Article

Friday, April 06, 2012

Allen & Heath iLive Console Chosen To Meet Live Needs Of Multi-Faceted Artist Amber Leigh

With a self-contained operation that played about 265 live dates in 2011, multi-instrumental country music artist Amber Leigh recently acquired an Allen & Heath iLive-T112 digital mixing console with a companion iDR-48 MixRack.

“It’s really important for us to have consistency and to be self-contained, so we needed something we could travel with whether we’re on a bus or an airplane,” says Holland Ryan, the band’s sound engineer. Playing fiddle, guitar and mandolin while blending country, Irish and bluegrass influences into a seamless pop-country sound, Leigh splits time between her south Florida base and Nashville.

Ryan first became familiar with the iLive while mixing at the Old School Square Performing Arts Center in Delray Beach, FL. “In redoing our sound system there, it worked out that the iLive was best for our needs and budget,” he notes. “I had some training on it and learned what a great system it is. So when Amber and I realized that we needed a console we could carry everywhere with us, it was an easy choice.”

With the flexibility afforded him by the iLive system, Ryan handles both the front of house and monitor mixes for the band. “There’s really no reason not to,” he explains. “We’re a five-piece band with 37 inputs, and everyone being on in-ears means the monitor mix will be consistent night after night. With the iLive, it’s almost too easy.”

Leigh adds the artist’s perspective, stating, “With the iLive, each night I step on stage, I know my monitors will sound just as incredible as they did the show before, and that the audience will get to hear the show the way it should sound every time.”

The band’s output requirements fit easily into the architecture of the iDR-48 MixRack. The FOH outputs start with the standard left/right stereo, with frontfill and subwoofer outputs added as needed. Ryan also uses another stereo output to record each show.

On the monitor side, there are stereo outputs for seven IEM mixes (five band members, plus Ryan and a crew tech). “There’s also one extra sub output for the drummer’s butt-kicker, which still leaves us extra outputs for a videographer plus anything else that might happen,” he says.

But beyond the basics of inputs, outputs, size and weight, what Holland Ryan really likes about the Allen & Heath iLive system is its flexibility and sound quality. “I really do love the flexibility of the surface and how easy it is to use,” Ryan states. “And as far as sound quality, it’s really clean and clear, equal to if not better than any other digital desk, even the more expensive ones. In fact, we plan to use it for recording Amber’s next album.”

To accommodate that need, he specified a Dante interface card when purchasing the iLive-T112. This allows a separate split of all inputs to be sent to an outboard recording device. “Obviously, we’ll be using it for live tracking,” he says, “but with the clean sound of the preamps and effects, it just makes sense for studio recording as well. We can’t wait to get started on that.”

Allen & Heath
American Music & Sound

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Wednesday, April 04, 2012

Church Sound: Clarifying The Mysteries Of Equalization

“What does it do?” “When and how do I use it?”

Some of the most frequently asked questions we get have to do with equalization: “What does it do?” “When and how do I use it?”

For whatever reasons, most church sound staffs desperately want to put EQ to the test and into action. Perhaps it’s that adjusting equalization controls provide immediate sonic response — for better or worse, you can instantly hear what you’ve done.

Equalization can be used in different ways, including: musically (that is, to “sweeten” certain frequencies of an instrument/vocal so that it is smoother sounding or more present in the overall mix); correctively to make a voice or instrument sound more like it should when inadequate microphones, acoustics or the instrument/voice itself needs help; or creatively as a special effect to make voice/instruments sound unnatural and otherworldly, very different from the original sound source.

We usually don’t want to create such sounds during services, though, as we want to understand what’s being said, and instrumentation usually plays a supportive role rather than an outrageous-sounding feature role.

In the church, generally, EQ is used to modify a voice/instrument so it is clearer and more representative of the original sound source. But suppose your original sound source is lacking depth or presence? Well, proper equalization can also correct a potential dilemma, such as making a nasally, bottom-heavy or thin voice sound more “natural” and thus, more intelligible.

What is equalization? A simplistic explanation of EQ is essentially the boosting/cutting of specific frequencies so that a sound source sounds more like itself despite acoustic, microphone or instrument or vocal shortcomings.

In essence you are adding or subtracting volume to specific frequencies. You can make people and instruments sound muddy and dark or thin and brittle. You can also by raising the level of specific frequencies create lots of feedback. That’s one major reason why you need to be careful when equalizing sound sources.

Now, of course, in order to properly adjust a voice/instrument via EQ you need to know what that sound source sounds like naturally, and if your microphones happen to boost or cut particular frequencies as part of their design. This is critical to your EQ success: know your sound source.

Most church folks I speak with don’t really know what the guitars, keyboards or choir voices sound like naturally, without sound reinforcement, and, therefore, can not adjust EQ to help solve an audio challenge. They have no starting point of reference. What to do?

Make it a point to rehearse with your choir and praise band. Ask the pastor if you can listen while a sermon is being rehearsed. In short, sit and listen so you learn what everything – voices and instrument – sounds like in the sanctuary or fellowship hall without the support of the sound system. Now you have a reference point.

In addition, learn the frequencies of the church instruments and voices. Equalization controls cover a broad spectrum of frequencies, while most instruments and voices cover narrower spectrums. This means that boosting or cutting certain frequencies of certain sound sources may do nothing, or worse, may create problems, such as feedback. Those frequencies you’re boosting/cutting simply may not exist in that particular sound source.

Don’t, for example, waste your time trying to boost a low-end 200-Hz frequency on a piccolo. It isn’t there. The frequency chart like the one to the left details the frequency ranges musical instruments and the human voice produce. Study the chart (click to enlarge it), make copies and hang it in the sound room for reference. Once you know what instruments and voices are supposed to sound like you’ll be able to makes EQ adjustments quickly and effectively.

Note that some consoles, according to country of origin, will have the “EQ”, “EQ On” or “EQ In” (active) button set differently — on some the EQ will be active when the button is in the Up position (out), while on others EQ will be active when the button is in the Down position (in). Keep your owner’s manual handy. Use the EQ In/Out button to compare what affect your equalization adjustments have created to the overall sound.

Let’s look at just some basic EQ terms and controls common to many consoles:

—Low-Pass Filter: Insert this filter into circuit to allow low frequencies to pass into the sound system. Also called a High Cut Filter.

—High-Pass Filter: Insert to reduce low-frequency rumble (e.g., from heating & air conditioning, moving pianos, footsteps on stage). Essentially shouts out low frequencies below a variable set point and allows frequencies above (higher) that point to pass through. Also called a Low Cut Filter.

—Sweepable EQ: Generally used in the mid frequencies. Set a certain amount of boost/cut and then “sweep” through the frequencies to find the sweet or sour frequency.

—Parametric EQ: Found in more sophisticated mixing consoles parametric equalizers offer more possible frequency adjustments, with separate control for center frequencies, bandwidth (difference between the upper and lower cutoff frequencies) and boost/cut parameters.

Always start with EQ controls set to “out” or off. Always check that EQ is out of the circuit before you start a service or rehearsal. Start each service with EQ “normaled,” which means out of the circuit, and all EQ controls set to off. That way you won’t inadvertently hit the button and send the congregation running for the exits. Generally, the noon or 12 o’clock position (zero) on the knob indicates no cut or boost for that group of frequencies. Turning the knob to the right (plus + side) boosts those frequencies, while turning that knob to the left (minus - side) cuts those frequencies.

