Mixer

Thursday, January 12, 2012

New iConnectMUSE Palm-Sized Digital Audio Mixer For iOS Devices

iConnectivity has introduced iConnectMUSE, a palm-sized personal mini-mixer that combines several music accessories into one iOS controlled product

iConnectMUSE offers quality audio processing, improved audio latency and the ability to share and mix audio through an iOS device, and even charge the iOS device at the same time.

Featuring six stereo inputs and six stereo outputs, as well as two USB device ports, hub-able USB Host Port, two input and two output 1/4-inch audio port jacks, a headphone monitor jack, Ethernet connection for network sharing, a MIDI pass-thru interface and limited DSP processing.

“iConnecMUSE is the future of audio mixing,” said Michael Loh, CEO of iConnectivity. “Its portability, expandability and enhanced functionality through software integration provides the conduit for music creativity for all musicians.”

iConnectMUSE, available Q2 of 2012, will retail for $229.99 (MSRP). It will be on display at the upcoming NAMM 2012 Show in Anaheim at booth 1671.

iConnectivity

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Posted by Keith Clark on 01/12 at 03:14 PM
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In The Studio: Mixing In Mono

Listening to mixes in mono can be very helpful
This article is provided by Home Studio Corner.

 
Lately I’ve been listening to mixes from members over at MixWithUs.com.

A number of times I’ve suggested to people that they check their mixes in mono.

To clarify, everybody mixes in stereo. Stereo simply means the mix has two channels (left and right).

A mono mix is simply one channel. You combine (or sum) the left and right channels into a single channel.

Listening to mixes in mono can be very helpful. I’ll explain why.

Phase
Whenever you combine multiple signals together (especially similar signals), you run the risk of having phase issues. (See What is Phase?)

When using multiple microphones, whether on an acoustic guitar or a full drum kit, the more mics you use, the more careful you have to be.

As more mics are picking up the same signal, those signals, when combined, can cause cancellations at certain frequencies (if they aren’t perfectly in phase with each other).

When setting up microphones, it can be difficult to hear phase issues if you’re listening in stereo. Listening in mono lets you hear them more easily.

The same applies to mixing. You may have a really cool guitar sound panned to the left, and another great guitar sound panned to the right.

In stereo it sounds amazing, but in mono it suddenly becomes thin and hollow. It’s not necessarily a bad thing, but I’ve found that mixes that sound good in mono sound GREAT in stereo.

Bass Issues
This is especially true when dealing with the ever-problematic low end in your mix. You think you’ve done the perfect amount of EQ. The bottom end is full, but not muddy.

Then you check the mix in mono and BAM…it’s all muddy again.

Why? Because the tracks you have panned left and right in your mix all have little bits of low end that you don’t hear as clearly when spread out in stereo.

But when you “fold back” everything to mono, these little bits of low end add up to a big boomy sound.

While listening in mono, make the necessary EQ changes until everything sounds nice and balanced…THEN switch back to stereo. Wow! It’s pretty astonishing how good it sounds.

You didn’t know there were low frequency issues before. Listening in mono pointed it out, then when you fixed it and switched back to stereo, everything sounded cleaner and more professional.

How To Do It
There are lots of ways to listen in mono:

—Pan your master fader L and R sliders to the center.
—Use a plug-in like TT Dynamic Range Meter on your master fader. It has a big “Mono” button.
—Use a monitor management box, like the PreSonus Monitor Station. These have mono buttons on them as well.

Some people would tell me that mixing in mono serves no purpose, since people will always be listening to your mixes in stereo. Good point, but I view mixing in mono as one extra way to discover problems in my mixes.

Sometimes I’ll mix an entire song in mono (on accident), then switch it back to stereo right at the end…and WOW…it sounds amazing. Give it a shot sometime.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

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Posted by Keith Clark on 01/12 at 09:25 AM
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RE/P Files: Construction Of A Live Echo Chamber

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine comes a wealth of knowledge about echo chambers which first appeared in the July / August 1979 issue.

A live echo chamber can be a considerable asset for any recording studio, that is providing that it is a good one.

That’s the problem — how do you construct a good echo chamber? When someone builds a chamber, they hope it will turn out great and pray it won’t turn out absolutely dreadful and good for nothing but storing echo plates.

The truth is there are a number of complex variables which will make each chamber unique.

These factors which effect the chamber include the type of wall construction and the selection of materials used on the inside surface.

Probably the most important consideration is the cubic volume and physical proportions of the chamber.

This leads to the first question to be asked before a chamber can be built. What space is available?

Space
Most times echo chambers occupy surplus space. The more space that is available to start with, the easier the construction job since the builder won’t have to deal with odd angles or cramped building conditions.

The size of the chamber can generally vary from 1,000 cubic feet to about 2,000 cubic feet of internal volume. 1,500 cubic feet seems to get excellent results within a workable space.

A small chamber won’t get optimum results and the largest chambers are a luxury since they occupy a space large enough to be usable for other purposes.

When determining if enough area is available, it is necessary to remember to allow for figuring the wall thicknesses.

Figure 1. Click to enlarge.

There is also a need to consider a wide enough passageway around the structure for a hammer to be swung. If this isn’t planned for putting on the exterior sheeting is going to be a difficult proposition.

It is also suggested that the space be large enough to accommodate a chamber with a minimum interior dimension of 7 feet. A chamber with a side shorter than this will usually give unsatisfactory results.

The Floor Plan
Once it has been decided that there is enough space, the next step is to design a floor plan (Figure 1). When laying out the sides of the room none of the walls should be parallel or even near to parallel. The ceiling should not be parallel to the floor. This is very important if the room is to have maximum random reflections and a smooth decay. It’s at this stage that a bit of math should be introduced (1).

T = Const X M/α S

where:
T = Reverberation time
V = Total volume of the room in cubic units
S = The total combined surface area of all sides in square units
α = The average absorption coefficient Constant:
.049 if measurements are in feet.
.161 if measurements are metric.

The American National Standards Institute (in S-l. 1-1960) defines reverberation time of a chamber as the time it takes for the mean square sound pressure level to decrease to 60 dB after a steady state signal has ceased (2). Generally this level is referenced to 500 Hz although some information relates the level to 1 kHz.

The following equation for figuring decay time was developed by Wallace Clement Sabin (1900). Since his time there have been a number of alternate equations developed but the original equation continues to be the most popular. This is due partly to the simplicity of the computation and the similarity of the resulting data.

Figure 2. Click to enlarge.

Sabin determined that the reverberation time was related to the volume of a room, its surface area and the total amount of absorption.

As the formula would indicate, the time is directly proportional to the volume and inversely proportional to the surface area of the room. With this being the case, the longest echo will be obtained when the required volume is achieved with a minimum of interior surface area.

Trying to achieve more diffusion in the chamber by building accordion type splays will have the effect of cutting down the delay time because the total surface area will have been increased.

Another point to be remembered when figuring out the length of the sides is that none of the dimensions between any two opposing walls should be the same or a multiple or fraction of any other two opposing walls.

There are a number of ratios which serve as guides when figuring acceptable proportions, including the Golden Section; (5^1/2 + 1): 2 :( 5^1/2 - 1).

This relationship was proposed by the Greeks and divides a line in such a manner so that the smaller dimension is to the greater as the greater is to the whole. There are four or five other ratios which have been proposed and accepted to varying degrees, (3) but the one most often used is Sabin’s 2:3:5 relationship. (Figure 2) (3)

Wall Angles
The angles used for the intersecting corners should not be severely acute. In practice, the simplest way to arrive at the wall angles is to build the sides so that two of the joining sides meet at right angles.

Their opposing walls are constructed similarly, but without permanently nailing down the floor plate. Once these walls have been framed, the entire unit can be angled inward. For the average chamber moving one end of each of these two walls in by a foot or so should be sufficient.

Rigidity And Isolation
Two things which are important in making a good live echo chamber are maximum interior wall rigidity, and the total isolation of the chamber from its surroundings.

The more rigid the wall is, the less energy dissipated when a sound wave hits it, hence the surface is more reflective. The isolation is important since it’s essential that the rooms broadband ambience be very low.

Figure 3. Click to enlarge.

Echo chamber wall systems which achieve these goals can be constructed from a variety of materials. There are a few chambers that have been built with walls, floors and ceilings of poured concrete. Such a design if properly executed will get very good results, but will be expensive to build, very permanent and extremely heavy. It will be there forever, providing you haven’t underestimated the strength of your sub- floor.

The second most popular approach is concrete block walls. They are easier to build, and a bit easier to tear down, but once again you have the weight problem.

The most popular and cheapest type of construction is a wood frame design made of 2” x 4” and 2” x 6"s. It is the easiest to build with hand tools and more importantly is relatively light compared to concrete and block.

