Article

Monday, July 12, 2010

Tech Tip Of The Day: Using M-S Stereo For Mobile Recording

How do I flip one M-S microphone input out of polarity without using a specially built preamp?

 
Q: I’m going to be doing some remote recording with a flash recorder and I’d really like to use the M-S stereo microphone technique.

However, my recorder doesn’t have any kind of M-S setting for flipping the one input out of phase.

Is there any way to do this without dragging along an external preamp?

A: First of all, congratulations on branching out from old recording staple of X-Y! No, seriously!

As you’ve undoubtedly discovered, M-S calls for two outputs from a bi-directional (figure-8) microphone (though other options are possible), one of which is out of phase.

As to how one accomplishes this?

First, you need to get two outputs from the microphone. There are a couple of ways to accomplish this. You can use a “Y” cable between the microphone and the mixer / recorder. You can buy or build a mic cable that is terminated to two XLR male ends.

Or, the preferred method (though not possible in your case) is to use a dedicated high quality microphone preamp and split the signal after it comes out (this is much more robust than splitting the signal out of the microphone itself).

Second, you need to reverse the polarity of one of the outputs (not change the phase as you say above). In the “Y” cable example this is as simple as reversing the wires going to pins “2” and “3” on one of the male XLR connectors.

If using a preamp you would need to reverse those pins on one of the cables connecting the preamp to your mixer or recorder. Many preamps and mixers have a handy polarity reverse switch on each channel you can use instead of modifying cables.

Also, it is possible to buy adapters that will reverse the polarity of a microphone signal, but it really is just as simple as reversing pins “2” and “3” somewhere.

Incidentally, whether you’re buying specialty polarity reversal cables or building your own, be sure to label your cable. That’s not the “Y” cable you want to accidentally grab when you’re setting up in a rush and need a regular “Y.”

For further information on stereo microphone techniques, check out this recent article brought to you by Shure.

As always, we welcome input from the PSW community and would love to know how you would handle M-S in a mobile environment. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

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Friday, July 09, 2010

Pulling The Drum Mix Together & Putting On The Finishing Touches

An excerpt from Bobby Owsinski's book "The Drum Recording Handbook".

This article is the second in a series on drums, excerpted from Bobby Owsinski’s “The Drum Recording Handbook”. Other articles in the series are available here.

Once you’ve miked your individual drums in a way you like, it’s time to listen to the kit as a whole.

First, stand about five to ten feet (2–4 m) before the kit and listen acoustically while the drummer plays.

Listen to the balance of the kit. Next, go back in front of your speakers and listen to the drummer play.

Does it have the same balance as when you were standing before the kit? Try to balance the microphones mix so that you achieve the same blend that you were hearing acoustically.

Step 1: Drum Kit Balance Technique 1
The best way to approach getting a drum sound is to think of the drum kit as one instrument, not seven or eight separate instruments.

Listen for a unified overall sound with no one element standing out from the rest. Hearing “lead hi-hat” or “lead tom-tom” can be very distracting to your audience and take all the pleasure out of listening to your recording.

The drum mix is all about balance—balance between all the drum mics and then balance of the drum mix itself against the rest of the band.

There are several schools of thought on where to start your mix from. I usually start from the kick first, although some engineers start from the overheads and some start from the toms.

Wherever you start, the idea is the exactly the same—to blend all the different drum mics into a cohesive single drum sound.

If the kit was recorded with the overheads miking the cymbals more than miking the entire kit, I’ll bring the fader of the kick drum up first so that the meter reads about -2 on the peaks.

At this point, you might want to add a little compression (1dB or so) to even out the peaks. If you want a more aggressive sound you can add more later after the entire kit is balanced.

Try to refrain from adding any EQ at this time. This is another thing that’s best left until you have the whole kit balanced.

Then, bring up the snare until the level is about the same as the kick. Again, you can add a little compression to even things out a bit, but this is a matter of personal taste.

You can bring up the hat and toms to a level that matches what you heard in the room, being careful to make sure that the level is the same on each drum fi ll. You might have to automate some of the fills to even things out.

Next, bring the overheads up until you just start to hear them. You want to make sure that the cymbals are not overpowering the rest of the kit, and the only way to know that is to have all the mics in the mix since they all have a little cymbal in them.

Lastly, bring up the room mics to the point where you can just start to hear them. This will fill in the sound a lot and glue together the kit balance.

It’s popular in rock music to heavily compress the room mics, but you should be careful about doing this without considering the sound of the room. This only works well when the room really sounds great to begin with.

Heavy compression also changes the balance a lot as the cymbals come to the forefront and can become too loud.

Now it’s time to add the EQ. You should only add some if you want a little more definition. You might want to attenuate around the 4–500 Hz range if anything sounds a little boxy. But don’t feel inhibited by our suggestion here. If you believe a lot of EQ works well for you, by all means, use it!

Balance Technique 2
If your overheads are in an XY or ORTF configuration, then you might have to approach the mix a little differently since this is where the main sound of the kit is coming from.

First, bring up the faders on the overheads so that the meters read about -6 dBFS at the peaks. This should give you a pretty even-sounding drum kit already.

Now bring up each drum track until you can just barely hear it. You might start with a kick drum and then add in a bit of snare.

The idea is to fi ll in the sound and add some punch. When you add the rest of the instruments, you’ll probably have to add a bit more of some of the drums.

Now add EQ and compression if desired as in technique #1.

Balance Technique 3
This technique is used when the tom fills and sound is very prominent in the song.

Start with the toms by bringing the faders up until the meters read about -6 dBFS at the peaks. Make sure that the sound of all of the toms are pretty much the same by adding or subtracting EQ as needed.

You should go through the song to make sure that every fi ll is balanced, automating the tom that’s too loud or too quiet as needed.

Now, build the mix around the toms, starting from whichever drum you’re comfortable. Add EQ and compression if desired as in the previous techniques.

This method makes sure that your toms are always in the forefront of the song.

Step 2: Drum Mix Panning
Regarding panning and positioning of the drums across the stereo spectrum, I personally set my drum mixes the way I am looking at the front of the drum kit.