It’s always a good idea to “cut before boost.” That is, try to remove an offending frequency rather than boosting a bunch of other frequencies. One example might be to cut out some of the bass frequencies in a muddy sounding voice rather than boosting the mid-range to try and make it cut through the muddiness.

Also, if you can’t hear a particular instrument well enough, rather than boosting 3,000 Hz to make it stand out, try to figure out what other instrument is getting in its way and cut 3,000 Hz out of the offending musician or singer. That way you reduce the potential for feedback in the PA system while increasing intelligibility of the mix… always a good thing.

Equalization can be your friend or foe depending on how much control you take before you use it. Your goal is to use EQ in the church to make instruments/voices sound more intelligible and natural in the environment.

Whether you cut or boost frequencies your adjustments should always support the sound source.

Most mixing boards will have one or two controls that are sweep EQs. That is, they can be tuned to add or subtract any desired frequency within their sweep range.

Now the thing that makes these controls a little tricky is that the two knobs interact. That is, you have to use them both or it will be useless. Don’t understand how they work, then look at the picture below/right.

Note that in addition to the top and bottom knobs marked LOW 80 Hz and HI 12K, there’s also two knobs in the middle with mysterious marking.

The upper middle control is marked with a U in the center and -15 and +15 markings at the counterclockwise and clockwise positions.

The lower middle knob is marked as 100, 200, 800, 2K, and 8K. Those are actually frequencies in Hz (Hertz) or cycles per second. If you refer back to the frequency chart on page 1, you’ll note that the middle A on a piano is 440 Hz, and each octave up doubles the frequency, while each octave below halves the frequency.

So how does it all work? Glad you asked. The HI and LOW controls do just what they say. The 12 o’clock top position marked as U refers to unity gain. That is, at unity gain there is no effect on those frequencies, so whatever sound source you put in comes out exactly the same. British mixing boards usually are labeled as 0 (zero) at the top, which is exactly the same thing. Each one of the notches in the circle is another 3 dB (decibels) of BOOST or CUT.

So if you twist the HI control clockwise to first notch on the right,  which is +3 dB, you’re boosting frequencies around 12 kHz by 3 dB, essentially doubling the high-frequency energy. If you twist it counter-clockwise to the first notch, which is -3 dB just left of the U at the top, you’re cutting the high-frequency energy by 3 dB, or half the power. Each notch will add or subtract 3, 6, 9, 12, or 15 decibels.

You may remember that this equates to 2, 4, 8, 16 and 32 times the original power when going to the right-plus side, or 1/2, 1/4, 1/8, 1/16 and 1/32 of the power when going to the left-minus side.  If you grab the LOW control and turn it to the second notch on the right, that’s 6 dB of boost around 80 Hz and below, which is 4 times the bass power that was sent into the strip from the original sound source. Going to the second notch on the left is cutting the power to 1/4 of the original level. Same goes for the HI control but at the 12 kHz and above frequencies. Seems simple enough, doesn’t it.

As you rotate the HI and LOW controls from center to the far left and right, listen to what the instrument sounds like. Too many highs will make it shrill, while too few highs will make it dull sounding. Too much bass boost will make it boomy, while too little bass will make it sound too thin. 

Here’s a brief list of different frequencies and how they affect our perception of a particular sound.

—Sheen: 10 kHz to 14 kHz

—Sibilance: 7 kHz to 8 kHz

—Presence: 3 kHz to 5 kHz

—Main vocals: 800 to 2,000 Hz

—Nasal honk: 400 to 600 Hz

—Tubbiness: 200 to 400 Hz

—Boom: 80 to 200 Hz

—Deep Bass: 30 to 80 Hz

So if you turn the HI control to the BOOST or CUT position, you’re only affecting the “Sheen” part of that sound, while the LOW control only affects the “Deep Bass” of the sound. The middle sweep EQ can affect everything from boomy sounds around 100 Hz through nasal honk around 500 Hz through sibilance at 7,000 Hz (7 kHz).

Now the real fun begins. The middle two controls are paired together, which you can see from the little boxed lines to the left that connect the controls. The upper Boost/Cut control affects the particular frequency area that’s set by the lower Freq control. It doesn’t do just that one frequency, but rather a haystack looking thing tuned on the frequency of interest.

Below, the left image shows various boost/cut curves if you leave the Freq control centered on 1 kHz, while the right image shows how your Freq control can move the haystack left to right across the spectrum.

So by setting the Boost/Cut to say +6 dB and sweeping the lower Freq control from 100 Hz to 8,000 Hz, you can easily listen to what that EQ control is doing to the sound of that particular instrument. Larger format mixing boards commonly have two of these sweep frequency sections, a total of four controls that adjust boost/cut for low-mid and upper-mid groups of frequencies.

Note that you can essentially re-tune the sound of a snare, making it sound like a big Led Zeppelin snare, by boosting around 1,000 Hz (1 kHz) or a Red Hot Chili Peppers snare by boosting around 3,000 Hz (3 kHz).

You can also try this experiment on bass guitar, cutting around 3 kHz if you want to get the big 70s sound of heavy bass, or perhaps boosting between 4 kHz to 5 kHz for that modern jazz bass sound.

How much boost and cut at the various frequencies is up to you, and subject to much debate. However, know that one method of finding obnoxious frequencies is to simply boost the mid/sweep control and listen for maximum offending sound while sweeping the Freq control, noting the frequency you don’t like. Then simply rotate the cut/boost control counter-clockwise by 6 dB or so, removing what you don’t like.

Another good trick is to eliminate frequencies from the channel that the particular instrument isn’t producing anyway, essentially getting rid of everything below the lowest fundamental frequency on that channel. So go ahead and cut all of the bass out of your flute channel. Any bass that comes in that mic is either wind noise from the flute player or bass leakage from your drummer or bass player. That’s the best way to start cleaning up the mix.   

Refer again to the image on page 1 of this article to see the fundamental frequencies we’re talking about for common instruments. For instance, you can see that boosting the bass control on a violin or flute doesn’t actually boost any bass from that instrument. You’ll only be adding unwanted noise from other instruments that will muddy the mix. The rule “if it doesn’t make a frequency, leave it out” applies.

The high frequencies are a little more complicated since some instruments produce lots of harmonics that add to their musical value. You can hear that while a bass guitar doesn’t go much above 400 Hz with its fundamental tones, there are indeed harmonics 10 times that at 4,000 Hz and up.

It’s up to you and your musical experience to decide just how much or how little of those harmonics to include in the mix. Hey, but that’s what mixing music is all about.

As we like to say “It’s all in the mix”, and it’s true. A great mix and equalization can elevate a good song or sermon to greatness or plunge it to the depths of mediocrity.

The only way to guarantee your best work is to practice, practice, practice.

Hector La Torre and Mike Sokol run the HOW-TO Church Sound Workshops, dedicated to growing ministries through audio education and applied technology through the national, 36-city, annual HOW-TO Church Sound Workshop tour. Find out more here.