Walls
In recording studio construction it is very popular to use 2” x 4” staggered stud construction since the results achieve a better transmission loss than a conventional wall. (Figure 3) A staggered wall uses two rows of studs 8” apart (2” x 4” alternating 16” centers) on 6” top and bottom plates. Every other stud is flush to the opposite edge of the plates. In this way the wall sheeting of the two sides is only connected at the top and bottom plate.

A standard stud wall uses a 2 x 4 plate. The sheet rock covering both sides is connected through each of the common studs. The transmission loss between these two types of construction with insulation is STC 36 dB for the standard wall and STC 49 dB for a staggered wall.4

The staggered wall would be the preferable wall design if transmission loss were the only consideration, however, rigidity is a more important factor in the design of the inside shell, hence standard construction is preferred. (Figure 4)

Figure 4. Click to enlarge.

The reason for this is that staggered construction does not allow enough space for cross bracing. The interior shell should have a cross brace splitting the span of every stud, joist, and beam. There should be no unbraced span longer than eight feet.

The preferred layout is to use a staggered exterior wall, and a standard interior one. These two walls should not be coupled in any way and hopefully will not only be decoupled from each other but from the rest of the building.

Floating Walls and Floors
A very important part of the de-coupling of an echo chamber is isolating it from the building it sits in. What sort of isolation is needed will depend on the specific situation and the funds available.

The drawings shown use Celotex for de-coupling. As can be seen, the outer shell is de-coupled from the floor of the building, while the echo chamber has a completely floated floor.

If the location requires extreme de-coupling then it might be necessary to construct the outer shell on its own completely floated floor. It is also likely that something other than than Celotex will be needed.

Machine rubber is a good alternative to Celotex. It seems to work acceptably when used in a quiet environment but it does seem to compress a great deal and might break down with time. A thickness of or more is necessary.

In an extreme isolation situation involving low frequency vibration more severe measures will be necessary. The worst case may need a floated concrete floor on spring isolators, but once again you have considerable weight and expense.

The best material which has been found for most situations are Fiberglas decoupling blocks. As shown, they are two inch cubes, covered with latex (to keep the moisture out) and are specially designed for floating floors. They can be used for both concrete and timber designs.

When they are used with concrete, a sheet of plywood is laid over them and a border runs around the plywood to create a pouring form. Be sure all cracks in the form are sealed so that none of the concrete will seep and re-couple the floating floor to the structural one.

When the blocks are used with wood construction, they should be set under three or four 2” x 6” headers. The timbers form the base for the echo chambers floor joists. Isolators should be placed about a foot or so apart along the entire length of the headers.

The only problem with the block spacers is the gap it leaves between the floating floor and the structural floor. The solution is to fill the space with Fiberglas and run the sheetrock down to %” away from the floor caulk.

These blocks are available from a number of suppliers. The blocks can also be used to isolate ceilings from walls. For this application the blocks seem to work best if they are split in two.

A chamber built on a concrete ground floor can be further de-coupled by using a concrete saw to cut a slot around the perimeter of the chamber. This helps quite a lot in isolating the rest of the building from the chamber. The newly formed slit can be stuffed with Fiberglas but a hard sealer should be avoided since it will slowly harden and compress and re-couple the slabs.

Insulation, Sheetrocking and Sealing
All the walls should be liberally stuffed with foil backed Fiberglas insulation (3%” #R-11). Note: when working with the glass, be sure to always wear gloves, goggles, a mask, and clothes that won’t allow the glass to touch your skin.

Both sides of each wall are covered with sheet rock. Two layers of y2” sheet rock is specified for the interior wall but two layers of 5/8” will work just as well or better. When two layers are used one should overlap the other so that none of the seams coincide.

All seams should be taped and sealed so the room will be airtight. In addition to standard seam Hydroseal which is a black gummy adhesive is highly recommended.

It is normally used as a roofing sealer and will stick to anything. It should be used liberally at every structural intersection of the sheeting or the stud construction. You end up using gallons of this stuff and practically glue the building together.

The purpose of the Hydroseal is to close every crack or seam. The smallest crack should not be ignored. If you have a 1/32” crack along the floor, the actual total area of that hole is considerable.

Figure 5. Click to enlarge.

A final note on Hydroseal. It is suggested that trowels are unnecessary for the application. Wood shims made from construction debris works a lot better. These are good for only about one or two applications. It is almost impossible to clean Hydroseal off anything, including the applicator.

The Chamber Surface and Its Application
After the walls have been built and the last inside layer of sheet rock has been taped and sealed, the chamber is ready to have its reflective surface treatment applied. The reflective walls of the echo chamber can be made from a number of different materials.

The key to how good a particular material is for this application depends on how rigid it becomes after installation and what its absorption coefficient is. Referring back to the earlier formula, the length of the echo is inversely proportional to this coefficient.

The absorption coefficient of any material is defined as “the ratio of sound energy absorbed by a given material to that which arrives at the surface from the source.” (5)

Hence, a porous surface will have a much higher number than a reflective one. This figure will not only be different for each material but vary widely depending on the frequency. Figure 5 shows an absortion coefficient chart of materials versus frequency. As can be seen, plaster has a remarkably low number and is relatively flat. The plaster which Scott uses is Keen Cement (absorption coefficient of .015). He says, “when it is properly applied, it is as smooth as a baby’s bottom.”

Before plastering the surface, the sheet rock has to be prepared so that it will hold the plaster. This is done by nailing on the actual buttonboard.

The actual plastering involves the application of two different layers. The bottom one is a brown coat layer and is put on as a preparation for the top coat of Keen cement. It is strongly suggested that the actual plastering be done by a very good professional.

The mixing of the two cements is nothing more than following the directions on their respective bags. One of the most important things to remember about the plastering process is the amount of time required for proper curing.

As often happens the chamber is complete after the final coat of plaster is applied and there is a desire to use it right away. If the chamber doors are closed prematurely, ventilation will stop as will the drying of the walls. How long it takes for a wall to properly dry changes with the temperature, ventilation, and humidity of the environment, and can vary from a few days to a few weeks.

The longer you can continue to ventilate a new chamber, the better. Scott suggests hooking up a dehumidifier in the room during the drying. He added, “you’ll be surprised how much water comes out of those walls.” This is a crucial part of getting the best end results.

“I have been in chambers that were built years ago that have walls that have never completely cured.” Needless to say such a room has a poor reverberation.

Some chambers sound bad only because the plaster coat was improperly applied. In such cases all that might be necessary is a re- plastering. It is also likely that an additional layer of buttonboard will be necessary so that the new plaster will have a firm base.

Connections And Doors
The chamber should have an IN and OUT opening so that the speaker and microphone lines can be kept separate. The pipe used should be flexible and have an I.D. of 3/4”.

Increase the diameter if there is a need for more than two or three lines per pipe. The smaller it is, however, the less possibility there is for any leakage. Plastic hose works well because it is flexible, has gentle curves and there are no ridges inside for wires to hang-up on.

Make the hole through which the hose is run as small as possible since you want to maintain the integrity of the wall system as much as possible. Once the hose has been installed, any gap between the hose and the wall should be completely caulked with Hydroseal.

A door is obviously needed somewhere in the chamber. Be sure the access to the door and the passageway leading to it is large enough to get a good sized speaker in and out of the chamber.

Figure 6. Click to enlarge.

The two doors used should be of solid construction, not hollow. They are mounted on completely separate frames and jambs to coincide with the de-coupled inner and outer walls. (Figure 6) Machine rubber should be used between the two frames where they almost touch (%” to between the walls.

Rubber should also be used completely around the jambs of the two doors. When they close the rubber should compress and make a tight seal. Since these doors are generally used infrequently, an elaborate closing is unnecessary.

Additional Construction Notes
A light should be installed somewhere inside the room. The switch for it can go anywhere convenient including right on the fixture. Be sure that de-coupling practices are maintained while running the AC conduit and mounting the fixture.

All the walls are built with studs on 16” centers. When laying out the studs, be sure to take into consideration that the drywall is 8’ or 12’ high and 4’ wide and the centers of every third stud needs to line up with the edges of the drywall. #16 nails are used for all end nailing and #8 for toe nailing.

Drywall nails are used on the first layer of drywall, but because of the added thickness #8 nails are used on the second layer. The buttonboard should be nailed on with #8 nails.

Be sure to estimate lumber lengths thoughtfully, to limit the amount of scrap.

However what is left should be used for blocking. The greater the waste, the more the chamber (and for that matter any type of construction) will cost.

Speaker(s) and Microphone(s)
Changing the speakers and microphones or altering their placement will change the sound heard in the control room. Deciding what type of speaker sounds best or what microphone should be used becomes a matter of taste.

