Technician

Thursday, January 16, 2014

See The Upcoming NAMM Show Webcast Live By PreSonus

Company sending "ninja team" to the show looking for fun, music, and new products to present to viewers

PreSonus is presenting three full days of the upcoming NAMM show in Anahaim—Webcast live for view.

The webcast will run Thursday, Friday, and Saturday (January 23—25), from 10 am to 6 pm Pacific time, and then it will be immediately repeated each evening for overseas viewers (and insomniacs in North America).

PreSonus is sending it’s “ninja team” of social media manager Ryan Roullard and associate creative director Cave Daughdrill to the show looking for fun, music, and new products to present to viewers.

So if you’re snowed in, stuck in the office, can’t get a pass, or are otherwise unable to make it to Anaheim, visit www.presonus.com/videos/presonuslive to follow the NAMM show live.

PreSonus

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Posted by Keith Clark on 01/16 at 06:02 PM
AVLive SoundRecordingChurch SoundNewsVideoWebcastAVBusinessEducationEngineerSound ReinforcementStudioTechnicianPermalink

Once Upon A Dream: Bridging The Tech Gap With The Rascals

Imagine a wildly successful blue-eyed soul group of the 1960s, pushing back against the British Invasion with many chart-topping hits including “Good Lovin’” and “Groovin.”

They became household names thanks to Top 40 radio, The Ed Sullivan Show, and the power of television. They performed to sold-out audiences across North America and Europe, and would eventually be inducted into the Songwriters Hall and the Rock and Roll Hall of Fame. Yet they were unable to escape the disillusionment that came with all this success; the principals simply disbanded by 1970 and did not play together as the original foursome for another 40-plus years. 

This is the story of the Rascals, aka the Young Rascals, who placed their careers on hold in the midst of a technological revolution, our revolution, only to return after the performance audio industry was born, developed, and matured. Now imagine the original members, who began their careers using primitive PA systems, reuniting after four decades of technological innovation had occurred, including developments by professional audio equipment manufacturers, solutions by touring sound companies, and the combined expertise of 40 years of house and monitor engineers. The contrast would be profound.

It may surprise some that high-impedance microphones plugged directly into guitar amps were the sound reinforcement systems of the day. Mark Prentice is musical director and bass player for the recent “Once Upon A Dream” tour, and has played with Rascals organist Felix Cavaliere for many years. He personally witnessed a Rascals show as a teenager, and recalled a system typical of the period.

Not so young but still kickin’—the Rascals in concert presenting “Once Upon A Dream.”

“I’m a fan as well as a guy in the band,” he told me when I met up with the tour in Toronto. “When I saw them in 1967 at Watertown (NY) High School, and the only reason I know this is because a friend of mine recently showed me a photo from that show, I think they were singing through a couple of Fender Bandmaster cabinets. Maybe a 4-channel Shure mic mixer running into a dual Showman head. No individual EQ on mics or anything, only on the guitar amp head. Possibly high impedance Shure microphones. There was certainly nothing resembling a monitor, and absolutely no one was running sound from offstage. I don’t think anyone conceived of that until Woodstock.”

Unlike many Broadway pop music revivals, these musicians are playing as a foursome with all of the original members—Eddie Brigati, Dino Danelli, Gene Cornish and the aforementioned Cavaliere. Assisted only by two sidemen and three backing vocalists, “Once Upon A Dream” is combination musical retrospective and 60s counterculture multimedia extravaganza.

The marquee for tour dates at Chicago’s Cadillace Palace Theater.

Miles And Miles
Directed by Bruce Springsteen guitarist Steven Van Zandt, with concert design by veteran Marc Brickman, the show leverages technology in a manner that simply could not have been imagined when the band cut their teeth playing tiny clubs in New Jersey. Almost every piece of equipment we take for granted would look foreign to these four when they released their first record in 1965.

No parametric EQs, no solid-state power amplifiers, no condenser mics built to survive the road, no networked system control, no in-ear monitors, no hanging loudspeakers, no digital…well, anything. Shure hadn’t even released the Vocal Master system when these guys started out.

“There was nothing in those days, oh no,” notes Danelli, the band’s drummer. “It’s come miles and miles, that’s for sure. I never sit down and think about it too much, because you just get caught up in the trip of it all.”

Fortunately, the tour is made possible by generations of sound system improvements, improvements we use and take for granted every day, guided by a fine 4-person audio staff charged with reinforcing a musical tour-de-force consisting of 30 songs and Brickman’s first-class video retrospective.

Monitor engineer Mark Hutchins pre-show at an Avid VENUE console, with Avalon 737 compressors applied to vocals mounted below.

Mark Hutchins serves as monitor engineer, and the technology he uses provides an ideal contrast between the stage of today and the performing environment of 1965. He keeps the band comfortably ensconced in an all in-ear environment essential to creating the right performing conditions, managing stage levels and facilitating timing with video content.

Mixes are done on an Avid VENUE digital console with every source miked. The deck is wedge-less save one tiny back-up monitor on the drum riser. Guitars and Leslies are isolated offstage in sound absorptive enclosures. Bass and keyboards are taken direct on DIs. A significant departure from their 60s upbringing, the Rascals stage is almost silent except for drums and percussion.

“This is not a simple monitor gig,” Hutchins states. “It’s taken some time to get them comfortable. We’re talking about musicians that haven’t been on ears their whole lives, they don’t want to be on ears. Gene looked at me the first week we worked together and said, ‘I want a monitor, I want a monitor.’ Eventually we got everybody happy.”

Musical director Prentice explains that the challenge of transitioning a band that used no vocal reinforcement beyond guitar amps to the highly devised performance environment they enjoy today was a seminal task. “In-ear monitoring is really the only way to do these shows.

Leaping from a zero monitor situation throughout their successful career to a potentially sterile laboratory environment with ears, and having to figure out how to get them feeling the music, and enjoying themselves and believing they are part of it, is the job and I think we’ve got there.”

A Matter Of Balance
After watching Hutchins mix a couple of tunes, and solo a couple of mixes, I learned that fundamentally, the primary issue is balancing Danelli’s drum kit, as the only non-isolated source onstage, with everything else. Hutchins hails from an extensive live television background, and was brought into rehearsals already underway when the band was not satisfied.

RF coordinator Brian Kingman in his world adjacent to the monitor mix position.


“I came in to observe what was going wrong, initially (under the guise of being) a video guy,” he notes. “The band wasn’t happy. It’s the old story of (balancing) a loud drummer and vocalists. I’m a drummer, and I wanted to get it right for Dino initially, so I spent a whole day playing his kit, with Brian Kingman (RF coordinator for the tour) mixing, to get the drum sound in Dino’s ears the way I thought he would like it. He came in the next day, sat down and played for 20 minutes by himself, and then looked at me and said, ‘that sounds fantastic.’ We had started to build some trust.

“Then it was a matter of understanding each of their ears,” he continues. “Gene (lead guitar) likes lots of top end, and Felix likes a midrange-scooped Steely Dan-type of sound. Very little low mids. Gene and Dino both have pretty aggressive rock mixes lots of kick and snare. Eddie doesn’t want to have any drums at all. He prefers to hear himself, some keyboard, and the background vocalists, leaning to a very unique, isolated blend of what is almost like folk music. Not like the other guys, but it works for him.”

Sennheiser ew300 IEM receivers for all performers, staged and ready to go.


Brigati is the lead singer of the Rascals and composer with Cavaliere of many of the group’s hit records. “Vocals, in my humble opinion, are supposed to be a glaze on the surface of the instruments,” Brigati states. “In rock and roll, you start with the bass drum and then build on that. I’m trying to get used to ears. You don’t hear the ambiance in the room in the same way (as) the earphones block out the ambiance in the room. An individual is feeding you a blend, but when it’s right, (IEM technology) helps me be a better singer.”

A Sennheiser A5000CP passive circulary polarized antenna for the wireless systems.


Hutchins describes the vocal treatments developed for the tour. “The only thing I’ve got going on gear-wise is two Avalon 737 compressors on Felix and Eddie’s vocals.  We went through a lot of vocal mics initially, and settled on Telefunken M81s. Felix sounded best on a Neumann KMS 105, but it just brought in too much off-axis stuff to be practical.

“Eddie needs something with a lot of rejection, but also has crooner elegance to it. A full range mic that is warm and inviting. The M81 is a good compromise, they can work around it but it also has a tight pattern. Those Telefunken mics are pretty cool.”

Adapting Realities
Chris Edwards mixes front of house for “Once Upon A Dream.” Originally the engineer at the Capitol Theater in Port Chester, NY, he joined the tour after working the initial out-of-town tryout at the Capitol to the satisfaction of director Van Zandt.

“I walked in and the theatre’s production manager said, ‘it looks like you’re going to be mixing the Rascals’,” Edwards recounts. “Steven sat with me every single night. He knows every note of every one of their records. He understood that I was a musical mixer and not just another dude in a bad Hawaiian shirt. Steven definitely had input. Trying to grasp 30 songs of new material, I didn’t hit it on the head every time, but as soon as we were cool, we were cool.”

As noted, Edwards is a music mixer by training, and has had to adapt to the realities of managing a highly-cued, theatrical type show. “The show has extensive narration that accompanies the video portions between songs,” he explains. “Many of the initial narration came from different sources with inconsistent levels and EQs, adding that getting various pieces of narration to sound right through the system was challenging: “I had never used any kind of snapshots, but (initially) I just dove right in using them to manage dialog levels and EQ.”

Later, Geoff Sanoff from Van Zandt’s Renegade Studios re-worked the narrative post-production audio to make it more consistent. Edwards: “The first time I heard the remixed dialog I hugged him. I later chose to abandon using snapshots altogether. I have a lot of experience working in old analog studios with no automation or Pro Tools, and these skills have been very useful to me in this production.

Self-described “musical mixer” Chris Edwards at his Midas PRO6 at front of house.


“I always approach the mix to honor the music,” he adds. “I’m a musician and deeply rooted in music, and have a great respect for these artists. I spent eight years as a stage tech with Levon Helm and recorded the Rambles at his barn. For me, it’s an honor to mix this show; I’m just trying to place all the parts where they should be dynamically, and pay homage to all the nuances.”

Edwards mixes on a Midas PRO6 digital desk supplied by Firehouse Productions, using loudspeaker systems provided locally by the venue in order to manage production costs. I had the pleasure of hearing two performances at Royal Alexandra Theatre during my visit to Toronto, and can testify first-hand that Edwards provides mixes with great vocal and instrumental clarity, while enhancing subtleties in the arrangements resulting in a believable, entertaining presentation.

The tour had “racks and stacks” provided locally, including Martin Audio MLA in Chicago, supplied by On Stage Audio.