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Tuesday, April 03, 2012

Church Sound: The Ten Axioms Of Mixing

Important points to keep in mind when you're up to your neck in knobs and faders

 
It’s easy to get lost in the nuts and bolts of mixing.

Remember these 10 axioms; because when you’re up to your neck in faders and knobs, much of what you are doing comes down to dealing with these 10 points.

1) There will occasionally be someone who thinks it’s too loud, no matter how carefully you watch your volume levels. Don’t take this as a sign of failure. Every person has their own idea of a proper volume level. 

Factoring in their age, personal music preferences, and views on church worship, it’s no wonder there will be people that disagree with the volume.

2) Feedback kills the mood. No matter how well you mix a band, or the entire service for that matter, a nasty feedback episode breaks the mood of the event. The pastor and the musicians want a specific mood established. 

In fact, after a service, a feedback episode might be the only thing the congregation remembers. Taking the proper precautions during setup, you should avoid 99.99 percent of feedback problems.

3) Missing a cue can kill the mood. Mixing is more than using technology to produce a great sound. Part of producing that great sound is paying attention to more than what you hear. You must watch the stage and the service schedule for what is going to happen next. 

As a tip, if you do miss a mic cue, turn the fader all the way down, turn the channel on, and then raise the volume. It’s better than the sudden jarring of a loud voice because the person on stage might try talking louder to compensate.

4) You aren’t mixing for a recording. You can easily spend every moment tweaking the EQ and effects. The congregation isn’t there to hear a recording; they are there to worship and sing. They won’t compliment you on an exceptional mix. 

Focus on getting the sound right during practice and then during the first song of the service. Anything else should only be done to significantly improve the mix to the degree in which the congregation would notice.

5) Vocals rule the day. The congregation listens to vocals for two reasons; follow along when singing and listing when they aren’t singing. Therefore, treat them in your mix as the most important element the congregation should clearly hear. 

You can reduce the lead vocal a little when the whole congregation is worshiping to a well-known song. This creates a more unified sound in the sanctuary.

6) Proper gain structure is the key to creating a great mix. This provides less noise in the channel (signal noise) and provides the best volume level control when using the faders. It also keeps feedback at bay.

If you don’t have the proper gain structure, you’re in for a lot of headaches.

7) Proper microphone usage is essential to good mixing. Sound coming into the mixer needs to be as pure as possible. Therefore, placing microphones as close to the source as possible (both instruments and vocals) is of utmost importance. 

Close mic’ing can provide a great deal of sound isolation and keep away feedback issues.

8) Mixing loud is not the same as mixing soft. Sounds that occur outside of the mid-range frequencies aren’t heard as loudly as those that are. That’s why a song cranked on the stereo might sound better than if it was turned down low. 

Therefore, remember when mixing a softer song to adjust your EQ and volume settings for instruments that are in those high/low frequencies.

9) You must confine the individual sounds to a smaller sonic space with a greater the number of instruments and performers. Likewise, the fewer number of performers, the greater sonic space they should occupy.  Part of mixing is spreading the sounds across the sonic space so the mixed sound is full and deep. 

You can’t spread out nine instruments on top of each other and make it sound great.

10) What would you add?

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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Monday, April 02, 2012

Carnival Cruise Lines Opts For Soundcraft Si Compact Digital Consoles

Carnival Cruise Lines utilizes Soundcraft Si Compact digital consoles for entertainment systems aboard its fleet of 12 ships, including two with Si Compact 16 consoles, and 10 with an Si Compact 24. There is an accompanying stage box for each console on board.

Erik Zvanut of Dobbs-Stanford, Soundcraft’s regional representative for Florida, suggested the Si Compact series because the layout is similar to an analog board and would be easy for the onboard staff to learn.

“Once I showed everyone the Si Compact series they fell in love with it pretty quickly,” Zvanut says. “The ease of use and the amount of functionality you get with the board is so much for the compact size.”

On each ship, the boards are mostly used in the theater and nightclub space, which offers live band karaoke. “The fact that the board is small and can be transported easily is huge. There isn’t much extra room on each ship, so having the small desk makes it easy to fit in a corner wherever it is needed. Carnival also wanted a desk that was easy to set up right out of the box without the need to retrain staff members,” states Zvanut.

“The price point was perfect. Since the musicians usually operate the sound systems, it’s great to have an easy-to-use board that doesn’t need much background training,” says James Keaton, entertainment audio supervisor for Carnival Cruise Lines. “With a few of the vessels being older, it also helped to connect the stage boxes with CAT5 cable. Much of the copper wire is out of shape, and having the ease of CAT5 allows us to utilize many more inputs and outputs for whatever we may need. We tried to find a universal product that would benefit everyone, and the Si Compact consoles do the trick.”

Many of the settings and features of the Si Compact series benefit Carnival Cruise lines on a daily basis. “Many times we have a group of rotating bands that perform on different vessels,” adds Keaton. “Having the ability to save settings and presets for each group makes the transition between acts very easy. Whoever is operating the desk just needs to upload the settings and they are set to go.”

The addition of the Si Compact consoles is the start of a long-term upgrade to the entire system onboard the vessels. “It’s great that the stage boxes are compatible with the large Vi series consoles. It gives us options to expand in the future, which is our goal,” Keaton says.

Soundcraft
Harman

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In The Studio: Tips For Mixing Rap Vocals (Includes Audio)

An approach to conceptualizing a vocal treatment

 

If I had to pick the most frequent question I get asked on a regular basis - it would have to be “how do I mix rap vocals?” Or some variation thereof. At least once a week, if not more often.

I mix a new rap vocal four or five times a week - much more if you count different rappers on the same song. I have developed an approach - sort of a formula to create a formula.

In truth, we know that all songs, vocals, captures, and performances are different. There can never be one formula to mix all vocals effectively.

And there are many approaches to conceptualizing a vocal treatment - mine is one of many.

The Concept

It all starts with the concept. I say this time and time again, and it only gets more true as I say it - in order to mix anything - you need an end game. There has to be some kind of idea of where the vocal is going to go before you start getting it there. That idea can and probably will change along the way, but there has to be some direction or else why do anything at all.

The big problem most people have with mixing rap vocals is that they think of the word “vocals” without considering the word “rap.” Rap is supremely general - there are big differences between 1994 NY style rap vocals, and 2010 LA style rap vocals.

Even within that you have A Tribe Called Quest - “1nce Again” vs. LL Cool J - “Loungin’”. Both are laid back smoother rap songs, but the mixing is totally different. (compare below)

Loungin’ is a quintessential Bad Boy style sound, mixed by Rich Travali - you can hear the similarities between that and 112, Total, Mariah Carey and later Biggie tracks.

1nce Again is a prime example of a Bob Power mix - a sound which pretty much dominated early NY rap.

I bring up this distinction because I hope you’ll compare the two. Notice how in Loungin’ the vocals are up in the mix - level with the snare - and have a “shiny” and smooth top end, great clarity and a really open yet detailed upper midrange.

Meanwhile in 1nce Again, the vocals are just under the snare and have an extremely forward and aggressive mid-range, and a grittier rolled off top end, and a steep hi-pass filter on the low end. The shape of the vocal is also different - the compression is much easier on Loungin, and again, very aggressive on 1nce Again (particularly Phife’s voice).

