Friday, April 19, 2013
Sound Check? What Sound Check?! Working Folk Festival Sound
Hands-on advice to capably handle a myriad of duties
I’m a long-time retired broadcast engineer, and since about 2006, I’ve been a sound engineer at the New England Folk Festival in Mansfield, MA. Working sound at a folk festival has unique challenges, e.g., unfamiliar equipment, fast-paced setup changes and no time for sound checks, and I thought it would be useful to provide an overview.
We are all volunteers, including the musicians, other performers and crew. It isn’t money that’s keeping us there. It’s love of the music, and of helping to make it happen for the performers and audience, whether they are singing, dancing or listening.
We’re not paid, but that doesn’t mean we are not professional. Most of the crew are retired sound mixers, or work as musicians or in sound reinforcement as their “day job” and help the festival pro bono.
We have to keep things moving. There is at most 10 minutes to change sets, move microphones and loudspeakers if needed, and see that mic lines, DI lines and 117V AC lines are where they are needed. All of this is being done while the audience is moving in and out and performers are setting up.
The equipment has been checked out before the festival starts, so there is little need (or time) for troubleshooting, beyond the basics of replace-or-bypass. Still, some basic tools are handy. You might consider carrying:
1) A small penlight—some mixing areas are not well-lit.
2) An AC outlet tester—there are many extension cords and spider boxes in use.
3) Multi-screwdriver for tightening the odd XLR connector.
4) Your own headphones (do you really want to use the last guy’s “ear buds”?)
In a typical two-hour shift, you will:
A) Take over from the previous sound mixer in the middle of a set.
B) Check to see what’s needed for the next two sets. We are given a layout book that shows instruments, mic placement, DIs, etc., for each set. In 10 minutes you may go from a square dance “caller” playing music from a CD to a 12-piece dance band.
C) Time keep to let the current act know when to get off stage.
D) Swap out equipment and get ready for the next show in the 10 minutes you have.
E) Get a pleasing sound out to the audience, or to dancers at all parts of a stage.
F) Make sure the musicians can hear themselves—there may be four channels of stage monitors for the musicians. With no time for sound checks, all of this has to be mixed on-the-fly, and often adjustment is needed from song to song.
G) At any time, field questions about “What group is this?” or “Who’s on next?” or “Which way is the main hall?”
Repeat it all at the end of this set, and then be ready to turn over control to your replacement. Do all this with possibly unfamiliar equipment. Our festival uses the same boards and other equipment from year to year…until it doesn’t!
Fortunately most of our festival venues are either all-dancing or all-listening, but if they’re not, you’ll have to re-tune your brain and your ears (which when combined are the best test equipment you have!) from “what a listening audience needs” to “what the dancers need.”
If you’ve not run sound for dancers before, here are a few tips:
—If there is a dance caller, she or he wins. The dancers must be able to hear her/him. As far as possible give the callers what they want.
—If the caller uses a wireless mic, as with all RF wireless feeds, there may be pick-up of RF interference when she/he turns it off. Changing channels (if possible) may help. Also, look for a “squelch” control on the receiver and turn it up until the audio mutes when the wireless mic is shut off. Another solution is to have the caller “mute” the mic, while leaving the RF carrier on. The final solution is to mute the wireless mic at the mixer as needed, but this is the worst solution, as you will have to ride the control all through the session, to the annoyance of you, the caller, and the dancers.
—The beat is essential. This should be obvious, but don’t bury the beat to bring out the nifty fiddle. Having said that, the melody, along with the caller, tells the dancers where they are in the dance, so they need to hear that, too.
—Walk around the floor to get a feel for what the curtains and or live walls are doing to the sound.
—Dancers make noise! Compensate.
As part of the sound crew, you’re also the one who summons emergency response teams or stops the show and helps clear the building in case of emergency. We’ve never had to, but we’re ready.
Remember to keep a smile on your face because you represent the festival, and we want the audience to come back next year!
Based in New England, Tom Padwa is a retired broadcast engineer who has worked in AM, FM and TV, and he’s a senior member of the Society of Broadcast Engineers and a Certified Electronics Technician (Audio).
The Old Soundman: Dealing With Indoor & Outdoor Venue Issues
Think it’s a picnic running sound inside a club? Think it’s nothin’ but a party running sound outside? The OSM has news for you!
I occasionally run sound for a band that tends to play local hole-in-the-wall venues.
Okay, we feel sorry for you, now move on!
The “stage” for the band is always in one of 2 places: in a nice boomy corner, or, better yet, right in front of that brick or paneled wall.
These are the times that try men’s souls!
I guess you might be a female, so no offense intended. I don’t know what “Stip” is short for. I am pretty sure that Jacquie (below) is…
One of many problems I run into (including the lead guitarist who insists he hears better with his knees)...
I know that guy! and I think half our readers at home do, too. He must have cloned himself a dozen times in each and every state of the union!
...is cymbal bleed-thru on the vocal mic’s. If I try to spare the audience the shrill ring of these upper frequencies by pulling back the highs on the board, I seem to lose clarity in the vocal.
That is not an illusion, Stip. That is, indeed what is happening, you are perceiving it correctly.
This problem gets worse when the guys are playing at a particularly loud stage volume, and I need to crank a little more vocal, which of course starts to feed back when the ring of the cymbals hit the mic’s, then come thru the monitors and hit the mic’s again…
You know the sad, sad story.
I do indeed know the sad story. And even sadder is the fact that the list of remedies is a very short one. I’m a straight shooter, Stip.
Move back the drum riser. Can’t. You’re stuck in this little club with a stage the size of a saltine.
Now that you mention it, some cheese and crackers would really hit the spot right about now! Wait a minute, you were saying something about cymbals …
The drummer can be asked to use lighter cymbals with a shorter decay time. But since he is a club guy, getting paid very little beyond the endless chain of longnecks he consumes, he probably only has his local music store’s finest, thickest bang-a-langa models.
Don’t tell me he wears those warm-up things on his wrists? You do have it rough, Stip.
I would be fired if I mentioned a brand name here, but it is kosher for me to tell you that you want a hypercardioid mic for your singer, and he needs to stay right on top of it.
The most radical thing you could do would be to ask the band to buy an infrared gate device to put on the mic, so that when his head moves away, it mutes the mic.
However, this has the undesired effect of really changing your mix, since that is the loudest mic on stage.
When that cymbal noise becomes the evil frosting on the cake of a monitor mix, isn’t that just the worst? You can try to identify as narrow a band as possible to reduce, on the graphics for the affected mixes.
I’m not gonna lie to ya, Stip, everything I have said boils down to band-aids. I am pretty much doctor dan the bandage man here. Stip, it is hellish there where you are. But the bigger gigs are hellish in different ways.
Okay, I’m just trying to cheer you up! on the big stages, it is really fun, sonically, when the drum riser is a mile behind the singer.
Would it make you feel better to hear how Jacquie gets treated? Sure it would!
Just had an outdoor gig. Singer was freaking out, saying “the sound sucks” when in actuality it didn’t suck at all. Tried to tell him (from my limited experience) that running sound outdoors is quite a bit different from running sound indoors.
Since I’m a rank amateur at this, is there anything specific I can tell him to shut him up? He’s a great singer, but like most musicians, he has high end hearing loss.
Thanks mucho. Dig your site. You crack me up.
Thank you, Jacquie! My, what excellent taste you have in humor. I am a much funnier man than others, am I not?
What you are going through reflects the agony of having a limited number of clients. If I read between the lines correctly, you don’t want to just tell this guy to take a hike.
Most of the self-righteous hornblowers over on the live audio board would be real quick to say that you should proudly tell this character off, and then march off into the sunset, with your pride intact, and your wallet quite empty.
Well, I guess some of the more sensible ones who read a lot of self-help books would advise you to talk to the guy when he is calmer (since right after a gig is a notorious time for musicians to make ludicrous remarks, usually due to their lack of confidence in their own abilities.)
In the past, I believe that the lads and lasses of the L.A.B. have recommended gently informing your yodeler that there is no “suck” knob on your console. And, that the way for him to win in life is to express himself as clearly as he can, to the limits of his ability.
He may continue to say “wull, I dunno, Jacquie, it just sucked, y’know?” most of us would shake your hand if you just hauled off and slugged him then. But we live in a very litigious society, so it is best not to.
What you are digging for is him saying something like “there was too much low end” or “it was too trebly.” Precise technical terms like that. Is he criticizing the monitor sound or the house?
Hey, you know what? You sound like you have your head on straight. I think you’re gonna go far, with or without this dullard! You rule, Jacquie!
Making It Match: An Introduction To Transformers In Audio Devices
Key principles of transformers and how they shape audio equipment and systems
In the dawning days of audio, transformers played a vital role in the functionality of first-generation all-tube based electronic circuitry. It was circa 1920 and radio broadcasts for the general populace had just begun, generating a rapid rise in the demand for broadcast audio systems, all of which needed transformers to function.