This is equally true as to where they are placed in the room.

Providing there was adequate space to start with and care was taken with the design, construction, and isolation considerations, it is probable that the chamber will end up sounding very good.

With a little bit of luck, it might turn out to be the sort of chamber that gains a reputation for itself as well as the studio that has it.

References:
1 - Sound Recording Practices, J. Borwick; “The Acoustics,” by Alex Burd, Oxford Press, page 27.
2 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 25.
3 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 87.
4 - Handbook of Multi-Channel Recording, F. Alton Everest, Tab Publications, page 261.
5 - The Audio Encyclopedia, H. Tremaine, Sams Publications, page 44.

Scott Putnam has been extensively involved in studio construction, having designed and built a number of echo chambers including two for Kaye- Smith in Seattle, and a pair for the Record Plant. As a builder he has collaborated on various projects with a number of acoustic consultants and architects including Jack Edwards, George Augspurger, and Scott’s Father, Bill Putnam.

Downloadable Media
Original Article (pdf)

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Posted by admin on 01/12 at 08:22 AM
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Wednesday, January 11, 2012

Blondie Takes To The Road With Help From Big House Sound & DiGiCo SD8

Big House Sound of Texas supplies SD8 digital console for recent tour by Deborah Harry and mates.

Deborah Harry and her band Blondie are still going strong nearly 40 years since they got together. The lineup these days includes core members Chris Stein on guitar and drummer Clem Burke, along with Matt Katz-Bohen on keyboards, Lee Fox on bass, and Tommy Kessler on guitar.

The group took to the road in 2011 in support of Panic of Girls, their ninth studio album. At the tour’s audio controls was Rod Nielsen of Austin, TX-based Big House Sound on a DiGiCo SD8/DRack system.

Nielsen actually purchased two SD8s over three years ago to fill the bill for the Blondie and Blues Traveler tours, as well as to accommodate specific rental requests. With the release of the new Blondie record, Nielsen packed up the SD8 last spring for nearly five weeks of gigs throughout Europe and the UK.

The console’s diminutive size gave him the luxury of shipping it with the rest of the backline—and allowed him to keep the band’s sound consistent from venue to venue without being at the mercy of whatever rental gear was available at each gig. Size notwithstanding, Nielsen found so much more to love about the console.

“Functionality-wise, the SD8 is great,” he states, “but the biggest reason I like the console is it really does sound quite a bit better than other consoles in its price range. The high frequencies on the console don’t seem to fall apart as easily as other consoles. It seems to be very open; the high-end is there but it’s not shrill and the low end is tight and warm. The ‘verbs are very nice as is the compression.

“The fact that the console can be setup any way you want; the inputs can be anywhere at anytime, you can have them in multiple places at anytime, control groups and returns in any place you want them to as well. Setting up the console makes a lot of sense to me. All of my banks have the FX returns needed for those specific inputs built into them. So for example, I can bring up my drum bank and all the drum inputs and FX are all right there.

“Everything that I use is right there on each page or bank of faders. The left side of my console is the side that I rotate through; my right side is vocals and all the things I want to stay up all the time. Other consoles, when you flip through pages all the faders flip. On the SD8, it’s just the ones you want to. So functionality wise, it seems a lot faster to me. And seems like it makes a lot more sense.”

Nielsen found the frequency-selective compression a boon when mixing Harry’s vocal. “Most of the time, I don’t have to do much EQing on her but there are times when I have to… she’s one of those singers that sings off the mic and on certain notes, she likes to eat the mic a little bit. Because of that proximity effect as she moves into the mic, low-end gets created and it needs to be taken care of somehow.

“Using the frequency selective compression, I can compress those frequencies that tend to stick out as she gets closer to the mic. That alone is a godsend and it changes the way I feel about her vocals. It’s specifically incredible.”

Another technique he employs to beef up the band’s sound is to double-bus the drums. “My entire drum mix goes to two separate stereo buses and both of them are compressed differently. One is compressed so that it takes the entire kit and levels the whole thing out super-smooth, almost too smooth for live. For the second set, I use really light compression so the drums really pop out. Mixing those two together gives me a drum mix that sits right in the pocket yet still has a lot of life to it, with a punchy kick drum and punchy snare.”

For the guitar sounds, Nielsen takes both mic signals from the guitar amps and a Radial JDX box that sit between the amps and the speaker cabinets.

This allows him to create the sound he’s looking for, but the delay between the direct and mic signal has to be corrected.

On the digital console, he uses the input delay to make this correction. “Other consoles don’t give you the fine-grained delay on the inputs that you need to get things to really line up and the DiGiCo does that. It’s unbelievable compared to what other consoles do,” he says.

In addition to the plethora of onboard effects, Nielsen’s carrying a rack of his favorite outboard gear. “Which doesn’t mean that the reverbs in the DiGiCo don’t sound good, its just there are a few things that have a specific sound that I like to add in. For example, I like the sound of analog compression and have two dbx 162SLs.

“I also have a Bricasti M7 reverb, an Eventide DSP7000 for a bit of thickening and that Antares sound when needed, a Dolby Lake Processor, as well as a TCD2 delay because I like some of the 2290-style delays that it can do. These are effects that I could probably get on the console but I have this comfort feeling using them and have been for a long time. The Bricasti reverb is just a beautiful verb and I want that specific sound.”

Nielsen makes use of a Motion tablet for the 8in/8out Dolby Lake Processor housed in his rack. “The DLP is used for system EQ, delay and distribution and receives three AES inputs from the SD8 that include Left, Right and Sub, with outputs including Left, Right, Sub 1, Sub 2, Front Fill, Center, Delay and Production Feed.  Using the DLP this way frees up AES and analog outputs on the SD8 for other uses and allows me to do my room correction with Lake and the tablet.

The analog inputs and outputs on the SD8 are used for two dbx 162SL’s inserted on Debbie’s vocal, bass subgroup, and Left & Right as well as send and return of the TC D2, the send of the Bricasti M7 and FOH talkback.  The AES ins and outs are the Eventide DSP7000, RME UFX interface and the returns of the Bricasti M7. In addition, the entire rig is clocked using an Apogee Big Ben clock with feeds to the console, DLP and the RME.”

On a gig at the Isle of Wight Festival last year, the SD8 proved handy last minute in solving some challenges posed while interfacing with the festival remote recording truck. The SD8s onboard MADI capabilities saved the day. “Last minute, we were asked to unpatch all of our stuff—we carry a full two-way split with one going to monitors one to FOH with no third split—as they wanted to run the signal through their active split first and then feed back into our split. The time was so tight and we were worried about all of our gains changing and just general issues going through a remote truck.

“We quickly discussed our options and I found out that the truck was running off MADI so we were able to come off the MADI B output of the console back down our 6-channel BNC snake and went back to truck that way. The truck was also able to send me their clock into the mic pre’s onstage and I clocked both systems off that so we were all running off the same clock. It was a very quick fix for something that was going to cause us possibly a lot of issues.

“In the end, they were happier with all of that and their mixes ended up being really nice and clean because they were getting the same thing that I was right off my mic pres. As MADI becomes more of a standard, it’s nice to have a console that’s more compatible than just about any other systems out there.”

The console’s ability to do virtual soundchecks was also a boon to Nielsen for a variety of reasons. “They’re definitely the kind of band that wants to soundcheck every day in order to feel comfortable in the room they’re in, but on the occasion when they can’t—because of interviews or transportation issues—it’s nice to have the virtual soundcheck for that. I record into Logic, which gives me the ability to go home and listen to the inputs and do rough mixes for the band.

“Also, at the beginning of the US tour, I couldn’t do the first week because I had to be in Austin for the ACL Festival and the solution was very simple. We found a replacement, Tom Heinisch of SK Systems who was very DiGiCo savvy, and I sent him my files from the consoles and also a hard drive that had the shows on it. He was able to listen to the shows, learn where all the changes and the solos were, and it gave him a grasp of what his mixes were going to be before he even met the band.”

For everyday recording purposes, Nielsen records all 42 channels with an RME MADIface card into a 17-inch MacBook Pro. “I put up a couple of additional Shure KSM 32 audience mikes and a Shure VP88 downstage center right behind Debbie’s wedges facing the audience. I also use a couple of Shure SM81s at FOH to catch the audience roar and excitement.”

In hindsight, Nielsen and Big House partner Roy Kircher have been more than satisfied with their SD8 purchases. A decision that was a result of hearing the console in action at the Austin City Limits Festival a number of years ago. He says although the deciding factors at the time was the buzz on the console and its growing popularity, it proved to sound and perform better than most of the consoles they’d had in stock. “So we bought two,” he laughs.