About half the engineers I know mix from the drummer’s perspective, but I personally prefer that the listener have the “audience” perspective. Therefore, with a righthanded drummer, the hi-hat would be panned to the right, the snare would be just off-center to the right, and the bass drum would be centered.

As for the tom-toms, if you have three toms in your kit, you would pan the high rack tom to the right at three o’clock, the next lower tom to the center at twelve o’clock, and the lowest tom to the left at nine o’clock.

This gives you a nice stereo spread. You may want to experiment by pulling the panning positions a little more toward the center for the tom-toms because if your drummer is playing a lot of tom fills you might find that listening to the wide panning makes you dizzy!

With the overheads, I pan the right overhead hard right and the left overhead hard left. The same goes for the room mics.

Step 3: Minor Mix Adjustments
You may want to add a little reverb to the snare and toms, but I suggest keeping them natural by not using over 2.0 seconds of decay time on whatever type of reverb that you choose. You may want a longer decay time for a ballad.

Try a gated reverb on the snare if you want a cool effect or want to sound like Phil Collins (remember him?). To get this, use the preset on your reverb unit or DAW reverb plug-in called inverse room, non-linear, or gated.

On some types of music it sounds good to use a separate reverb on the snare from the one on the toms. This usually makes the snare stand out a bit more.

Final Drum Sound Checklist
Like the foundation of a house, the drums are the foundation of a recording.

On a strong foundation, you can build almost anything you or your clients can imagine.

A little effort and time spent miking the drums and getting the sound just right can result in a recording that sounds great.

We strongly encourage you to take risks, experiment, take notes on what works for you and what doesn’t, be creative and most of all have fun!

Here’s a list of things to check if you think things just don’t sound as good as you think they should.

Remember that each situation is different and ultimately the sound depends upon the drums, the drummer, the song, the arrangement, and even the other players.

Sometimes things are simply out of your control. These are not hard and fast rules; they are just a starting place. If you try something that’s different from what you read below and it sounds good, it is good!

1. Do the drums sound great acoustically? Make sure that you start with a great acoustic drum sound with the drums well tuned and a minimum of sympathetic vibrations.

2. Are the mics acoustically in phase? Make sure that tom mics and room mics are parallel to each other. Make sure that any underneath mics are at a 90-degree angle to the top mics.

3. Are the mics electronically in phase? Make sure that any bottom mics have the phase reversed. Make sure that all the mic cables are wired the same by doing a phase check.

4. Are the mics at the correct distance from the drum? If they’re too far away, they’ll pick up too much of the other drums. If they’re too close, the sound will be unbalanced with too much attack or ring.

5. Are the drum mics pointing at the center of the head? Pointing at the center of the drum will give you the best balance of attack and fullness.

6. Are the cymbal mics pointed at the bell? If the mic is pointed at the edge of the cymbal, you might hear a swishing sound of the cymbal as it moves back and forth away from the mic.

7. Is the hi-hat mic pointed at the middle of the hat? Too much toward the bell will make the sound thicker and duller. Too much toward the edge will make the sound thinner and pick up more air noise.

8. Are the room mics parallel? If you’re using two room mics instead of a stereo mic to mike the room, make sure that the mics are on the same plane and are exactly parallel to each other. Also make sure that they’re on the very edge of the kit looking at the outside edge of the cymbals.

9. Does the balance of the mix sound the same as when you’re standing in front of the drums? This is your reference point and what you should be trying to match. You can embellish the sound after you’ve achieved this.

 
Editor Note: This article is the second in a series on drums, excerpted from Bobby Owsinki’s “The Drum Recording Handbook”. Other articles in the series are available here.

To acquire this book, click over to the ProSoundWeb Book Store. NOTE: ProSoundWeb readers receive free shipping when entering promotional code PSW at checkout. (offer valid to U.S. residents, applies only to media mail shipping, additional charges may apply for expedited mailing services).

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Church Sound Files: The Basics Of Using Compressors In Your System

What compression can do to benefit a system, and the basic principles of operation

What does an audio compressor actually do?

Most types of signal processors, such as reverbs, equalizers, and delays, are designed to make an obvious change in the sound.

But a compressor’s action is much more subtle; when used properly, most listeners won’t be aware that signal processing is being used. Only if you hear the original dynamic range of a signal and compare it to the compressed version will the effect be noticeable.

A compressor/limiter is essentially an automatic volume control. Imagine a sound operator/engineer with a hand on a mixing console fader and eyes on an input level meter. As long as the meter stays below a certain point (the threshold), the engineer leaves the fader all the way up and the gain is unchanged.

But the instant the sound gets louder, the engineer pulls down the fader by a certain amount. After the sound gets soft again, the engineer will push the fader back up.

This is what the compressor is doing, except much faster and more accurately than humanly possible.

Paradoxically, by cutting the peak levels, a compressor allows us to raise the average level of a sound using the output control to make it sound louder.

By using the threshold and ratio controls, we can set a stable sound that will hold its position in the mix whether the singer is whispering or screaming.

What The Controls Do
Let’s go back to the “engineer with hand on a fader and eyes on the meter” analogy, using it to explain how a compressor works. Think of it as an “automatic engineer in the box.”

The front panel controls simply tell the engineer (compressor) what rules to follow.

THRESHOLD tells him how high the input meter can rise before he has to start pulling down the fader: if it’s turned full clockwise, he won’t pull down his fader until the highest red LED comes on; if it’s turned counter-clockwise, he’ll have his hand on the fader even before the lowest green LED lights.

RATIO tells him how far he should “pull the fader down"when the signal is above the threshold level: should he pull it down just a little bit (compression) or pull the fader as far down as necessary to make sure the output level is never higher than the threshold (limiting)?

The OverEasy switch found on dbx compressors such as the model 160SL shown here, affects how he reacts as signal approaches and travels through the threshold: does he reduce it exactly by the ratio only after it crosses the threshold, or does he gradually ease into the full ratio as it passes through?

The LEDs of the gain reduction meter tell you how much the “engineer” is pulling down the “fader” at any time. If these LEDs aren’t on, his hands are in his pockets.