Very Comfortable
Jeff Child is an independent systems tech provided by Firehouse Productions, managing another pile of gear no one could have imagined 40 years ago. Child usually tours with technology-savvy Ultrasound accounts including Dave Matthews, Further, and Phil Lesh and Friends. He struck me as very comfortable in this setting, managing adjustments for the house-supplied d&b audiotechnik Q Series line arrays with two B2 subs left and right. Q7s handled in fill and front fill duties.

“Stacks and racks are what we usually pick up. The balance is provided by Firehouse or owned by the band. Both Mark and Chris have extensive house engineer, broadcast, and studio backgrounds, so I bring a touring rock sensibility to this,” Child explains.

As noted earlier, Kingman is responsible for RF equipment and frequency coordination, and also handles earpieces and beltpacks for the artists. He uses Intermodulation Analysis Software from Professional Wireless Systems and a WinRadio spectrum analyzer to coordinate frequencies.

The audio crew at the drum riser, left to right: Hutchins, Kingman, Edwards and system tech Jeff Child.


“Frequency coordination here in Toronto has been easy,” he tells me. “I was informed that no licenses were required. To date, I’ve only had to change one frequency. The loudest thing onstage is Dino’s drums, and keeping drums, shakers, and tambourines out of vocal mics is the greatest challenge. Being older guys, the in-ear environment is very different. Our main role is to let them know ‘we are here to make you comfortable’.” In-ear electronics are Sennheiser ew300 IEM G3 systems with a Sennheiser combiner and helical antenna. All artists are on Ultimate Ears UE-11 earpieces.

Prentice notes that fortunately, the Rascals have adapted well to the profound changes in performance technology. “They’ve all become really, really comfortable in that environment. Now we just stick these little things in our ears and do a show, and I think you miss all the technological magic that has to exist to make that happen.” Fortunately for the Rascals, that technological magic happens every day because of innovation and a talented crew offering them a supportive musical environment in sharp contrast to when they first began.

“This whole phenomenon that we’re enjoying now, this re-visitation of almost 50 years ago, is about young guys that got together and cooperated and protected each other, and created together, and it was like a chance at peace,” concludes Brigati.

Danny Abelson is a consultant that specializes in the design and construction of technology systems in professional and collegiate sports facilities.

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Posted by Keith Clark on 01/16 at 05:29 PM
Live SoundFeatureBlogConcertConsolesEngineerMicrophoneMonitoringSound ReinforcementTechnicianWirelessPermalink

Wednesday, January 15, 2014

In The Studio: An Interview With “The Drum Doctor”

This article is provided by Bobby Owsinski.

 
We all know that the drums are the heartbeat of a song, and a wimpy drum sound will make the engineer work so much harder during the mix. That’s why it’s so important to get the drums to sound great acoustically before the mics are even placed. That said, it’s surprising how little many engineers actually know about making a drum kit sound great acoustically.

Following is an excerpt from The Recording Engineer’s Handbook, Third Edition that features an interview with the famous “Drum Doctor,” Ross Garfield, who’s been responsible for the actual drum sound on a multitude of huge records by some giant artists. Ross gives some hints on how to take almost any kit and make it ready to record.

Anyone recording in Los Angeles certainly knows about The Drum Doctors, the place in town to either rent a great sounding kit or have your kit fine-tuned. Ross Garfield is the “Drum Doctor” and his knowledge of what it takes to make drums sound great under the microphones may be unlike any other on the planet. Having made the drums sound great on platinum selling recordings for the likes of Bruce Springsteen, Rod Stewart, Mettalica, Marilyn Manson, Dwight Yokum, Red Hot Chili Peppers, Foo Fighters, Lenny Kravitiz, Michael Jackson, Sheryl Crow, and many more than what can comfortably fit on this page, Ross agreed to share his insights on drum tuning. 

What’s the one thing that you find wrong with most drum kits that you run into?

I think most guys don’t know how to tune their drums, to be blunt. I can usually take even a cheap starter set and get it sounding good under the microphones if I have the time. It’s really a matter of people getting in there and changing their heads a lot. Not for the fact of putting fresh heads on as much as the fact that they’re taking their drums apart and putting them back together and tuning them each time. The repetition is a big part of it. Most people are afraid to take the heads off their drums. 

When I get called into a session that can’t afford to use my drums and they just want me to tune theirs, the first thing I’ll do is put a fresh set of heads on. 

How long does it take you to tune a set that needs some help?

Usually well under an hour. If I have to change all the heads and tune them up it’ll take about an hour before we can start listening through the mics. I try to tune them to what I think they should be, then when we open up the mics and hear all the little things magnified, I’ll modify it. Once the drummer starts playing, I like to go into the control room and listen to how they sound when he plays, then once the band starts I’ll see how the drum sound fits with the other instruments.

What makes a drum kit sound great?

I always look for a richness in tone. Even when a snare drum is tuned high, I look for that richness. For example, on a snare drum I like the ring of the drum to last and decay with the snares. I don’t like the ring to go past the snares. And I like the toms to have a nice even decay. Usually I’ll tune the drums so that the smallest drums have a shorter decay and the decay gets longer as the drums get bigger. I think that’s pleasing.

What’s the next step to making drums sound good after you change the heads?

I tune the drums on the high side for starters. For tuning, you’ve got to keep all of the tension rods even so they have the same tension at each lug. You hit the head an inch in front of the lug, and if you do it enough times you’ll hear which ones are higher and which are lower. The pitch should be the same at each lug, then when you hit it in the center you should have a nice even decay. I do that at the top and the bottom head.

Are they both tuned to the same pitch?

I start it that way, and then take the bottom head down a third to a fifth below the top head.

I’ve been in awe of the way you can get each drum to sound so separate without any sympathetic vibrations from the other drums. Even when the other drums do vibrate, it’s still pleasing. How do you do that?

Part of that is having good drums and that’s the reason why I have so many; so I can cherry-pick the ones that sound really good together. The other thing is to have the edges of the shells cut properly. If you take the heads off, the edges should be flat. I check it with a piece of granite that I had cut that’s perfectly flat and about two inches thick. I’ll put the shell on the granite and have a light over the top of the shell. Then I’ll get down at where the edge of the drum hits the granite. If you see light at any point then you have a low spot. So that’s the first thing; to make sure that your drums are “true.” 

The edges should be looked at anyway because you don’t want to have a flat drum with a square edge; you want it to have a bevel to it. If you have a problem with a drum, you should just send it in to the manufacturer. I don’t recommend anyone trying to cut the edges of their drums themselves. It doesn’t cost that much and it’s something that should be looked at by someone who knows what to look for. 

Once you get those factors in play, then tuning is a lot easier. I tend to tune each drum as far apart as the song will permit. It’s easy to get the right spread between a 13 and a 16 inch tom, but it’s more difficult to get it between a 12 and a 13. What I try to do is to take the 12 up to a higher register and the 13 down a little. The trick to all that is the snare drum because the biggest problem that people have is when they hit the snare drum there’s a sympathetic vibration with the toms.

The way I look at that is to get the snare drum where you want it first because it’s way more important than the way the toms are tuned. You hear that snare on at least every two and four.

The kick and snare are the two most important drums and I tune the toms around that and make sure that the rack toms aren’t being set off by the snare. The snare is probably the most important drum in the set because for me it’s the voice of the song. I try to pick the right snare drum for the song because that’s where you get the character. 

Do you tune to the key of a song?

Not intentionally. I have people who ask me to do that, and I will if that’s what they want, but usually I just tune it so it sounds good with the key of the song. If there’s a ring in the snare, I try to get it to ring in the key of the song, but sometimes I want the kit just to stand on its own because if it is tuned in the key of the song and one of the players hit the note that the snare or kick is tuned to, then the drum kind of gets covered up, so I tend to make it sound good with the song rather than in-pitch with the song.

Would you tune things differently if you have a heavy hitter as opposed to someone with a light touch?

Yeah, a heavy hitter will get more low end out of a drum that’s tuned higher just because of the way he hits, so I usually tune a drum a little tighter. I might move into different heads as well, like an Emperor or something thicker. 

How about the kick drum? It’s the drum that engineers spend the most time on.

It’s weird for me because I always find them to be pretty easy because you muffle the kick drum on almost every session and when you do, it makes tuning easier. On the other hand, a tom has as much life as possible with no muffling. 

What I would recommend is to take a down pillow and set it up so that it’s sitting inside the drum touching both heads. From there you can experiment, so if you want a deader, drier sound then you push more pillow against the batter head, and if you want it livelier, then you push it against the front head. That’s one way to go.

Another way to go is to take 3 or 4 bath towels and fold one of them so it’s touching both heads. If that’s not enough then put another one in against both heads on top of the first one. If that’s not enough then put another one in. Just fold it neatly so that they’re touching both heads. That’s a good place to start, then experiment from there.

Do you prefer a hole in the front head?

It makes it easier. I do some things without holes in the front head, but having it really makes it easy to adjust anything on the inside. No front head is good too. It’s usually a drier sound and you’re usually just packing the towels against the batter head. Just put a sandbag in front to hold the towels against the head. 

How about cymbals?

One thing for recording is that you probably want a heavier ride but you don’t want that heavy of a cymbal for the crashes. You also have to be careful when you mix weights. For example, if you’re using Zildjian A Custom crashes you don’t want to use a medium.  You want to stay with the thins rather than try to mix in a Rock Crash with that because the thicker cymbals are made for more of a live situation. They’re made to be loud and made to cut and sometimes they can sound a little gong-like to the mics. On the other side of the coin, if you playing all Rock Crashes and the engineer can deal with the level, that’s not so bad either because the volume is even, but a thinner cymbal mixed in with those would probably disappear. 

What records better; big drums or smaller ones?
I depends what you want your track to sound like. When I started my company, people would always say to me “Why would someone want to rent your drums when they have their own set?” For one simple reason; most drummers have a single set of drums. If they’re going for a John Bonham drum sound, they’re not going to get it with say a “Ringo” set.

A lot of times when they go into the studio, the producer says, “You know, I really heard a 24” kick drum for this song. I hear that extra low end,” but the drummer’s playing a 22, so it’s important to have the right size drums for the song. If you’re going for that big double headed Bonham sound, you really should have a 26. If you’re going for a Jeff Porcaro punchy track like “Rosanna,” then you should probably have a 22. That’s my whole approach; you bring in the right instrument for the sound you’re going for. You don’t try to push a square peg into a round hole.

How much does the type of music determine your approach?

The drums that I bring for a hip-hop session are actually very close to what I bring for a jazz session. Usually the hip-hop guys want a little bass drum like an 18 and that’s what’s common for a jazz session, to have an 18 or a 20. Then maybe a 12 or a 14 inch rack tom, which is also similar to the jazz setup. The big difference is in the snare and hi-hats and the tuning of the kick drum and the snare.