“In central Texas, we’re one of the only rental houses that have these consoles and we get rental requests not just from bands but also from other sound companies as well. They’ve filled a niche for us in this biz as a higher-end, specialty console.

“Plus, they’re great sounding and a step above everything else we have in their price range. The big thing with digital consoles is everyone’s worried about how reliable they are. Does it handle the road, does it work everyday and can you rely on it? And the answer is Yes. We’ve had NO issues with either one of our consoles, whether its reliability or crashing issues, and they’ve paid for themselves time and time again.”

DiGiCo

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Posted by Keith Clark on 01/11 at 10:54 AM
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Church Sound: Working Within Volume Limits

You’re better off dealing with the root cause of the problem rather than trying to figure out how to stay below an arbitrary number
This article is provided by ChurchTechArts.

 
The topic for this post comes from a reader who wants to know what he should do when faced with the requirement to mix no louder than 85 dB SPL peaks.

That’s right, 85 dB peaks.

Why I Hate Volume Limits
I used to own a video production company. We were often hired to do video based on length.

I always tried to talk the client out of imposing a length limit on a project saying, “The video needs to be as long as it needs to be, then it’s done.” I feel the same way about volume.

Ideally, the worship leader, front of house engineer and church leadership are all on the same page when it comes to volume.

In that ideal world, the music will be mixed as loud as it needs to be to convey the power and energy (or lack thereof) required. The band, the song and the crowd will tell you how loud it needs to be. Go over that and it’s too loud; go under and it’s too soft.

Imposing a arbitrary limit on volume to me seems a bit like telling the pastor his sermon needs to be 3,000 words, no more, no less. But we live in a less than ideal world, and we have to live within arbitrarily defined volume limits. So what’s a sound guy to do?

Investigate
The first thing I would do when faced with a limit like that is find out where the number is coming from.

Is it based in an inaccurate reading of OSHA hearing protection guidelines? If so, educate yourself and have a rational conversation with your pastor. Help him to understand that 8 hours of exposure to 85 dBA SPL in a machine shop 5 days a week is a whole different animal than 85 dBA SPL peaks for 15 minutes of worship music.

If that’s not the case, dig a little deeper and see where the number came from. Did someone wander by the booth one day and see 85 on the meter and think, “That sounds about right?” Are people complaining that it’s “too loud?” Is it really too loud or are there spectral balance issues? Or perhaps the setter of the number doesn’t like electric guitar. Or drums.

Acoustic drums will generate 85 dB peaks with the PA turned off, so you need to figure out where this is coming from.

System tuning and spectral balance are huge issues that can be addressed and give you a to more leeway in mixing at an appropriate level. 85 dBA mixes can still be excruciating, while 100 dBA can be enjoyable if done well.

Live Within Your Means
Or in this case, your leadership. In my current church, I have a different definition of “too loud” than my senior pastor does.

Since his is lower, I have to adapt my mixing style to suit him—he’s the boss after all.

The challenge for me is that his definition changes week to week.

I’ve been told it’s “awesome” one week at 92-94 while it could be “too loud” at 90-91 next week.

So I’ve spent a lot of time working on getting my mixes right, the balance correct and the system tuned to his liking.

I’ve also adapted a different way of metering my loudness. I use a software program called LAMA, which can display both a standard SPL readout (I use A, Slow) and an average (I have chosen 10-seconds).

LAMA allows me to set colors at various levels so I have my average number turn yellow at 85 dBA SPL, and red at 91, which gives me a “corner of the eye” indication as to where I am.

I keep an eye on the standard readout as well, and occasionally my peaks run into the low to mid 90s, but for the last month and a half, if I keep my 10-second average below 90, my pastor is happy.

Personally, I’d be happier if it was louder. But I’m not paid to be happy; I’m paid to make him happy. I often say, “If you can’t abide by the limitations your leadership puts on you, then you need to leave.” Same applies here.

Again, I would talk to my pastor and find out where this is coming from. As him if it would be OK to try mixing to a 85 dB 10-second average and see how that feels. Address the spectral and mix balance issues; you might be surprised.

Beware Compression
The reader asked if he should compress the inputs, and bus compress the mix to give him the power he wants, while staying under the “legal limit.” To me, that’s a little like putting your phone on speaker and holding it in front of you while you drive.

Yes, you could compress the inputs a few dB, then bus compress a few more, then compress the master another a little further, and compress it again in the DSP. That would certainly raise your average SPL while keeping your peak below 85.

However, it’s very likely that this technique will result in the perception that it’s even louder, which may cause your limit to be lowered further.

You could also hard-limit your DSP so you can’t exceed 85; but again, if you suck all the dynamics out of the music, all the life goes with it, and it will also sound louder. This would be self-defeating on two fronts.

At the end of the day, I think you’re better off dealing with the root cause of the problem rather than trying to figure out how to stay below an arbitrary number.

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 01/11 at 10:09 AM
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Monday, January 09, 2012

Church Sound: Your Audio Assignments For 2012

The more you keep doing what you’ve always been doing… well…you aren’t going to improve your sound
This article is provided by Behind The Mixer.

 
You need to get out of your normal audio routines. 

The more you keep doing what you’ve always been doing… well…you aren’t going to improve your sound. 

Therefore, I’m starting out 2012 by giving you seven assignments. 

These assignments will make you really think about your mixing, your equipment, and your expectations. 

Best of all, these assignments are relatively easy!

1. [Mixing] Consider two vocalists singing the same lines. Try creating contrast between their voices. Now try removing much of that contrast and creating one voice. Which sounds better? 

Experiment with different songs and find out where they should be blended as one versus contrasted. What would you do if they were singing different lines?

2. [Mixing] Walk around the sanctuary and carefully listen to the music at different points in the room. Note any places in the room where the music sounds different. Perform this during a band practice. 

How can you alter the mix so you get a more uniform sound throughout the whole room?

3. [Mixing] Ask a fellow audio tech, a friend, and a musician about the mix. Compare their responses. How are they different? How are they the same? 

What did you learn about your mix? What did you learn about that person’s preferences?

4. [Mixing and Equipment] Take an inventory of all the microphones; note the make and model, the type (condenser or dynamic) and the polar pattern. Focusing on the type and the polar pattern, where do you think each would work best? 

Try different setups during practice until you find the one that gets the best sound. Take note of that and use that instrument/mic and vocal/mic setup from then on.

5. [Equipment] Use a sound meter during your church services. Note the average volume levels of the pastor and of the band. Track this in a spreadsheet.

Whenever you get a volume complaint, look at that day’s average volume compared to other days. It might be it was louder or softer than normal. It might be that person just wasn’t in the mood for the music.

6. [Equipment] Follow the signal path of all cables coming out of the mixer. Create an easy-to-understand schematic of how the sound booth is wired. Track all wires going into other components and where they go. 

The next time you have a problem with equipment in the booth, use the schematic to determine where the problem likely originated and what other components might be involved.

7. [Expectations] If you attended a concert, what would you expect from the audio tech? If you were in a band, what would you expect from the audio tech?

How do your answers to these questions reflect what you are currently doing as a sound tech?

BONUS: [Mental Anguish] Avoid the “constant tweak” mentality by asking yourself this question…will the congregation notice what I want to do?  They are not going to notice a slight mid-range bump in the guitar EQ. 

They will notice if you add clarity to an instrument so it sits better in the mix.

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 01/09 at 09:44 AM
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Thursday, January 05, 2012

The Old Soundman: Studio Versus Front Of House

You’ve gotta be kidding me!

Dear Old Soundman:
Studio engineers verses front-of-house engineers - can they be one and the same?

Studio guys don’t want to let us get a shot at their jobs, because we would wipe the floor with them! Unfortunately, we would also wipe the floor with their clients!

Do I really want to spend three months listening to some rock star (who tries to pretend he’s a man of the people) reveal how completely dependent he is on the click track, and the Pro Tools, and his bloody re-amping?

The lights must be kept turned down nice and low, while his sycophantic entourage sits on the couch, smoking ganja and getting crumbs from their vegan pastries on my carpet.

Only the love and devotion I feel for my family keeps me from climbing on star boy and beating his head in with an RE-20 while shrieking, “play a damn song, why don’t you!”

If I ever did that, David, I would make sure to have the theme music from “Psycho” playing very loudly in the background.

Don’t even get me started on pretty boy from the record company, and his cowardly criticizing of my mixes behind my back. Hanging is too good for the likes of him! I’m thinking a year in solitary confinement, with the same Melvins record playing over and over again, 24/7.

By the way, you are honored and respected, OSM.

Righteous!

Signed,
A frontman/singer well cared for by his own OSM.
David P.

See, here is a smart guy who didn’t employ a younger sound buddy who would have asked for less money but doesn’t know his expletive deleted from a hole in the ground!

