behind a console.
Connectivity is what it’s all about, and today’s digital consoles have really hit the mark with their routing, patching and interfacing effectiveness.
All of these features and benefits are not just limited to large digital boards – many compact models also offer quite extensive interface and routing capabilities, so let’s have a look via this Photo Gallery Tour, taking stock of the connections that are onboard as well as expansion capabilities they have in terms of inputs and outputs.
And because many of these models only hit the market recently, we’ll also provide some additional details on their overall facilities.
Craig Leerman is senior contributing editor for Live Sound International and is the owner of Tech Works, a production company based in Las Vegas.
Guitarist Steve Stevens Relies On sE Electronics VR1 Microphone For Private Production Studio
Guitarist and songwriter finds Voodoo VR1 passive ribbon microphones essential to private production studio setup.
An sE Electronics Voodoo VR1 passive ribbon microphone is an essential component in the recording signal chain at a private production studio owned by guitarist and songwriter Steve Stevens, best known for his collaborations with Billy Idol over the past three decades.
Stevens, a virtuoso guitarist who is equally skilled at hard rock, pop, blues, progressive rock, new wave, jazz fusion and flamenco, is currently writing new material for the next album release by Idol, with whom he recently toured the United States.
Stevens, who has also recorded with Michael Jackson, Ric Ocasek, Robert Palmer and many others over the years, reports that fellow guitarist Pete Thorn, a self-styled “guitar nerd,” initially recommended the VR1 microphone to him.
“Pete’s got impeccable ears, and if he says you should check something out, chances are it’s going to be really, really good,” he says. “sE were totally helpful in recommending the right microphone for my needs. I’ve got nothing but good things to say about the microphone and the company.”
Stevens often works alone in his Avid Pro Tools and Apple Logic Pro-based songwriting studio and consequently has to engineer all of his own recordings.
“I have other ribbon microphones, and they require—for me, at least—exact placement. Pete said a selling point of the VR1 was that you could put it anywhere on any speaker and it would sound really, really good. I thought, that sounds exactly like what I need!”
He continues, “I’m a songwriter, not an engineer, and whatever gets me there the quickest is what I’m going to use. That’s why I like this microphone—it’s a no-brainer.”
When recording in his project studio, Stevens typically uses a combination of two microphones on his guitar cabinet, an industry standard dynamic mic and the VR1, positioned in the center of the loudspeaker cone.
The VR1 is brighter in tone than another more expensive ribbon microphone with which he has recorded at other studios, reports Stevens, and as a result more of the track recorded using the Voodoo tends to end up in the mix.
The VR1 offers a wide-open frequency response of 20 Hz—20 kHz, due to the implementation of a patent pending mechanical device designed by Siwei Zou, the CEO of sE Electronics.
“That other ribbon mic is really dark. Usually you record it in combination with a 57 and only end up using about 25 percent of the ribbon to 75 percent of the 57. But with the Voodoo, sometimes I end up really favoring it in the mix,” he says. “It’s a bit more forgiving than that other mic.”
Stevens first came to the public’s attention with his guitar playing and songwriting on Billy Idol’s breakthrough hits of the early 1980s, which included “White Wedding,” “Hot in the City,” Rebel Yell” and “Eyes Without a Face.”
He later played on “Dirty Diana,” on Michael Jackson’s “Bad” album, also appearing in the music video; recorded several solo albums; played and co-wrote songs on the debut solo album by Vince Neil of Mötley Crüe, “Exposed;” and recorded two albums with super-group Bozzio Levin Stevens, alongside drummer Terry Bozzio and bass player Tony Levin.
Most recently, Stevens once again teamed up with producer and film soundtrack composer Harold Faltermeyer on an instrumental anthem for the Flying Bulls, Red Bull’s collection of vintage aircraft, aerial acrobatic pilots and skydivers.
Faltermeyer is best known for two iconic pieces of film music: “Axel F,” which he wrote and produced for the 1984 film “Beverly Hills Cop,” and 1986’s “Top Gun Anthem,” on which Stevens played, sharing the Grammy Award with Faltermeyer for Best Pop Instrumental Performance.
The pair premiered their new Flying Bulls anthem at the Scalaria Air Challenge in Austria in mid-July, performing on the wing of a moored Dornier seaplane.