The ATTACK and RELEASE controls involve the speed of the engineer’s response.

Short attack times order the engineer to get his hands on the fader 1/10,000th of a second after he sees a too-loud signal; long attack times tell him to let transients less than 1/5th of a second pass.

Release tells the engineer how quickly he should push the fader back up again after a loud signal has stopped; when it’s turned counter-clockwise, he pushes the fader back up instantly, and when it’s full clockwise, he’ll take longer to push his fader back up to unity gain.

The OUTPUT control is simply a gain control located after our “automatic engineer in the box.” The INPUT/OUTPUT switch allows us to see the levels before the engineer does his job, or after.

The most important controls are the threshold and ratio knobs. They both interact to get the desired effect, and that requires some experimenting.

For example, if your average input signal is 0 dB, a ratio of 2:1 with a threshold of -12 dB will give you 6 dB of gain reduction, as will a ratio of infinity with a threshold of -6 dB. But the latter setting will sound more “squeezed” than the former.

Common Compressor Mistakes
Extreme settings will lead to extreme results. If you set an infinite ratio and turn the threshold down to -40 dB, the compressor will do what it’s being told to do: turn the level way down.

Trying to compensate by cranking the output control to its maximum amplifies the noise of the mixer, EQ, microphone preamp, and the compressor itself. This noise will fade in whenever the input signal stops, resulting in the classic “pumping” and “breathing” problems.

In short, noise is present in every system, and improper use of any compressor will amplify it to an obnoxious level.

If the ratio is set to 1:1, it doesn’t matter where the threshold control is set: the compressor is being told not to change the gain at all, even if it’s above the threshold level. None of the reduction LEDs will light, and the compressor may as well be in bypass mode.

Similarly, if the ratio is infinite and the threshold is high, or the input trim of the mixer or microphone preamp is too low, there will be no compression. Further, if the output level control is raised, the noise floor will be amplified. 

For low noise operation, make sure your mixer, compressor, and amplifier settings are set properly. As a general rule, you want as much gain as possible in the front of the system (at the microphone preamp), so that a good line level signal is traveling through the whole signal path.

If you have a weak signal to start with, and then amplify it at the end of the signal path (by turning the main outputs of the mixer all the way up, for example) it will be excessively noisy.

When using a compressor with a live sound system, improper settings can cause feedback. Make sure that a channel is well below the feedback point when there is no gain reduction active. If you hear feedback every time the music stops, you must lower the overall level of the system.

Pumping and Breathing
When a compressor is making large changes to the input signal (10 to 12 dB or more); the noise floor will also rise and fall with the signal level.

When this noise signal rises and falls drastically between signals, such as a heavily compressed, noisy drum track, you might hear the noise level “breathing” between drum hits.

One solution to this breathing problem is to turn up the release time. This way, the noise floor won’t have time to rise between drum hits.

However, if the release time is too long, lower level signals after the peak will be lost as the compressor slowly stops reducing gain.

This is called “pumping” as the lower level signals (noise included) slowly fade back up to their normal signal level.

The secret to avoiding these problems is to achieve a balanced release time on the input signal.

This article is courtesy of dbx professional.

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Tech Tip Of The Day: The Top 10 Ways To Fry Your PA Speakers

Are you guilty of any of the following?

 
10. Frequent occurrences of hard feedback

If you say “ouch” after that last feedback squeal, chances are your speakers were hurt too! Using the speakers after they have probably been damaged makes it worse.

9. Improper bi-amplification: crossover too low or tweeter amp too high

Always check your speaker specs for the best crossover points when bi-amping.

8. Not enough speaker systems for SPL requirements or proper coverage

Instead of using higher wattage with the same speaker complement for more SPL, add extra channels of amplification with additional speakers.

7. Trying to cover an outdoor gig with your indoor system

Outdoor gigs require at least 12dB (16x’s the power) more sound output than indoors, and as much as 20dB (100x’s power!) to really do it right.

6. Excessive EQ

The classic “smile” EQ curve is actually an evil grin for your speakers! Keep in mind that EQs are best used for cutting, not boosting the signal. Need more highs? Reduce the bass…need more lows? Reduce the highs.

5. Incorrect use of compressors/limiters

Excessive compression and limiting squeezes the life out of your music and your speakers!

4. Not enough amplifier headroom

If your amp has too little power (not enough headroom), clipping may occur, which will damage your speakers.

3. Sudden transients while the speakers are hot

Transients are thumps caused by powering off and plugging/unplugging mics, etc.

2. Clipping the signal before it gets to the power amp

This is caused by improper mixer gain distribution, or a line signal that’s too hot.

And . . .

1. Continuing to Use Your Speakers After Damaging Them…

And failing to have the crossover parts checked for damage after abusing the speaker system by continued use. Let your ears be your guide. If you hear distortion from clean inputs, damage has likely occurred.

 
For more tech tips go to Sweetwater.com

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Thursday, July 08, 2010

A Practical Guide To Good Bass: Part 1, Acoustical Concepts Of Subwoofers

Part 1 of an ongoing series focusing on subwoofers - how they work in various arrays, concepts and techniques for getting good bass, and more

In sound systems, it would be terrific if loudspeakers worked like spotlights: find the loudspeaker boxes with the right directional patterns, aim them where you want sound to go, and you’re done. Of course, that’s not the way it works, especially for bass. 

Ordinary bass loudspeakers are very nearly omnidirectional over their working ranges, but when you stack up a few of them, the pattern becomes more directional and more complex. Imagine if lights worked that way—a bare light bulb would illuminate the whole room, but four of them in a row would only light up some parts of it. 

To make things worse, when you use multiple woofer stacks—stage left and stage right, for example—it produces wave interference (also called “comb filtering”), causing peaks and nulls in different places in the room at different frequencies. If light worked that way, then when you lit up a room with two white lights spaced some distance apart, the room would be illuminated with a rainbow of different colors.

Even beyond that, there’s the problem of reverberation, which adds its own kinds of confusion and coloration in the time dimension.  That effect that doesn’t even have a parallel in lighting.

In the face of all these phenomena, how do audio professionals design subwoofer arrays and drive schemes that provide required qualities of coverage and fidelity?