Our friend went and got an OSM who knows the drill, and it pays off in the long run because now David P. can exist in a superior, relaxing audio environment rather than wondering what input is about to go wildly intermittent or explode into hellish feedback.

We old guys change the batteries - you know what I’m saying? We not only check things, we double-check them! Maybe at one point in our lives, we did that mainly because we were all amped up on controlled substances deleted, so to speak, but now we do it because we know it’s the right thing to do!

And it pays off for our artists. If they want some Chippendale-looking guys, they can damn well go to Chippendale’s!

I like being the absolute ruler of my little acre. When you and pretty boy and Mr. Royal Rock Star come into my club, you do things my way or the highway!

Luv
The Old Soundman

There’s simply no denying the love from The Old Soundman. Check out more from OSM here.

{extended}
Posted by admin on 01/05 at 01:45 PM
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Church Sound: “Cheating” In The Mix To Dial In Unfamiliar Performers?

Preparing for the best possible result - on the fly

In the eyes of some people, I “cheat” when mixing. 

This cheating usually takes place when I’m working with a band and/or singers that I don’t know that well.

One of the common things that we face in church production today is “the worship team” - usually four to five singers that vocally lead the service or presentation.

Too many times to count, I’ve found myself mixing and not knowing a single person on the team. It’s also common not to have an opportunity to rehearse with them beforehand.

It’s a less than ideal situation, but it happens.

And this is when I turn to cheating, or, as I prefer to call it, prepare for the best possible result:

1) I make sure that all the vocalist have matched mics, and ideally a mic I’m very familiar with (like a Shure SM58);

2) I preset the gain on the channel to all match at the anticipated level needed;

3) I preset the channel EQ of those mics to the anticipated EQ based upon what I know about the performance characteristics of the mics;

4) I set all of channel faders to the same level;

5) And, this is the one where I get called the most on cheating: I use meters to initially set the mix, not my ears.

The how of this is a bit more complex than the why. The minute the singers start singing, I “solo” (or pre-fade listen, PFL) each of them, one at a time.

This is done without headphones on because I need to be very quick to get the mix under control.

During the PFL process, I look at the meters, and work to match the level of all of the singers. If I know who the leader of the group is (if there is one), I set his/her gain 3 dB hotter than the others.  The logic behind this: when I look at my faders while mixing, I have an accurate visual representation of the mix. By the way, I do the same for the band.

My personal goal is to have a live able mix with in the first 30 seconds.

Once I get a “livable’ mix, I go back to PFL, but with headphones this time. With Sharpie in hand, I listen to each vocalist separately, mark his/her channel with either arrows pointing to that person’s respective location on stage, or better yet, if I have four-part vocals I will label them S (for soprano) A (for alto) T (for tenor) and B (for bass). 

With the vocals now properly identified and level matched, I start to use my ears to fine-tune the mix.

Maybe you call this mixing by meters, rather than cheating, but it’s served me very well and I plan to keep on doing it, no matter what it’s called.

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

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Posted by Keith Clark on 01/05 at 10:16 AM
Church SoundFeaturePollConsolesEngineerMixerTechnicianPermalink

Monday, December 26, 2011

Drum Tuning Tips From The Famous Drum Doctor

These quick drum tuning tips could be lifesavers in your next session if you're new to tuning drums
This article is provided by Bobby Owsinski.

 
If you’re doing a session in Los Angeles and you want your drums to instantly sound great, then your first call is to the Drum Doctors to either rent a fantastic sounding kit, or have your kit tuned.

Ross Garfield is the “Drum Doctor” and you’ve heard his drum sounds on platinum recordings from Bruce Springsteen, Rod Stewart, Metallica, Dwight Yokum,  Red Hot Chili Peppers, Foo Fighters, Lenny Kravitiz, Michael Jackson and many, many more.

Ross was kind enough to sit down for an interview when I wrote The Recording Engineer’s Handbook, but I’ve featured some of his tips in other books as well including The Drum Recording Handbook, The Touring Musician’s Handbook, and The Music Producer’s Handbook.

So, I like to think his tips are worth sharing! Here are a few of his quick drum tuning tips, which can be lifesavers if you’re new to tuning drums.

If the snares buzz when the toms are hit:
• Check that the snares are straight.
• Check to see if the snares are flat and centered on the drum.
• Loosen the bottom head.
• Retune the offending toms.

If the snare drum has too much ring:
• Tune the heads lower.
• Use a heavier head like a coated CS with the dot on the bottom or a coated Emperor.
• Use a full or partial muffling ring.
• Have the edges checked and/or recut to a flatter angle.

If the kick drum isn’t punchy and lacks power in the context of the music:
• Try increasing and decreasing the amount of muffling in the drum, or try a different blanket or pillow.
• Change to a heavier, uncoated head like a clear Emperor or PowerStroke 3.
• Change to a thinner front head or one with a larger cutout.
• Have the edges recut to create more attack.

If one or more of the toms are difficult to tune or have an unwanted “growl”:
• Check the top heads for dents and replace as necessary.
• Check the evenness of tension all around on the top and bottom heads.
• Tighten the bottom head.
• Have the bearing edges checked and recut as required.

If the floor tom has an undesirable “basketball-type” after-ring, try this:
• Loosen the bottom head.
• Check the top heads for dents and replace as necessary.
• Loosen the top head.
• Switch to a different type or weight top or bottom head like a clear Ambassador or Emperor).
• Have the bearing edges recut to emphasize the lower partials.

If the cymbals are cracking or breaking with greater frequency, try the following:
• Always transport the cymbals in a top-quality, reinforced cymbal case or bag to avoid nicks that can become cracks.
• Use the proper cymbals felts, washers and sleeves at all times.
• Avoid over-tightening the cymbal stand.
• Use larger or heavier cymbals that you won’t have to overplay to hear.

Hopefully these tips are useful to you in your next session!

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

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Posted by admin on 12/26 at 12:48 PM
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Ensuring You’re Recording At The Proper Levels

You're probably tracking too hot. Here's why...
This article is provided by MASSIVE Mastering.

 
I’m going to try to keep this very “fool proof”.

This was born out of the rantings of hundreds and hundreds of posts on a dozen or more audio forums exploding like a volcano recorded with lots of headroom.

I hope to instill a basic understanding of why certain trends and common beliefs are just plain bad.

And by the time you’re done reading, and perhaps doing a little experimentation based on this, you won’t need me to prove it. You’ll know it yourself.

Is this a “miracle cure” for bad recordings?

Normally, I’d say no.

But with the dozens and dozens - easily now into hundreds of e-mails, phone calls, letters, forum posts and other forms of communication I’ve received about how this advice has completely changed a persons view of recording recording, I figured this information is worth sharing with a wider audience.

The sad part is this should be common sense. To anyone that grew up “on tape” it probably is. To those brought up in 1’s and 0’s, it might not be so obvious.

So, if you’ve been struggling with recordings that sound “weak” or “small” or too dense or “just not ‘pro’ enough” then please, read on. If this is about you, you might think differently soon.

As a mastering engineer, I work on recordings from pretty much every level of experience. A few years ago, I noticed something unusual.

“Ultra rookie” recordings, that is those made by people with little or no experience, sounded fine. They didn’t know any better, so they didn’t have enough rope to hang themselves with.

Then, at the other end of the spectrum, “Pro” recordings sounded fine. They know what they’re doing and/or are using gear with obscene amounts of usable headroom (explained later).

The “middle of the road” engineers with a year or two - or much, much more experience—Those are the recordings that sounded “small” and spectrally challenged. So after quizzing these people over months and months, I came up with the following conclusion…

You’re probably recording too hot.

And it’s absolutely ruining recording after recording after recording. And it’s the simplest thing in the universe to correct.

I know, I know - “It says in the manual to record as hot as you can without clipping.” Well, I’m going to flat-out call that B.S. and I’m going to back it up with a simple (if not somewhat time-consuming) experiment.

Also as a mastering engineer, let’s get something straight - I don’t like the “loudness war” going on.

However, I’m as guilty as the next in contributing to it. I can’t fight it, as much as I try. Hopefully it’ll be over some day. Unfortunately, with the quest for loud, there are a lot of engineers out there shooting themselves in the foot before they even know how to aim.

They think that tracking loud and mixing loud contributes to a louder recording after the mastering phase. This is absolutely untrue and it’s generally the best way to make sure that your recording will not have the “loudness potential” of the average commercial release.

Clean recordings, or those made with low distortion and good spectral balance, are the ones that handle the “abuse” of the mastering phase with flying colors.

This article isn’t intended to give you some secret way of making louder recordings. But it will almost undoubtedly give you the ammo needed to make better recordings. And those better and cleaner recordings are the ones that can be louder recordings in the end.