Posted by Julie Clark on 09/03 at 11:12 AM
Tuesday, August 27, 2013
Church Sound: Mixing Like A Pro, Part One—Gain
Getting it right can change your mixes for the better
One of the more common audio mistakes we see in churches is improper setting of input gain levels on the house mixer. Getting input gain right is one of the most critical steps in creating good audio mixes.
Setting the input gain affects absolutely everything we do on the console and it can make or break your house mix, monitor mix, recording mix, etc. Before we start mixing the house on the channel faders or do anything else, every channel’s input gain should be dialed in.
A Simple Analogy
Think of your audio channel as a garden hose, and the gain knob (a.k.a., trim or HA/head amp knob) as the water spigot. If we were going to fill up a bucket, there is an ideal amount to open the spigot.
Opening it up too little means we get very little water and while it works, it takes a while to fill the bucket and is not terribly effective. Opening it up too much often means it’s going to get messy with water going everywhere.
This concept holds true with audio. Too little gain will give us weak, inefficient sound. It’s not that there isn’t sound, but it doesn’t sound big and full as we’d typically like things to sound. Overcompensating on the input gain to get your weak level back up to where it should be also introduces unpleasant noise into the system.
We also make a mess when we turn our gain up too much. Pushing too much signal through causes clipping—not a pretty sound!
How Do I Know When Gains Are Set Right?
Unfortunately it’s a little different on every console, but there is a pretty easy indicator that you can use to find the sweet spot. On most consoles, analog or digital, you usually find an input meter—or at the least a little level indicator light.
On the meter, you usually see three different colored sections: green, yellow and red. Even if you have a single light, typically it’ll show green if you have signal, yellow if you’re getting a lot of signal and red if you’re getting too much.
On most consoles, a great target for your gain is right where the green and the yellow lights meet. On many consoles that’s the number 0 on the meter, or 0 dB. For others it’s a different number (for example, on Yamaha digital consoles this happens around -18 dB).
Regardless of the number, setting your gain to where the green light meets the yellow light should give you a big, full sound with plenty of head room to avoid clipping.
Remember, the first thing you need to do is set input gains before anything else. In fact, my ideal way to do begin sound check is to have the band run through 1-2 minutes of the biggest song on their list for that week.
You could do each instrument individually, but in order to save time and verify that the band and singers are giving me their full effort, I’ve found that having them run through something big for two minutes is enough time for me to quickly move through each channel and set gains accordingly. I don’t touch anything else in this period of time—in that 120 seconds I’m only looking at gain, making sure every channel is giving me enough and not too much.
Once completed, all house volume changes are made at the fader and monitor mix changes are done at the auxiliary (AUX) knobs.
Don’t Touch That Dial
Once I feel good about my gain, I do my very best to not touch it again. Sometimes someone jumps all over his/her mic or instrument and you have to decrease your gain a bit, but if at all possible I leave the gain alone at this point and just adjust their fader to change the house sound.
Why? If I adjust the gain at this point, I change the house mix, monitor mixes, recording mixes and everything else on the console. At that point I’ve also taken away the musician’s reference point. What she thinks was ground zero for her volume is no longer true. As she tries to play dynamically, she no longer knows whether he can trust what he is hearing.
We want big, full sound from our instruments and vocals in order to produce a good mix. Our musicians need consistency in their monitor mixes in order to be comfortable and know what they are playing is translating well in the mix. We want our live recordings and web/broadcast feeds to have good consistent sound.
All of these things are affected by how we set the input gains on our house mixer. We need to get our input gains set right at the beginning of the sound check by setting levels close to 0 dB, or between the green and yellow signal lights on our mixer. Once we get the gains set, we leave them alone to keep mains, monitor, recordings, etc., consistent.
Take the time to get you’re gain right, it’ll change your mix for the better!
Duke DeJong has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. CCI Solutions is a leading source for AV and lighting equipment, also providing system design and contracting as well as acoustic consulting. Find out more here.
Tuesday, August 13, 2013
SSL Matrix Console Takes Key Role In Depeche Mode’s ‘Delta Machine’
“The demo’s had this great sound to them — because he was using all these old synths and mixing them through the Solid State Logic Matrix.”
Producer Ben Hillier recently specified Solid State Logic’s Matrix SuperAnalogue mixing console with software controlled analogue patch system and multi-layer DAW control to anchor tracking on Depeche Mode’s eagerly-awaited thirteenth studio album, Delta Machine.