Later, as equipment for live sound reinforcement began to emerge, transformers again proved indispensible as the only means of matching microphone impedances to vacuum tube preamplifiers.
Transformers were also used as inter-stage devices in amplifiers, for line output drivers, and for matching a power amplifier’s output stage to the impedance of a loudspeaker voice coil – just as they are still used today in audiophile tube-based amplifiers and musical instrument amps.
Eventually, widespread use of transistor-based preamps and power amplifiers lessened the need for transformers, but as any electric guitarist will tell you, tubes “just sound better.”
However, transformers do much more than just impedance matching. They can differentially balance a microphone or line-level signal at the source, and then de-balance the same signal at the destination (or more properly stated, the “load”). In the process, electromagnetic interference (EMI), the cause of all-too-familiar hum and buzz, is cancelled out as a function of the transformer’s common-mode rejection ratio (CMRR or CMR).
Line-level transformer isolators. (click to enlarge)
The term “common mode” refers to any stray field that is common to both the plus and minus poles of a balanced line. Add the word “rejection” and it describes exactly what the transformer is doing: it phase-cancels the EMI because the poles are 180 degrees opposed to each other, thus rejecting any unwanted field induced in the cable.
This is pretty important stuff. When a line is not balanced, it becomes vulnerable to picking up all kinds of stray energy. Usually this takes the form of 60 Hz (or 50 Hz in Europe), along with related harmonics. The interference is induced in the cable from nearby alternating current (AC) power cables or from AC rectifiers in electrical devices.
Problems can also be caused by radio transmitters and other high-power devices that generate unwanted energy fields, such as diathermy machines and wood welders (yes, there really is such a thing as a wood welder). In the case of higher frequencies, the invasive energy may not be audible, but it can wreak havoc in a sensitive broadband mic preamp if the energy is not cancelled out by a precision-grade transformer (or by other means).
High-grade audio transformers, such as those manufactured by Lundahl in Sweden and Jensen in the U.S., exhibit high CMRR values across a broad spectrum of frequencies; low-grade transformers may help reduce hum and buzz a bit, but their CMRR is rarely sufficient to solve any significant problem. When it comes to hum and noise rejection, precision high-grade transformers are an invaluable insurance policy.
Audio transformers not only provide differential balancing of signals, they eliminate ground loops as well. In fact, very well.
Ground loops occur when two points should be at the same ground potential, but aren’t. They’re caused by improperly installed equipment (or sometimes improperly designed equipment), with the result being noise and interference in a system. Merely plugging AC power cords into outlets that do not share the same ground path can create a ground loop when one piece of equipment is connected to another.
When many devices are interconnected, or when devices are far apart, the likelihood of a ground loop increases significantly. Ground loops are what keep audio professionals working through the dinner break. They can be difficult to sort out because different manufacturers of consoles and signal processing gear use different grounding techniques.
Enter the transformer. It stops ground loops cold, without resorting to the dangerous practice of lifting the AC ground pin on (one or more) power cords. Those who know this carry a handful of “barrel” style line-level transformer isolators to gigs.
Matter Of Magnetics
A unique feature of transformers is the ability to transfer a signal from the primary winding to the secondary winding without any galvanic connection, thus the input and outputs are said to be galvanically isolated from one another.
Galvanic isolation refers to isolating one branch, or section, of an electrical system from another, preventing current flow. No conductivity is permitted between the sections, yet energy and information are still exchanged. This perfectly describes the function of an audio transformer. Instead of a galvanic connection, the input signal induces a magnetic field in the transformer’s primary, which in simple terms is a coil of wire wrapped around an iron core. The transformer’s secondary consists of another coil of wire wrapped around the same core, but insulated from the primary coil.
Power (VI) applied to the primary coil is magnetically transferred to the secondary coil, hence the galvanic isolation. (click to enlarge)
The two coils may have the same number of turns, or one coil may have greater or fewer turns than the other. This is known as the “turns ratio” and determines whether the transformer steps-up the applied voltage, steps-down the applied voltage, or does not re-scale it at all, but serves only to isolate.
When an AC signal is applied to the primary coil, it generates a field that is magnetically induced in the secondary windings, thus reflecting the properties of the applied signal, but not making direct electrical contact. This is the means by which ground loops are broken and unwanted current is prevented from flowing between two or more units that share ground conductors.
For this same reason, no transformer will ever pass direct current DC, because DC does not generate an alternating magnetic field. Note: DC blocking does not apply to auto-transformers that are sometimes used in 70-volt/100-volt distributed loudspeaker systems, because the primary and secondary windings of auto-transformers are connected galvanically.
Lost In Translation
It’s important to understand that even though transformers are passive devices, they are actually quite complex with internally distributed resistance, capacitance and inductance.
Thus they still exhibit varying electrical and performance characteristics, just as active circuits do. They are, in fact, not dissimilar to loudspeakers – except there are no moving parts.
Earlier, we referred to CMRR (common-mode rejection ratio), which is one of the most important performance characteristics that audio product designers and system integrators look for when specifying transformer-coupled interconnects.
But an impressive value of CMRR does not fully define a transformer’s overall performance. It must also get high marks in each of the following characteristics if the end result is to be clean and transparent sound quality.
Distortion. Like an active circuit, a transformer will inevitably introduce a measure of distortion. In a high-grade design, distortion will be quite low, but present nonetheless. Transformer distortion is a function of level and frequency; the lower the frequency and the higher the signal level, the more distortion. In particular when operating limits of the transformer are exceeded (high level at low frequencies), and core saturation is eminent, distortion will increase rapidly. Core saturation occurs when a transformer’s iron core cannot absorb any additional magnetism, thereby “clipping” the signal.
Linearity. Rarely published as a specification, linearity describes how the other parameters (frequency response, distortion, phase response and transient response) will stay stable (or not) under a range of input levels.
Level. This specifies the maximum input and/or output levels at a specified frequency (normally 20 Hz or 50 Hz) before saturating and becoming non-linear.
Frequency Response. Like any other audio device, there is an upper and lower response limit. Within those limits the response may be perfectly flat – or it may deviate a little – or a lot.
Phase Response. This parameter describes any deviation from a flat phase response within the specified frequency range. Even among the leading transformer manufacturers, phase response data is not always available. When it is, it’s usually presented as a graphical plot, often with the frequency response depicted on the same graph, as they are proportionally related. In low-grade products, usually very little data is supplied.
Transient Response. Specifies how fast the transformer can respond to a short signal burst (or the leading edge of a continuous signal), and how quickly it stops emitting energy after the applied signal has stopped. Like phase response, this data is not always available, but can be extrapolated by examining frequency response and self-resonance data (pulse transformers for digital audio may delve into this specification more deeply).
Transformers that generally exhibit good frequency response and effective CMRR can still color the sound, sometimes dramatically, due to a slow initial response and significant overshoot at the tail end. Sometimes this can be pleasing to the ear, providing a sense of “warmth,” but most times it is not. In any case, it’s an inaccurate representation of the signal.
It’s better to use a plug-in, or a signal processor intended for such an effect, rather than to infect the system with a sub-optimal transformer. As with everything else in audio, when evaluating a transformer, it pays to spend time listening.
Many microphone types utilize transformers, either for step-up (typically ribbon transformers) or step-down (typically tube microphones), or 1:1 for dynamic microphones.
Some manufacturers, such as Cascade, offer upgraded transformer options for many of their ribbon mics. The company even specifies the transformer brand and model, which indicates how important the transformer’s contribution is in achieving optimal sonic quality.
Another common stage source is the direct box, or DI. For many years, all DIs were transformer based, and with good reason. An instrument plugged into a DI has one ground reference through the instrument amplifier, while the sound system has a different ground reference. As a rule, a transformer-based DI will solve a ground-loop problem faster and easier than an active DI.
Transformer-based mic splitters incorporate specialized transformers, with one primary and one, two, three, or more secondaries. We mentioned transformers as a form of insurance policy earlier. Nothing equates to more “audio insurance equity” than a well designed and built transformer-based mic splitter.
Mic preamps are another key application. While many brands offer only electronically balanced inputs, a good number of premium products are either transformer-based from inception, or offer transformer versions as an option.
A Lundahl transformer at the heart of a Focusrite ISA One mic preamp. (click to enlarge)
Moving along the signal path, some high-end consoles employ transformers, and/or offer transformer options on line outputs. Considering that the console is where all signals come together, this is a good place to consider specifying transformer options, when available.
The console, in turn, feeds signal processing of all types: loudspeaker management systems, outboard equalizers and limiters, and/or banks of self-powered loudspeakers. All are candidates for transformer usage, especially in situations that vary from day to day, when there’s little time available to solve induced EMI or ground loop problems.
Present day audio transformers are appreciated and revered for their unique problem solving capabilities; they are perhaps more valuable now than ever before. Today’s audio systems have become incredibly complicated, with many interconnected devices comprising even a small system, and a staggering number of devices in large-scale systems.