On a jazz session I would keep the kick drum tuned high and probably not muffled. On a hip-hop record I would tune the kick probably as low as it would go and definitely not have any muffling so it has that big “boom” as much as possible. I would also have a selection of snares from like a 4 by 12 inch snare, 3 by 13 and maybe a 3 by 14. On a jazz record, I’d probably send them a 5 by 14 and a 6 1/2 by 14. The hi-hats on a jazz record would almost definitely be 14’s where a hip-hop record you’d want a pair of 10’s or 12’s, or maybe 13’s.

Obviously it’s open to interpretation because I’m sure a lot of hip-hop records have been made with bigger sets, but when I’ve delivered what I just said, it usually rocks their boat.

Go here for more on The Drum Doctors.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. Get the The Recording Engineer’s Handbook, Third Edition here.

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Posted by Keith Clark on 01/15 at 06:25 PM
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Church Sound: Stopping Hums, Buzzes And Shocks On Stage — Volts

Provided by the No Shock Zone.

 
Most musicians really don’t want to learn about electrical engineering, or even how basic electricity works. Everyone, however, should learn how to test for and avoid electric shocks on stage.

Guitar amps and mixing boards as wired from the factory are inherently safe, but they can become silent-but-deadly killers if plugged into an extension cord or wall outlet that’s improperly grounded. This is because guitars are held in moist hands while wet lips are touching another electrical circuit, the microphone.

It’s up to you, the sound tech and musician, to make sure a guitar or microphone is never electrified due to poor maintenance, bad connections or a broken-off ground plug.

The so-called “Hot-Chassis” problem is what causes a tingle or shock when you touch the mic with one hand or your lips while holding a guitar with your other hand. The cause is that a chassis has become “hot” through a wiring fault.

What Is This Volts Thing?
What’s so hard to understand about electrical shocks in general is that they don’t seem to happen for any obvious reason. For instance, you can watch a pigeon on a power line that’s not being shocked, yet sometimes just holding a guitar while standing on wet ground can bring you to your knees. Why is that?

The first thing to understand about electricity is the concept of voltage. Think of voltage as electrical pressure, just like the pressure in a tank of water.

Now in a tank of water we measure pressure in something called PSI (pounds per square inch), which will, of course, increase if we get a deeper tank. This pressure is caused by the pull of gravity from the Earth and if you hook up a hose to the tank, the water will flow toward the ground.

So while 10 PSI of water pressure from a short tank might give you a trickle of water when hooked up to a hose, 100 PSI of water pressure from a really tall tank gives you a stream that will spray much farther.

Click to enlarge.

Water — and electricity — tries to flow to the side of least pressure. You can imagine that if a pipe is connected between two tanks with exactly the same water level and pressure (say, 100 PSI) there will be no flow of water through the hose. It just sits there and does nothing because the system is equalized.

However, if you connect one tank with 100 PSI of water pressure to another tank with 10 PSI of water pressure, water will flow from the high tank to the low tank. We measure this water flow in gallons per minute.

Under Pressure
The same thing happens with electricity. You’ve often heard of “completing an electrical circuit,” but think of it as pipes between different electrical pressures.

Click to enlarge.

Getting back to the pigeon on the power line, if both of the bird’s feet are on the same wire, they’re at exactly the same electrical pressure. Because they’re at the same pressure, there’s no electrical current flowing through the bird.

If, however, the pigeon is unlucky enough to touch one foot on a power line and a wing to the grounded metal power pole, then his foot will be at 1,000 volts (think PSI of water pressure) and his wing at 0 volts (think an empty tank with zero pressure).

This will cause a lot of current to flow through the bird, which we’ll measure in amperes. And indeed, 1,000 volts across a pigeon can cause a bird explosion.

Hot Chassis Shocks
Now, consider your guitar. Sometimes you may feel a shock when you touch one hand on the guitar with the other hand on the mic.

What’s happening is that there could be an electrical voltage (think pressure) on the strings of your guitar, which is waiting for some different electrical voltage level to head towards. If your entire body is at the same voltage, then like the pigeon every part of you is at exactly the same voltage. And like the pigeon, there’s no current flow and you feel no shock.

However, if your one hand is on the mic at essentially zero volts and your other hand is on your guitar at 120 volts due to a wiring problem, you become the pipe and the different electrical pressure (volts) will push current (amps) through your hand, arm and chest cavity, then out through your other hand.

If your hands are dry, there might be so little current flow that you might not even feel it. But put a damp hand on your guitar strings and wet lips on the mic and you’ve made a good connection from the power plug of your guitar amp to the ground of the PA system.

In the case with an ungrounded guitar amp, a lot of current will flow through your body, which you’ll quickly recognize as a shock and potentially an electrocution.

Heart To Heart
The dangerous part of shocks is when this electrical current flow goes through your chest cavity since right in the middle of you is your heart, and hearts don’t like to be shocked. That’s because your heartbeat is controlled by electricity which comes from your own internal pacemaker.

And just like a clock radio can be scrambled by a nearby lightning strike, even a small amount of electricity passing through your heart can cause it to start skipping beats and cause a heart attack. Just how little? I’m glad you asked.

I’m sure by now you’ve seen the 20-amp marking on a circuit breaker. That means it can supply 20 amps of current flow when asked to do so. Again, think of it as gallons per minute of flow, and amps are indeed a count of electrons per second flowing through a wire (think pipe).

Much more on that later, but it takes less than 5 milliamps of current to cause your heart to go into fibrillation mode.

That’s just 5/1000 of an amp or 0.005 amps of alternating current to cause what’s essentially a heart attack. It takes just 30 volts of alternating current (AC) to stop your heart if your hands are wet.

On the strange-but-true side of the coin, while 60 Hz AC is what comes out of your wall outlet) can cause your heart to go into fibrillation and stop pumping blood, the emergency rescue crew will use direct current (DC) of several hundred volts to reboot your heart and get it beating regularly again.

That’s what they’re dumping through the paddles placed on your chest — direct current from big capacitors like you see charging on the TV dramas before they yell “Clear!”

Play It Safe
The first rule of staying safe from electrocution is to keep your heart out of the current flow. You can see that getting shocked from hand to hand or hand to lips or feet is about as bad as it can get.

That means if you’re plugging in your guitar amp with one hand, the last thing you want to do is hold onto the metal rail around the stage with your opposite hand or be kneeling on the wet ground. If you have two points of contact and something goes wrong (like you touch a bare wire), the current will flow to your opposite hand or feet, passing through your heart in the process.

So always use just one hand when plugging or unplugging your power cords for your amps. Not doing so is to invite death by electrocution, and, really, who wants that?

Keep Grounded

Take a look at a typical 120-volt grounded wall outlet, shown in the image at left. The top half of the illustration shows the sideways slot of a 20-amp outlet, while the bottom half shows a more common 15-amp outlet.

In both versions you’ll see a Hot connection (the short blade), a Neutral connection (the tall blade), and a U-shaped Ground connection (called the safety ground).

Click to enlarge

Those ground blades are on the power outlets and plugs for good reason. If something goes wrong internally with the amp (say a wire shorts to the chassis or a power transformer gets leaky), that ground blade is supposed to divert the voltage from the strings of the guitar through the ground in the power panel, which will then trip the circuit breaker.

If the circuit breaker doesn’t trip because you’ve eliminated the safety ground by breaking off the ground blade of your power cord, then you may have an electrically hot guitar or microphone in your hands. And you may not realize it’s electrically hot until you touch something else that’s grounded with your other hand or lips, just like the bird holding onto the power line with his feet doesn’t get shocked until his wing touches the grounded metal power pole. Then it’s lights out!

So if you circumvent that safety ground by cutting off the ground blade or using an adapter plug like you see on the left in an attempt to stop hums or buzzes in your sound system , you can put your heart in the middle of the ground path and risk your life every time you plug in your amp.

Don’t do it.  Always ground your amps and PA system properly.

Make your stage a No~Shock~Zone
By grounding every amp and mixer in your sound system properly you will help create a “No Shock Zone” on stage, making it a safe place to perform without fear of getting shocked or electrocuted.

So take this seriously… if you or anyone in your band is getting shocked by a guitar or mic on stage or even in your practice basement, now is the time for action.

Quick Tips

  • Use only one hand to plug or unplug any power cables for your amps.
  • Don’t cut off the ground blade of your amp or mixer power plug to stop a hum in your PA.
  • Never stand or kneel on wet ground while touching a guitar, keyboard or microphone.
  • If you feel a shock on stage, avoid further contact until you can determine the source of the problem.

Mike Sokol is the chief instructor for the HOW-TO Sound Workshops and the HOW-TO Church Sound Workshops. He is also an electrical and audio expert with 40 years in the industry. Visit the No Shock Zone Website for more electrical safety tips.

This article is provided as a helpful educational assist with sound system setup and musical performance, and is not intended to have you circumvent an electrician or qualified audio technician. The author and the HOW-TO Sound Workshops will not be held liable or responsible for any injury resulting from reader error or misuse of the information contained in these articles. If you feel you have a dangerous electrical condition in your PA system or instruments, contact a qualified, licensed electrician or audio installer.

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Posted by Keith Clark on 01/15 at 05:09 PM
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Tuesday, January 14, 2014

Church Sound: Eighteen Live Audio Mixing Tips & Tricks

This article is provided by Behind The Mixer.

 
These gold nuggets of mixing/audio production wisdom are insights into doing something small to make a huge impact. My notebook is filled with audio mixing tips and tricks from the Gurus of Tech 2013 conference earlier this year. 

That tells you two things; the conference was great, and I still go old school with a paper notebook. If I wasn’t writing down something I thought was useful, I was writing down something I thought you’d find useful.

1. Consider building your mix off of a template. 

—Consider all of the instruments and singers in the worship band. Consider a template of presets with the following in mind;

—Engage the HPF (high-pass filter) for channels which usually benefit from a HPF.

—What channels would likely benefit from compression? Set their threshold but don’t engage it yet.

—Start all faders at unity.

—Consider where the vocalist sits in the mix – are they more high, mid, or low-range singers? Now you know where to carve out space for them in the other channels.

This template concept is a great way to build a mix from scratch. You could add to the above points if you think about it.

2. Use compression for producing a well-rounded sound.

Having multiple channels with a wide-range of volume dynamics makes it difficult to produce a well-rounded sound. Use compression to even out many of those volume spikes.

3. Hear what your live microphones hear.

Listen via PFL/SOLO to a vocal microphone and pay attention to all of the other sounds the microphone is picking up. This gives you an idea of other stuff your microphone is picking up and why microphone proximity to the sound source is so important.