If we succeed, then:
• The bass will be clear and will have constant tonal balance over the entire listening area.
• The bass sound level will be in correct balance with the midrange and high-frequency over the entire listening area.
• Negative effects of reverberation and reflection will be minimized.
• Efficiency of the equipment (sound power output per unit cost) will be maximized.

This article offers concepts and techniques for getting good bass. Our focus will be the frequency range from approximately 20 Hz to 150 Hz.

Wavelength
Just about everything having to do with loudspeaker array acoustics is relative to wavelength. A box or array is “large” if its dimensions—or some of its dimensions—are more than about 1.5 wavelengths across. A dimension is “small” if its dimensions are less than about a third of a wavelength.

Here are some typical wavelengths:

 
For normal air temperature, pressure, and humidity, the formulas for wavelength are:

Basic Directivity Rule
For ordinary sound sources, directivity is inversely related to dimension. If an object is small, its directivity is wide;  if large, its directivity is narrow. (See Figure 1) Remember that “small” and “large” are measured in wavelengths, not feet or meters.

Horizontal-Vertical Independence
The basic directivity rule applies independently in the horizontal and vertical planes. For example, a horizontal line of subwoofers might be large horizontally and small vertically. Therefore, its directivity would be narrow horizontally and wide vertically, as shown in Figure 2.

Multiple Sources and Lobing
Many, if not most, subwoofer installations use two separate arrays on opposite sides of the stage. Sometimes these arrays are stacked on the floor, sometimes they’re flown. 

 

 
Either way, the multiple sources exhibit what physicists call “wave interference”, and what audio people call “comb filtering” or “lobing”.

Figure 3 shows the directivity of a single EV Xsub woofer at 50 Hz. In this example, size of the stage is 40x20 feet. The red trace is the polar pattern. Circles are 6 dB apart. The Xsub is essentially omnidirectional.

Figure 4 shows what happens when another Xsub is added at the opposite side of the stage. The result is very different—and not better!

Because the woofers are omnidirectional, everyone in the room hears both woofers. However, the distance from each woofer to the listener is different, except in the middle. Where the distance difference equals an odd multiple of a half-wavelength, the sounds from the two woofers cancel, and the listener hears no bass, at least not directly from the woofers.

These lobes will produce uneven bass tonal balance and level in the venue. In indoor venues, the tonal balance problems are partly masked by reverberation, but the lack of clarity remains. Outdoors, there is no reverberation, and the problem is usually quite obvious.

Figure 5 shows performance of two practical cases - groundstacked rows of subwoofers, and flown subwoofer line arrays.

 

 
The only region that is lobe-free at all frequencies lies along a line running directly out from center stage. Along this line, the bass is strongest and clearest. This is the familiar “power alley” effect that makes the bass sound very good at the mix position, but does not give the mix engineer a good idea of what the rest of the audience is hearing.

The best solution for lobing problems is to use a single center cluster instead of separate left-right stacks. This works for both horizontal and vertical arrays. However, it is not often a practical solution for staging and rigging reasons. 

When left-right stacks are used, lobing problems can be reduced using stacking, beamforming and/or gradient woofers. In all cases, the idea is to minimize interference between the coverage areas of the two stacks.

Beamforming
Beamforming is a technique by which the sound wave emitted by a large array can be aimed and shaped. In a beamformed array, the loudspeakers are driven separately (or in small groups), and each drive signal has its own delay and level. 

Figure 6 and Figure 7 illustrate a typical effect of beamforming on a typical medium-sized subwoofer array. The illustrated array is four EV Xsub subwoofers. Figure 6 shows the array with no beamforming. In Figure 7, the delay values are chosen to direct the bass radiation offstage.  This is a typical technique for increasing side coverage.

Beamforming only works on arrays that are large (as defined above). Controlling directivity of small arrays requires gradient techniques, which will be addressed in my next installment of this article.

 

 
Gain Shading
The term “shading” means modifying array drive parameters for the elements on or near the ends of the array. “Gain shading” means adjusting—specifically, reducing—the drive gain for one or more elements at either end of an array.

For long arrays, shading takes the form of a gradual tapering of gain from 0 dB to about -6 dB over the last two or three elements at each end. The effect of the shading is to make the coverage pattern more regular and less frequency-dependent.

Next time, I’ll be discussing various woofer array types and applications.

NOTE: The polar patterns illustrated in this document have all been produced by the Electro-Voice LAPS 2.2A line array design program.  Starting with release 2.2A, LAPS includes a subbass pattern modeling page. 

Jeff Berryman served as the director of Jasonaudio, a touring sound company based in Canada, and is a senior scientist with Electro-Voice.

Related Articles by Jeff Berryman:
What Really Defines Good Bass In Sound Reinforcement?
Discussion & Analysis Of A Variety Of Bass Coverage Patterns

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Tech Tip Of The Day: Patch Bays

What type of patchbays should I be using in my studio?

 
Q: I’m currently in the process of building a home studio and just at the point of wiring all the raw cable to patch bays.

However, I’m a bit stuck because I’m not sure what patch bays to get!

Is one “style” of patch bay better / more durable / ore popular than another? I guess the same question goes for the patch cables themselves.

A: Wow, is this ever a loaded question.

Because the industry can’t come to a general agreement on the type of connectors we should all use, there are a number of different patch bay and cable configurations.

Since you’re asking because you’re building a patch bay, the best answer is for you to look around your studio and take stock of the types of connectors you have on the back of your gear.

Manufacturers often include both balanced and unbalanced connectors (not always, be sure to check), so your first step is in deciding whether you can use a balanced or unbalanced patch bay.

It’s best to first select a patch bay that suits your specific needs and then pick what patch cable it uses if there are options.

If you can, balanced will allow longer cable runs before signal degradation. Now, in many studios, off the shelf patch bay and patch cables from are perfect.

One of the most common connections you’ll see is 1/4” balanced (TRS) or unbalanced (TS). However, you’ll also find patch bays that have Bantam / TinyTelephone (TT) and XLR connections.