First, let’s get through a little nomenclature,

dBFS
Deci-Bel (one tenth of a Bel) Full Scale, on the digital recording scale, -0dBFS is the hottest signal you can have.

“All ones.” Top of the scale, can’t get hotter, etc. Always “minus” as you can never go higher - So the reading will always be a specified amount below 0.

Line Level / 0dBVU
Just what it says. Line level. 0dBVU on an analog VU (volume unit) meter. Pro (+4dBu) or consumer (-10dBv) level, it’s line level.

We can also refer to this as RMS (Root Mean, Squared), or a level over a specific amount of time. You can go above or below 0dBVU.

It’s simply a nominal level to which basically everything audio is related to.

Headroom
The space between a nominal signal (in this case, line level) and the point where the circuit fails.

In digital, basically anything under full scale (-0dBFS) would be considered headroom.

In analog, it’s the space between 0dBVU and the point where the circuit clips (failing completely). In analog, there can be a big difference between “headroom” and “usable headroom.” We’ll get into that in a bit.

Steak
From the old Norse “steik” meaning “roast”. A slice of meat, typically beef, usually cut thick and across the muscle grain and served broiled or fried (thank you, Wikipedia).

So…

You have a microphone and a preamp going into a converter or sound card. Those converters are calibrated at line level.

In most cases, over the last several years, most I’ve seen are calibrated to -18dBFS = line level (or 0dBVU).

In other words, if you run a steady signal (a sustained note on a keyboard for instance) through a preamp and turn up the preamp gain until the VU meter reads 0dBVU, at the converter, it will read -18dBFS (or -18dBFS(RMS) - full scale, but measured over time).

This is where your gear is designed to run. This is where it’s spec’d at. You will have a decent amount of headroom, the lowest distortion, the best signal to noise ratio, etc., etc., etc. around this level or lower.

Some gear - usually very high-quality stuff, has a good amount of usable headroom above this level.

A lot of “budget friendly” gear does not. So all of this advice is more important if you’re using “okay” gear at the input. Even your digital converters are analog components up to the converter itself. They don’t want to be “beat up” all the time either.

Let’s look into headroom. Above that 0dBVU/-18dBFS range, digital headroom is simple - perfect, perfect, perfect, perfect, CLIP.

The signal is “what it’s supposed to be” until the point of failure. Analog gear (your preamp, compressors, outboard signal processors, etc.) isn’t like that…

It’s more like “Perfect, a little noisy, “tight” sounding, spectrally distorted, CLIP.

The converter’s job is simple - reproduce the signal it’s fed digitally, whether that signal is clean and dynamic or distorted and squishy.

The analog chain’s job is anything but - typically, you’re adding 20, 30, maybe 50dB of gain to the incoming signal.

The preamp is working - not just “passing” the signal.

And that signal can start to suffer from noise, distortion and dynamically dependent (varying along with volume) spectral imbalance (a skewing of the overall spectrum from an EQ standpoint).

In other words, a nice, thick, chunky guitar tone (for example) might have different characteristics depending on how hot the signal is.

The highs might be open and airy and then the signal gets loud for whatever reason and the highs either get swallowed up, or perhaps get very harsh and strident.

In any case, it’s an inconsistency that isn’t’ there when the levels are more “normal.”

Even though the analog gear probably has spec’d headroom well above digital’s full-scale, it doesn’t mean that signal actually has the integrity it should up to that level.

So what happens is simple, a signal is recorded that’s too hot (usually to “use all the bits” which again, is a bunch of BS).

It overdrives the input chain not unlike a guitar preamp overdriving a Marshall stack (well, not that much, but the premise is the same).

Now, after all the other tracks are recorded, all of them need to be attenuated by 12, maybe 15dB or more so the mix doesn’t clip. Those distorted, spectrally questionable, squishy, noisy tracks all get turned down.

Are you seeing my point yet?

When you take a steak and cook it until it’s burnt, it’s burnt.

If you pour ice cubes all over it, it doesn’t make it more rare - it makes it a cold, wet, burnt steak. No matter what you do, it’s still burnt. Just like if you record too hot.

But if you cook a steak a little too rare, you can always heat it up a bit later.

You can microwave it without it turning into leather. You can pan-fry it for a few minutes and it’s still a tasty, savory piece of steak.

When you use up all your headroom right away, you don’t get it back by turning it down.

It’s gone forever. Sure, you can increase mix headroom or the headroom at the buss - but it’s not going to make the track less distorted or fix the skewed S/N ratio.

Here’s Your Experiment
You’ll need a few Y-cables (let’s not get into the technical aspects of splitting a mic signal - It’s an experiment) and at least one stereo (2-channel) preamp.

Record a song using as many tracks as you feel fit. The more, the more apparent. You’re going to split the mic signal and record each twice simultaneously.

On one channel of the preamp, set the gain so it peaks between -18 and -12dBFS at the converters and record them to odd numbered channels.

On the other, set it as high as possible without clipping and record them to even numbered channels. Record some guitars, drums, maybe piano, of course some vocals, keyboards, go nuts.

Set all the odd numbered (“normal” level) channels to unity and toss up a rough mix to a stereo buss - which should be a piece o’ cake.

Switch over to the even numbered channels and figure out how much you’re going to have to attenuate them all so the main buss isn’t clipping constantly.

It might be a lot. Could be a 10-15dB cut on all channels before you can even think of starting to do anything else. Send those to a stereo buss. Solo the busses, one at a time, and try to match the levels between the mixes.

You’ll probably immediately notice that the “normal” mix is much more open, dynamic, airy, clear, clean, with much more “sonic space” between the instruments than the “hot” mix.

Now, add a limiter on the main buss. Run the “hot” mix into it and bring the level up until it starts to obviously distort and fall apart sonically.

Then switch over to the “normal” mix - which should now be “rammed” by the same amount. If your experience is pretty much like everyone else’s, the “normal” mix is still much more open, airy and dynamic with less distortion and more “crankability” than the other.

The “Dumbed Down” Version
Stop recording so hot. Instead of trying to get your tracks to peak at -2dBFS, have them peak between -20 and -12dBFS and your recordings will almost undoubtedly sound better.

Mixing will be easier. EQ will be more effective. Compression will be smoother, more manageable and predictable. You’re in the age of 24-bit digital recording - relax and enjoy the headroom.

Even if your only concern is the volume of the finished product (which would be a shame, but it happens), recordings made with a good amount of headroom are almost undoubtedly better suited to handle the “abuse” of excessive dynamics control.

Quieter recordings have more potential to be loud later. It’s because they’re usually better sounding recordings in the first place.

John Scrip is the Owner and Engineer at MASSIVE Mastering.

{extended}
Posted by admin on 12/26 at 10:16 AM
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Monday, December 19, 2011

Church Sound: Knowing When To “Trash” Your Mix

Getting to a better place while saving time and focusing on the fundamentals
This article is provided by Behind The Mixer.

 
WHAT’S WRONG WITH IT?!

Have you ever mixed the band and felt something was wrong with the mix? You tweak and tweak but the sounds just don’t gel.

Time to trash your mix and start over.

The Two Benefits of Trashing Your Mix

1. You save time. It’s hard to start over on a mix when you consider how much time you spent building it. However, when you can’t identify the one or two elements in your mix that are off balance, you will spend less time if you start over than if you tweaked for an eternity.

2. You are forced to focus on the fundamentals. Yes, this is a benefit because it forces you to think about what you are doing and why you are doing it. Just like professional basketball players still practice dribbling and passing, focusing on the basics of your work keeps you sharp.

How to Rebuild the Mix in Nine Steps

Here is a basic outline for what you should do…

1. Reset everything. Reset all your EQs, turn off your effects, turn off any compression, and take channels out of sub-groups. You are now back to a simple baseline mix with only your channel gains being set. I mention removing sub-groups as you might have EQ specific for subgroups.

2. Review your channel gains. I’ll say 99 percent of the time, my gain levels are good. It’s usually a vocal microphone I might boost a little. Boosting gain means you can kill your monitor mixes or blow out the ears of a musician with IEMs, so make sure you notify the band if you need to boost gains.

Here are a couple methods for setting gain:

How to Set Gain Structure
Dave Rat’s method for Setting Gain (An Alternative Method)

3. Set your general volume balance with all your musicians and singers. There are two ways of doing this; starting with drums or starting with the vocals. You will want to start with drums. move from drums to bass to guitars to vocals.

Starting with vocals is helpful when you are dealing with a strict decibel restriction. Setting the vocal level’s first, you are ensuring they will be prominent in the mix.

4. Mix in your drums and your bass. The kick drum can cover up the bass and vice versa if you don’t mix them properly. Decide which instrument owns the low end sound. Using a sweepable mid on an analog board, consider sweeping the bass mid’s far down [250 hz range] and then cutting/boosting. I find I cut in the low-end eq and then using the sweepable mid, I can give the bass a distinct clear sound.