In the past, transformers were the only way to get things done; now, in many cases, they may be the only right way.
Editor’s Note: This is Chapter 1 in our ongoing series on transformers. Chapter 2: Transformers–Insurance Against Show-Stopping Problems, a new white paper that continues the discussion, has just been made available as a free download. Get it here.
Ken DeLoria is senior technical editor for Live Sound International and ProSoundWeb, and has had a diverse career in pro audio over more than 30 years, including being the founder and owner of Apogee Sound.
For the last several years, I’ve had the opportunity to mix on digital consoles.
One of the benefits of most digital consoles is the EQ section; typically a full 4-band parametric plus a variable high-pass filter. Frankly, I’ve gotten so used to it that it’s tends to be a bit of a shock when I work on an analog desk that’s not so equipped.
It occurred to me the other day that many of you live in that world all the time so I thought I would share some thoughts on making the most of limited EQ.
The first thing to keep in mind is that getting good sound at the source is of utmost importance. Often times, simply moving a mic an inch or two to the left or right; or closer or farther will clean up 80 percent of what you need to fix.
Choosing the correct mic is also important. For example, if your vocalist has a sibilant voice, perhaps there is a mic in your locker that rolls off some high end. Swap mics and you may no longer need that narrow notch at 8 kHz in the EQ that you can’t have anyway.
Sometimes you’d like to get more punch from the kick mic; instead of turning EQ knobs, try a different mic that has the sound you’re looking for. It’s also important to have the system tuned as well as it can be. The closer the system is to sounding good, the less board level EQ you need to do.
I’ve actually walked into churches and seen a cut in the high mids of all the input channels on the console only to look at the room EQ and noted a big boost across that range. Ideally, you want the PA to sound good without any channel EQ, then use the channel EQ to tweak anomalies in the source.
Often times the easiest way to upgrade the sound of your system is to upgrade your system processing (either with a new one, or by properly setting up what you have). The reason this is important is because with just a few bands of EQ, you have to choose your battles carefully. On a smaller board with only four fixed bands of EQ, you really don’t have a lot of options. If you find yourself consistently cutting the high band, make the change at the system level.
Sometimes, you might find that you have a particular source that is tough to get sounding good. You may have a pastor who has a nasal voice, or a worship leader who always sounds muddy no matter what you do.
If you’ve already tried different mics and haven’t achieved the desired result, you have one option left, inserting an EQ into the channel. For not a lot of money, you can pick up a new or used analog graphic EQ and simply insert those into the problem channels.
We did this a few years ago on our pastor’s channel. He had a really difficult voice and even with decent analog EQ on the board, we needed some external help. It made a huge difference and we finally got him sounding good.
Now, there is a caveat to this; it might be tempting to buy a whole bunch of outboard EQs and insert them all over the board. If you’re going to do that, consider upgrading other parts of the system (perhaps including the console) first.
This is a surgical strike, something you do when you’ve done what you can do with what you have. Adding outboard EQ adds complexity and increases the opportunities to really make a mess of the sound. So do this carefully with great consideration.
With some simple tweaks to the entire audio signal path, you can greatly improve the sound quality of the whole system.
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.
Once everything has been recorded, and before you start mixing, do you edit the audio? Do you fix things? Do you think it’s cheating?
Here’s my take.
What is editing?
An audio editor is much like a book editor. He takes the original audio file and adds or removes bits and pieces to make it better. An audio editor at a radio station will take a spoken-word commercial that’s 34 seconds long and trim it down until it fits into a 30-second spot.
A book editor will read the manuscript and suggest that certain parts be taken out…or certain parts be stretched out.
With regard to music production and recording, editing involves any changes made to the audio between the recording phase and the mixing phase. This can involve normalizing audio files, correcting timing issues, removing unwanted sections, or even changing the actual performance itself.
Editing is not just a digital thing. Back in the “analog days,” engineers would regularly cut and splice tape between two different takes.
Is it cheating?
One could argue that the musician’s performance should remain untouched. If the performance wasn’t perfect, that’s okay. That’s reality. That’s how the musician really sounds.
Others like to take a good performance and “touch it up” here and there to make it even better. A prime example? AutoTune. Some people rant and rave against AutoTune. They write things in the liner notes like “AutoTune was not used anywhere on the album.” Others swear by AutoTune.
First, let me make a point here. You should be recording good musicians. No amount of editing tools or software or magic voodoo will make a crappy musician sound good. Let’s just assume we’re talking about good musicians and good performances.
So…is it cheating to take a good performance and try to improve it? Is it wrong to “pocket” the drums so they’re a bit tighter and more in sync with the click track? Is it wrong to pocket the bass, making it “lock in” with the kick drum? Guitars? Keys? Background vocals?
I can’t tell you if it’s cheating or not. But let me give you my take.
I’m creating a product
When I’m working on a recording project, the end result (most likely) is a finished CD or album. I’m producing something that’s going to have my name on it. I want it to sound as good as possible. That’s why I work with good musicians.
However, what if there are mistakes in the audio? What if the bass comes in a little too soon in a few spots? Well, I ask myself, what would be best for the song? Would it sound better if the bass was playing with the kick drum rather than a few milliseconds before?
My answer? Yes.
What’s best for the song? That’s what I ask myself. What will make this product I’m creating sound its best?
In my opinion, all these editing tools are just that…tools. Just like in any other industry, I use the tools I have at my disposal to make the best product I can. That means I almost always pocket the drums, then the bass, then the guitars and other rhythm instruments.
Even an amazing performance can stand a little tweaking here and there. I’m not talking about changing the performance entirely. I’m simply trying to enhance the performance. Chances are every change I make is exactly what the musician was trying to do, but didn’t.
At a live show, the band can be really tight, and it sounds great. On a recording however, the little sloppy parts are much more noticeable…so I fix them.
So…like I said, I’m creating a product. I’d much rather listen to a song that sounds amazing and doesn’t have any distracting parts. It doesn’t have to be perfect, and you should never overdo any editing. You’re simply allowing the song to be the focus, rather than the individual components. Again…it’s all about the song.
Next I’ll present some scenarios where it makes sense to edit tracks, followed by some scenarios where it doesn’t.
Where it makes sense:
1. Fix Noticeable Timing Issues. This is probably the most obvious reason to edit, but it’s worth mentioning again. There will come times when you’re recording (either yourself or someone else) where there will be those trouble spots, places where the guitar just got really out of time with the drums, or the bass came in a half-second early.
These things happen. Sometimes you miss them during tracking. Sometimes this is just the best your going to get out of the musician. (Let’s be honest, sometimes we don’t have the luxury of recording A-list musicians.)
In these cases, the timing issues are obvious, and most people are going to notice them. It’s in your best interest to fix them. It shouldn’t take long. These types of timing issues can usually be fixed with just a minute or two of editing without negatively affecting the quality of the audio.
2. Tighten Up A Good Performance. Once you’ve fixed the major timing issues, is there any room for more pocketing? Or lets say you are recording extremely talented musicians, is it still possible that you’ll want to pocket those tracks as well? A lot of times the answer is yes.
Whether you agree with it or not, a lot of musicians and their fans are expecting a very polished, tight recording. Even if the musicians nail their parts in tracking, there may be small, subtle timing difference between the various instruments.
While the tracks may sound fine without any editing, a few hours of pocketing can push them over the top in terms of tightness and a (perceived) “professional” sound.
Don’t believe me? Nearly every professionally-produced album that comes out of Nashville (particularly in the country music industry) has gone through this pocketing process. These session musicians are insanely talented, but their tracks still get pocketed. At the very least, it’s something to think about.
3. Get Rid of Unwanted Noise. It’s very possible to produce a recording out of your home studio that sounds like it was done in a professional facility. One of the tell-tale signs of an “unprofessional” home recording is the unnecessary noises that somehow don’t get removed somewhere along the way.
Things like lip noise from the vocalist, the sound of the musicians moving around between sections of the song, pops and clicks in the audio from edits without crossfades, can distract the listener and make them question the quality of the recording.
You may not think this is an issue, but those little noises get amplified quite heavily once you compress and master your final mix. Suddenly a little annoying noise becomes a lot more noticeable…and therefore distracting.
Where it doesn’t make sense:
1. The Artist Doesn’t Approve. Everything you do in the studio should be done with the artist/client in mind. Whether he/she is paying you or not, you’ve been hired to take their songs and turn them into great-sounding recordings.
Sometimes artists don’t want you to mess with their performances. They have strong opinions that they want the final recording to sound exactly like what they performed. This is understandable, and you should respect their wishes.
However, take into account what genre of music they’re performing. If it’s a straight-forward country album, you may want to remind them that most country albums have been edited/pocketed pretty heavily, and that it might be in their best interest to do the same in order to compete.