Many feel that the 1/4” connectors are the most durable, but often that’s a matter of opinion. Also, it’s important to remember that the cables will be in a studio, not on the road, so in general patch cables typically do not need the durability and flexibility of primary guitar cables or microphone cables.

They do need good shielding and good sonic quality, so I don’t mean to infer that you can go cheap here. Plus, cosmetics do not matter a whole lot, since no one will ever see them except you.

So, in many instances shorter, well-built cables fit the bill.

As always, we welcome input from the PSW community and would love to know how you would handle patch bays. Feel free to let us know in the comments below.

For more tech tips go to Sweetwater.com

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Wednesday, July 07, 2010

Tech Tip Of The Day: Soffit Mounting Monitors

Can nearfield monitors be soffit mounted just like farfield monitors?

 
Q: I’m using a pair of nearfield monitors in a small studio suite in my house.

As the room is so small I would like to recess them into the wall so I can move my mixer forward and create more client space behind me.

Before I cut holes in my walls, is there anything special I should know?

A: This is something to avoid if you can.

When you mount near field monitors flush with a wall, the wall essentially becomes a giant baffle, which will radically change the sound of your speakers both in terms of tonality and stereo imaging.

They just aren’t designed for soffit mounting.

However, if you must do this, there is a way to do it where you can minimize the problems.

Cut the hole out so there is an extra foot or so of space all the way around where the monitor will sit. If possible use a speaker stand behind the wall, or use a lot of very thick foam to decouple the speaker from the wall.

You can use some type of grill cloth to cover the giant-sized holes.

Mechanically isolating the speaker from the wall and keeping the wall surfaces away from the front (baffle) of the speaker by at least a foot will greatly reduce acoustic anomalies that will detract from the accuracy of your monitor and help retain some of the soundstage in your listening.

You are still better off with near field monitors on stands out in the room, but this is a workable compromise.

As always, we welcome input from the PSW community and would love to know how you would handle this situation. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

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Friday, July 02, 2010

Using Equalization To Improve Church Sound Systems

The nuances of church architecture, and any enclosed room for that matter, pose unique acoustic hurdles. Proper equalization is key.

So what is happening when we put a sound system in a church or in any enclosed room?

Why is the sound sometimes clear in one part of the church but only a few seats away we can hardly hear anything at all while another few seats down it is very loud, or if we hear we cannot understand what is being said?

Why is it when the speakers were brought in and demonstrated on stands they sounded great on the demo music that was played, but when they were hung up higher and installed, the spoken word just is not easy to understand?

The Enemy
When we speak outdoors or hear a loudspeaker outside, the sound is allowed to travel free in any direction. As soon as we put walls around a sound system, the sound starts to bounce off one wall and then to another wall, and so on.

Cancellation and acoustic amplification at certain frequencies can occur. Therefore, much care has to be taken with how the walls are placed, the size of all the structures, what they are made of, the position of the congregation of the church, and so on.

We do not want a church to sound “acoustically” dead for either speech or music. We want a little “warmth” or reverb (not to be confused with echo) but too much can affect speech intelligibility.

Some of the bad things that can happen are:

• Too much reverb smearing the speech.
• If the width, height and length of the room are divisible by the same number, then “standing waves” result. These cause a cancellation of frequencies at a certain area of a room. You may hear everything perfectly where you are sitting but two seats down the pew you can barely hear at all.
• Feedback is that horrible loud squealing sound that can come from your loudspeakers. This occurs when the sound from your speakers re-enters your microphone. Feedback “modes” are very narrow and can be numerous in a church with poor acoustics. They usually occur between 50 Hz and 2500 Hz.
• The room “rings.” This is different from feedback. What you hear is a tone-like or ringing sound that seems to be added to almost every word. What is happening is one or more of the frequencies from the loudspeaker are exciting the physical architecture of the room itself. Ring “modes,” like feedback modes are also very narrow but different as they do not have the phase characteristics to cause the system to self-oscillate (or feedback).

Fixing It
You may say to yourself, “well we’ll get an architect to design the church and that will solve all our problems.” However if the church does not also retain an acoustic consultant at the same stage as the hiring of the architect, trouble could result. Be aware that the architect usually hires the acoustic consultant.

As a result there are far too many cases where his advice was not taken and fundamental acoustic treatment and high quality audio components were left out in lieu of solid brass door knobs, exotic wood furnishings, etc.

A lot of designs may look good but if your walls are flat and parallel, worse yet, made out of glass, the room could simultaneously be an architectural splendor and acoustic hell.

Some of the most gorgeous and famous “modern” churches seem to be getting a new sound system every other year (and will continue to do so until the architecture is corrected).

My advice is to have the acoustic consultant work directly for the church with veto power over the architect’s designs as they affect the acoustic performance of the rooms and budget for acoustic treatment and sound system equipment.

Then you might say, “Okay, we will just get an audio equalizer and that can fix all these acoustical architectural errors.”

The truth is if you have an extremely well-designed acoustical room with high-quality speakers covering their designed areas, then the need for sound system equalization is greatly minimized.

And even though I love to sell equalizers, the less you need the better. Besides, of our list of “enemies” above, an equalizer can only help the last two, feedback and ringing.

Pastors & Music Ministers
And then there is the biggest ongoing conflict I see in churches’ sound systems today. No, it is not God versus Satan. It is the pastor versus the music minister.

At many churches I visit, these two people seem to be at cross-purposes. The pastor just wants to be heard clearly. But the music minister wants a “rock ‘n roll” sound system that would rival the Rolling Stones.

One of the problems facing sound system designers is that the equalization curve for a pure speech system is very different for a music system.

What we are starting to see in many churches is an idea that came from movie theaters and the Broadway stage.

Designers are using a left-center-right (LCR) speaker system. In a true LCR installation the center channel cluster is “equalized” for speech intelligibility, and the two left and right speakers are set up for music.

Beware that some mixing consoles are advertised as being an LCR mixer, but are not.

A true LCR mixer pans from left to center and from center to right. Some companies have relabeled their “monaural” or mono output to “center.”