Let’s take a more modern track, say Nicki Minaj’s “Massive Attack.”

Here you have super clear presence and treble in the vocals, the vocals are up in the mix, and there isn’t as much lower mid range as say “Loungin.”

Each of the 3 examples does something very specific:

   
• “1nce Again” is edgy and aggressive sounding - quintessential to the early NY sound, and rap’s image at the time.
   
• “Loungin’” is very intimate and smooth - it’s almost like an R&B song sonically.
   
• “Massive Attack” has the vocals clear as crystal, but leaves plenty of room for the low range drums to dominate the mix - which is good for clubs.

The point is, the what and why are just as important as the how when it comes to mixing vocals. Who is the artist’s audience, what is the artist’s style, where is the song being played - and what can you as the engineer do to encapsulate that?

So you’ve determined what you want… but how do you get there?

The Cleanup

Before mixing, many rap vocals need a little cleaning. Common issues are - the vocals were recorded in an unideal location, such as a closet (I get that one all the time) or a bathroom. I know it sounds weird but the myth has gone around that recording in a closet or a bathroom is a good idea. Generally speaking, it’s not.

The other common issue - the vocals were recorded too hot. Again, a myth has seemed to perpetuate that it’s a good idea to record the signal as loudly as possibly. This is totally untrue, particularly in the age of 24-bit audio.

Cleaning up is a little rough at times - because the scope of what you can do is limited. For audio that came in too hot - ie, is clipping, distortion removal software such as iZotope’s Rx De-Clipper is ideal. Also, that distortion will create frequency center resonances, which can be eased off with an EQ.

For vocals tracked in a reverberant space, subtle gating, and careful EQ can suppress the room sound - or you can use software like SPL De-Verb. The other option is to mix the track in a way that makes the reverb appear deliberate.

For vocals tracked in closets or corners, the issue will be comb filtering. One trick for easing off comb filtering - if there are doubles of the vocal, pitch shift them up or down a slight amount. This will change the frequency bands that are being filtered, so that when layered with the main vocal, the same bands will not be missing all across the board. The backups will “fill in” the missing bands. The comb filtering will still be there, but it won’t be as readily apparent.

Processing

Now you have the vocals clean (or maybe they came in clean to begin with). It’s time to decide what to do with them. Now, I can’t write how you should or should not process your vocals, but I can give some insight into things to consider and think about.

Balance

Figuring out the relationship between the vocals and other instruments in the same frequency area is extremely important. Quintessentially, Hip Hop is all about the relationship between the vocals and the drums - and the number one contestant with the voice is the snare. Finding a way to make both the vocals and the snare prominent without stepping on each other will make the rest of the mix fall nicely into place.

In “1nce Again,” you’ll notice that the snare is a little louder than the vocals, and seems to be concentrated into the brighter area of the frequency spectrum, while the vocals are just an inch down, and living more in the mid range. This was a conscious decision made in the mix. But mixes like Loungin’ have the vocals on par with the snare. And Massive Attack has the vocals up - but it’s not really a snare, it’s a percussive instrument holding down the 2 and 4 that lives primarily in the lower mid region.

“Air”

Hip Hop vocals generally do not have much in the way of reverb. There are three reason for this primarily. 1) Rap vocals tend to move faster and hold more of a rhythmic function than sung vocals - and long reverb tails can blur the rhythm and articulation. (2) The idea of Hip Hop is to be “up front and in your face”, where reverb tends to sink things back in the stereo field. (3) Everyone else is mixing their vocals that way. Not a good reason, but kind of true.

However, vocals usually do benefit from sense of 3-D sculpting, or “air.” A sense of space around the vocals that make them more lively and vivid. Very short, wide, quiet reverb can really do the trick here. Another good thing to try is using delay (echo) - and pushing the delay way in the background, with a lot of high end rolled off of it. This creates the sense of a very deep three dimensional space, which by contrast makes the vocal seem even more forward.

Lastly, if you are in a good tracking situation, carefully bringing out the natural space of the tracking room can be a good way to get super dry vocals with a sense of air around them. Compression with a very slow attack, and relatively quick release, and a boost to the super-treble range can often bring out the natural air.

Shape & Consistency

A little compression is often nice on vocals, just to sit them into a mix and add a little tone. On a sparse mix, a little dab’ll do ya. The most common mistake people make when processing vocals for Hip Hop is to over-compress. High levels of compression is really only beneficial to a mix when there is a lot of stuff fighting for sonic space. When you read about rapper’s vocals going through four compressors and really getting squeezed it’s probably because there are tons of things already going on in the mix, and the compression is necessary for the vocals to cut through. Or because it’s a stylistic choice to really crunch the vocals.

Filtering

What’s going on around the voice is just as important to the vocals as the vocals themselves. Carefully picking what to get rid of to help the vocals along is very important. For example, most engineers hi-pass filter almost everything except the kick and bass. That clears up room for the low information.

But often the importance of low-pass filtering is overlooked. Synths, even bass synths, can have a lot of high end information that is just not necessary to the mix and leave the “air” range around the vocals feeling choked. A couple of well placed low-passes could very well bring your vocals to life.

Also, back to the subject of hi-passing, unless you are doing the heavy handed Bob Power thing, you really don’t need to be hard hi-passing your vocals at 120 Hz. The human voice, male and female, has chest resonance that goes down to 80 Hz (and even under sometimes). Try a gentle hi-pass at around 70 or 80 Hz to start with if you’re clearing up the vocals. Or maybe no hi-pass at all…

Presence

Deciding where the vocal lives frequency wise is important. Mid sounding, “telephonic” vocals can be cool at times, low mid “warm” sounding vocals certainly have their place. Commonly, the practice is to hype the natural presence of the vocals by getting rid of the “throat” tones and proximity build up which generally live around the 250-600 Hz range (but don’t mix by numbers, listen listen listen). This in turn exaggerates the chest sound, and the head sound - particularly the sounds that form at the front of the mouth, tongue, and teeth - these are the tones that we use to pronounce our words and generally live in the upper midrange (2-5 kHz, no numbers, listen listen listen).

Matthew Weiss records, mixes, and masters music in the Philadelphia, New York, and Boston areas. Find out more about him here.

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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Thursday, March 29, 2012

ACIR Professional To Host Yamaha CL Demo At Trump Taj Mahal In April

ACIR Professional will host a demo of the new Yamaha CL Digital Console Series on Thursday, April 5 from Noon to 4 pm at the Trump Taj Mahal (Atlantic City), Xanadu Theatre.

Yamaha systems application engineer Kevin Kimmel and regional district manager Bob Quinones will be on hand to demo the new CL console and will also present the new Rupert Neve Designs Portico 5045 Primary Source Enhancer and Dugan-MY16 card.

All audio professionals are welcome. RSVP to Lisa Young via e-mail (.(JavaScript must be enabled to view this email address)) by April 2, 2012.

Yamaha CL Series Digital Console Series: The Yamaha CL Series is a Dante network-based console featuring remote I/O for a faster, more responsive Yamaha system solution. All three CL models in the Centralogic series, only differentiated by frame size and input capability, feature 24 mix buses, 8 matrix buses, plus stereo and mono outputs, and 16 DCAs.