4. Know what you COULD be boosting.

Microphones on the stage can pick up a variety of background sounds. In particular, boosting the high-end frequencies of a vocal microphone can pick up drum cymbals and unintentionally accentuate them.

5. Pull your male singers out of the mud.

Cut your male vocals in the 325-350 Hz range to clear up your vocals. Often, the 325-350 Hz range is where the muddiness exists.

6. Use reverb for vocal separation.

Using a lot reverb, you can push a singer into the background. Using a little, you can make it stand out in the mix. Use your ears to find out what’s best for your situation.

7. The kick drum and bass can work together on your low end sound.

Try letting the bass give you the tone of the low-end while letting the kick drum win on the attack.

8. If you are ever prone to hitting a piece of equipment to make it work…

…once is maintenance, twice is abuse.

9. Keep your headphone volume down by using delay on your solo buses.

Slow down the solo bus with delay to sync your headphones with the PA so you don’t have to run headphones overly loud. If you’re 75 feet from a speaker cluster, try a 75 millisecond delay. Using an analog board, run the headphone out to an external delay unit and then into a headphone amp and then back to the headphones.

10. Don’t let a power outage take out your system.

Use APC units for keeping the power going to your vital audio equipment. Power can go out for a number of reasons, and using an APC unit can keep your system powered and your service going.

11. Plan on equipment failure.

Ask yourself questions like, “what could fail?” “how could we get around it?” and “what’s the least amount of equipment we need to keep the system going?” Make a plan for equipment failure so when it does happen, you’ll be prepared.

12. Use a 911-microphone.

Wireless systems go out. Wireless mic batteries die. And even DI boxes can go bad at the worst time. Set up a wired vocal microphone on a microphone stand, with a long microphone cable. Place it just off-stage. Grab an extra DI box and 1/4-inch cable and place that at the base of the microphone stand. 

The next time a microphone or DI goes out during a service, you (or a musician) can pull out the 911-microphone or your emergency setup – whatever you want to call it. And in the case where it seems all the equipment comes crashing down, a church audio system only needs one channel and one microphone.

13. Don’t forget about the HPF and LPF.

The high-pass filter allows high frequencies to pass through while the low-pass filter allows low frequencies to pass through. If you don’t need low frequencies out of a channel, then engage the HPF. If you don’t need high’s from a channel, use the LPF. In the case of HPFs and LPFs that have controllable frequency points, sweep the point until it’s noticeable in the mix, then back off a little.

14. Take control of your house EQ by controlling the Q-value of your cuts and boosts.

On a standard 32-channel rack EQ, the Q value is the same with the exception that it might automatically tighten up if a cut is below 3 dB.  Therefore, if you run a digital mixer, use the on-board master EQ to alter the house EQ. This gives you the ability to also control the Q-value of your cuts and boosts.

15. Use a ducker for background music and announcements.

A ducker, on a digital console, will automatically cut the volume of a channel when it detects sound on another channel. Therefore, it’s great for the “one-man operation” when you’re running all over and you have background music and the pastor starts talking when you aren’t in the sound booth. 

You can set the delay for the period of time in which the music channel comes back up after they stop talking. This way, if they’re taking a breath before talking again, the background music could say low in volume.

16. Don’t discount frequency bands of an instrument.

For example, try adding a lot of high-end on your toms.  Even the bass guitar has usable sounds that aren’t just in the low end.

17. Use meaningful distortion.

Distortion can work on more than a bass or a guitar. It can even work on a snare drum. Distortion can sound different depending on how you use it and set the appropriate parameters. When you do use it, use it because it helps the overall sound of the song.

18. Don’t forget about gating.

Try focusing your gating around a frequency range, if possible. Not only do you benefit from only broadcasting the sound once the input reaches a certain volume level, you can know it’s when you are getting the frequencies you desire. Imagine what you could do with a kick drum or a tom.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians.  He can even tell you the signs the sound guy is having a mental breakdown.

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Posted by Keith Clark on 01/14 at 04:14 PM
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SynAudCon Web-Based Training Sales Continue Growth

Success attributed to a variety of factors, including very high information retention rate

SynAudCon announced that its web-based training sales increased 15 percent in 2013.

The web-based training program is headed up by Pat Brown, who uses the educational methods he has found work best after years of teaching popular in-person audio education courses. SynAudCon online courses include animation, graphics and sound, analogies, interactive calculators, and scenarios that enhance the learning process and add a dose of fun as well.

“We attribute the success to four major things: people being more receptive to web-based training, the multi-media presentation, the flexibility, and most importantly, students retaining a high percentage of the information,” explains Brenda Brown of SynAudCon.

SynAudCon offers five web-based training courses, including How Sound Systems Work, Principles Of Audio, Transformer-Distributed Loudspeaker Systems, Audio Applications – System Optimization and Equalization, and Sound Reinforcement For Designers.

Each course includes a series of lessons and quizzes that help insure the materials are processed and understood. Graduates receive a certificate of completion, and courses are approved for Continuing Educations Units (including RUs).

“The convenience of web-based training consistently ranks high on our evaluations,” adds Brenda Brown. “Participants can take the training when it works for them. Students have up to 45 days to complete the course, and they can repeat lessons as often as needed to fully grasp the principles.”

She points out that follow-up evaluations of students shows an 80 percent retention rate on average, with many up to 90 percent. “We’re extremely pleased with these numbers. It’s a higher percentage than most of the research that is published on retention using seeing and hearing,” she notes.

Each course comes with a one-year membership to SynAudCon, providing access to blogs, articles and the organization’s noted online community of audio professionals.

SynAudCon

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Posted by Keith Clark on 01/14 at 02:07 PM
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Monday, January 13, 2014

BT To Emcee NAMM Foundation’s 29th Annual Technical Excellence & Creativity (TEC) Awards

Chad Smith of the Red Hot Chili Peppers, Everclear’s Art Alexakis, George Clinton and more scheduled as presenters

The “mad scientist of electronic music,” known simply as BT, will host the NAMM Foundation’s 29th Annual TEC Awards (Technical Excellence & Creativity), Friday, Jan. 24, in Anaheim during the upcoming NAMM Show

The accomplished technologist and composer of hit Hollywood films Monster, The Fast and the Furious, Go and others will oversee the evening’s ceremony. He will also perform at the awards, which has become the foremost program recognizing the achievements of audio professionals.

Confirmed presenters include famed drummer and music-education advocate Chad Smith of the Red Hot Chili Peppers, electro-funk icon George Clinton, “Chilli” and “T-Boz” from the band TLC, Everclear’s Art Alexakis, Al Schmitt, who has recorded and mixed more than 150 gold and platinum albums, producers Glen Ballard (Alanis Morissette) and Ed Cherney (Bonnie Raitt, Eric Clapton), as well as four-time Grammy-winning engineer Jimmy Douglass, to name a few.

The NAMM Foundation selects TEC Awards presenters from a field of highly accomplished musicians, recording engineers and songwriters.

A precursor to the Grammy Awards, which are held the same weekend, the TEC Awards recognizes Outstanding Technical Achievement in Product Design across 22 categories and Outstanding Creative Achievement in Sound Production in eight categories.

Recording artist Todd Rundgren will receive the foundation’s highest honor – the Les Paul Award. Two honorees will be inducted to the TEC Awards Hall of Fame - sound enforcement pioneer/audio engineer John Meyer, and the man whose drumming recordings have been heard on 5,000 records, Hal Blaine. Title sponsors include Harman Professional and the Les Paul Foundation. A complete list of nominees can be found here.

NAMM Foundation

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Posted by Keith Clark on 01/13 at 04:35 PM
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In The Studio: Injecting Some Soul Into Your Click Track

This article is provided by the Pro Audio Files.

 
Click tracks are like Kryptonite to musicians. There isn’t a lot of enjoyment one gets from recording with them.

Click tracks sound like nails being hammered into a coffin. A coffin you’ll find the remnants of your soul whimpering in. OK, maybe that’s a little harsh, but clicks aren’t fun and can suck the life out of a groove sometimes.

However, click tracks are often vital to a recording session. They’re a necessity, but one that can rob a session of its vibe.

Connoisseur Of 8th Notes

One of the shortcomings of a click track is that it’s impartial to wherever the 8th note may be sitting. Which means, it really has no feel. Anyone who has spent a lot of time cutting records can write a dissertation on how much the 8th note may vary between grooves. It’s a subtle, but hugely important detail.

There is a lot of blank space in between two clicks. A lot can happen in that great canyon of space. A lot of interpretation can be made.

In theory, you could lock to a click and still not have nailed the feel. But how, if it’s in time?! Playing in time is only part of the job. Negotiating the distance in between the clicks is a far more difficult challenge.

This is one of the things that finessed drummers do. They define the space. I work with a great drummer named Doug Yowell on occasion. His sensitivity to the placement of the 8th note is super refined. I would say he’s a connoisseur of 8th notes. He has a masterful control over the what the youngsters call “quantization’ (or what we old types call “feel”). It’s the space between the notes.

Lava Lamp

My aim is to create a mood while recording. “Click, Click, Click, Click…” may create a mood, but not one of happiness and comfort. A traditional click is often restrictive to an artist’s performance.

This led me to the idea of creating my own loops to use as a click track. I would clap my hands, tap keys on glasses, sweep a broom on the floor, stomp my feet and use whatever was around as percussion.

The goal is to create not only a groove in time, but one with some sort of feel related to the song.

Of course, once you start doing this, it doesn’t give you a lot of flexibility in moving the tempo around. This is why before you even think about recording you should spend some time with a metronome to lock the tempo. Do your homework first!!

Wild Horses

This was the technique I used on the Lonely and the Moose record “And All Of The Space In The Whole Wide World.” We recorded the album in a cabin at a horse ranch in Colorado. Take the song “Lonely” as an example: Listen

The homemade click loop worked so well we decided to keep it on the final recording. On occasion, these loops can become part of the ambience of a song.

Method

To create the homemade click track for the “Lonely,” I did the following: stomp for 4 bars, find the best bar, cut it and loop. I would tap on the countertop for 4 bars and pick the best bar. Then loop it. Next? A broom on the floor to create a scraping sound… and how bout some crumbling paper? I kept going until it felt “vibey.”

Drive Thru

If you’re not good at recording found sounds, you can also use loops. On a recent session for a Christmas song (“XXX Mas Song” by Bryan Dunn and Andi Rae Healy), I didn’t want to use a regular click.

For this session, it was easier for me to just drop in a drum groove using EZ Drummer, especially since live drums were not going to be recorded for the first session. I didn’t use a stock midi pattern, though. I wrote in my own part.

The reason I don’t like a lot of pre-fab midi grooves is they tend to be busy. Since I grew up originally as a drummer, it’s easy for me to program. If that’s not in your skill set and you must use a pre-fab groove (nothing wrong with that), start off with something simple and add slowly.