Check out these articles for more information:

Foundation of the Perfect Mix
The Art of Bass EQ

5. Mix in your keyboards. Keys can be used for everything from a grand piano sound to 80′s synth pads.

Keyboards that are used for a large pad sound can get lost in a bit of the bass and the toms.

Start with enabling the HPF, high pass filter, and then listening for the frequency space the keyboard is filling.

Make your adjustments accordingly so the pads have their own sound and aren’t lost with the other instruments.

6. Mix in the guitars. Start with the electric and the move to the acoustic. All the time, think about how the existing sounds in the mix are blending with what you have on the stage. You might not have a bass player. In that case, give that electric guitar some space in the lower frequencies.

A key point to remember is you should make a small band sound big and a big band sound small. That is to say, in the case of a big band, tighten up the frequency range of each instrument so you can easily pick it out in the mix.

A popular topic in guitar mixing is whether or not to use the HPF. Mixing is an art and therefore, there is no black and white answer as to when to use the HPF on a guitar. Use your ears. You know what you want to hear, listen for it and use what you have to reach that sound.

EQ an Acoustic Guitar
Mixing Multiple Guitars
Common Acoustic Guitar Mixing Mistakes

7. Mix the vocals. Start by using the high pass filter to cut the very low end frequencies. This will help eliminate any extra low end coming from instruments like the drums.

Next, try boosting in the 6 kHz range to give some brightness and clarity to the vocals. Finally, consider cutting the problem frequencies that can exist in the 2.5 kHz to 4 kHz range.

General Vocal EQ Settings and Theory
More on EQ’ing Vocals

You might have other instruments that I haven’t listed. If so, mix them in where they seem appropriate. For example, mix in the violin after the acoustic guitar.

8. Mix across channels. You know it’s not as easy as mixing instruments individually and then you are done. During the process of mixing all sound together into the right sound you want, keep in might the arrangement of the song.

For example, if the song calls for the keyboard to be the focal point, then make it the focal point by giving it room in the mix. You can do that not only by using volume but by making slight frequency cuts in instruments that share the same frequencies as the keyboard. You don’t have to cut them out completely… just think of it as making room for the keyboard to be obvious as the lead instrument.

9. Add your effects. This can be done after you mix across all channels or during that mixing. A simple rule of thumb is do everything you can with your EQ work and then use the effects to fill in what is missing.

Otherwise, you’ll find yourself battling against the EQ and the effects because you are trying to get the effects to do something that’s better done with the EQ.

Summary

Van Gogh painted over his own artwork because he didn’t like what he’d painted the first time. C.S. Lewis likely had a few drafts of Narnia chapters that got tossed in the fireplace.

There will be times when you don’t nail the mix and your next move shouldn’t be reworking that mix. Your next move should be starting over.

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 12/19 at 07:49 AM
Church SoundFeaturePollConsolesMixerSound ReinforcementPermalink

Monday, December 12, 2011

Test Your Live Sound Fundamentals In Our Latest Quiz

Think your live sound knowledge is at the appropriate level? Time to find out!

Working with in live sound isn’t easy by any measure, which is why it’s always wise to be testing your skills.

Take a read and see how you stack up against our questions, and be sure to check out our other quizzes as well.

As always, some of these questions may have more than one correct answer.

image

 
 
1. Accurate gain structure:
A: Is the optimum adjustment an gain adjustments.
B: Will maximize output levels.
C: Will minimize noise.
D: All of the above.

 
2. CMRR means:
A: Common Mode Rejection Ratio.
B: Noise is cancelled by equal noise voltage in both conductors.
C: Both.
D: Neither.

 
3. VCAs (Voltage Controlled Amplifiers) are:
A: Also know as sub-groups.
B: Only used in automated consoles.
C: Never used without equalization.
D: None of the above.

 
4. A transformer isolator is needed to:
A: Create a floating input.
B: Create a unbalanced input.
C: Create a VCA.
D: None of the above.

 
5. The “VU” in VU meter means:
A: Volume Unlimited.
B: Voltage Unit.
C: Vector Unlimited.
D: None of the above.

 
image

 
 
6. Any channel-insert cable must:
A: Have three conductors.
B: Carry both send and return signals.
C: Both.
D: Neither.

 
7. LCR is:
A: Level, Control, Return.
B: Often used in theatrical applications.
C: Left, Center, Right.
D: Both B & C.

 
8. The preferred “return” connection for effects is:
A: Directly into a sub-mix input.
B: Directly into an aux input.
C: Directly into a channel input.
D: None of the above.

 
9. Unity Gain is:
A: The amount of electrical energy used by a mixing console.
B: When an audio circuit is adjusted to neither amplify nor attenuate an audio signal.
C: The input level demanded by any outboard signal processor.
D: The difference between output level and output resistance.

 
10. Microphone input gain control is:
A: Also know as input attenuation.
B: Frequently referenced as pad.
C: Often called trim.
D: All of the above.

 
image

 
 
11. Headroom is the:
A: Difference (in dB: between a circuit’s nominal level and maximum level.
B: Another term for signal-to-noise ratio.
C: Slang term for in-ear monitor mixing.
D: Electrical calculation defined by CMV.

 
12. Foldback is a:
A: Noise-canceling circuit.
B: Term for a monitor mix.
C: Common insert-snake wiring scheme.
D: An obsolete mono mixing technology.

 
13. In-ear monitor mixes are:
A: Usually mixed in stereo.
B: Illegal in Germany (health reasons).
C: Always transmitted via wireless systems.
D: All of the above.

 
14. A monitor mixing desk:
A: Must be placed on the stage.
B: Always receives the “first” signal spit.
C: Is always used to create a matrix.
D: Is always configured for mono mixing.

 
15. A signal split is a conventional method for:
A: Reliable grounding.
B: Dividing input signals for multiple mixing functions.
C: Measuring potential gain.
D: Feedback control.

 
image

 
 
16. An analog mixing console must contain:
A: Summing amplifiers.
B: Voltage controlled amplifiers.
C: Phantom power ranging 9V to 52VDC.
D: None of the above.

 

17. Unwanted mixer noise can be caused by:
A: Misadjusted gain structure.
B: Silicon controlled rectifiers.
C: Poor room acoustics.
D: Both A & B.

 
18. A mixing console’s equalizer section can:
A: Create a sub mix.
B: Cause system-wide distortion.
C: Create a post-fader aux send.
D: None of the above.

 
19. A “direct out” allows for:
A: Converting mono inputs to stereo outputs.
B: A matrix output.
C: A potential split for live recording.
D: All of the above.

 
20. Any aux send (output) in pre-fader mode will:
A: Be unaffected by fader movements.
B: Never reach unity gain.
C: Require equalization.
D: Both A & B.

 
More Fun Quizzes on PSW:
Test Your Knowledge Of Bass & Subwoofers
Take Our “Real World” Stage Monitoring Quiz
Test Your Knowledge Of Power As It Relates To Sound Systems
Quiz To Rate Your Audio Skills, Knowledge & Personality Type

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Posted by admin on 12/12 at 01:09 PM
Live SoundFeaturePollStudy HallDigitalEducationInstallationLoudspeakerMixerSignalSound ReinforcementSystemAudioPermalink

Soundcraft Releases “Guide to Mixing” App For iPad

Takes users through all aspects of setting up and using an audio mixer

Soundcraft has announced that it is released its popular tutorial, “The Soundcraft Guide to Mixing,” as an iPad app.

Derived from the popular “Guide to Mixing” booklet, it provides an introduction and explanation about what an audio mixer is, what its functions are, and the basics of setting up and using a PA system.

“Since its introduction, ‘The Soundcraft Guide to Mixing’ has become a popular reference for learning about what a mixer does and how to use it, and there are more than a million copies in use worldwide,” said Keith Watson, marketing director of the Harman Mixing Group. “From its original booklet form it has evolved to become available as a web page, as a downloadable PDF, on DVD and online video formats. Making this industry-standard reference available on the iPad was the inevitable next step.”

“The Soundcraft Guide to Mixing” is tailor-made for the iPad, and uses the same video clips that have drawn more than 350,000 views on YouTube.

The guide takes users through all aspects of setting up and using an audio mixer including: explaining the various controls and functions, connecting microphones, instruments and outboard effects, basic and advanced mixing techniques, using mixers for live sound and for studio recording, and many additional topics.

The full version of “The Soundcraft Guide to Mixing” is available for $2.99 here or search iTunes for “Soundcraft.” A lite version will soon be available free of charge that contains one full chapter and excerpts from the other chapters.