Do not be fooled, this is not the same. The point is that the pastor’s voice should only come from the center, even if he is singing. In order to minimize comb filter effects, the mix to the left and right speakers need to be treated as stereo, and not a mono program panned to the left and right.

The First Acousticans
So, who were the early practitioners that discovered equalization?

We have to go all the way back to the Middle Ages when the first pipe organs were being installed in large cathedrals. The pipe organs were put together in workshops, and then taken apart and reassembled in the cathedral.

When the organ was played they found that certain notes “took off,” or blared loudly. These were referred to as “bull” notes (a term still used today).

So the pipe organ maker, by ear, would tune that pipe down in level so it would not excite or cause the room to resonate. This was a long painstaking task, since every pitch had to be perfect.

Plus the ornate structures and curves of the ancient churches did a lot to help diffuse acoustic reflections, and intentionally or not, made pleasing acoustic mediums for the organ music. So the art of sound equalization goes back a long way.

Today we have many tools at hand. I will not go into real time analyzers - TEF, Smaart, etc. - since a discussion of these devices would require their own article. Instead, we will talk about filters.

Third-octave and parametric equalizers all use filters, the difference is their shape and characteristics. We found out earlier that feedback and ring modes are extremely narrow, so we just want to rid ourselves of them without affecting the overall sound system.

The most frequently seen EQ is the 1/3-octave equalizer. Prices can range from $200 to $2,000. It is interesting to note that a very famous “loudspeaker” manufacturer teaches its sales force to always use the best quality equalizer available, and if you have to cut costs at all in the sound system the last thing you cut is the quality of the equalizer. Bless their hearts

Much goes into the quality of an equalizer. Aside from careful attention to noise and distortion specifications, the first thing the designer needs to choose is the filter’s shape or architecture.

The two most common today are referred to as constant-Q and variable-Q. The Q in equalization deals with the width of the filter as it is raised and lowered. (See Figures 1 & 2).

Notice the constant-Q maintains the overall width of the filter as it is adjusted, whereas the variable-Q varies its width as it is adjusted from wide at +/-3 dB to narrow at +/-9 dB. These architectures or “transfer functions” were developed in the 1930’s. Companies that use them will make their own minor modifications to how they work, but generally speaking this is how they perform.

Constant-Q was espoused at a time when analog parametric equalizers were very noisy (most still are), to get rid of feedback without taking music out of adjacent frequencies. The fallacy of this claim is that feedback rarely occurs on ISO (Industry Standards Organization) centers. These are the common frequencies everyone has agreed to use for uniformity on equalizers. (See Figures 3 & 4)

Look at the transfer function of two adjacent frequencies of constant-Q and variable-Q equalizers, and see the gap between the two frequencies on the constant-Q, whereas the variable-Q sums as one. What if the feedback mode was between those two frequencies?

Most professional fixed-installation installers now use 1/3-octave equalizers more for tonal balance, not feedback control. They may move three or four adjacent frequencies up or down a few dB for tone control. (See Figures 5 & 6)

Look at the ripple effect with constant-Q. It looks like the intelligibility distortion known as “comb-filtering.” Now look at the variable-Q example and see how the frequencies sum more smoothly. Which do you think sounds better?

Constant-Q is marketed strongly through the “lower-end” of the rock ‘n roll market. One engineer from a well-known audio company has privately said their older equalizers sounded much better than their new constant-Q equalizers but they changed because of marketing pressure.

Another difference is what is known as designing a “band-pass” type equalizer. In this type of equalizer (which requires extremely high-quality componentry) all the EQ filters are always in the circuit regardless of the position of the faders. This guarantees that the overall noise level will not change much as the filters (or sliders) are moved up or down.

Compare this with companies who make their +/- 0 dB setting a bypass circuit.

They do this because their circuitry is not as good (we are talking about much less expensive equalizers now).

Every time you move one of their filters, noise is introduced to the system in a cumulative effect.

Even when a filter is cut, the noise rushes in. This is probably the major price difference in analog equalizers.

Parametric EQ & Bowling
The ideal filter would be a digital parametric filter for control and low noise.

Parametric filters let you vary the Q any width you want from many octaves wide, to as narrow as 1/70th of an octave.

And, believe it, those feedback and ring modes are even narrower than that.

The idea of ridding feedback with a 1/3 octave EQ in a fixed installation system is like rolling a bowling ball to knock down only one of twenty adjacent toothpicks.

You get the one you want but you crush the rest. A 1/3 octave equalizer can diminish the feedback and destroy all the music and sound of the neighboring frequencies.

A parametric equalizer can selectively “notch” the offending frequency without harming the overall sound.

We now have much higher quality digital signal processors with built-in parametric equalizers and even stand-alone digital parametric equalizers for reasonable prices that operate very quietly.

Fixed installation narrow-banded EQ is available to anyone on almost any budget, although it still usually takes a professional to do it correctly.

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Tech Tip Of The Day: Studio Kit Maintenance

How do I know when it's time to replace the heads on my studio drum kit?

 
Q: So, I should probably preface this by saying that in no way am I a drummer.

However, in my small studio, I have a drum kit that is available for use to my studio customers during sessions.

I’ve noticed that the heads are starting to look rather worn, but I don’t really know what “worn” means to a drum head.

I mean, my SM57’s don’t get replaced just because they have a little wer and tear.

Can you offer any insight on how often I should replace the heads?

A: Well, it’s rather impossible to give an exact answer (“after XXX number of hits…”) However, signs that it is time for new drumheads include:

• The coating is worn off a coated head.

• The head is severely dented.

• The head is “dished” into the drum, rather than being flat. This indicates the head has stretched.

• The head can’t be tuned to the pitch you want. This indicates the head has lost elasticity.

• You hear buzzing or distortion from the head. This means the head has stretched and is no longer in complete contact with the shell.

Most importantly, listen to what your drummers are saying and pay attention. If they’re talking about needing new heads and the set looks like something you wouldn’t play yourself, then it’s probably time to retool your kit.

 
For more tech tips go to Sweetwater.com

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Thursday, July 01, 2010

Tech Tip Of The Day: Half Power

Is a 1000-watt amplifier really that much more powerful than a 500-watt amplifier?