Click Replacement Therapy

If you don’t have EZ Drummer or BFD, you can program a simple percussion track using a tambourine or shaker. But don’t just set the click track to play a tambourine on the beats. You want to create something with a feel similar to the song.

If you have a sampler like Kontakt or EXS, you can take single hits of sounds you’ve sampled and use them later for other clicks at different tempos. You can build a library of found sounds that make wonderful click tracks.

I’ve also used loops from iTabla, which is an app for the iPad. Within the app, there are rhythms for tablas with good samples. Sometimes, it’s perfect for the song’s vibe.

The Math

You should figure out the lowest common denominator of the song and include that in your loop. If it’s an 8th note feel, there should be 8th notes in you’re homemade click. If it’s a 16th note feel, there should be 16th notes in your homemade click.

This is especially important if there isn’t a drummer present as an ambassador to the space between notes which we call “feel.”

All Together Now

If you’re playing with a loop or homemade click track, you can include it in everybody’s cans. That’s another weird thing about click tracks. Some engineers will only feed the click to the drummer.

So you’re following the drummer and the drummer is following the click. Sometimes not everyone is seeing eye to eye and the drummer has to negotiate which direction he leans since he’s the only one hearing both.

What do I mean? Say there is a bridge that has a lot of energy. Naturally, everyone wants to push a little. Perhaps it’s on the verge of rushing, but in a natural, emotional kind of way.

Without a click track it’s easy because everyone is following each other. But when the drummer is chained down, they have the click pulling them one way and the band the other.

Why Ya Gotta Be So Mean To The Drummer?!

By having a cool loop/click track and placing it in everybody’s headphones, everyone will play together. Yes, there is still going to be moments when everyone may push forward or behind the beat feel-wise, but everyone has the same relation to the click. Everybody feels the same push and pull.

Tick-Tock

It may seem like a major time suck, but it’s worth spending the time to set up. It sets the mood for the feel of the song. Feel is everything! You’re better off spending extra time at the beginning of the session rather than risking the musicians getting frustrated later on.

Your goal is to get great performances. Prep time is your best ally to achieve this goal.

Click forth, my good people!

Mark Marshall is a producer, songwriter, session musician and instructor based in NYC.

Be sure to visit The Pro Audio Files for more great recording content. To comment or ask questions about this article, go here.

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Posted by Keith Clark on 01/13 at 03:58 PM
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Friday, January 10, 2014

Nine Things Systems Integrators Can Count On In 2014

This article is provided by Commercial Integrator

 
Variables aren’t variables if you count on them, so let’s lay out market factors to consider in strategic planning. Here are nine things that systems integrators can be absolutely certain about in 2014.

1. Technology will continue to advance rapidly, making it more difficult to keep up with change. Integrators will need to be far more “engaged and informed,” and become experts at anticipating the next generation of product offerings they need to incorporate to remain relevant to clients. Industry expertise will become absolutely necessary to become or remain profitable.

2. Numbers and financial metrics will become your best source of information. With no room for error and with eroding margins, it will become a necessity to know the true cost of each and every project. You will have to know your labor utilization ratios and per-employee revenue numbers based on industry average versus actual.

Identify and bid based upon the true cost of labor on projects, as well as what it costs to “roll a truck.” You will learn to determine whether your margins and markups are ahead of or behind industry averages. You will start to compare wages, benefits, and operating expenses with similar companies. (To benchmark yourself against the industry, find NSCA’s Labor Installation Standard on nsca.org.)

3. Health-care reform will not be repealed. Even if Republicans upset the pollsters and win both houses in November, they’ll never get the two-thirds majority in each house needed to overturn the Affordable Care law. They can defund it and chip away at it, but it’s not going away. It’s the law, so plan accordingly this year because the employer mandate will happen on Jan. 1, 2015.

Be sure to consult experts who can not only help with the insurance issues, but also help identify your future cost model. Also, beware that even with the delay of the online SHOP enrollment, there are still avenues for small businesses to participate in Affordable Care if need be.

4. Interest rates will stay low, and then rise slightly. The federal funds rate, which is the rate the Federal Reserve uses to influence interest rates and the economy, is at 0.25 percent, a historic low. The Fed is beginning to taper its easing as the economy heats up. It has promised no rate increases while the U.S. unemployment rate remains above 6.5 percent. The economy will grow this year, which means that rates will not go down.

To minimize inflation, the only effective way for the Fed to try to control the flow of money leaving its $4 trillion balance sheet is using interest rates. Interest rates could go up sooner if it’s not managed effectively. I doubt this will be significant, but it could be costly to business owners who don’t lock in rates soon.

5. There will be no significant tax increases. I see no significant tax increases on the table this year. In 2013, we absorbed increases to capital gains and individual rates (now at 39.6 percent for top earners), decreases in deductions, and added taxes for Medicare and unearned income. And we’ll keep paying those in 2014. But there’s nothing significantly new on the horizon.

6. It should be easier to get financing. The banking industry has recovered from the last crisis. The economy has moderately improved. Rates are low. Banks’ balance sheets look better. Your balance sheet looks better.

The venture-capital industry is flush and looking for more opportunities. There were many initial public offerings in 2013, and many more scheduled for 2014. This will be a good year to look for cash, new financing, or investors. A healthy balance sheet is still key to your banking relationship.

7. You will pay your employees more. The U.S. unemployment rate is down. Economic activity is moderately rising. Wages have been depressed for years. But in 2014 the competition for good people will continue to heat up. Skilled workers will go at a premium. Others will ask, and receive, better increases than in prior years. It’s quickly becoming a seller’s market for employees, and that means business owners will pay a premium this year.

8. Your cost of doing business in the cloud will continue to decrease. Research firm Gartner forecasts that the market for software as a service applications will top $22 billion through 2015, up from more than $14 billion in 2012. Cloud-based applications are proliferating. The number of companies that offer cloud-based managed services is increasing. And so is the number of small companies that are embracing these technologies.

Companies like Amazon Web Services are cutting monthly fees for services that will be popular for small businesses. Costs are declining and will continue to go down in 2014. This year, you move more to the cloud.

9. Your customers will be even more educated. End users, and especially clients and projects with a strong IT influence, will continue to educate themselves on your systems and solutions. Buying decisions will be influenced by this, and more buyers will test your expertise and service capabilities.

We will need to stay ahead of this and continue to become IT-savvy solutions providers. We’ll be discussing this topic in detail at our 16th annual Business & Leadership Conference in Dallas on Feb. 27—March 1. Plan to attend to learn more about what your clients will expect from you in regards to IT.

Chuck Wilson worked as a sound contractor for more than 20 years and is now the director of the National Systems Contractors Association (NSCA).

Go to Commercial Integrator for more content on A/V, installed and commercial systems.

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Posted by Keith Clark on 01/10 at 03:46 PM
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Monday, January 06, 2014

SynAudCon Announces Spring 2014 In-Person Training Schedule

SynAudCon announces 2014 in-person training schedule.

 
Synergetic Audio Concepts (SynAudCon) has released their in-person seminar schedule for the Spring of 2014.

SynAudCon is renowned for their real-world audio educational offerings through web-based and in-person training offered worldwide.

SynAudCon will offer the three-day “Sound Reinforcement for Technicians” (SRT) in Portland, Oregon on February 24-26, 2014 and again in Cincinnati, Ohio on April 2-4, 2014.

SRT instructor Pat Brown provides hands-on exercises which allow attendees to use a tool kit (that includes meters and other items that are needed) to test and troubleshoot systems. The class also goes into detail on how to use modern dual-channel FFT measurement platforms. On day three, SRT demonstrates the setup of a 3-way triamped loudspeaker, including polarity testing, equalization, crossover selection and signal alignment.

“SynAudCon Digital”, a three-day seminar, will be presented April 28-30, 2014 in North Haven, Connecticut. “SynAudCon Digital” is designed to provide a comprehensive introduction to digital audio, digital signal processing and digital audio networks. The materials presented shorten the learning curve for understanding everything from data formats to networked audio systems with an emphasis on the practical. The seminar is taught by Pat Brown, Steve Macatee and Bradford Benn.

For more specific information about the 2014 schedule, seminar agendas, and online registration, visit the SynAudCon website.

SynAudcon

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Posted by Julie Clark on 01/06 at 03:00 PM
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Thursday, January 02, 2014

What Have You Done For Your Ears Lately?

Chances are you make at least part of your living with your ears. Stop and think about it. Could you perform your job as well…would your income level be the same…would your professional reputation be intact if you suffer severe hearing loss?

Both musicians and the live sound technicians who work with them need to be able to hear things. Not just hear them well, but hear them better than the average person. This should make us stop and consider our own hearing health, and the environments that we work in.

What have you done for…(and to) your ears lately?

Work-Related Hazards
Did you have your head deep inside a bass bin, listening for a 60-cycle hum, when somebody pushed “play” on the CD player? Were you walking past the tri-amplified sidefill stack, with your ear at compression driver level, when the lighting crew’s ladder knocked the center stage vocal mic stand over into the floor wedge to induce non-stop feedback? Did the drummer hit his primary crash cymbal, hard, 3 inches from your ear, while you were on the drum riser adjusting the hi-hat microphone?

Each of these typical events can be a daily occurrence on a typical concert stage, but any one of them might be the accident that causes you to have either temporary or permanent hearing loss. This could result in a shortened career and a decreased ability to earn a living with your chosen skill.

Accidents are one thing. Constant and intentional exposure to high sound levels is yet another. Did you just finish a 50-show run in tiny concert clubs with that new speed metal band? Was your powerful cue monitor wedge placed on end only one foot from your right ear as you mixed stage monitors for that entire world tour? Do you check 64 house mic line inputs every day with a ragged set of stereo headphones while listening to a clipping headphone amp?

Chances are good that your ears at least need a rest; but there are also certain techniques that can be employed to offer the maximum amount of protection to your hearing as you continue to do your job.

Hearing Protectors
Earplugs are now in use more and more frequently by ushers, security guards, video crew persons, and others who must work at their job while surrounded by the high-level sound intensity of today’s rock music concert programs.

Throw-away foam-type plugs are often issued on a daily basis at arenas and auditoriums for the working crews; some facilities have a nurse or public health official who will provide these items to any member of the general public audience who complains about loud sound levels.

If you’re a technician who works around powerful sound systems, but is not actually responsible for mixing sound during the show, it is a good idea to have some sort of hearing protection device handy.

The same is true if you are a sound professional who is waiting around for your band to come on while listening to someone else operate a loud system. Here are some basic options:
Disposable Foam Plugs. This type of hearing protection device comes in a small cardboard or plastic pouch, and several can easily be stuffed in a shirt pocket or a briefcase pouch. They are disposable, intended for one-time use. Common brands are E.A.R., and DeciDamp from North Health Care. Such devices offer a noise reduction rating of about 12-20 dB, depending on frequency. These plugs mainly reduce high frequencies.