“’The Soundcraft Guide to Mixing’ has become an invaluable educational tool for students, small bands, musicians, novice sound engineers and people who are starting their careers in pro audio,” Watson adds. “The iPad is the perfect portable and interactive format for the guide.”

Soundcraft

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Posted by Keith Clark on 12/12 at 12:29 PM
AVLive SoundRecordingChurch SoundNewsPollProductAVConsolesMixerSoftware • (1) CommentsPermalink

Friday, December 09, 2011

Selecting The Right Contractor For Your Church Sound System Upgrade

Tips on selecting the company which will best serve your church audio needs.

Let’s pretend . . .

Imagine you walk into your next committee meeting and to your amazement discover that they have approved your suggestion to seek bids for a new sound and video system. 

You leave the meeting excited! 

Finally all of you problems will go away. 

No more buzz, no more hum, no more having to dim the lights and shut all the shades just to barely make out what you are projecting on the screen! 

You figure by next Sunday all of your frustrations will be gone. The next morning you immediately look in the yellow pages and see a large ad for Audio Services. 

You call and talk with “Blair” who informs you that he has on his shelf two of the latest, greatest speakers that will work in any room and deliver equal sound pressure and all frequencies.

To top it off he can have his guy there Friday to do the install! 

Then to seal the deal (or your fate) he throws out a price that is well within what your committee said you could spend.  Bingo we have a Deal! 

Blair and his hatchet men show up Friday afternoon.  You get out of work and excitedly head over to see the progress that has been made. 

To your surprise Blair and crew are walking around examining the sanctuary. They haven’t started a thing…

After many heated questions and answers, it comes to light that these particular speakers won’t possibly work in a room of this size (ie: average) without a needlessly costly add-on. Feeling boxed in, you agree.

Upon the completion of the “installation,” not only does the system not work, but now you’ve spent more than the committee approved. 

In fact you are so embarrassed you donate enough to make up the difference so that no one will know of you blunder….other than they hear it every week!

Rewind: What should you have done?
When deciding that it is time to upgrade the old sound system there are many options as to how to proceed. 

The first choice should be to decide to hire a design build contractor or a consultant to design the system.  Needless to say there are some very bad design build contractors and some very bad consultants. 

However there are also some very good ones!  My general rule (variables such as acoustics and complexity of the system also play into this decision) is that if the project is under $100,000.00 and in a room under 1000 seats I would explore a good design build contractor.  Most design build firms have good experience in these size rooms.

You also have the choice of using the local music store.  In general, unless the music store has a specific division that focuses on installation and has a strong proven track record, I would steer clear. 

Many of the poor designs and implementations that I see are from good intentioned “guitar shops” that have a great passion and understanding of gear and technology, but do not understand the laws of physics and just how difficult it can be to install a successful sound system in a larger room.

What does a design build contractor do?
A design build contractor should function in much the same way as a consultant. 

The only difference is that he is not going to bid the project out at the end of the design (the contractor will also be the installer).  I recommend that you once again do your home work and select the right contractor. 

What does a consultant do?
A good consultant will first and foremost find out who you are, what your ministry is like, future plans for the ministry and current challenges you are facing.

The consultant should take time to interview the sound technicians, musicians, worship leader and senior pastor to get a good understanding of your churches over all needs.

They should also take a good physical inventory of what equipment you already own and also your experience level with sound equipment. 

In addition blueprints and field measurements should be gathered to aid in the design of the system. 

Once the consultant has gathered this information the issue of budget must also decided. 

The best case scenario is to have the consultant help guide and establish the budget.  However this happens all too infrequently.

After establishing the budget the consultant should be prepared to do the first round of design. 

The design should be based on using tools like EASE (computer design simulation software that help predict many aspects of how a sound system will perform in a give space). 

The consultant should provide and equipment list as well as a descriptive narrative of how the system should perform. 

There are 4 main types of “contractors”

Music store
In general the music store is the least qualified contractor.  The person on the floor at the music store has more than likely not been exposed to all the tools and products that are available in the Professional contracting arena. 

In addition they have probably had very little if any training in system design, safety standards, proper grounding techniques and most importantly rigging. 

In addition they probably do not have software tools such as EASE or Star draw to aid in the design and documentation. 

Finally, you know that statement, “I wish I knew then what I know now.”  That usually applies greatly in this situation.  The person working the music store floor likely is unaware of what information they are lacking. 

Generally, unless the music store has a contracting division with all the proper tools and professional equipment lines, I’d steer clear.

Low Voltage Contractor
A low voltage contractor is a step in the right direction. They may (or may not) have the expertise required to do the project. 

Again, this has a great deal to do with whether or not they have a sound contracting division.

These firms usually work with telephones, networking, security and control systems. Their installation work is usually very good (unlike a typical music store install).

However, unless they have the tools and the correct people they may not be very adept at system design.  They may be good at copying the last installation they did, but they will probably have a hard time designing a system specific to your need.

Sound Contractor
The sound contractor is likely qualified to handle your project.  At the very least. they should be qualified to bid on a design if you go the consultant route.

When looking at the sound contractor is important to note the product lines that they carry and the type of work that they perform. 

Are they mostly a commercial contractor doing paging and public address systems?  Or are they geared towards night clubs and entertainment? 

Find out from their reference list and past jobs what they seem to do the most of and what type of systems they like to put in. 

For example if every install you visit is Brand X speakers, your installation will be using brand X speakers.  That may not be bad but you should be aware of it. 

With the sound contractor make sure you meet the people who will work on your project. 

They may have a great sales person and terrible engineers and project managers, so push to meet the entire team.

System Integrator
A true systems integrator should be able to take care of your design build needs.  They should have multiple product lines available to them and engineers on staff that have implemented many projects like yours. 

It is your responsibility to check the track record and confirm that these guys are qualified. 

Do some research and visit their installation and most importantly talk to as many users of their systems as you can find. 

Again with the systems integrator meet everybody that will be on the team of your project!

How do I select who to use?!

1. Decide if you are going to use a contractor or consultant

2. Select 3-4 reputable firms to ask them to present their capabilities to you on your project.  (ask around for names, visit the National Systems Contractor Association’s website NSCA.org to find members in your area)

3. Meet with the firms to explain your needs. Analyze…. Do they listen?  Or are they only interested in selling how great they are?  Do they give input as you describe your needs…?  “You may want to consider……….” or are they just taking down your ideas?

4. Set a specific date for the proposal to be turned in by.  (If they do not turn in a proposal on time chances are they will not install the system on time!)

5. While you are waiting for the proposal to come back.  Do your home work in finding out even more about these companies.

6. Let the companies present the proposal to you.  Listen carefully to see if they can articulate how they are meeting your specific needs. 

7. Evaluate the proposals based on how well the system meets your needs.  If the highest price is the only one that meets your needs you need to take it.  If it doesn’t meet your needs you need to change your stated needs so that you can afford them.

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

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Posted by Keith Clark on 12/09 at 02:19 PM
Church SoundFeatureBlogGary Z's Church SoundPollStudy HallAVEducationInstallationLoudspeakerMixerSignalSound ReinforcementSystemAudioPermalink

Thursday, December 08, 2011

Hal Leonard Publishes Microphones & Mixers, Second Edition

Covers the critical first steps in a truly organized path through the entire recording process

Hal Leonard Books has published Microphones & Mixers ($39.99) by Bill Gibson, an updated second edition of Book 1 in the 6-book Hal Leonard Recording Method series and includes a DVD ROM and online media.

As the first book in the series, Microphones & Mixers covers the critical first steps in a truly organized path through the entire recording process.

Topics include how professional microphones work, which to choose and why (plus accepted techniques for using them), understanding the signal path from mics to mixers and how to operate these critical tools to capture excellent recordings, as well as explanations of the most up-to-date tools and techniques involved in using dynamics and effects processors.

From initial considerations to mix-down, mastering, and replication, this method provides important considerations and techniques every recording musician needs to know.

In Microphones & Mixers, Gibson augments his straightforward writing style with diagrams, product photos, graphs, and charts to make complex concepts easy to grasp.

Gibson has spent more than 30 years writing, performing, recording, producing, and teaching music, and has written more than 30 books and produced several videos covering important audio concepts. He is currently an instructor at the Art Institute of Seattle and teaches online sound courses for Berklee College of Music in Boston, for which he has developed the curriculum.

Microphones & Mixers is the first book in The Hal Leonard Recording Method, which is the first professional multimedia recording method to take readers from the beginning of the signal path to the final master mix. Other books in the series include Instrumental & Vocal Recording, Recoding Software & Plug-ins, Sequencing Samples & Loops, Engineering & Producing, and Mixing & Mastering.

Hal Leonard Books

 

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Posted by Keith Clark on 12/08 at 08:46 AM
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