The SSL Matrix has become a central hub of the Depeche Mode production process with frontman Dave Gahan, having set up a Matrix-based project studio at his New York City home, while Ivor Novello Award-winning principal songwriter, synthesist, and guitarist Martin Gore has added a Matrix to his home-based project studio in sunny Santa Barbara.
Having helmed the Depeche Mode production process so successfully for 2005’s Playing The Angel album and its Grammy-nominated Sounds Of The Universe follow-up in 2009, producer and Matrix owner Ben Hillier returned to the Depeche Mode production fold for Delta Machine.
The Matrix proved an invaluable element of the unusual Depeche Mode creative process as the central hub for a huge collection of analogue hardware and multi-DAW based workflow.
“At the end of ‘Sounds Of The Universe’ we’d encouraged Martin to shift his work method away from purely using soft synths,” says Hillier. “We really enjoyed using vintage analogue synths on that record.
“Martin didn’t have a massive collection to start with, but he collected a whole load more. As a result of that, he knew he needed to rebuild his home studio, so we set him up with an SSL Matrix.”
Fast forward to the genesis of Delta Machine and Gore was at it again, having assembled a monstrous modular synth system with well over 700 Eurorack modules.
According to Hillier, “One of the first things that [A&R] Daniel Miller said to me was, ‘The demos that Martin’s made this time are amazing!’ And he was right. They had this great sound to them — because he was using all these old synths and mixing them through the Matrix.”
Transforming those promising-sounding demos into a fully-fledged Depeche Mode album meant moving band, producer, and programming team into a self-built multi-DAW studio setup within the live room at Santa Barbara Sound Design.
“That studio has a great big live room, so we built a studio in there, because putting a band in a live room and me in a control room is not relevant to the way Depeche work.
“We had a Pro Tools rig with hardly any plug-ins as the main recording hub for our studio, which we sort of used like a multitrack, with a summing bus as a mix output, plus a load of laptop-based areas.
“We didn’t want to have everyone sitting around looking over other people’s shoulders at one computer screen while someone moved bass drums around, so it was a way of making programmed music in a slightly more sociable, slightly less navel-gazing way.
“We’d worked in a similar way on Sounds Of The Universe, but really missed having a mixing console, because it was quite difficult to reprocess stuff and doing things on the fly was harder, so that’s the reason why we got the Matrix.”
Next Depeche decamped to the East Coast to continue working in The Penthouse at the beautiful Jungle City Studios in NYC, with its SSL Duality equipped control room and inspiring live room.
“There’s really good monitoring in the main studio there, but we still rebuilt our own studio in the live room, so we could run both rooms simultaneously.”
The transportable Matrix-based workflow worked a treat, bringing several salient strengths to the tracking table.
“We were comp’ing everything on the master Pro Tools rig and running that through the Matrix, so we could just hit recall and get back to the same musical buzz every time, while the DAW control got us away from looking at the computer screen too much.
“We did all the summing in the Matrix — at least all the rough mixes on all the work-in-progress stuff, which worked well. The mix busses sound really great — there’s a gain pot on the top that’s especially handy when you’re setting up a mix and want to drive it a little bit more to make it glue together nicely.”
The key to mixing Depeche Mode’s successful sound arguably lies in balancing those copious quantities of synthesisers (and occasional guitar) alongside Dave Gahan’s distinctive baritone and Martin Gore’s more melodious backing (and occasional lead) vocals.
Here, too, the Matrix proved proficient when tracking the band’s most emotive elements.
“We had a large rack of outboard compressors and EQs — a few of my favourites and some choice pieces liberated from Martin’s studio — that were all patched into the Matrix’s software patch bay.
“Pretty much everything was put through some of them, and we had preset insert chains for things like Dave and Martin’s vocal sounds.”
Hillier must have been doing something right, for following another month-long stint in Santa Barbara Sound Design then an Autumnal return visit to Jungle City Studios tracking was complete. The Matrix proved to be the perfect production partner for Depeche Mode’s modus operandi.
“What’s most exciting about this record is that it seems like we’ve really managed to communicate what the band wanted,” Hillier concludes.