Re-usable Silicon Insert Plugs. These rubberized insert cushions conceal tiny metal filtering diaphragmatic mechanisms to attenuate sound levels. They are often seen in use by gun buffs, construction workers and heavy equipment operators. The Sonic Valve II comes in its own plastic storage case with a key chain attached, and offers about a 17 dB noise reduction rating. Often available in gun shops or industrial safety supply stores, a pair can run from $15-20.

Personal Custom-Fit Earmolds. The best hearing protection device, and the one most applicable to working around musical sound, is one that attenuates all frequencies evenly. When correctly designed and properly fitted, custom-molded flexible plastic earmolds can offer 15-20 dB of balanced noise level reduction; in other words, full-frequency sound is still heard, but at a reduced level. There are numerous suppliers, who provide custom fitting services as well, such as Sensaphonics.

Industrial Headsets. When maximum attenuation of very loud sounds is desired, particularly at low frequencies, the cushioned headset works well. Offering up to 30 dB of attenuation, hearing protectors from David Clark have cushioned headpads and tight-fitting earseals. This is also an option for person who do not wish to stick standard earplugs inside the ear.  This is the type of protection often seen in use on airport runways and in the cabs of tractors and heavy cranes at construction sites.

Protecting Your Hearing On The Job
Use mini-nearfield monitors as a cue system for live mixing instead of headphones whenever possible.

By placing one or two small, powered monitors at your mixing console position and giving them the output from your stereo cue bus, you are able to solo up a mic input or an output mix and hear the signal without having to put on regular stereo headphones.

Roland, TOA, Yamaha, Tascam and other musical-instrument oriented manufacturers offer a variety of compact products.

This is particularly handy during setup and sound check. Using this method, you’ll have less loud, direct sound putting pressure on your eardrums, yet you will still hear the needed information.

Dummy headphones can be used as a quick way to lower the sound level of what you hear. Simply put on your regular stereo headphones, but don’t plug them into anything. Run the cord into your pocket. This will offer isolation from the louder acoustical environment that surrounds you during a show, while your ears have a chance to rest.

Rests away from the job site should be taken whenever possible. Remove yourself from the noisy environment and take time to have a meal, a nap, read a book, or whatever there is to be done in a quieter space. Focus on finding a ‘quiet zone’...no blaring TV or Walkman headphones. This can mean a walk outdoors, finding a secluded dressing room, or whatever.

The important thing when working around loud sound levels is to give your hearing system and ear mechanism time to recover. If you work in a loud environment, your hearing will be more sensitive and ‘fresh’ if you take regular breaks like this.

Sound Level Meters
If you do not already include a hand-held, battery powered SPL meter in your working toolkit, get one. Don’t rely on assumed level readings from your 1/3-octave real-time analyzer unless you are absolutely sure that the correct microphone is in use, (mic sensitivities can vary greatly, causing erroneous SPL readings), and that the system is properly calibrated. It’s better to have a small portable unit that you can keep in front of you on the mix console, or carry around the venue with you as you check coverage.

These handy devices can range in price from $65 (Radio Shack) to $2,500 (Bruel & Kjaer). I recommend the General Radio 1565-B Sound Level Meter (about $600); this is a hand-held battery powered meter that is approved by US Government agencies for environmental noise measurements. With its OSHA certification sticker, it helps you stand up to noise regulation officials, many of whom may have less sensitive and reliable gear.

Almost any type of SPL meter will do what you need; the accuracy difference between the cheapest and the most expensive can be about 1-2%...this would mean a possible error, plus or minus, of 1-2 dB at around 100 dB SPL. The more sophisticated, expensive units are best for critical situations.

Learn the difference between ‘A’ and ‘C’ weighting filter scales (US Government agency guidelines stipulate the use of C-weighted measurements for noise environments dominated by frequencies below 500 Hz; A-weighted measurements are most useful for making comparative readings in live show environments and discussing levels with others).

Use the sound level meter to get useful information in the front rows, the high balcony, the back of the hall, at the console…wherever you need to know the actual, average sound pressure level of your show.

Find the ideal ‘pocket’ where your show mix is as exciting and powerful as it needs to be, yet where you do not get audience complaints about excessive volume.

Use your meter as a daily reference guide, regardless of the type of acoustical environment.

Paying attention to the level of your system’s operation will be one more step toward protecting your own hearing, as well as that of others.

Long-Term Effects Of Loud Sound
We have probably all experienced TTS (Temporary Threshold Shift) after being exposed to very loud music or other sounds. This is the sensation that someone has stuffed cotton in your ears after you have already walked out of a loud environment; after one or two hours of high-level listening, your shifted hearing threshold may compensate as much as 40 or 50 dB.

In other words, your ears have ‘shut down’ to reject the extra-loud sounds that you have exposed them to. Recovery may take from a few hours to several days.

Prolonged exposure to very loud music can bring on tinnitus, which is a ringing sensation that you hear in your ears, even though no loud sounds are present around you. If you experience this ringing several days after exposure to a powerful sound system, consider that to be your own body’s way of giving you a danger signal. Heed the warning.

Have a regular hearing checkup. Get to know your audiologist or hearing specialist. Once or twice a year, get checked for both air and bone conducted sound sensitivity, speech understanding, and make sure that your inner ear parts are functioning properly.

If your job involves working with live sound, and you want to continue doing it, take time to carefully consider what your own personal approach is going to be as you work to conserve your hearing. You are also preserving your livelihood in the process.

David Scheirman is vice president, tour sound at JBL Professional and is also a long-time contributor of pro audio and sound reinforcement editorial.

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Posted by Keith Clark on 01/02 at 03:04 PM
Live SoundFeatureBlogStudy HallProductionAudioEducationEngineerLoudspeakerTechnicianPermalink

Thursday, December 19, 2013

In The Studio:The Trouble With Cheap Mics

On the whole, inexpensive audio gear sounds better than ever and is a much better value than even a decade ago, and yet...
This article is provided by Bobby Owsinski.

 
In many ways we’re in the golden age of audio gear. On the whole, inexpensive audio gear (under $500) sounds better than ever and is a much better value than even a decade ago and way better than 20 years ago.

The same can be said for mics, as there is a large variety of cheap mics that provide much higher performance for the price than we could have imagined back in the 70s and 80s.

That said, there are some pitfalls to be aware of before you buy. Here’s an excerpt from The Recording Engineer’s Handbook, 3rd Edition, that covers the potential downside of inexpensive mics.

One of the more interesting recent developments in microphones is the availability of some extremely inexpensive condenser and ribbon microphones in the sub-$500 category (in some cases even less than $100).

While you’ll never confuse these with a vintage U 47 or C 12, they do sometimes provide an astonishing level of performance at a price point that we could only dream about a few short years ago. That said, there are some things to be aware of before you make that purchase.

Quality Control’s The Thing
Mics in this category have the same thing in common; they’re either entirely made or all their parts are made in China, and to some degree, mostly in the same factory. Some are made to the specifications of the importer (and therefore cost more) and some are just plain off-the-shelf.

Regardless of how they’re made and to what spec, the biggest issue from that point is how much quality control (or QC, also sometimes known as quality assurance) is involved before the product finds its way into your studio.

Some mics are completely manufactured at the factory and receive a quick QC just to make sure they’re working and these are the least expensive mics available. Others receive another level of QC to get them within a rather wide quality tolerance level, so they cost a little more. Others are QC’d locally by the distributor with only the best ones offered for sale, and these cost still more.

Finally, some mics have only their parts manufactured in China, with final assembly and QC done locally, and of course, these have the highest price in the category.

You Can Never Be Sure Of The Sound
One of the byproducts of the rather loose tolerances due to the different levels of QC is the fact that the sound can vary greatly between mics of the same model and manufacturer.

The more QC (and high the resulting price), the less difference you’ll find, but you still might have to go through a number of them to find one with some magic. This doesn’t happen with the more traditional name brands that cost a lot more, but what you’re buying (besides better components in most cases) is a high assurance that your mic is going to sound as good as any other of the same model from that manufacturer.

In other words, the differences between mics are generally a lot smaller as the price rises.

The Weakness
There are two points that contribute to a mic sounding good or bad, and that’s the capsule and the electronics (this can be said of all mics, really). The tighter the tolerances and better QC on the capsule, the better the mic will sound and the closer each mic will sound to another of the same model.

The electronics is another point entirely in that a bad design can cause distortion at high SPL levels and limit the frequency response, or simply change the sound enough to make it less than desirable. The component tolerances these days are a lot closer than in the past, so that doesn’t enter into the equation as much when it comes to having a bearing on the sound.

In some cases, you can have what could be a inexpensive great mic that’s limited by poorly designed electronics. You can find articles all over the web on how to modify many of these mics, some that make more of a difference to the final sound than others.

If you choose to try doing a mod on a mic yourself, be sure that your soldering chops are really good since there’s generally so little space that a small mistake can render your mic useless.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. Get the 3rd edition of The Recording Engineer’s Handbook here.

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Posted by Keith Clark on 12/19 at 06:01 PM
RecordingFeatureBlogOpinionStudy HallBusinessEngineerMicrophoneSignalTechnicianPermalink

Church Sound: Shrinking Buildings & What That Means For Worship Tech

We’re seeing a significant shift from big worship to an emphasis on groups
This article is provided by ChurchTechArts.

 
I’ve been saying this for a while, but there is a change a comin’ in the church. While some disagree with me, I suggest that we are nearing the end of the era for big church buildings with big production going on.

Not that big buildings will disappear all together, but there will be fewer of them and they will not be the sought-after goal of most churches.

I’m hearing similar thoughts from other church leaders, and the latest to chime in is Thom Rainer. Last week he wrote a post listing seven reasons church worship centers will get smaller.

Unlike the last time I referenced a post in an article, I’m pretty much in agreement with him on this one. I really believe this is coming, and it’s going to affect what we do as technical leaders. Let’s consider some of his points.

Multi-Site & Multi-Venue Churches Are On The Rise

We see this everywhere. More and more churches are discovering that they can have a significantly more powerful impact on their community by launching multiple, smaller campuses instead of one big one. Or perhaps they will do multiple venues with different worship styles. Either way, this tend is here to stay (until the next trend, anyway).

What does this mean for us? On the plus side, all of these venues will need at least a basic production technology package. Often, it will need to be portable. So we’ll have a lot of gear to manage.

However, with smaller campuses, come smaller congregations (that’s kind of the point, right?), and smaller budgets. Not many churches will be hiring full-time guys to run campuses. I suspect what we’ll see is churches hiring one or maybe two technology directors who will oversee all the campuses, helping recruit, train and keep volunteers going.