The footprint of all three CL consoles is small, yet powerful and has been developed specifically for sound reinforcement applications such as performing arts venues, theaters, houses of worship, touring, and remote broadcast. The high output bus count will be a great benefit to live broadcast and monitor applications.

Read more about the new CL Series here.

Rupert Neve Designs Portico 5045: The new Rupert Neve Designs Portico 5045 Primary Source Enhancer has been developed for, and is distributed exclusively by, Yamaha Commercial Audio Systems, Inc. Primarily used in live sound applications, the RND 5045 is a 2-channel analog, 1RU device that not only enables up to 16 dB of additional gain before feedback when conditions allow, but because it is a dynamics-only processor, the integrity of the original signal is preserved.

The device reduces background sounds and enhances the main audio source, providing clarity and warmth to podium, lavaliere, and headset mics, as well as in extreme situations, such as referee mics.

Dugan-MY16 Card: The Dugan-MY16 card is designed for current model Yamaha digital mixers and processors including Yamaha 01V96, DM1000, 02R96, DM2000, M7CL, LS9, DSP5D, PM5D, and DME24/64N. Dan Dugan Sound Design automatic mic mixing products eliminate cueing errors, reduce feedback and ambient noise pickup, allow for smooth transitions between talkers, and provide consistent system gain no matter how many mics are open. The card can handle up to 64 live microphones, and is remote controllable.

Yamaha

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Wednesday, March 28, 2012

Innovason Takes The Lid Off Pandora

 
Innovason debuted the flagship Eclipse GT digital console: named for the “all-gifted” figure of Greek mythology, Pandora is a new way of panning that provides audibly better results across the listening field than traditional stereo panning.

Tonmeister and renowned classical sound engineer, Carsten Kummel, explains: “The problem with traditional panning is the loss of signal on one side when you pan to the other. To resolve this, some people choose to mix in mono, but this isn’t really a viable solution, especially for complex shows where you are dealing with anything from 80 to 100 channels. So, I started playing around with delay and seeing if I could achieve better results that way.”

“I went away and did a whole series of testing and taking measurements and the results were astonishing – by using the tools available to us to reproduce the way in which the ear works naturally, we opened up a whole new world of possibilities,” continued Carsten. “I had been achieving similar results myself by using the functions that were available to me on the Eclipse, but it was just a way of working, and not an integrated feature of the desk.”

“At this point, it was time to consult Hervé de Caro, the godfather of Eclipse – when I discussed my findings with him, he said straight away that it was something he wanted to incorporate properly into the console. After some initial experimentation, being the genius that he is, Hervé and his team had developed a working prototype algorithm with the function available on a single pan pot in just two days!”

Hervé de Caro takes up the story. “This is an absolutely amazing function, even if I say so myself,” he says. “The world has been waiting for Pandora, and finally, thanks to our collaboration with Carsten, it’s here. Carsten has been slowly developing the idea over the last three years and testing it in real-world situations in the work that he does daily with major symphony orchestras across the world.”

“Without going into the technicalities, the difference it makes to the listener is amazing,” declares Hervé. “We have finally found a way to pan a signal from full left to full right to create the feeling of a stereo image for the entire audience, no matter where you are sat, WITHOUT losing any information for those sat at the extremes, and no loss of sound quality for those sat in the middle.”

“On the contrary, in fact - because we are reproducing the natural function of the ear, it actually sounds better. In practice this means that if you are sat at the extreme left of the auditorium whilst the signal has been panned to the extreme right, you can still hear it from the left, even though the sound image has moved to the right, and vice-versa.”

“With traditional panning, you would hear nothing from the left, and the sound from the right would sound very far away – in fact, you’d be losing half of the audio information. Even those in the middle would experience a loss of level and audio quality. Pandora puts an end to all that – you can now hear everything all of the time and STILL benefit from a true stereo image. How cool is that?!”

Innovason

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Thursday, March 22, 2012

New Horizons For Studiomaster At Prolight+Sound

 
Studiomaster is exhibiting the new Horizon 2012 in final production form for the first time at Prolight+Sound2012; introducing a new benchmark in power and portability, performance, quality, and value for money.

The Horizon 2012, resurrects the legend of the company’s world beating Horizon series mixers of the 1990s. A totally new design redefines the portable powered mixer and provides a new class of high build quality, high power and performance product. The mixer is unbelievable light weight and is carried on a single shoulder strap.

The new ergonomic form factor provides for the greatest possible strength and protection, and a fully professional control surface that is clear and uncluttered, while innovatively integrating a staggering 2000 watts of power from Class-D amplification. The design also incorporates a removable control cover and a retractable panel to angle the mixer at the best working position. Designed for the quickest possible set up and go, the mixer nevertheless offers a level of operational versatility and features that provides new sound reinforcement capabilities for portable PA.

The 12 input mixer can be used free standing or rackmounted. All mic/line input channels feature the famed VMS optical compressor, legendary Studiomaster 3-band EQ, with sweepable mid, and a total of four AUX sends. 60mm smooth faders, Mute, PFL buttons, and SIGNAL and PEAK LEDs complete the channel strip. Like the original Horizon powered mixers, a combined stereo and mic input channel effectively adds three more mic channels.

In line with standard Studiomaster practice, the custom designed twin FX features real studio quality reverbs and delays instead of the poor quality 128 program DSP found on so many powered mixers; the focus of the FX section is on enabling the user to achieve the desired effect with the highest quality and as quickly as possible. A 2-channel USB audio interface includes assignable signal source; playback from a computer can be routed to the main MIX output, or to a stereo channel giving access to EQ and auxiliaries. The output to the computer can be from the main MIX, to record a performance, or from the DSP effect sends to make use of plug-in FX.

Each amplifier channel supplies a colossal 1000 watts into 4 ohms using Class-D topology; stable into all loads with temperature & short circuit protection. Comprehensive three stage amp routing is selectable between normal stereo operation, split function (MIX to the left and Auxiliary to the right), or Monitor only. Stereo 9-band graphic equalisers feed the MIX output, which is routed to the amplifiers and the balanced XLR outputs. A second MIX2 output duplicates this signal.

Also located here is a switchable 2-way 150Hz 18dB/oct crossover. When selected, the main MIX only receives frequencies above 150Hz whilst the MIX 2 output gets everything below 150HZ – ideal for connecting additional subs to a system. An Amp Power switch reduces power in three stages down to 15%, providing high fader resolution in low volume applications. The new Horizon boasts two sets of led meters – one for the mixer and one for amplifier level.

Studiomaster

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Wednesday, March 21, 2012

Soundcraft Announces ViSi iPad Remote For Digital Console Range; Chance To Win An iPad

 
Prolight & Sound 2012 marks the unveiling and preview of the new Soundcraft ViSi Remote iPad app, which enables a single iPad to control multiple consoles from the Soundcraft Vi Series and Soundcraft Si Compact range on one wireless network.

The Soundcraft ViSi Remote app allows users to roam around in a venue and not only adjust input channel levels and mutes, but also adjust aux send levels and matrix sends, and level and graphic EQ settings on bus outputs. 

The app uses Harman HiQnet architecture to connect a network of consoles to a wireless router, which communicates with the iPad.

For even more flexibility, multiple iPads may be used on the same network, even on the same console, so that individual artists could control their own monitor sends, for example.