Solid State Logic
Posted by Julie Clark on 08/13 at 02:13 PM
Thursday, August 08, 2013
Church Sound: Understanding The Differences And Uses Of 1/3-Octave & Parametric EQ
Locating important frequencies, when to apply EQ, what type, and more
You’ve probably seen the ubiquitous “1/3-octave EQ” (One-third-octave EQ). This is the piece of equipment in the audio rack with all the little sliders on the front.
Unfortunately it will likely have all sliders set the same: A - smiley face; B - frowney face. These settings happen largely due either to the inexperience of the operator or a poorly designed sound system—or both.
First let’s look at the 1/3-octave equalizer and get an idea of how it can best be used.
Note the graphic of the 1/3-octave equalizer below, and notice the 31 sliders on the front panel. Each slider is set on a frequency.
Starting at the left slider, and moving right, the frequencies begin at 20 Hz and end at 20 kHz.
Look at the first slider (20 Hz) and count over to the right to the third slider, and you will find 40 Hz. This is one octave higher in frequency than 20 Hz. What that means is there are three sliders/ filters per each octave, meaning that each filter is 1/3-octave apart from the filter next to it.
With this information (and what I discuss here) about where certain important frequencies are, we can better EQ our system.
For example, we know that the range for vocalists is about 70 Hz for the lowest bass singer to about 1,400 Hz (1.4 kHz) for the highest soprano, so setting a smiley face on the 1/3-octave EQ does nothing more than boost all frequencies on either end of the vocalist range—or in essence—cuts all the frequencies where the vocals should be!
1/3-octave graphic equalizer
Next time you can’t hear vocals in the house mix (or vocalists complain that they can’t hear themselves in the monitors), check to see if your EQ is smiling at you.
It’s always my recommendation that the main house EQ, if being done with a 1/3-octave EQ, never be randomly adjusted. If your system was installed by a competent audio contractor. EQ should already be set for maximum performance and shouldn’t need to be changed.
Any changes you believe necessary for improvement should be done on the EQ section of your console.
However, this is not the case when using a 1/3-octave EQ for monitor mixes. Because the monitors often move to different positions on the platform, and the “set” and surroundings change as well, so too will EQ likely need some adjustment.
Still, take note that major adjustments (+/-10 dB or more) shouldn’t be necessary - if this is the case, you should look at other aspects to identify potential problems.
The other common type of equalization is parametric. It offers adjustable frequency filters, which means that instead of having filters on set frequencies - the case with 1/3-octave - the operator decides what frequency (or frequencies) need to be cut or boosted.
And not only can you decide what frequency needs to be adjusted, you can also decide how many frequencies around the center frequency will be affected by your adjustment.
Parametric EQ is not for the beginner, nor is it likely needed for each performance. It is, however, a very useful tool and one that every system operator should be familiar with.
On many parametric EQs, including the one shown here, there are at least three controls. One control allows selection of the frequency at which to insert a filter. Another control allows adjustment of how wide or narrow that filter will be.
A wider filter affects more frequencies around the center frequency, a narrow filter, less frequencies - more pinpoint adjustments and/or broader overall adjustments. Finally, the third control is for cut/boost of that particular filter.
Often following installation, the sound contractor or designer will use a parametric EQ for tuning the system. This assists in identifying problem frequencies and helping to correct them without affecting the surrounding frequencies.
It’s also helpful if the system needs a broad spectrum of frequencies adjusted, because this too can be accomplished using perhaps just maybe one filter.
I use a parametric EQ when doing overhead miking because it helps me identify frequencies that tend to feedback too soon, so I can then set filters to notch them out.
The only problem I have is finding the frequency of that out-of-tune choir member and trying to notch it out! But I keep trying…
Joe Wisler has worked in professional audio for more than 35 years and has also been involved with church sound and technical ministry throughout that period.
Tuesday, August 06, 2013
Signal Processing Fundamentals: Passive & Active Crossovers
The nature of sound, and a discussion of loudspeaker crossovers, a "necessary evil"
In space, no one can hear you scream ... because there is no air or other medium for sound to travel.
Sound needs a medium; an intervening substance through which it can travel from point to point; it must be carried on something. That something can be solid, liquid or gas. They can hear you scream underwater ... briefly.
Water is a medium. Air is a medium. Nightclub walls are a medium. Sound travels in air by rapidly changing the air pressure relative to its normal value (atmospheric pressure). Sound is a disturbance in the surrounding medium.