In this scenario, technical leaders won’t be nearly as hands-on; we won’t be able to be in five places at once. But we will need to be really good at putting together packages of gear that can survive being loaded in and out by volunteers each week. If you have holes in your technical systems knowledge, now is the time to fill them in.

We’re Seeing A Significant Shift From Big Worship To An Emphasis On Groups

Thom points out—correctly in my opinion—that churches are starting to move away from the worship service being the central event of the church. It’s not going to go away, but there will be more emphasis on groups. Churches will be needing to raise up more leaders who can lead groups.

What does this mean for us? We’re already seeing it. Churches are becoming less interested in hands-on techs and more interested in technical leaders who can train and develop others to do the work.

Again, this won’t be binary. There will likely always be churches with large tech staffs who do the work. But I suspect we’ll see a shift towards volunteer teams, even in larger churches. If you’re a hard-core tech with no people skills, this is going to be a challenging transition for you.

But if you’re a builder of people and teams, you will do well. Now is the time to start honing those leadership and discipleship skills; you’ll be needing them!

We Will Be Spending Less On Buildings, More On Ministry

Again, more and more churches are foregoing a large, expensive worship center (or sanctuary, auditorium or whatever you want to call it), so they have more funds to invest in community ministry programs. I’ve always been conflicted with how much production technology costs.

On the one hand, I believe if we’re going to commit to doing production, we should do it well, and that takes money. On the other hand, I wonder sometimes if our priorities are misplaced. I’m not settled on this, and I suspect we’ll always live in tension in this regard.

What does this mean for us? Budgets will continue to shrink. We’ll have to find ways to do more with less. We will need to get very creative in how we do production. It may be that we do less production, but do what we do very well.

Hard choices will need to be made, and this will be a problem for some. If you refuse to work on anything but a Grand MA2 or a DiGiCo SD7, this may be hard for you. But if you’re open to scaling back and still doing production with excellence, this is going to be a lot of fun.

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.

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Posted by Keith Clark on 12/19 at 05:35 PM
Church SoundFeatureBlogOpinionTrainingBusinessEngineerSound ReinforcementSystemTechnicianPermalink

Wednesday, December 18, 2013

Phase & Polarity: Causes And Effects, Differences, Consequences

The terms "polarity" and "phase" are often used as if they mean the same thing. They do not.

Polarity and Phase - these terms are often used as if they mean the same thing. They are not.

POLARITY: In electricity this is a simple reversal of the plus and minus voltage. It doesn’t matter whether it is DC or AC voltage. For DC, Turn a battery around in a flashlight and you have inverted or, more commonly stated, reversed the polarity of the voltage going to the light bulb. For AC, interchange the two wires at the input terminals of a loudspeaker and you have reversed the polarity of the signal coming from that loudspeaker.

PHASE: In electricity this refers only to AC signals and there MUST be two signals. The signals MUST be of the same frequency and phase refers to their relationship in time. If both signals arrive at the same point at the same time they are in phase. If they arrive at different times they are out of phase. The only question is how much are they out of phase, or stated another way, what is the phase shift between them?

The important point to note in these definitions is that you can reverse the polarity of one signal and you can measure this change. You need two signals to measure a phase shift.

For convenience, the word “speaker” will be used in place of the more correct term “loudspeaker” in the rest of this article.

A picture is worth 1,000 words… but a few words of explanation can help.

The following figures show the differences and some consequences of polarity and phase. Figures 1 through 12 show graphs of sine wave signals. Actually it is a sine wave from one signal source split two ways. Except for figure 1, one of the splits is “processed” by reversing its polarity and/or by delaying it (phase shifting it) as described. To put this in the real world, imagine two speaker systems side-by-side, each reproducing one of the signal splits. (More precisely, the graphs show what you would see on an oscilloscope looking at the output of a mixing console with each split going to a separate input after one of the splits has been “processed”.)

The vertical scale in the graphs is in arbitrary units of -2 to +2 with lines at each 0.5 interval. If you like, consider this as -2 to +2 volts. Because phase shifts are measured in degrees, the horizontal scale in the graphs is labeled in degrees with a vertical line at each 90-degree point. One full cycle or period of a sine wave is 360 degrees.

Assume that the signals shown are 1 kHz sine waves, in which case each vertical line represents 1/4 millisecond of time. Sound travels in air about 3.4 inches (85 mm) in 1/4 millisecond so each vertical line also represents this distance. Note that in the graphs the signals all start 1/4 millisecond or more from the left so you can clearly see when each signal starts. (The importance of this will be seen in figure 9.) There is no signal along the flat line from -90 to 0 degrees.

Signals In Polarity, In Phase
Figure 1: This shows 3 periods or 3 cycles of two simple sine waves. Both are +/-1 volt high at their peaks = total of 2 volts. One is shown in blue the other in red.

Figure 1: Sine Waves in Fig. 1 Added.

Figure 2: This is what happens when the two are combined (= added together). This is exactly what would happen on a line exactly between the two side-by-side speakers. The two signal beings being in phase and in polarity add up so the peaks are now at the +/- 2 volt lines = 4 volts or twice the original signals. Acoustically this is an increase of 6 dB = 20 x log(1+1).

Figure 2: Two Sine Waves - Same Polarity & Phase.

Signals Out of Polarity
Figure 3: This is like figure 1 but the second sine wave, shown in red, has been reversed in polarity. As you can see the + and - voltage points are exactly opposite from the first sine wave, shown in blue. This would be accomplished by reversing the +/- input connection on the speaker reproducing the red sine wave.

Figure 3: Two Sine Waves - Red = Polarity Reversed.

Figure 4: This is what happens when the two are combined. Each point of the two signals being in phase, but opposite polarity, adds up to zero. Acoustically this is an infinite decrease of output. Because you can’t take the log of 0 assume the difference is actually 0.0.01 volts (the dots = 58 more zeros). 20 x log of this number is -1200 dB. That should be pretty quiet. You can’t easily hear this with two speakers because of having two ears. But using a very carefully positioned microphone to measure this in a place with no sound reflections, you would find almost no signal.

Figure 4: Sine Waves in Fig. 3 Added.

Signals Ot of Phase

Figure 5: The second sine wave, shown in red, starts 1/4 millisecond later (90 degrees later) than the first one, shown in blue. Put another way, the second signal has been delayed by 1/4 millisecond.

Figure 5: Two Sine Waves - Red = Phase Shifted 90 Degrees.

Figure 6: This is what happens when the two are combined and it’s pretty interesting. First notice that the peaks are almost at the +/-1.5 volt lines. The value is actually +/-1.414 volts. This is a 3 dB increase. This would be like listening to two speakers but the one reproducing the red sine wave is 3.4 inches (85 mm) further away from you than the other. The first thing you hear is only from the speaker reproducing the blue sine wave. The black line starts when the sound from the second speaker is heard and this line is the combined signal of both speakers.

Figure 6: Sine Waves in Fig 5 Added

Suppose the speaker reproducing the red signal were only 2.25 inches (57 mm) further away. The signals would be shifted by only 60 degrees. The increase for the combined signal would be about 4.5 dB. So the amount of phase shift is important.

The second thing to notice is what happens at 1/4 millisecond or 90 degrees after the blue signal starts when the second signal “kicks” into the picture represented by the line turning black. There is a distinct change in the waveform.

The third thing to notice is that the entire waveform after the “glitch” is shifted in time compared to figure 7 about 45 degrees = average of 0 and 90 degrees.

Signals Out Of Phase And Polarity

Figure 7: The second sine wave, shown in red, is a combination of the sine wave in figures 3 and 5. The signal not only has its polarity reversed but it is shifted in phase by 90 degrees compared to the first signal, shown in blue. In this case the speaker reproducing the red sine wave has its +/- input connection reversed in polarity and is 3.4 inches (85 mm) further away from you than the one reproducing the blue sine wave.

Figure 7: Two Sine Waves - Red = Phase Shifted 90 Degrees & Polarity Reversed.

Figure 8: This is what happens when the two signals are combined. The picture is similar to figure 6 with two important differences. First the “glitch” at the point where the second signal starts is different. This is the point where the line turns black. Second is that the entire waveform is shifted by 45 degrees again but this time to the left of the original signal.

Figure 8: Sine Waves in Fig. 7 Added.

The “Glitches”
The glitches in figures 6 and 8 give an indication of what happens during the onset of a signal. While the so-called steady state portion of the combined signal (shown by the black portion of the lines) looks the same except for the amplitude change, these glitches will affect the transient attack of sounds. This is not to say that either will sound horrible, but a phase shift between otherwise identical replicas of a sound WILL make a difference in the sound of the initial transient attacks, depending on the frequency and amount of phase shift.

This is exactly the kind of phenomena that can occur in the crossover region of a speaker. This is because the distance from each driver to the listener is usually different and the crossover itself shifts the phase of the signal between the drivers. Speaker designers are often faced with a choice between something like what you see in figures 6 and 8. Neither is “correct” so a designer can only choose the one that “listens” better. Just looking at these two, I would bet the waveform in figure 8 might sound better and the choice would be to reverse the polarity of one of the drivers. These crossover “glitches” occur only over a small range of frequencies where both drivers reproduce the sound. It is well accepted by designers that this kind of “improvement” is sonically more significant than the fact that frequencies above and below the crossover point may be out of polarity.

Signal Phase Shifted 180 Degrees
This is where many get into trouble in thinking that phase and polarity are the same thing, meaning that it is often assumed that a 180 degree phase shift and reversing the polarity are the same.

Figure 9: In this figure each sine wave lasts for only 2-1/2 cycles. The second sine wave, shown in red, is shifted in phase 180 degrees from the first shown in blue. This is what would happen if the speaker reproducing the red sine wave were about 6.8 inches (170 mm) further away from you than the one reproducing the blue sine wave. You can see that between the 180 and 900 degrees the signals LOOK like they are simply out of polarity but they are NOT. It is VERY important to note that if you could not see the beginning or the end of these signals you could not tell whether they were out of polarity or 180 degrees out of phase. Too often this is what causes confusion between a polarity reverse and a 180 degree phase shift.

Figure 9: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 10: This is the result of combing the two signals. Unlike figure 4 where the signals are simply out of polarity, and completely cancel, there are clearly two positive halves of a sine wave visible before and after the two signals cancel along the black line between 180 and 900 degrees. The first is from the blue sine wave in figure 9 that occurs before the start of the red sine wave. The second is from the red sine wave in figure 9 that continues after the blue sine wave stopped.

Figure 10: Sine Waves in Fig. 9 Added.