The Soundcraft ViSi remote is expected to be available from the iTunes store at the end of April 2012.

By pre-registering on the Soundcraft website, users will not only be informed immediately when the app goes live, but will be entered into a prize draw to win an Apple iPad. (conditions apply, see website here for details).

Soundcraft
Harman

 

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DiGiCo Introduces The New SD5 Live Audio Mixing Desk at Prolight+Sound 2012

 
At the ongoing Prolight+Sound 2012 show in Frankfurt, DiGiCo has just unveiled the the SD5 live audio mixing desk, another new option that fits in terms of scope and scale between the SD7 and SD10 in the product line.

A decade after the launch of the D5 Live, the SD5 fits directly into the D5’s shoes, but benefits from the advancements made possible by DiGiCo’s proprietary Stealth Digital Processing

The SD5 offers a low noise, heat dissipation work surface benefiting from Hidden-til-lit (HTL) technology, yet its five digitally driven full color TFT LCD screens, three of which are touch sensitive, have a new configuration that allows easy access to single or multiple users.

There are also two interactive dynamic metering displays (IDM)) and instant access ‘quick select’ buttons which are positioned conveniently down the left side of the two channel screens for fast and intuitive navigation.

Incorporating the master screen into the work surface meterbridge design has allowed for complete user feedback, but maintained a lower profile meter bridge. This still allows clear visibility of those on stage for the user, with everything in close reach to the mix position.

As with all SD range consoles, the SD5’s headroom, dynamic range and audio quality are of paramount importance.

As standard, the SD5 comes with a 2 GB fiber optic system, which is capable of running 448 channels of I/O at 96 kHz, plus 56 console-to-console tie lines, allowing connection to up to 14 of the SD-Series racks.

There are three redundant MADI ports and local I/O includes eight microphone inputs, eight line outputs and eight AES I/O (mono).

The SD5 has 124 input channels; 56 configurable buses, plus up to 5.1 master; a 24 x 24 fixed matrix; DiGiTubes on every channel, buss and output; 24 assignable dynamic EQ; 24 multiband compressors; 24 stereo effects; 32 Graphic EQ; 10 x 4 (40) RGB backlit macro buttons; plus the ability to add a Waves upgrade.

Main Features:

—127 Input Channels
 
—56 Configurable Buses Plus Master Bus

—24 Dynamic Equalizers

—DiGiTubes on All Channels

—24 Multiband Compressors

—24 Digital FX

—32 Graphic Equalizers

—24 x 24 Matrix

—Multi Channel Folding

DiGiCo

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Tuesday, March 20, 2012

Church Sound: How To Power Your Mix With Musical Energy

The difference between a song with energy and one without is significant

 
Kent Morris said, “live music is about the energy.“ 

That’s so true. Where does that energy come from? How do you control it?  And why can it sometimes be so hard to find? 

All questions that will soon be answered.

Phil Coulter, on his Highland Cathedral album, has a song that’s dedicated to the drums. It starts out with a few lines of verse that highlights the importance of the drums since before the first lyrics were ever written. Drums have been giving energy to music for a long time. But they aren’t the only source.

Not all musical energy comes through the drums. For some people, they don’t like the drums at all.

Therefore, a great way to find out what’s best for powering your mix with musical energy is by learning where it comes from, how people respond to it, and how to maximize it in your mix.

Learning how to mix using musical energy comes primarily through:

Listening to powerful music and analyzing it. Listen to songs you think are powerful. These might be songs that inspire you, songs that you can’t help but sing or tap your foot. Listen to the songs and ask these questions:

—What instrument(s), if removed from the song, would drain the song of its power?

—What instruments could be taken out and the song still has energy?

—Is there anything in the vocal delivery that carries more energy than the instruments?

—Why does a vocal line create power when all the instruments stop playing (i.e., accapella on the 2nd chorus)?

Listening to live music (as in “be there” not as in “listing to a recording of live music.“)  Ask yourself these questions;

—At what point could you not help but sing or sing louder?

—What instrument(s) gave the song power?

—What part of the mix arrangement gave power (ex. reducing instrument level and pushing vocals in a passage)?

—Don’t mistake your emotional feelings for energy in that if a song ends with an acappella chorus and everyone is singing and you are “feeling it,” then it’s not because of what’s happening but what happened prior to that point.

Knowing your audience. Not all audiences like the same type of music.

Drums for some, bass for others, and piano for other. Some might focus on the vocals.

These can all depend on the denomination, the average age, the preferred style for the church (think mandate).

You can’t promote energy through a bass line if your audience doesn’t like the sound.

Reacting to your audience. Much the same way you should control your mix volume based on your audience, you should do the same with your mix.

If you think maximizing the energy comes from pushing the volume but your audience sings softer, stops singing, or stops moving around, then you haven’t opted for the best way to maximize the energy.

Application

There are a few ways you can bring energy through your mix:

Increase volume. This would be the easiest one to overdo. Raising the volume of the source(s) of energy can work but shouldn’t be the default method. Consider the others.

Make room in the mix. Clarify that instrument in the mix. Make the instrument/vocal stand out in the mix by cutting the frequencies of other instrument that share some of those frequencies. You can boost frequencies for that energized line but start with cutting others first.

Decrease volume. Decrease the volume of the other instruments/vocals that aren’t promoting that energy.

Remember your audience. If they aren’t the drum-energized crowd, then don’t think they will be energized by the drums. Look to a different instrument. A rhythm guitar is a good default but learn to experiment. It could be the shaker drives the song.

You can also work with the musicians. You don’t want to push a particular instrument if they want the focus to be on another instrument. Talk with them and see what they think gives power to their songs.

Summary

Using the above methods, you can bring more energy to your songs. The difference between a song with energy and one without is significant.

A song without that energy sounds uninspired. Give your mix energy!

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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Friday, March 16, 2012

Audio Basics: Stage Monitoring Simplified

Sage advice on mixing monitors and the house simultaneously.

Encompassed in today’s live show are several individual shows, for example, light shows, laser shows and more.

It’s not uncommon for each of these independent shows to have its own set of engineers.

The focal point is the band/artist, and they’re the reason all of the other shows are taking place.

And, it should be pointed out that most of these “shows within the show” are presented for the audience.

Good engineers realize that the monitor mix is a show in itself. It’s the only show that the musicians get to hear, and it certainly is the mix that most affects their performance.

Big shows usually have a separate engineer for the monitor mix, but for average shows one valuable individual functions as both the front of house (FOH) and monitor engineer.

To accomplish a good monitor mix, you must understand your particular mixing console; we can, however, examine some basic principles that apply unilaterally.

While working your magic behind the FOH console, most people don’t realize the work you’re doing for the musicians.

The musicians also have to hear the performance, except they usually want to hear something totally different than the house mix.

In fact, many bands have members that each want (or even require) a different mix than another band member. That means multiple mixes, all running at the same time, which can present some challenges to the engineer, because all of these mixes cannot be monitored simultaneously.

Most consoles allow toggling between each of the mixes, allowing you to make changes to each individually. This, of course, depends on what features the console has and how you, as an engineer, decide to accomplish your monitor mix.

Let’s first take a look at some obstacles you may encounter in your equipment.