A vibration that spreads out from the source, creating a series of expanding shells of high pressure and low pressure ... high pressure ... low pressure ... high pressure ... low pressure.
Moving ever outward these cycles of alternating pressure zones travel until finally dissipating, or reflecting off surfaces (nightclub walls), or passing through boundaries, or getting absorbed—usually a combination of all three.
Left unobstructed, sound travels outward, but not forever. The air (or other medium) robs some of the sound’s power as it passes. The price of passage: the medium absorbs its energy.
This power loss is experienced as a reduction in how loud it is (the term loudness is used to describe how loud it is from moment to moment) as the signal travels away from its source.
The loudness of the signal is reduced by one-fourth for each doubling of distance from the source. This means that it is 6 dB less loud as you double your distance from it. [This is known as the inverse square law since the decrease is inversely proportional to the square of the distance traveled; for example, 2 times the distance equals a 1/4 decrease in loudness, and so on.]
How do we create sound, and how do we capture sound? We do this using opposite sides of the same electromagnetic coin.
Electricity and magnetism are kinfolk: If you pass a coil of wire through a magnetic field, electricity is generated within the coil. Turn the coin over and flip it again: If you pass electricity through a coil of wire, a magnetic field is generated. Move the magnet, get a voltage; apply a voltage, create a magnet ... this is the essence of all electromechanical objects.
Microphones and loudspeakers are electromechanical objects. At their hearts there is a coil of wire (the voice coil) and a magnet (the magnet). Speaking causes sound vibrations to travel outward from your mouth.
Speaking into a moving-coil (aka dynamic) microphone causes the voice coil to move within a magnetic field. This causes a voltage to be developed and a current to flow proportional to the sound—sound has been captured.
At the other end of the chain, a voltage is applied to the loudspeaker voice coil causing a current to flow which produces a magnetic field that makes the cone move proportional to the audio signal applied—sound has been created.
The microphone translates sound into an electrical signal, and the loudspeaker translates an electrical signal into sound. One capturing, the other creating. Everything in-between is just details.
And in case you’re wondering: yes; turned around, a microphone can be a loudspeaker (that makes teeny tiny sounds), and a loudspeaker can be a microphone (if you SHOUT REALLY LOUD).
Crossovers: Simple Division
Loudspeaker crossovers are a necessary evil. A different universe, a different set of physics and maybe we could have what we want: one loudspeaker that does it all.
One loudspeaker that reproduces all audio frequencies equally well, with no distortion, at loudness levels adequate for whatever venue we play.
Well, we live here, and our system of physics does not allow such extravagance. The hard truth is, no one loudspeaker can do it all.
We need at least two—more if we can afford them. Woofers and tweeters. A big woofer for the lows and a little tweeter for the highs. This is known as a 2-way system. (Check the accompanying diagrams below for the following discussions.)
But with two speakers, the correct frequencies must be routed (or crossed over) to each loudspeaker.
At the simplest level a crossover is a passive network. A passive network is one not needing a power supply to operate—if it has a line cord, or runs off batteries, then it is not a passive circuit.
The simplest passive crossover network consists of only two components: a capacitor connecting to the high frequency driver and an inductor (aka a coil) connecting to the low frequency driver.
A capacitor is an electronic component that passes high frequencies (the passband) and blocks low frequencies (the stopband); an inductor does just the opposite: it passes low frequencies and blocks high frequencies.
Above, passive 2-way crossover, and below, passive 3-way crossover.
But as the frequency changes, neither component reacts suddenly. They do it gradually; they slowly start to pass (or stop passing) their respective frequencies. The rate at which this occurs is called the crossover slope.
It is measured in dB per octave, or shortened to dB/octave. The slope increases or decreases so many dB/octave. At the simplest level, each component gives you a 6 dB/octave slope (a physical fact of our universe).
Again, at the simplest level, adding more components increases the slope in 6 dB increments, creating slopes of 12 dB/oct, 18 dB/oct, 24 dB/oct, and so on.
The number of components, or 6 dB slope increments, is called the crossover order. Therefore, a 4th-order crossover has (at least) four components, and produces steep slopes of 24 dB/octave.