On the other hand, if you’re recording somebody like Jack White who doesn’t really conform to any genre or style, you’re better off leaving his stuff alone. That rough, seemingly disorganized sound IS what he’s going for.
As with most things, it all boils down to good communication. Do what’s best for the client.
2. The Song Sounds Amazing As Is. Sometimes things just magically gel in the studio, and the tracks sound absolutely fantastic. Even fantastic tracks can sometimes benefit from a little editing, but if you listen through the entire song and don’t hear any spots where things could be tighter, good for you! You can skip editing and go straight for mixing.
It might be a good idea to try editing a small section of the song just to be sure, but if your editing is hurting rather than helping, forget about it.
3. The Song Isn’t Finished. I’ve seen so many people jump in and start editing recorded parts before they’ve recorded everything. For example, they’ll heavily pocket the drums and bass before the guitars have been recorded. This can be a big waste of time for two reasons:
A) Without all the instruments, you can’t really hear the “groove.” If you can’t hear the groove, then you probably can’t hear exactly where and how to pocket the bass and drums. Wait until the guitars, etc. are recorded, THEN determine if pocketing is necessary.
B) Once everything’s recorded, you may not NEED to pocket anything. This happened to me recently. I recorded acoustic guitar and drums, and it wasn’t super tight, but I went ahead and recorded all the other parts, keyboards, bass, lots of guitars. Once the mix was really full, those subtle timing issues between the acoustic guitar and drums were masked by all the other instruments. I didn’t need to do much editing at all.
In the last few years, there have been so many digital mixing consoles released that it’s almost mind boggling to keep up with them.
But let’s look at some of the big-picture differences between analog and digital consoles. Perspectives first…If you’re the sound technician at your church, then what you may like about a console could be very different from what the money people or the congregation think about it.
Here are a some things to consider in terms of analog and digital:
Audio Quality. Ultimately, it’s all about audio and what it sounds like. With that in mind, regardless of all the snazzy features on a mixer, what really matters most is how good it sounds. If a console doesn’t have good preamplifiers, then regardless of all the other cool features and pretty lights, no one will be pleased with what comes out of the speakers, through monitors, or on a recording. Fortunately, almost all modern consoles, whether digital or analog, provide acceptable preamp circuits.
Number Of Channels. It’s important to consider where you are in your church’s growth curve. What might be just fine now may or may not last for a significant season. If you’re a start-up, or fairly new church with big plans to grow quickly, then you might want to go smaller to start with so you can move your console downstream to the youth, children or fellowship area. The reverse is true as well. If you’re growing rapidly, you might want to get more channels that you need right now to future-proof your main room setup
Presets, Scenes, and DSP. The main difference between a digital console and a traditional analog console is that digital consoles have lots of processing power in addition to the preamplifiers, subgroups, and auxiliary sends. Most have a strong complement of presets and scene memory along with multiple bands of equalization and effects processors.
The presets alone can make it worth purchasing a digital board for many churches. Imagine the ability to set a Sunday morning, Wednesday evening, funeral, and wedding preset. These can make midweek or Saturday events much easier when a trained sound tech can’t be there. Most digital consoles have moving/automatic faders as well. Some consoles, like the PreSonus StudioLive series, have presets and great DSP, but they don’t have moving faders.
Worship Style. If you have numerous worship services and they vary with different styles of worship, or if you have another congregation meeting in your facility, then a digital console can be a huge advantage. Each service can be customized and saved as a preset for easy recall and less headaches. This is particularly helpful if you have multiple worship team members who rotate throughout your seasonal schedule. Saving a particular vocalist’s or different preacher’s/teacher’s eq settings can be a lifesaver and really streamline the sound check process.
Control Freaks In The House? Sometimes tactile control of the console is not enough control because your mixing location is not in an ideal place. That NEVER happens in a church, right? So, many digital consoles have the capacity to actually control them wirelessly with an iPad or tablet computer from anywhere in the room that your wireless connection can reach.
What you hear and what the rest of the congregation hears are not always the same. This can be particularly helpful if you run traditional monitor mixes from the house console and you need to adjust them from the stage. You can actually walk onstage and make adjustments wirelessly while simultaneously amazing your worship team with how tech savvy you are! Even personal monitors can be controlled wirelessly by the team members or the sound tech can control for them as well.
The Bottom Line. If you don’t need all the whistles and bells that a digital console provides, then your analog desk options are varied in channel count, features and price. They range in price from under $500 up to five figures.
Digital consoles range in price from about $1,000 up to even six figures, depending on what your needs are. In alphabetical order, here are some of the top console manufacturers: Allen & Heath, Avid, Behringer, DiGiCo, Midas, PreSonus, Roland, Soundcraft, and Yamaha. Of course, there are others, but these are the most well-known and have been around the longest.
Jeff McLeod is managing director and a certified church consultant for Church Audio Video.
Church Audio Video specializes in the design, installation and support of high-quality and affordable custom audio, video, lighting, broadcast and control systems for worship facilities. For more information, visit their website.
Eliminating Potential Trouble & Getting The Noise Out Of A System
Replacing myth, misinformation, and mystery with knowledge and clear understanding
“A cable is a source of potential trouble connecting two other sources of potential trouble.”
The humor in this statement may be lost on those who regularly assemble sound systems. But a reality of sound systems is that a signal accumulates noise as it flows through equipment and cables. And once noise contaminates a signal, it’s essentially impossible to remove it without altering or degrading the original signal.
For this reason, no system can be quieter than its noisiest link. Noise and interference must be prevented along the entire signal path.
Delivering a signal from one box to another may seem trivial, but when it comes to noise, the signal interface is usually the danger zone, not the equipment’s internal signal processing. Many - if not most - designers and installers of audio systems think of grounding and interfacing as a black art. How many times have you heard someone say that a cable is “picking up” noise - presumably from the air like a radio receiver?
Even most equipment manufacturers often don’t have a clue what’s really going on. The most basic rules of physics are routinely overlooked, ignored, or forgotten. As a result, myth and misinformation have become epidemic!
It’s time to replace myth, misinformation, and mystery with knowledge and clear understanding.
How Quiet Is Quiet?
Of course, how much noise and interference is tolerable depends on how a system is used. A monitor system in a recording studio obviously needs much more immunity to ground noise and interference than a construction site paging system.
The dynamic range of a system is the ratio, generally measured in dB, of its maximum undistorted output signal to its residual output noise or noise floor - up to 120 dB of dynamic range may be required in high-performance sound systems.
By the way, with video systems, a 50 dB signal-to-noise ratio is a generally accepted threshold beyond which no further improvement in images is perceivable, even by expert viewers.
Of course, a predictable amount of “white” noise is inherent in all electronic devices and must be expected. White noise is statistically random and its power is uniformly spread across the signal frequency range. In an audio system, it is heard as “hiss.” Excess random noise is generally due to improper gain structure, a topic that really isn’t part of our discussion here.
On the other hand, ground noise, usually heard as hum, buzz, clicks or pops in audio signals, is generally much more noticeable and irritating. (And note that 10 dB noise reductions are generally described as “half as loud,” while 2 dB to 3 dB reductions are “just noticeable.”)
Myths About Earth Grounding
As electronics developed, the common return paths of various circuits were also referred to as “ground,” regardless of whether or not they were eventually connected to earth. In addition, a single ground circuit most often serves, either intentionally or accidentally, more than one purpose.
Thus, the very meaning of the term ground has become vague, ambiguous, and often quite fanciful. Some engineers have a strong urge to reduce these unwanted voltage differences by “shorting them out” with massive conductors - the results are most often disappointing.
Other engineers think that system noise can be improved experimentally by simply finding a “better” or “quieter” ground. Many indulge in wishful thinking that noise currents can somehow be skillfully directed to an earth ground, where they will disappear forever!
Here are some common myths about grounding:
Earth grounds are all at zero volts - presumably with respect to each other and to some “mystical absolute” reference point. This leads to whimsical ideas about lots of ground rods making system noises disappear! In fact, the soil resistance between ground rods is much higher (often tens of ohms) than a wire between them.
Impedance - symbolized as “Z,” it’s the apparent AC resistance of a circuit containing capacitance and/or inductance in addition to pure resistance. Wires have zero impedance, and, therefore, can extend a zero-voltage reference to many locations in a system, eliminating voltage differences. In fact, wires are quite limited:
The DC Resistance of a wire applies only at very low frequencies and is directly proportional to its length. For example, the resistance of 10 feet of 12-gauge wire is about 0.015 Ohms.
The inductance of a wire is nearly independent of its diameter (gauge) but is directly proportional to its length and increases at bends or loops.
Figure 1 (click to enlarge)
Our 10 feet of 12-gauge wire has an impedance of 30 Ohms at 1 MHz (AM broadcast band) as shown in the Figure 1.
Substituting a 1/2-inch diameter solid copper rod lowers the impedance only slightly to about 25 Ohms.