Snakes, Sends, & Returns
The snake, of course, is the multiple input cable that all of the instruments plug into on the stage. The cable then is plugged into the console to carry the signal from the stage to the mixing console.

The snake also has “returns,” which, as the name implies, route signal from the console back to the stage and musicians.

The amount of returns available will directly affect the amount of signal that can be routed back to the stage. For example, a 24 x 4 snake offers up to 24 pathways to the console from the stage, and four pathways back to the stage from the console.

With four return paths, the possible combinations of mixes are four independent mono mixes, two stereo mixes or one stereo mix and two mono mixes.

The more return paths, the more possibilities you have to run monitor mixes.

Bear in mind that these return paths also must carry the signal to the main loudspeakers.

Another limitation that can be encountered is the amount of pathways that a mixing console has to use as returns to the stage.

A 24 x 4 x 2 console has 24 inputs for instruments, four bus (or group) outputs, and a pair of outputs for the mains.

Bus or group outputs (sometimes called sub-outs) can be used as monitor return outputs.

Four-bus outputs would yield the same combination of possibilities as a four-return snake.

Obviously, more bus outputs equals more possibilities for this type of monitor mix.

Probably the method with the most possibilities is routing monitor mixes with the auxiliary sends. Like buses, auxiliary or aux sends can be used to route monitor mixes.

Although aux sends are used for routing signal to effects processors, they are very useful in running monitor mixes. For aux sends to be useful as monitor mixes, they must be able to be used in what is known as pre-fader mode.

There is usually a button next to the aux send pot on each channel that will allow you to switch between pre and post-fader modes. Keep in mind that for each mono return path, a separate amp at the other end for a power source is needed.

Stereo requires a two-channel power amplifier, or a separate amp for the left and right sides.

Have you checked your equipment for features? To make it easier to describe some basic techniques, we need a typical scenario.

Let’s assume we have a 24-channel input console with four bus outputs and at least four auxiliary sends. Our console is also equipped with a stereo headphone output, so we can listen to each mix separately without listening to (or affecting) the house mix.

Most consoles possess this capability because it is necessary for the engineer to listen to alternate mixes during a performance.

Our snake has 24 channels with four return paths. Let’s also assume that we will be running a stereo house mix. To achieve a good monitor mix, there are several ways to get there. So, let’s take the trip.

Get On The Bus
Each channel has a feature that will allow assignment of its signal to a group or bus out. These assignment groups include the main L/R bus.

The main L/R group will be used to route signal from the console’s main output jacks on the back of the board through the snake’s first two returns. The faders on the console labeled “mains” or “L/R” (or something similar) will control the amount of signal.

The returns from the snake on the stage side will be routed to the stereo power amp (or amps) that powers the main front of house speakers. This way, the signal from the power amps takes the shortest path to the speakers.

In addition to the main L/R outputs, a four-bus console has four additional outputs that correspond to faders that control the signal routed to them.

The faders are typically labeled “bus 1,2,3,4” or “Sub out 1,2,3,4” etc.

Each channel has a switch, which allows you to assign its signal to a particular bus.

With this configuration, we could run a stereo monitor mix.

If we decided to do this, we would assign each channel to buses 1 and 2.

The output of bus 1 will be routed through the snake’s return 3, and the output of bus 2 output will be routed through return 4.

Remember we’re using returns 1 and 2 for the main house mix. Return 3 will be routed to the amp that powers the left side of the monitor mix and return 4 to the right side.

I only recommend doing a stereo monitor mix if in-ear monitors are to be used. Most applications require the use of on stage loudspeakers (monitors or wedges).

For this reason, it is a good idea to run a mono monitor mix. Monitors on stage would normally be too complex to run a stereo monitor mix. By assigning all of the instruments to bus 1, we can be sure that all of the stage speakers represent an overall mix for all of the musicians on stage.

Because we are only using bus 1 for this mix, we could then use bus 2 for some accent monitors on stage.

For example, if all of the musicians can hear the overall mix, we could assign only the vocals and other lead instruments to bus 2 and route that bus to spot monitors for singers (who like to hear themselves heavily in the monitor mix) as well as other lead instrumentalists who cue off of each other.

If we had more returns in our snake we could even provide accent monitors for the bassist and drummer/percussionist, for example, by using buses 3 and 4.

One limitation of using buses for your monitor mix is that if you assign a particular channel to a bus, you might not be able to route that signal to another bus or at least not the bus you need.

Another limitation is that on some consoles, you cannot adjust the amount of signal that each channel contributes to each bus. That means that whenever the bus is assigned, the entire signal from that channel is routed to that bus assignment.

The amount of individual signal from each channel to the bus is determined by the position of the channel fader.

Many manufacturers design features that allow for more possibilities; however, that usually means more circuitry, hardware and board space for additional faders or pots.

More features means more expensive. This is a quick and easy way to run monitors, but there are other avenues that lead out of the console - the auxiliary send outputs (aux sends).

Working The Aux Sends
In addition to bus output jacks, mixing consoles also have output jacks that correspond to their auxiliary send pots.

Remember that our sample console has at least 4 auxiliary sends.

Each one of these channels has a level adjustment pot for each of the sends.

The console will almost certainly have an aux master section that will allow for control of the overall signal level of each aux output.

The aux 1 output jack should be routed to the snake’s return 3. Aux 2 out should be routed to the snake’s return 4.

We still need to use returns 1 and 2 for the house mix. By utilizing the aux sends, we can accomplish two mono mixes. Each of these mixes can contain any amount of each channel.

For example, if we want more of channel 9’s audio in the aux 1 mix, we simply turn up the aux 1 pot on channel 9.

By using the level adjustments for aux sends 1 and 2 on each channel, we can create a mix that is suitable for the lead instruments on aux 1 and a completely separate mix that is suitable for the rhythm section on aux 2.

One feature that our mixing console must incorporate is that ability to route the sends in “pre-fader” mode. In pre-fader mode the position of the volume faders on each channel has no effect on the level being sent down each of the aux sends.

This means that if we turn down the lead guitar in the house mix by lowering the channel’s volume fader, the monitor mix will not be changed. Once again, if our snake had more return paths, we could utilize additional sends for more on stage monitoring possibilities.

By assigning the aux mixes one at a time to the headphone outputs, you can make adjustments to each mix without affecting what the audience hears.

In addition, by routing the signal outputs of our effects processors to channel inputs, we can send some effects to each of the monitor mixes.

For example, by routing the outputs of a reverb processor to the inputs of channels 23 and 24 of our mixing console, we can send an adjustable level of reverb to each of the two monitor mixes by simply turning up sends 1 and 2 on those channels.

Be cautious of routing signal from channels 23 and 24 back to the inputs of the reverb unit as this will result in an electronic feedback loop.

If you use send 3 and 4 for the inputs to your effects processors while using channels 23 and 24 for your reverb returns, turning up sends 3 and/or 4 on these channels will create this kind of loop.

Using aux sends for monitor mixing is probably the best and most popular approach as it affords the engineer the most versatility and functionality.

Hopefully, this article will arm you with some of the knowledge critical in implementing some basic rules of audio. Always make sure you know the equipment you own and the equipment you intend to buy. Good luck, and have fun!