Signal Phase Shifted 180 Degrees And Reversed In Polarity

Figure 11: This is the same as figure 9 but the polarity of the red signal is reversed from figure 9.

Figure 11: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 12: This is the two signals in figure 11 combined. Between the 180 and 900 degrees, the signals add much like in figure 2. However there are significant differences in the overall 90 to 1080 degree signal. The first 1/2 sine wave of this signal is only from the blue sine wave from figure 11. The last 1/2 sine wave is only from the red sine wave in figure 11. You can clearly see that both of these 1/2 sine waves are only 1 volt at the peaks. This is a clear difference from figure 2 where all the peaks reach 2 volts.

Figure 12: Sine Waves in Fig. 11 Added.

The reason is that the two signals in figure 11, even though identical, are offset by 180 degrees. They add together only between 180 and 900 degrees when both are being heard. More importantly, during this time period DIFFERENT parts of the same signal have added together. For example you can see that between 180 and 360 degrees it is the second 1/2 of the blue signal’s first complete sine wave that adds to the first 1/2 of the red signal’s first complete sine wave.

Real Audio Signals
Sine waves are easy to look at to dramatically show the difference between polarity and phase. Armed with this knowledge you can look at figures 13 through 18 that show something like a real audio signal where the effects of polarity and phase are more difficult to see.

The signal shown in these figures was a generated by a mathematical algorithm that produces something close to a pink noise signal. Pink noise contains all frequencies with an equal amount of energy in each octave band. Real audio signals don’t look much different than pink noise (but one would hope they sound better!). The scales on these graphs are arbitrary. You can look at the vertical scales as +/-3 volts if you like. However, because of the way the signal was generated, there was no way to define absolute time or degrees along the horizontal scales. Suffice it to say that the phase-shifted signal used in these figures was shifted by one data point out of the 240 data points that make up the signal lines.

There is one important thing to understand about phase shift. The amount of time one signal is delayed from another will have different effects at different frequencies. Assume there is a 1 millisecond time difference between two identical signals. At 500 Hz the result will be as shown in figure 10 because at 500 Hz the 1 millisecond time difference is a phase shift of 180 degrees. The signals are offset by 1/2 a cycle.

At 1 kHz the signals will be offset by 1 complete cycle. In other words you would hear one cycle from the first signal then both combine then you’d hear the one cycle from the second signal after the first stopped. This is similar to what is shown in figure 12 (which shows only 1/2 cycle) but is not the result of the same conditions that were used to make figure 12.

At 250 Hz the effect would be as shown in figure 6 because a 1 millisecond time difference corresponds to a 90 degree phase shift at 250 Hz or an offset of 1/4 cycle. At lower frequencies the phase shift would be even less and the signals would tend to add as in figure 2, approaching but never quite reaching the 6 dB increase shown in that figure.

Contrary to phase, polarity affects all frequencies the same way. It makes the positive portions negative and the negative portions positive. Put another way, it simply flips the signal over the same way at all frequencies. With these things in mind, examine figures 12 through 18

Effects of Polarity and Phase On “Real” Audio Signals
Figure 13: This shows a pink noise signal generated as noted above.

Figure 13.

Figure 14: This shows both the original signal in blue and what happens when an identical but phase shifted signal is added to it, as shown in red. The red signal is similar to the combined signal shown in figure 6. Note the increases in signal level and the changes in the waveform (many glitches). However you can also see the combined signal follows the original fairly closely.

Figure 14.

Figure 15: This shows both the original signal in blue and what happens when the phase shifted signal is also reversed in polarity and combined with it, as shown in red. In this case there are huge differences between the original and combined signal.

Figure 15.

Figure 16: To better understand what is going on, this figure shows an averaged or integrated version of the pink noise signal in figure 13. This is basically what would you would see if you graphed the readings from a typical SPL meter for the signal in figure 13.

Figure 16.

Figure 17: This shows the averaged signal from figure 16, in blue, and the averaged combined signal from figure 14, in red. Note that there are primarily level differences (mostly increases). Otherwise the two lines look very similar.

Figure 17.

Figure 18: This really shows what is going on in figure 15. The blue line is the averaged signal from figure 16. The red line is the averaged signal from figure 15. The red line shows that the out of polarity and phase-shifted signal approaches a straight line. Because you are looking at a broad frequency range, you are seeing a severe cancellation of the lower frequencies due to the polarity reversal. However, unlike the low frequencies, the upper frequencies do not completely cancel due to the phase shift. The red line contains primarily high frequency energy. In the blue signal the higher frequencies are the small “bumps”. These can be clearly seen in the red signal and most of them correspond to those in the blue signal.

Figure 18.

Figure 18 is a prime example of what you would hear if you stand exactly between two speakers playing the same signal (i.e. mono) with one speaker out of polarity. The bass will disappear. But, there will always be a difference in distance between you and the speakers due to the spacing of your two ears and probably a slight overall difference in distance between you and each speaker. A difference in distance means a difference in the time arrival and thus there will be phase shifts between the sound from the two speakers. The amount of shift will vary with frequency. Because of the shorter wavelengths at high frequencies, the phase shifts allow most of the highs to be heard. They may be out of polarity but the effect is like what is shown in figure 8. Also, in a room you would also hear sound reflections from the floor, walls, and ceiling. You would only hear something like the red line in figure 18 outdoors away from any reflective surfaces or in an anechoic chamber.

Figure 19.

The small distance between your ears and any small difference in distance from you to each speaker do not cause appreciable phase shifts at low frequencies. This is because of the considerably larger wavelengths. The difference in your distance from each speaker might be only 1 inch (25 mm). However, the wavelength of even a 1 kHz sound is roughly 1 foot (300 mm) and at 100 Hz roughly 10 feet (3 m). At the lower frequencies the polarity difference predominates because the phase shifts due to the difference in your distance from the speakers is very small compared to the wavelengths of the low frequencies. Thus the lower frequency signals, being nearly in phase but out of polarity, will cancel like in figure 4. The lower the frequency the less the phase shift between the two speakers and the greater the cancellation.

A Polarity / Phase Field Trip!!
(As with all physical exercise, check with your doctor first, who might not recommend you do this for some reason.)

Find two railroad tracks, lie across them, and wait.

Two trains, one on each track, come along. Both are right side up and both hit you at exactly the same time. The trains are in polarity and in phase.

The same thing happens again and both trains hit you at exactly the same time. However, this time one train is upside down. That is a polarity reversal.

The third time both trains are right side up but one hits you first and the other hits you shortly after the first. That is a phase shift.

The last time the second train is upside down and hits you later than the first. That is both a polarity reversal and a phase shift.

Summary
So there you have it. Although this has only touched on a few areas concerning phase and polarity issues, it is hoped you better understand the difference between the two and a few of the effects of each. Remember that the audio frequency range covers wavelengths of over 30 feet (10 meters) at the lowest frequencies to less than an inch (under 25 mm) at the highest frequencies.

While a reversal of polarity will affect all frequencies identically, a difference in time arrival between two otherwise identical signals will have very different effects on the phase between them. The amount of phase shift will be different at different frequencies and this will depend on how much time difference there is between the arrival of the two signals.

{extended}
Posted by Keith Clark on 12/18 at 03:35 PM
AVFeatureBlogStudy HallAVEducationInstallationMeasurementProcessorSignalTechnicianPermalink

IK Multimedia releases UltraTuner For iPhone, iPad And iPod Touch

UltraTuner utilizes a new patented detection engine making it the most accurate instrument tuner available for iOS - 10 times more accurate than a mechanical strobe tuner

IK Multimedia is pleased to announce UltraTuner, the most precise digital chromatic instrument tuner app for iPhone, iPod touch and iPad.

UltraTuner is available as standalone app for iPhone and iPod touch or as an in-app purchase within AmpliTube for iPhone and iPad (in all versions).

UltraTuner features one of the fastest, smoothest and most responsive tuning engines of all tuner apps, and is precise to well below a hundredth of a cent - far beyond human perception of pitch, making it 10 times more accurate than a mechanical strobe tuner (considered the gold standard of tuners).

UltraTuner’s accurate pitch detection is achieved through a proprietary dual-analysis processing engine that allows the tuner to get a much more accurate reading of the frequency from the incoming note. The fast response of the tuner makes tuning a mechanical instrument a breeze. UltraTuner’s precision also makes it the perfect app tuner for intonating string instruments like guitar or bass, and by offering note precision down to less than .01 (1/100th) of a cent, making it ideal for calibrating electronic instruments like vintage synths.

UltraTuner’s precision and speed provides players and engineers with a pocket-sized professional instrument tuner always at hand and offers two basic modes of operation: Stage and Studio.

When in Stage mode, UltraTuner is a straightforward, ultra-accurate instrument tuner featuring a unique, simple interface that’s very easy to see in low light conditions. It features a graduated flat and sharp display to indicate degrees from pitch, and when the note is in tune, the display turns green. UltraTuner utilizes the entire device screen for tuning so it is easily visible from a distance on dimly lit stages.

In studio mode, UltraTuner provides pitch tracking - allowing players to monitor pitch over time. This is especially useful in the studio for monitoring pitch of non-chromatic instruments like vocals, violin and brass etc., on recordings over time, and very useful for vocal and instrument pitch training. Studio mode also offers two different visualizations: pitch over time and a proprietary oscilloscope wave form display.

The pitch over time display shows the accuracy of a note as it is played and decays. The oscilloscope wave form display indicates flat or sharp notes by animating the waveform to the left if the note is flat, completely still if the note is right on, and animating the waveform to the right if the note is sharp. These visualizations make it very quick and easy to reference notes while playing a non-chromatic instrument or singing in the studio.

In the settings window, players can quickly calibrate the “A” note, set the temperament type, set the root key for tracking and adjust the audio sensitivity. UltraTuner can use the device’s built-in microphone for audio input, and works with all analog or digital audio interfaces for iPhone, iPad and iPod touch, like IK’s iRig HD, iRig PRO and the iRig MIC series of products.

Key Features:

  Innovative patented tuning engine that provides the highest level of responsiveness and accuracy with 0.01 cent resolution
  Dual Mode: Stage and Studio
  Large Tuning indication visible from a distance (stage mode)
  Tuning History to see the tuning variations over time (studio mode)
  Temperament selection includes: Equal, Pythagorean, Just Major, 1/4 Comma Mean Tone, Kimberger III, Werckmeister III, Young and Kellner
  Useful not only for tuning, but also for non-chromatic instrument training

UltraTuner is available now as a standalone app from the App StoreSM for iPhone and iPod touch for $4.99/€4.49, or as an in-app purchase within AmpliTube for iPhone (Free/LE/Full versions), and AmpliTube for iPad (Free/Full versions).

IK Multimedia

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Posted by Julie Clark on 12/18 at 10:30 AM
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