A wire resonates (becomes an antenna) when its physical length is a quarter wavelength. For a 10-foot wire, this means it will essentially become an open circuit at about 25 MHz.
Are earth grounds really necessary for low-noise system operation? Think about all the electronics in an airplane!
Under fortuitous conditions, systems may be acceptably quiet in spite of poor techniques. But physics will ultimately rule and noises may appear for no apparent reason!
Once we understand how grounding systems and interfaces actually work and how noises couple into signals, finding and fixing problems becomes simple and logical.Perhaps the most important aspect of troubleshooting is how you think about the problem.
Without a methodical approach, chasing noise problems can be both frustrating and time-consuming. For example, don’t fall into the trap of thinking something can’t be the problem just because you’ve always done it that way. Remember, things that “can’t go wrong” do!
Don’t start by changing things! Because many problems reveal themselves if we just gather enough clues, gather as much information as possible before you change anything.
Ask questions! Did it ever work right? What symptoms tell you it’s not working right? When did it start working badly or stop working? What other symptoms showed up just before, just after, or at the same time?
Be alert to clues from the equipment itself! Operation of the equipment’s controls, along with some simple logic, can provide very valuable clues.
For example, if the noise is unaffected by the setting of a volume control or selector, logic dictates that it must be entering the signal path after that control.
If the noise can be eliminated by turning the volume down or selecting another input, it must be entering the signal path before that control.
Write everything down! Less than perfect memory can waste a lot of time.
Sketch a block diagram of the system! Show all signal interconnecting cables, including digital and RF, and indicate their approximate length. Mark any balanced inputs or outputs. Generally, stereo pairs can be indicated with a single line. Also note any equipment that’s grounded via its 3-prong power plug, and note any other ground connections such as cable TV or DSS dishes.
Work through the system backwards! As a general rule, and unless clues suggest another starting point, always begin at the inputs to the power amplifiers and sequentially test interfaces backward toward the signal sources.
1. Long delays, reverb predelays, or reverb decay push the sound further away if they’re loud enough.
2. Shorter reverbs (less than 1 second) and shorter delays (less than 100 ms) makes the sound bigger.
3. If delays are timed to the tempo of the track, they add depth without being noticeable.
4. If delays are not timed to the track, they stick out.
5. Reverbs work better if their predelay and decay time are timed to the tempo of the track.
6. Layer reverbs by frequency with the longest being the brightest and the shortest a being the darkest, or vice-versa.
7. Return the reverb in mono and pan accordingly. All reverbs needn’t be returned in stereo.
8. Make things sound big with reverbs and get the depth from delays, or vice versa.
9. Use a bit of the longest reverb on all major elements of the track to tie all the environments together.
10. EQ your effects to make them fit in the track better by rolling off the highs and lows, and/or scooping out the mids in the 2kHz range.
Keep these tips in mind during your next mix and see how much better your mix sounds.
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.
Church Sound: Eliminating Distractions That Negatively Impact Sound Reinforcement
Avoiding the types of mistakes that tend to be fruitful and multiply!
When working with sound at church, we all know just how many things can go wrong. The kicker is usually when they go wrong, which invariably seems to be at the worst possible moment.
The church I belong to, like most, doesn’t have a great sound system. We sure would like to have one, but like many, we’ve chosen to make do over the years.
One day, I asked the senior pastor what his primary goal would be if we could get a new system. His reply? “We need something that would cause no distractions.”
Of course, I was expecting him to mention things like audio quality, ease of use, uniform volume levels at every seat, wireless features, and so on. Thus his answer surprised me at first. But after thinking it over, I realized he was exactly right. As people who work with sound, we focus first on features and technology.
On the other hand, pastors often consider aspects that can have a negative impact on a service, which ultimately detracts from worship – and that’s why we’re all there in the first place! Everything else comes second.
I’ve always thought of the church sound operator as a referee at a sporting event. Most of the time, when either does their job well, no one notices. That’s the way it should be - no distraction. But when something goes wrong, everyone takes note – distraction.
My train of thought continued. Can a well-designed modern sound system, with simplified controls and intuitive applications, lead to fewer problems and therefore less distractions?
The majority of modern audio components perform far better than their predecessors, due to superior design and manufacture. Not to mention they’re newer and thus less susceptible to problems. As to the issue of whether they’re “easier” to operate, I believe that’s a subjective opinion of each system operator.
But this did lead me to consider another potential source of distractions, and where they often originate when it comes to sound: the training (and lack thereof) of system operators. Beyond training, how well are most churches equipped to schedule and manage volunteer (or even paid) system operators?
Now, let’s backtrack for a moment. As noted, the church I belong to doesn’t have a “whiz-bang” sound system, but it does get the job done, and we work very hard to make sure it causes as few distractions as possible. This is because we invested in quality components, which were installed by a qualified A/V systems contractor.
Not only did we choose to go this direction with the system when it was new, but we also rely on this professional to handle any upgrades of components, to fix problems that come up, and to assist with “check-ups” on a regular basis. A little preventative maintenance goes a long way.
I understand the temptation to try to purchase new systems and products in the least expensive manner possible, and to “self-install” them. This is natural – we all, churches included, want the best bang for our buck.
But if there’s one absolute fact I’ve learned after working in audio for more than 30 years, it’s this: one of the best ways to eliminate potential distractions is to have a system designed and installed by trained professionals.
Installation mistakes such as poor grounding, sloppy wiring and terminations, improper cable selection and a host of other little things, can all add up to one gigantic mess. And these types of mistakes tend to be fruitful and multiply!
Worse yet, I’ve walked into churches and have seen loudspeakers that are not designed to be suspended being hung by eyebolts screwed into the side of their particleboard cabinets. I just hope that these “accident waiting to happen” distractions don’t occur during a service.
Here’s a checklist for evaluating distraction potential:
1) Was your sound system designed by a reputable audio consultant who understands the needs of the church, the acoustical properties of the sanctuary, and the capabilities of those who operate the system?
2) Was your sound system installed by a certified individual employed by a reputable systems contracting firm?
3) Is the company that installed the system still in business, and involved in your additions and changes?
4) Has your system been installed in phases or added to over time?
5) Are system operators well trained and knowledgeable?
6) Does your sanctuary’s physical layout require a lot of audio equipment to be moved around and re-connected between services?
7) Does your church struggle to find trained, motivated people to run the system?
8) Does your system produce random hums and buzzes, level changes, dropouts, crackles, distortions, pops, feedback or other noises that seem to go unexplained?
9) Do you own and consult instruction manuals and documentation on your equipment and system?
10) Is your system subject to regular maintenance inspections?
If you answered “yes” to checklist items 1, 2, 3, 5, 9 and 10 – and no to the rest – then your system is probably in good shape.
If not, it’s time to consider taking the proper steps in making sure your church is a distraction-free place to worship.
Last month I finally found the time to finish one of the DIY projects I’ve had on the go for months: turning a Sony Walkman into a nasty distortion box.
The concept is simple. Remove the mechanical guts, replace jacks and overload the Walkman amplifier to create distortion.
It’s not a tape distortion, but I can do that in a different project.
Any portable cassette player will work. I like the Sony Sport for it’s great rugged case and memorable look. A player with EQ controls, bass boost or other enhancements could be helpful.
Step 1 – Remove Guts
Remove the tape mechanism, tape head, radio antenna, and extra plastic parts. Take note of where the tape head attached to the board and points that the play switch connected.
Step 2 – Power
A power supply is needed for testing and it should be removed before any soldering. Batteries heat up fast! Attach the power with alligator clips to the circuit board. Solder the play control points to the on position to keep the power always on. Solder a switch on one of the battery leads to cut power.
The most difficult part of the build was creating a battery holder for the two AA. Several hours of trial and error went by before I found a reliable solution. If you attempt this project I recommend buying a AA battery holder because they’re very tricky to fabricate.
Step 3 – Solder Jacks
Replace the 1/8-inch TRS jack with 1/4-inch TS jack. Simply desolder the jack, run leads into the board from the jack.
The input jack replaces the tape head. I soldered right to the legs of the opamp. If you connect the ground first, you can just poke around with the hot lead until input reaches the output. This is a good way to troubleshoot or experiment. I also removed the volume control pot and shorted it to full volume.
Step 4 – Test & Experiment
Now that you have the power, input and output connected, run some sounds through it. Mine did very little with a guitar connected directly to it going to a small passive speaker. Taking a signal from my audio interface in gave me plenty of level to work with, in fact I could drive an 8-ohm 12-inch speaker at a healthy level. This is a good time to try circuit bending. I kept things pretty simple and just added a switch to bridge the channels for increased gain.
Step 5 – Mount Jacks & Switches
Get the circuit board back in the case and figure out where there is room to safely drill a few holes for jacks and switches. Drill, trim, reassemble, done.
I’m very happy with how this turned out and is a very usable effect for some sources. I think it works particularly well for dirty synths like the Monotron. The entire build would have been much faster if only I had that battery clip. You could probably do this in under an hour if you don’t get carried away with testing and making noise like I did.