 
Q: My buddy told me that if I go to a 500-watt power amp from a 1000-watt power amp, the volume will be half as loud.

But someone else said they are really about the same. Or, that the difference wouldn’t be large enough to notice.

Who’s right?

A: All other things being equal, and they rarely are, a 1000-watt power amp will indeed be ~3dB louder than a 500-watt amp.

As you’ve mentioned, 3dB is a noticeable difference, and is the standard difference used to define half power.

So, if you choose to use the 500-watt amplifier in lieu of the 1000-watt amplifier, you should expect to see a halving of your overall output assuming everything else actually is equal.

What do I mean? If your speakers couldn’t handle 1000-watts in the first place and you were running the 1000-watt amplifier at less than full power, you will obviously notice less of a difference, if any.

 
For more tech tips go to Sweetwater.com

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Wednesday, June 30, 2010

Tech Tip Of The Day: Setting Front-of-House Delay

How can I easily set front-of-house delay to match backline sound?

 
Q: I work with a lot of smaller bands in local clubs and have been doing so for quite a while now. I’d say my mix chops are really starting to

 
However, lately I’ve been in a few places where I really needed to dial in the delay between the mains and the backline, but I really didn’t know how.

Can you help?

A: As you’ve noticed, in many smaller club settings your main sound system can face competition from the signals generated by the backline - the drums and amplifiers onstage.

These project into the audience and can cause timing-related problems that are perceived as “smeared” audio.

There’s a relatively simple way to combat this and produce a cleaner, more pleasing FOH sound.

First, it’s worth noting that large sound companies use sophisticated room analysis software to calculate the correct alignment times necessary for their FOH systems to sound their best.

This function is built into many boxes, like the DriveRack and other popular models. However, for every average club engineer out there, here’s a much less scientific - and more approximate - method.

Since the idea is to counter the sound coming off the stage, start by selecting the loudest acoustic source onstage. This is usually the snare drum. Have your drummer play single strokes on the drum, about one per second. Make sure he or she plays at “gig” level!

Start with the approximate formula that 1 foot equals 1 millisecond (rounding the speed of sound down to 1000 feet per second). Measure the distance from the snare to the drivers of your sound system and set the delay that’s connected to your FOH system accordingly.

Be absolutely certain your sound system volume is as close to equal the acoustic snare’s volume as possible. This won’t be your gig level; it’s just for purposes of setting the delay.

Now use your ears and add or subtract delay amounts until you hear the closest possible attack consonance between the stage sound and the speaker sound. You’ll get better at this with practice, it will improve your ear training, and it won’t cost you a dime.
 
For more tech tips go to Sweetwater.com

 

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Monday, June 28, 2010

Phase Cancellation - The Drum Sound Destroyer

An excerpt from Bobby Owsinski's book "The Drum Recording Handbook".

This article is the first in a series on drums, excerpted from Bobby Owsinki’s “The Drum Recording Handbook”. Other articles in the series are available here.

One of the most important and overlooked aspects of drum miking is making sure that the mics are all in-phase.

This is really important because with only one out-of-phase mic, the whole kit will never sound right, and if not corrected before all the drums are mixed together, it can never be fixed.

So just what is phase anyway? Without getting into a heavy explanation, it just means that all the microphones are pushing and pulling together.

If one mic is pushing while another is pulling, they cancel each other out. Check out the diagram of Figure 1.

In this figure, both mics are pushing and pulling together. Their signal peaks happen at the same time as does their valleys.

As a result, their signals reinforce one another. In Figure 2, when mic 1’s signal peaks, mic 2’s signal valleys. They cancel each other out and result in a very weak sounding signal when mixed together.

Acoustic Phase Cancellation
There are two types of phasing problems that can happen—electronic and acoustic.

An acoustic phasing problem occurs when two mics are too close together and pick up the same signal at the same time, only one is picking it up a little later than the first because it’s a little farther away.

With acoustic phase problems, the sounds won’t cancel each other out completely, only at certain frequencies. This usually makes the mix of the two together sound either hollow or just lacking depth and bottom end.

The way to eliminate the problem is by moving mic 2 a little further away from mic 1, or if the mics are directional, make sure that each one is pointing directly at the source that they’re trying to capture.

Keeping the mics parallel to each other, or at a 90-degree angle for mics underneath drums will also really make a difference.

Electronic Phase Cancellation
Why would there be an electronic phase problem? Almost all of the time it’s because a mic cable is mis-wired; it was either repaired incorrectly or originally wired incorrectly from the factory (which is rare).

There are two ways to check the electronic phase.

Checking Phase The Easy Way
Here’s a very easy way to check mic phase, although not as precisely as method #2 shown later.

After you get a mix balance of the kit together, flip the phase selector on each mic channel one at a time either on your console or on the DAW.

Whichever position has the most low end, leave it there. Do this on every mic in the kit (select the overhead and room mics in a pair, but check the left mic against the right as well).

Checking Phase The Slightly More Difficult Way
This method takes a bit more work, but you’ll know for sure if you have a mic cable that’s wired backward. Also, you really have to have another person with you to make this work. It’s a two-man operation.

First you have to pick a mic and make it your “reference.” Any mic on the kit will do, but it’s easier to pick an overhead or a mic that can easily come off the stand.

Now take your reference mic and put it next to another mic on the kit, say the kick drum mic, as in Figure 3. Make sure that each mic is at the exact same volume level in the speakers, not fader level.

Now have someone talk into the mic while you switch the phase selector on either the console or DAW. Again, choose the selection that sounds the fullest.

Do this to each microphone. Any channel that has its phase selector different from all the others has a mis-wired cable. Make sure you mark it so you don’t have the same problem again!

Times When You Might Want The Phase Reversed
There are times when you should definitely consider flipping the phase before you start mixing.

As we said before, there may be some acoustic phase issues as well because even though a mic may be further away than another, it may still be picking up the same source.

In the following cases, the phase should be flipped to overcome an acoustic phase problem.

An Under-Snare Mic: The under-snare mic should just about always be flipped out of-phase.

Any Under-Drum Mic: Anytime a mic is placed underneath a drum, it’s phase should almost always be flipped
out of phase.