The steeper the better for most drivers, since speakers only perform well for a certain band of frequencies; beyond that they misbehave, sometimes badly. Steep slopes prevent these frequencies from getting to the driver.
You can combine capacitors and inductors to create a third path that eliminates the highest highs and the lowest lows, and forms a mid-frequency crossover section. This is naturally called a 3-way system. (See diagram)
The “mid” section forms a bandpass filter, since it only passes a specific frequency band. Note from the diagram that the high frequency passband and low frequency passband terms are often shortened to just high-pass and low-pass.
A 3-way system allows optimizing each driver for a narrower band of frequencies, producing a better overall sound. So why not just use passive boxes?
The single biggest problem is that one passive cabinet (or a pair) won’t play loud enough and clean enough for large spaces. If the sound system is for your bedroom or garage, passive systems would work just fine—maybe even better. But it isn’t.
Once you try to fill a relatively large space with equally loud sound you start to understand the problems. And it doesn’t take stadiums, just normal size clubs. It is really difficult to produce the required loudness with passive boxes.
Life would be a lot easier if you could just jack everyone into their own cans amp—like a bunch of HC 4 or HC 6 Headphone Amps scattered throughout the audience. Let them do the work; then everyone could hear equally well, and choose their own listening level.
But life is hard, and headphone amps must be restricted to practice and recording.
Monitor loudspeakers, on the other hand, most likely have passive crossovers. Again, it’s a matter of distance and loudness. Monitors are usually close and not overly loud—too loud and they will feed back into your microphone or be heard along with the main mix: not good.
Monitor loudspeakers are similar to hi-fi loudspeakers, where passive designs dominate ... because of the relatively small listening areas. It is quite easy to fill small listening rooms with pristine sounds even at ear-splitting levels.
But move those same speakers into your local club and they will sound thin, dull and lifeless. Not only will they not play loud enough, but they may need the sonic benefits of sound bouncing off close walls to reinforce and fill the direct sound. In large venues, these walls are way too far away to benefit anyone.
So why not use a bunch of passive boxes? You can, and some people do. However, for reasons to follow, it only works for a couple of cabinets. Even so, you won’t be able to get the high loudness levels if the room is large. Passive systems can only be optimized so much.
Once you start needing multiple cabinets, active crossovers become necessary.
To get good coverage of like-frequencies, you want to stack like-drivers. This prevents using passive boxes since each one contains (at least) a high-frequency driver and a low-frequency driver. It’s easiest to put together a sound system when each cabinet covers only one frequency range.
For instance, for a nice sounding 3-way system, you would have low-frequency boxes (the big ones), then medium-sized mid-frequency boxes and finally the smaller high-frequency boxes. These would be stacked or hung, or both—in some sort of array.
A loudspeaker array is the optimum stacking shape for each set of cabinets to give the best combined coverage and overall sound.
You’ve no doubt seen many different array shapes. There are tall towers, high walls, and all sorts of polyhedrons and arcs. The only efficient way to do this is with active crossovers.
Some smaller systems combine active and passive boxes. Even within a single cabinet it is common to find an active crossover used to separate the low- and mid-frequency drivers, while a built-in passive network is used for the high-frequency driver. This is particularly common for super tweeters operating over the last audio octave.
At the other end, an active crossover often is used to add a subwoofer to a passive 2-way system. All combinations are used, but each time a passive crossover shows up, it comes with problems.
One of these is power loss. Passive networks waste valuable power. The extra power needed to make the drivers louder, instead boils off the components and comes out of the box as heat—not sound. Therefore, passive units make you buy a bigger amp.
A couple of additional passive network problems has to do with their impedance.
Impedance restricts power transfer; it’s like resistance, only frequency sensitive.
In order for the passive network to work exactly right, the source impedance (the amplifier’s output plus the wiring impedance) must be as close to zero as possible and not frequency-dependent, and the load impedance (the loudspeaker’s characteristics) must be fixed and not frequency-dependent (sorry, not in this universe; only on Star Trek).
Since these things are not possible, the passive network must be (at best), a simplified and compromised solution to a very complex problem. Consequently, the crossover’s behavior changes with frequency—not something you want for a good sounding system.
One last thing to make matters worse. There is something called back-emf (back-electromotive force: literally, back-voltage) which further contributes to poor sounding speaker systems.