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.
News & New Products From Prolight+Sound/Musikmesse 2013
All of the latest from the Prolight + Sound show in Frankfurt
Last week’s Prolight+Sound and Musikmesse 2013 show in Frankfurt, German drew a reported record high 113,000-plus visitors (from 142 countries), surpassing the previous record for attendance set in 2009. Last year’s figures were 109,481 visitors from 120 countries.
“This visitor response has far surpassed our own expectations and that of the exhibitors”, says Detlef Braun, managing director of Messe Frankfurt. “With 113,000, the visitor numbers not only exceeded last years results by three percent, but it also surpasses the all-time record high from 2009 (112,478 visitors).
“This is a fantastic result for these two innovative and very market-active industry sectors at their leading shows here in Frankfurt am Main.”
The visitor growth was divided equally over the trade and public days. Accordingly, the increase was also equally distributed in terms of national and international visitors.
“Especially noteworthy is the fact that with over 42,300 overseas visitors, the increase of international visitors is comparable to the increase in national visitors,“ explains Braun. The visitors came mainly from Germany (71,200 visitors; 2012: 68,267) as well as from the Netherlands, France, Italy, Switzerland, Belgium, Austria, the Russian Federation, Sweden and Poland. The increase can be mainly attributed to new visitors from Italy, Belgium, Austria and Russia
More than 30,000 products were on display, and as always, new products were a primary focus. We’re presenting many of them here, and be sure to check back during the week for updates.
What are the main parts of a loudspeaker cable, and what does each one do?
Typically a loudspeaker cable has two stranded copper conductors, covered with insulation, twisted together with fillers and sheathed with an overall jacket.
How big should the conductors be?
The required size (or gauge) of the conductors depends on three factors: (1) the load impedance; (2) the length of cable required; and (3) the amount of power loss that can be tolerated. Each of these involves relationships between voltage (volts), resistance (ohms), current (amperes) and power (watts). These relationships are defined with Ohm’s Law.
The job of a loudspeaker cable is to move a substantial amount of electrical current from the output of a power amplifier to a loudspeaker system. Current flow is measure in amperes. Unlike instrument and microphone cables, which typically carry currents of only a few milliamperes (thousandths of an ampere), the current required to drive a speaker is much higher; for instance, an 8-ohm speaker driven with a 100-watt amplifier will pull about 3-1/2 amperes of current.
By comparison, a 600-ohm input driven by a line-level output only pulls about 2 milliamps. The amplifier’s output voltage, divided by the load impedance (in ohms), determines the amount of current “pulled” by the load. Resistance limits current flow, and decreasing it increases current flow. If the amplifier’s output voltage remains constant, it will deliver twice as much current to an 8-ohm load as it will to a 16-ohm load, and four times as much to a 4-ohm load. Halving the load impedance doubles the load current.
For instance, two 8-ohm loudspeakers in parallel will draw twice the current of one loudspeaker because the parallel connection reduces the load impedance to 4 ohms.
(For simplicity’s sake we are using the terms resistance and impedance interchangeably; in practice, a loudspeaker whose nominal impedance is 8 ohms may have a voice coil DC resistance of about 5 ohms and an AC impedance curve that ranges from 5 ohms to 100 ohms, depending on the frequency, type of enclosure, and the acoustical loading of its environment.)
How does current draw affect the conductor requirements of the loudspeaker cable?
A simple fact to remember: Current needs copper, voltage needs insulation. To make an analogy, if electrons were water, voltage would be the “pressure” in the system, while current would be the amount of water flowing. You have water pressure even with the faucet closed and no water flowing; similarly, you have voltage regardless of whether you have current flowing.
Current flow is literally electrons moving between two points at differing electrical potentials, so the more electrons you need to move, the larger the conductors (our “electron pipe”) must be. In the AWG (American Wire Gauge) system, conductor area doubles with each reduction of three in AWG; a 13 AWG conductor has twice the copper of a 16 AWG conductor, a 10 AWG twice the copper of a 13 AWG, and so on.
But power amp outputs are rated in watts. How are amperes related to watts?
Ohm’s Law says that current (amperes) times voltage (volts) equals power (watts), so if the voltage is unchanged, the power is directly proportional to the current, which is determined by the impedance of the load. (This is why most power amplifiers will deliver approximately double their 8-ohm rated output when the load impedance is reduced to 4 ohms.)
In short, a 4-ohm load should require conductors with twice the copper of an 8-ohm load, assuming the length of the run to the loudspeaker is the same, while a 2-ohm load requires four times the copper of an 8-ohm load.
Explaining this point leads to the following oft-asked question:
How long can a loudspeaker cable be before it affects performance?
The ugly truth: Any length of loudspeaker cable degrades performance and efficiency. Like the effects of shunt capacitance in instrument cables and series inductance in microphone cables, the signal degradation caused by loudspeaker cabling is always present to some degree, and is worsened by increasing the length of the cable.
The most obvious ill effect of loudspeaker cables is the amount of amplifier power wasted.
Why do cables waste power?
Copper is a very good conductor of electricity, but it isn’t perfect. It has a certain amount of resistance, determined primarily on its cross-sectional area (but also by its purity and temperature). This wiring resistance is “seen” by the amplifier output as part of the load; if a cable with a resistance of 1 ohm is connected to an 8-ohm loudspeaker, the load seen by the amplifier is 9 ohms. Since increasing the load impedance decreases current flow, decreasing power delivery, we have lost some of the amplifier’s power capability merely by adding the series resistance of the cable to the load.
Furthermore, since the cable is seen as part of the load, part of the power which is delivered to the load is dissipated in the cable itself as heat. (This is the way electrical space heaters work!) Since Ohm’s Law allows us to calculate the current flow created by a given voltage across a given load impedance, it can also give us the voltage drop across the load, or part of the load, for a given current. This can be conveniently expressed as a percentage of the total power.
How can the power loss be minimized?
There are three ways to decrease the power lost in loudspeaker cabling. First, minimize the resistance of the cabling. Use larger conductors, avoid unnecessary connectors, and make sure that mechanical connections are clean and tight and solder joints are smooth and bright.
Second, minimize the length of the cabling. The resistance of the cable is proportional to its length, so less cable means less resistance to expend those watts. Place the power amplifier as close as practical to the loudspeaker. (Chances are excellent that the signal loss in the line-level connection to the amplifier input will be negligible.) Don’t use a 50-foot cable for a 20-foot run.
Third, maximize the load impedance. As the load impedance increases it becomes a larger percentage of the total load, which proportionately reduces the amount lost by wiring resistance. Avoid “daisy-chaining” loudspeakers, because the parallel connection reduces the total load impedance, thus increasing the percentage lost.
The ideal situation (for reasons beyond mere power loss is to run a separate pair of conductors to each loudspeaker form the amplifier.
Is the actual performance of the amplifier degraded by long loudspeaker cables?
There is a definite impact on the amplifier damping factor caused by cabling resistance/impedance. Damping, the ability of the amplifier to control the movement of the speaker, is especially noticeable in percussive low-frequency program material like kick drum, bass guitar and tympani.
Clean, “tight” bass is a sign of good damping at work. Boomy, mushy bass is the result of poor damping; the loudspeaker is being set into motion but the amplifier can’t stop it fast enough to accurately track the waveform. Ultimately, poor damping can result in actual oscillation and loudspeaker destruction.
Damping factor is expressed as the quotient of load impedance divided by the amplifier’s actual source impedance. Ultra-low source impedances on the order of 40 milliohms (that’s less than one-twentieth of an ohm) are common in modern direct-coupled solid-state amplifiers, so damping factors with an 8-ohm load are generally specified in the range of 100-200.
However, those specifications are taken on a test bench, with a non-inductive dummy load attached directly to the output terminals. In the real world, the loudspeaker sees the cabling resistance as part of the source impedance, increasing it. This lowers the damping factor drastically, even when considering only the DC resistance of the cable. If the reactive components that constitute the AC impedance of the cable are considered, the loss of damping is even greater.
Although tube amplifiers generally fall far short of sold-state types in damping performance, their sound can still be improved by the use of larger speaker cables. Damping even comes into play in the performance of mixing consoles with remote DC power supplies; reducing the length of the cable linking the power supply to the console can noticeably improve the low-frequency performance of the electronics.
What other cable problems affect performance?
The twin gremlins covered in Understanding Microphone Cable, namely series inductance and skin effect, are also factors in speaker cables. Series inductance and the resulting inductive reactance adds to the DC resistance, increasing the AC impedance of the cable. An inductor can be thought of as a resistor whose resistance increases as frequency increases.
Thus, series inductance has a low-pass filter characteristic, progressively attenuating high frequencies. The inductance of a round conductor is largely independent of its diameter or gauge, and is not directly proportional to its length, either.