 
Scott Foulkrod is the Audiovisual Coordinator for the Houston Rockets in Houston, TX.

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Thursday, March 15, 2012

In The Studio: Tips For Controlling Vocal Sibilance

Keeping the problem from becoming a musical distraction

 
Vocal sibilance is an unpleasant tonal harshness that can happen during consonant syllables (like S, T, and Z), caused by disproportionate audio dynamics in upper midrange frequencies. Sibilance is often centered between 5 kHz to 8 kHz, but can occur well above that frequency range.

This problem is usually caused by the actual vocal formant, but can also be exaggerated by microphone placement and technique. This article will discuss some ways to control vocal sibilance, and keep the problem from becoming a musical distraction.

Sibilance at the Source
(best read with sibilant whistle)

In phonetic terms, sibilance comes from a type of vocal formant called a fricative consonant. During these sorts of utterances, the airway (usually the mouth) is drastically constricted by two anatomical features, like the teeth, tongue, or palette.

This pressurization causes some amount of noise that forms the consonant sounds we would recognize from a phase like, “Sally sits sideways on the tennis trolley.” Sibilance is a very necessary feature of human speech, but when there’s (subjectively) too much noise created during these consonants, we get a very distracting harshness.

It isn’t really practical or productive to address micro-muscular vocal technique during a session, so your best bet to mitigate sibilance at the source is microphone selection and placement. Here are a few suggestions:

—Every vocalist is remarkably different, so don’t pre-suppose that anything you’ve tried before will or will not work again.
—Be sure to leave some space between your vocalist and the microphone. Twelve to eighteen inches would be a nice starting point.
—A pop filter won’t do anything to help with sibilance.
—Once you find a microphone and distance combination that helps, try angling the microphone downward 10 to 15 degrees to place the 0-degree axis toward the throat instead of the sibilant source.

Audio Dynamics Processing

Vocal sibilance is a phenomenon of disproportionate dynamics within an isolated frequency range. In other words, it is a problem of too much loudness contrast within a small frequency range of a waveform that has a dynamic profile of its own.

‘De-essing’ is the classic compressor technique used to address vocal sibilance through processing. In fact, de-essing is just one example of many uses for compression that is conditioned on a limited frequency band, or a modified harmonic profile.

De-esser Signal Flow

Audio dynamics processors like compressors and expanders contain two signal paths:

1) The audio path, which is subject to conditional gain reduction and;

2) The sidechain or ‘key’ path, which the gain reduction is conditioned on.

In short, gain reduction happens (or not) in the audio path based on the interaction between the sidechain signal and the detector settings (i.e. threshold and time constants). By placing an EQ in the sidechain path, we can further condition gain reduction on user definable frequency conditions.

The de-esser technique typically uses a narrow peak EQ in the sidechain path to boost the most offensive sibilant frequencies. This EQ exaggerates the dynamic difference between the sibilant band and the rest of the vocal waveform, making it much easier to achieve gain reduction during those consonants (and only then).

A pre-configured de-esser may provide an interface as simple as a compressor threshold and the peak EQ center frequency. These often work just fine. For more detailed control, one could patch an EQ into the sidechain of a relatively fast compressor, or use any number of compressor plug-ins that provide detailed EQ in the sidechain path.

There are lots of great techniques based on this signal flow, so spend some time with it. Frankly, de-essing is the least of what you can do by adding frequency conditions to your gain reduction.

Other Precautions

When you’re recording a vocal performance that may have a sibilance problem, resist the urge to compress the signal in the channel path. Over-compression can exaggerate sibilance. Instead, try using a fader to level the vocal performance, or just record with an adequate amount of headroom.

The same applies to the mixing process. Once you’ve done your best to control vocal sibilance, try using a fader and automation to maintain a consistent vocal volume in the mix. If you simply must instantiate a compressor on every vocal track, keep the attack time slow (> 30ms), and the ratio low.

Finally, don’t listen too loudly when you mix. That’s good general advice, but quality control issues like sibilance highlight its importance. Try a control room volume of 78-83 dB(C) SPL. You might be surprised how much detail you’re suddenly able to hear.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

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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Numark Now shipping N4 DJ Controller

Numark announces that N4, a four-channel DJ controller with built-in mixer, is now shipping to stores.

Featuring four decks of software control plus a mixer that can be used with or without a computer, N4 is designed for DJs who want powerful capability in a lightweight, portable package.

This complete four-channel controller has everything DJs need to perform at their highest level: large, touch-sensitive platters, four decks of software control with looping and effects controls, a USB audio interface and a comprehensive mixer section with EQ and gain.

N4 will come with both Serato DJ Intro software and a four-deck version of Virtual DJ LE.

N4 is designed for flexible control of virtually any music source. Its integrated DJ mixer allows DJs to bring music from any external device into their set, including turntables, CD players, MP3 players, even compatible phones. DJs are able to instantly switch from controlling four decks of software to controlling two decks of software plus two channels of external source. N4 employs ultra-high-resolution 14-bit MIDI that virtually eliminates latency, giving DJs the tight response they need.

With its ability to use time code, N4 sets a new standard for four-channel DJ controllers in its class. N4 gives DJs the ability to use their CD player or turntable to control DJ software by using either turntables with time-coded vinyl or CD players with time-coded CDs.* 

“We created this controller with the mobile DJ in mind,” said Chris Roman, Numark Product Manager. “N4 is lightweight, full-featured and just looks incredible — that combination is unheard of at this price.”

N4 is now available from musical instrument retailers with an MSRP of $699 and an estimated street price of $499.

Numark

 

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Wednesday, March 14, 2012

Ashly Audio Enters The Commercial Sound Electronics Industry

 
With the introduction of the new TM-360 mixer/amplifier, Ashly Audio has entered the commercial sound electronics industry.

This is Ashly’s first product in this genre and heralds the company’s commitment to providing quality products to this growing segment of the audio marketplace.

The TM-360, is a 3-input, 60-watt mixer/amplifier that offers input and output flexibility coupled with energy efficiency. It has the capability to automatically switch into a stand-by mode if no audio input is received for 25 minutes, thus reducing the current draw.

The TM-360, along with other new products in this series, will be covered by Ashly’s five-year warranty.

“Our review of the commercial sound electronics market showed a lack of innovation and quality products,” explains John Sexton, Ashly Audio vice president of sales & marketing. “Many of the companies in this sector offer only a one-year or three-year warranty. The energy-efficient TM-360 delivers Ashly-level quality backed by our five-year warranty at a competitive price point.”

The TM-360 uses a Class D amplifier topology for added energy efficiency. Input options include selectable mic or line level, telephone, and dual RCA sum-to-mono. The transformer-isolated output offers a choice of Low-Z (4-ohm), 25-volt or 70-volt.

The 230-watt international version features Low-Z, 70-volt or 100-volt output options. There are two separate zone outputs – a one-watt output for driving a remote loudspeaker and a pre-amp output that can be used to drive a separate amplifier. Rear panel dip switches allow you to select the mix going to each zone. A front panel 1/8-inch mini-jack makes it easy to plug-in a back-up music source.

The TM-360 is in stock and available for immediate shipment. The rack-mount kit (model RMK-360) is available as an accessory.

Ashly Audio

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