This is the phenomena where, after the signal stops, the speaker cone continues moving, causing the voice coil to move through the magnetic field (now acting like a microphone), creating a new voltage that tries to drive the cable back to the amplifier’s output! If the speaker is allowed to do this, the cone flops around like a dying fish. It does not sound good!
The only way to stop back-emf is to make the loudspeaker “see” a dead short, i.e., zero ohms looking backward, or as close to it as possible—something that’s not gonna happen with a passive network slung between it and the power amp.
All this, and not to mention that inductors saturate at high signal levels causing distortion—another reason you can’t get enough loudness. Or the additional weight and bulk caused by the large inductors required for good low frequency response. Or that it is almost impossible to get high-quality steep slopes passively, so the response suffers.
Or that inductors are way too good at picking up local radio, TV, emergency, and cellular broadcasts, and joyfully mixing them into your audio.
Such is life with passive loudspeaker systems.
Active crossover networks require a power supply to operate and come packaged in single-space, rack-mount units or more often in recent years, built into loudspeakers with power amplifiers.
Looking at the accompanying diagram shows how active crossovers differ from their passive cousins.
For a 2-way system instead of one power amp, you now have two, but they can be smaller for the same loudness level. How much smaller depends on the sensitivity rating of the drivers.
Likewise a 3-way system requires three power amps. You also see and hear the terms bi-amped, and tri-amped applied to 2- and 3-way systems.
Active crossovers cure many ills of the passive systems. Since the crossover filters themselves are safely tucked away inside their own box, away from the driving and loading impedance problems plaguing passive units, they can be made to operate in an almost mathematically perfect manner.
Extremely steep, smooth and well-behaved crossover slopes are easily achieved by active circuitry.
Above, active 2-way crossover, and below, active 3-way crossover.
There are no amplifier power loss problems, since active circuits operate from their own low voltage power supplies. And with the inefficiencies of the passive network removed, the power amps more easily achieve the loudness levels required.
Loudspeaker jitters and tremors caused by inadequately damped back-emf all but disappear once the passive network is removed.
What remains is the amplifier’s inherent output impedance and that of the connecting wire. Here’s where the term damping factor comes up. [Note that the word is damp-ing, not damp-ning as is so often heard; impress your friends.] Damping is a measure of a system’s ability to control the motion of the loudspeaker cone after the signal disappears. No more dying fish.
Siegfried & Russ
Active crossovers go by many names. First, they are either 2-way or 3-way (or even 4-way and 5-way). Then there is the slope rate and order: 24 dB/oct (4th-order), or 18 dB/oct (3rd-order), and so on.
And finally there is a name for the kind of design. The two most common being Linkwitz-Riley and Butterworth, named after Siegfried Linkwitz and Russ Riley who first proposed this application, and Stephen Butterworth who first described the response in 1930.
Up until the mid `80s, the 3rd-order (18 dB/oct) Butterworth design dominated, but still had some problems. Since then, the development (pioneered by Rane and Sundholm) of the 4th-order (24 dB/oct) Linkwitz-Riley design solved these problems, and today is the norm.
What this adds up to is active crossovers are the rule. Luckily, the hardest thing about an active crossover is getting the money to buy one.
After that, most of the work is already done for you. At the most basic level all you really need from an active crossover are two things: to let you set the correct crossover point, and to let you balance driver levels. That’s all.
The first is done by consulting the loudspeaker manufacturer’s data sheet, and dialing it in on the front panel. (That’s assuming a complete factory-made 2-way loudspeaker cabinet, for example. If the box is homemade, then both drivers must be carefully selected so they have the same crossover frequency, otherwise a severe response problem can result.)
Balancing levels is necessary because high frequency drivers are more efficient than low frequency drivers. This means that if you put the same amount of power into each driver, one will sound louder than the other. The one that is the most efficient plays louder. Several methods to balance drivers are always outlined in any good owner’s manual.
Dennis Bohn is a principal partner and vice president of research & development at Rane Corporation. He holds BSEE and MSEE degrees from the University of California at Berkeley. Prior to Rane, he worked as engineering manager for Phase Linear Corporation and as audio application engineer at National Semiconductor Corporation. Bohn is a Fellow of the AES, holds two U.S. patents, is listed in Who’s Who In America and authored the entry on “Equalizers” for the McGraw-Hill Encyclopedia of Science & Technology, 7th edition.