Skin effect is a phenomenon that causes current flow in a round conductor to be concentrated more to the surface of the conductor at higher frequencies, almost as if it were a hollow tube. This increases the apparent resistance of the conductor at high frequencies, and also brings significant phase shift.
Taken together, these ugly realities introduce various dynamic and time-related forms of signal distortion which are very difficult to quantify with simple sine-wave measurements. When complex waveforms have their harmonic structures altered, the sense of immediacy and realism is reduced. The ear/brain combination is incredibly sensitive to the effects of this type of phase distortion, but generally needs direct, A/B comparisons in real time to recognize them.
How can these problems be addressed?
The number of strange designs for loudspeaker cable is amazing. Among them are coaxial, with two insulated spiral “shields” serving as conductors; quad, using two conductors for “positive” and two for “negative;” zip-cord with ultra-fine “rope lay” conductors and transparent jacket; multi-conductor, allegedly using large conductors for lows, medium conductors for mids, and tiny conductors for highs; 4 AWG welding cable; braided flat cable constructed of many individually insulated conductors; and many others.
Most of these address the inductance question by using multiple conductors and the skin effect problem by keeping them relatively small. Many of these “esoteric” cables are extraordinarily expensive; all of them probably offer some improvement in performance over ordinary twisted-pair type cables, especially in critical monitoring applications and high-quality music systems. In most cases, the cost of such cable and its termination, combined with the extremely fragile construction common to them, severely limits their practical use, especially in portable situations.
In short, they cost too much, they’re too hard to work with, and they just aren’t made for rough treatment. But, sonically, they all bear listening to with an open mind; the differences can be surprisingly apparent.
Is capacitance a problem in loudspeaker cables?
The extremely low impedance nature of speaker circuits makes cable capacitance a very minor factor in overall performance. In the early days of solid state amplifiers, highly capacitive loads (such as large electrostatic speaker systems) caused blown output transistors and other problems, but so did heat, short circuits, highly inductive loads and underdesigned power supplies.
Because of this, the dielectric properties of the insulation used are nowhere near as critical as that used for high-impedance instrument cables. The most important consideration for insulation for loudspeaker cables is probably heat resistance, especially because the physical size constraints imposed by popular connectors like the ubiquitous 1/4-in phone plug severely limit the diameter of the cable.
This requires insulation and jacketing to be thin, but tough, while withstanding the heat required to bring a relatively large amount of copper up to soldering temperature. Polyethylene tends to melt too easily, while thermoset materials like rubber and neoprene are expensive and unpredictable with regard to wall thickness PVC is cheap and can be mixed in a variety of ways to enhance its shrink-resistance and flexibility, making it a good choice for most applications. Some varieties of TPR (thermoplastic rubber) are also finding use.
Why don’t loudspeaker cables require shielding?
Actually, there are a few circumstances that may require the shielding of loudspeaker cables. In areas with extreme strong radio frequency interference (RFI) problems, the loudspeaker cables can act as antennae for unwanted signal reception which can enter the system through the output transistors. When circumstances require that loudspeaker-level and microphone-level signals be in close proximity for long distances, such as cue feeds to recording studios, it is a good idea to use shielded loudspeaker cabling (generally foil-shielded, twisted-pair or twisted-triple cable) as “insurance” against possible crosstalk form the cue system entering the microphone lines.
In large installations, pulling the loudspeaker cabling in metallic conduit provides excellent shielding from both RFI and EMI (electromagnetic interference). But, for the most part, the extremely low impedance and high level of loudspeaker signals minimizes the significance of local interference.
Why can’t I use a shielded instrument cable for hooking an amplifier to a loudspeaker, assuming it has the right plugs?
You can, in desperation, use an instrument cable for hooking up an amplifier to a loudspeaker. However, the small gauge (generally 20 AWG at most) center conductor offers substantial resistance to current flow, and in extreme circumstances could heat up until it melts its insulation and short-circuits to the shield, or melts and goes open-circuit, which can destroy some tube amplifiers.
Long runs of coaxial-type cable will have large amounts of capacitance, possibly enough to upset the protection circuitry of some amplifiers, causing untimely shut-downs. And of course there is enormous power loss and damping degradation because of the high impedance of the cable.
• Ballou, Greg, ed., Handbook for Sound Engineers: The New Audio Cyclopedia, Howard W. Sams and Co., Indianapolis, 1987.
• Cable Shield Performance and Selection Guide, Belden Electronic Wire and Cable, 1983.
• Colloms, Martin, “Crystals: Linear and Large,” Hi-Fi News and Record Review, November 1984.
• Cooke, Nelson M. and Herbert F. R. Adams, Basic Mathematics for Electronics, McGraw-Hill, Inc., New York, 1970.
• Davis, Gary and Ralph Jones, Sound Reinforcement Handbook, Hal Leonard Publishing Corp., Milwaukee, 1970.
• Electronic Wire and Cable Catalog E-100, American Insulated Wire Corp., 1984.
• Fause, Ken, “Shielding, Grounding and Safety,” Recording Engineer/Producer, circa 1980.
• Ford, Hugh, “Audio Cables,” Studio Sound, Novemer 1980.
• Guide to Wire and Cable Construction, American Insulated Wire Corp., 1981.
• Grundy, Albert, “Grounding and Shielding Revisited,” dB, October 1980.
• Jung, Walt and Dick Marsh, “Pooge-2: A Mod Symphony for Your Hafler DH200 or Other Power Amplifiers,” The Audio Amateur, 4/1981.
• Maynard, Harry, “Speaker Cables,” Radio-Electronics, December 1978,
• Miller, Paul, “Audio Cable: The Neglected Component,” dB, December 1978.
• Morgen, Bruce, “Shield The Cable!,” Electronic Procucts, August 15, 1983.
• Morrison, Ralph, Grounding and Shielding Techniques in Instrumentation, John Wiley and Sons, New York, 1977.
• Ott, Henry W., Noise Reduciton in Electronic Systems, John Wiley and Sons, New York, 1976.
• Ruck, Bill, “Current Thoughts on Wire,” The Audio Amateur, 4/82.
Most Importantly, Qualified: A Conversion From Punk To Useful Human Being
Back when I knew everything...
Back when I knew everything, I dragged an obscenely heavy rig up an intolerably tall mountain in a woefully underpowered truck to an impossibly small theater in the backwoods of northern Pennsylvania.
Since this was when I knew everything, I had the somewhat surly local crew stack my compact TMS 3s two wide, four high, with the 15s coupled for maximum bass energy. Because stacking that box four high without a fork was sort of like pushing a dump truck up a hill with the brakes on, some of my help that day were engaging in a spirited debate regarding my experience level and the wisdom of my stacking choice.
Later, after the tinkling banjos from Deliverance had faded away, I began tuning the PA, carefully twisting up the low-frequency band-pass on the crossover while listening to selected cuts from Back in Black. I was shortly satisfied that when my client arrived and listened to my subtle, intuitive tuning, he would turn to me with a grateful smile and request the EQ across left and right be removed since there was obviously no need for it. I practiced my thank-yous in a bathroom mirror backstage to achieve a proper balance of humility and steadfast resourcefulness.
The artist that day was Frankie Valli. About an hour later Jim Sanders (a.k.a. “Redford”) strolled in, walked out to front of house, looked at the PA, looked at me, looked at the PA again then stuck out his hand and said, “I’m Redford…please don’t tell me you have the 15s coupled.”
This is not in the script. “Um, yeah… Since we don’t have additional subs, that’s what we do,” I uttered with humility and steadfast resourcefulness.
Redford: “Kid, don’t your parents have any Frankie Valli records? This is a coverage show, not an AC/DC gig.”
Me, summoning all of the forces of good, told him not to fear, I’d just re-stack the PA. He peered at me with kind eyes that spoke the words, “thank you, you are truly a professional.” Then his mouth added something like “buddy, that crew is gonna play tether ball with your intestines if you tell ‘em to pull that PA down.”
Since I knew everything, I was sure that my wizened 22-year-old brain would have a solution soon… if I just gave it a minute or so. Unaware of the delicate cerebral dance transpiring in my head, Redford continued, “Ah, screw it, I’ll deal. Otherwise you appear to be about the right size for one of those (expletive) guys to gaff tape a plumbing fixture to your head and stick you in a trashcan at load out… Come on, let’s get a drink. You like scotch?”
I do now.
These encounters have always reminded me of a late-night talk show with people like Redford as the unexpected surprise guests. You know, you’re expecting the guy from the San Diego Zoo to come out with a cockatoo that’s destined to crap on someone’s shoulder when suddenly, Bono appears to do an acoustic version of the first four songs from Joshua Tree .
They’re like snow days in July. Situations that are dripping with the hallmarks of misery by audio, but miraculously are rescued by something very simple—in this case, a personality. Inside of a short three-minute conversation, Redford had managed to convince me to join his team. Instead of me versus him, with me grudgingly doling out small concessions to his wishes, I began suggesting more work for myself: “Redford, you want I should tie into the house delays? You want the front fills in stereo? You want me to put a bag over Frankie’s head so he stops making that noise?”
To be clear, I was not an engineer sycophant when I was a puppy. But I had concluded - based upon my hours of experience - that force of personality and the imitable ability to hang typically superceded any true mixing ability.
This mangled Aristotelian logic brought me to some simple conclusions: the louder and friendlier a guy was, the less qualified he was. Quiet and unassuming? Well obviously still waters run deep. Time would later prove this and many of my other theories worthless, but I was 22 and knew everything.
Turned out Redford would be the beginning of my conversion from punk embryo to useful human being. He demonstrated to me that in addition to being loud, funny, and sarcastic, he was most importantly qualified.
He was batting in the Show and I was pretty impressed with myself for hitting off Little League pitchers. Never once, though, did he point that out to me. He so dwarfed me in skill ... yet had absolutely no need to mention it. In fact, quite the opposite, during the day he would ask me questions and solicit my opinions.
At night, he’d spin up a mix that made Frankie sound way better than he had any right to, finish his drink, shake my hand assuring me we’d “fooled them again,” and then disappear. I’d spend the rest of load-out striking the PA while collecting murmured compliments from the exiting audience. With each one I dutifully disavowed my role in the night’s success promising to pass the praise on to the mix engineer. And I always did. Or I mostly always did. Secretly I hoarded small pieces of each note and nod, hoping later to puzzle them together into a map that might point me to the Show someday.
I’ve lost track of Redford. But no matter. I’ve never lost track of the course he accidentally or purposefully set me on. It took 18 years of hindsight to truly appreciate the beauty of his unintended education.
Had he spoken at me, I would have turned all medieval and disaffected twenty-something on him, demonstrating my lethal ability to maim with a single contemptuous roll of my eyes. Instead, he included me as an equal part of his adventure, casually pointing out things he had discovered along the way. I was free to participate or not, but I was welcomed along if I wanted to come.
When it came time for the show, he would continue to point out the foibles of the world with one hand while he made the PA broadcast his right to speak with the other.
I think about Redford sometimes when I’m faced with someone young, eager and already in possession of the secrets of the world. It causes me to pause and consider inflicting patience on them instead of strychnine.
Don’t get me wrong, I’m perfectly comfortable vetting young engineers by drowning them under mountains of broken cables and castors to see how badly they really wanna play… But I know in doing that, I’m striking a bargain to pass on information that was passed on to me. The method of delivery, though, will determine how well they absorb those lessons.
In the end, they’re the next round, no matter what we say. My generation will pull the bus door shut after our final load-out, and then it’ll be their turn to live out all of the Jackson Browne songs. Might be nice though if we offered them a drink when we see them off to the Show.
Editor’s Note: Find out more about Jim “Redford” Sanders here.
Sully is a veteran live sound engineer and really has no clever off-hand remarks for this space at this time.
Fall To Grace: Digital Paths For Paloma Faith On Tour
Reinforcing a distinctly musical style at London's Hammersmith Apollo
Much talked-about UK actress-turned-songwriter Paloma Faith recently brought her distinct musical style and sizeable, glitzy set to London’s Hammersmith Apollo for a slick headline show that punctuated a successful concert tour.
Faith first started making waves in the UK music scene in 2009, with her first three singles all making the top 20. She’s been nominated for three BRIT Awards, and her second album, Fall to Grace, went platinum, peaking at number two on the UK album chart in the process.
For the tour, which saw Faith backed by a band of accomplished players, front of house engineer Huw Richards chose L-Acoustics KARA line arrays with flown subwoofers, and has been impressed. “It’s my first time using KARA, and I like it a lot; it’s a very compact and light weight system,” Richards tells me during sound check in the near empty 5,000 capacity venue. “We needed a system that would be flexible but that still packed a punch, as the venues on the tour are all different shapes and sizes.
“We’re pushing it to its extreme here (at the Apollo),” he continues, “but because of the way we are able to configure it, it actually covers a room like this better than (L-Acoustics) V-DOSC or K1 would.”
SSE Audio Group, the sound reinforcement company for the tour, supplied 48 KARA boxes, 36 of which were utilized at this London show (two hangs of 18 per side). They were joined by three hung SB18 subwoofers per side, with all loudspeakers driven by L-Acoustics LA8 amplified controllers.
SSE system tech Perttu Korteniemi (left) and engineer
Hew Richards at the house system DiGiCo SD10 console. (click to enlarge)
“We haven’t had to ground stack this system once, and the hung subs are more than enough; to be honest, KARA has not just delivered, it’s gone way beyond that,” Richmy choice of PA company. They serve us particularly well, knowing exactly what we wanted and getting it just right.”
Another aspect supplied by SSE is Richards’ DiGiCo SD10 digital house console, joined by a companion SD Rack as well as some other interesting pieces.
“Miles (Hillyard, SSE senior project manager) gave us some cool bits and bobs play around with, including the Waves PuigChild (hardware compressor/limiter), which works beautifully on Paloma’s vocal because she has quite a peaky mid range,” he explains. “We have it on a setting that works well for her; it just catches the peaks and holds them back, and also fattens up the vocal nicely. We’ve also got a Waves Maxx BCL applied across the PA.”
Monitor engineer Saul Skoutarides at his Midas PRO2 console. (click to enlarge)
Richards has favored DiGiCo consoles for some time for functionality and sound quality reasons, and he offers specifics about how he’s using the SD10 on this tour.
“I’m running 54 channels,” he explains, “and what’s great about this model is that I have 12 faders in each bay rather than eight, which is fantastic for drums. We have several snares and kicks, but I can pack the whole kit into one bay, which is ideal.
“I mix very hands-on – like an instrument, really – and this console allows me to do that,” Richards continues. “We use snapshots for faders and mutes, but that’s all.”
Perttu Korteniemi of SSE, the tour’s system tech, always gets to the venues early (“with the lampies,” he notes, grinning) to ensure all of the array hang points are in the right place.
“I tend to work from a blank canvas every day, though I’ve been here at the Apollo before, so I had the room file, which made things a bit easier,” he says. “I use L-Acoustics SoundVision software 3D prediction software, and it’s excellent, though there’s no wizard element to it.
“A lot of this stuff is IP-based now. If you have the dimensions of the room and good drawings, you can do a lot of prep work, which obviously gives you an advantage,” he continues. “With this (KARA) system, we only need two local crew members to hang it, because once propagated, it goes up so quickly. And after the show, the boxes are often down before the drum kit is, which is pretty incredible.”
It also points to the radical changes currently underway in live sound reinforcement. This was further reinforced as Korteniemi showed me the system racks, noting that the signal path stays digital in traveling from the SD Rack to the SD10 to Lake processors at front of house and at the stage, all the way through to the LA-8 amplifier controllers.
Perspective of the stage at the Hammersmith Apollo prior to show time. (click to enlarge)
“When the microphone and line level inputs hit the SD Rack on stage, their signals don’t go back to analog until it hits the speaker cables,” Richard says. “This method keeps it all simple, and level-wise, if it’s the input coming in and all faders are at zero, there’s no setting up of gains and no hiss. It’s all nice and clean, so when you open up the PA, there is no interference whatsoever.”
At the monitor position, Saul Skoutarides works from a packed-out Midas PRO2 console, utilizing 54 channels and eight auxes in providing mixes to eleven Sennheiser G3 in-ear monitoring systems utilized by Faith and her backing band. Stage fill mixes are also supplied to L-Acoustics ARCS positioned to each side of the stage.
“I used to use G2s, which were great, but these G3s have better stereo imaging, better RF, better gain structure, and a better sound board. They’re the best system out there right now,” Skoutarides states.
Faith favors a Sennheiser MD 5235 dynamic microphone element on an SKM 5200 mk II handheld transmitter for her vocal, which Skoutarides paired with a Sennheiser EM 3732 wireless receiver.
Sennheiser G3 IEM systems racked up at the monitor position. (click to enlarge)
“The 3732 has an AES output, which is great if you want to keep it digital all the way,” he says. “It’s in the ultra-wide NGB range, which goes from 606-790 (MHz), so I can also use it as a scanner for the entire band, because my in-ear systems also fall into that band. That’s a real treat.”
The Apollo show in its entirety also proved a treat, with the expert work of the sound team resulting in a technically flawless audio production. Credit should also go to Paloma Faith – this is an artist that knows what she wants, from the set to the set list, and the show oozed meticulousness. A capacity crowd went home very satisfied after witnessing a concert that sounded every bit as good as it looked. Paul Watson is the editor for Europe for ProSoundWeb and Live Sound International.
An absolute essential to a successful recording is getting things right at the source, and here, he focuses on doing it with an electric guitar. He also contrasts the sounds of a couple of very different electric guitars for illustrative purposes.
You can see the process as well as listen along to really hear the difference.
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