Room Mics: Depending upon where they’re placed, how much room reflection they’re receiving, and how high they’re used in the mix, sometimes the room mics sound a lot better if the phase is reversed.

Overhead Mics in Extremely Rare Cases: Once again, it depends upon how high they’re placed above the kit, what kind of reflections they’re receiving and if they’re the main sound of the kit, but on rare occasions it might sound better (meaning fuller) if the phase is flipped.

 
Editor Note: This article is the first in a series on drums, excerpted from Bobby Owsinki’s “The Drum Recording Handbook”. Other articles in the series are available here.

To acquire this book, click over to the ProSoundWeb Book Store. NOTE: ProSoundWeb readers receive free shipping when entering promotional code PSW at checkout. (offer valid to U.S. residents, applies only to media mail shipping, additional charges may apply for expedited mailing services).

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Friday, June 25, 2010

Variable-D And Beyond: Classic EV Microphone Design & Evolution

Lou Burroughs would demonstrate the 664’s ruggedness by smacking it on a two-by-four...

The Electro-Voice (EV) model 664 microphone, introduced in the mid-1950s, was designed for typical sound reinforcement applications of that era.

EV employees with the company at the time recall that one of the reasons for the 664’s development was to answer the considerable success of the Shure model 55 (Unidyne).

Yet the 664 was hardly an imitation. It’s the first microphone to employ the company’s renowned Variable-D design principle, which is still at the heart of some EV mics popular to this day.

Patented by Alpha (Alphie) M. Wiggins in 1963, Variable-D mics use frequency selective rear ports to achieve a cardioid pattern. This results in considerably less proximity effect in comparison to single-D designs.

Whether this is a positive or a negative is in the ears of each beholder, but suffice to say that Variable-D has earned its place in the “microphone hall of fame.” (Well, if one actually existed.)

The 664 also earned the nickname “Buchanan Hammer,” a moniker paying homage to the company’s then headquarters in Buchanan, Michigan as well as some serious durability.

Word has it that EV co-founder and mic guru Lou Burroughs would demonstrate the 664’s ruggedness by smacking it on a two-by-four board. A later (unconfirmed) demo reportedly had Lou using the mic to actually drive a nail into the board.

The model 635A later assumed the “hammer” nickname, but it pales in comparison: think tack hammer versus framing hammer. However, the 635A did carry on the tradition of being able to withstand brutal punishment. (Editor’s Note: while working at EV, I indeed drove large nails into boards with the head of a 635A, and it still performed just fine - in addition to providing a chuckle.)

A short time after the 664 debut, EV introduced the models 665 and 666 for broadcast. The 665 looks and feels like a 664, but its finish is non-reflecting gray rather than chrome.

 

And while the 664 was capable of high- and low-impedance operation, the 665 and 666 were low-impedance only.

The 665’s connector is an XLR instead of the dreaded 91-series Amphenol four-pin used on the 664. Meanwhile, the 666 was the premium broadcast model, outfitted with a Cannon UA series connector (which looks vaguely like an XLR, but is larger and “D shaped”).

Where the 664 and 665 could attach directly to a mic stand, the 666 required a specialized clip.

Although the 666 was discontinued by the late 1960s, it still commands a premium price on eBay, and many live sound engineers still prefer it for kick drum and bass.

 

It’s also an excellent horn microphone, and I happen to like the 666 (and its newer incarnations) for electric guitar amps.

The subsequent model 667 combined the 666 design with a transistorized preamplifier. This preamp could supply extra gain if needed, and offered equalization switches for the low and high ends of the spectrum.

A separate on/off switch could be used to add in a presence peak, if desired. The preamp used a mercury battery; it predates phantom powering by many years.

Later, the preamp was abandoned on the models 667A and 668, replaced with internal equalization settings that allowed frequency response to be tailored with use of several pins that “programmed” the equalizer.

The 667A and 668 were primarily intended as boom mics - and -they were the first mics to make use of the Continuously Variable-D principle, with that patent credited to Harold S. Mawby.

The model RE15 came along to replace the 666, but those who knew still preferred the 666, establishing its beginnings as a cult object of present day.

A popular myth goes that the 666 was discontinued because of the satanic implications of the model number, but the people who were there at the time say this just isn’t so. Competition, not the devil, was the end of this microphone.

The RE15 also offered a Continuously Variable-D design, meaning that it had even less proximity effect than the 666, and its polar patterns were very consistent with frequency.

 

Although the RE15 never attained the cult status of its older brother, it was a favorite with broadcasters because of its smaller size.

Laugh if you want, but The Lawrence Welk Show used a bunch of RE15s to replay its 666s. The more uniform polars contributed to less acoustic phase interference in the finished mix, and the resulting cleaner sound was not lost on the ABC television network’s technical crew or on Welk’s people.

The final chapter in our story is the model RE20 - ever have heard of it? The late Tom Lininger was the principal designer, and it was originally conceived as a “condenser killer.” It was quickly adopted it for a variety of tasks in the studio, mostly relating to things that were either loud or low.

Broadcasters also found it to be a very good announce mic, and it’s still popular in that application today, as is the RE27N/D, which incorporates a neodymium element.

 

Oh - and let’s not forget that the RE20 (sometimes also branded as the PL20) is still one of the most popular kick drum mics in sound reinforcement some 35-plus years after its introduction.

Take a look at the RE20 response curve, and you can see that it indeed offers the high end of a good condenser.

While history says that it didn’t really “kill” the condenser genre, the RE20 has nonetheless more than earned its place in the “mic lockers” of many. (There’s an interesting marketing lesson to be found here.)

Rick Chinn is a long-time audio professional and history buff. He heads up Uneeda Audio. Find out more about Rick and the company at www.uneeda-audio.com.

References
Telex Communications and Electro-Voice
Allied Radio Catalogs: 1954, 1955, 1956, 1957
US Patents: 3,115,207, 3,378,649

And “Those Who Were There” - the author’s heartfelt thanks to all of the following EV folks for helping with 50-plus-year-old memories:
Jim Long, senior sales support engineer (current)
Bill Raventos, product manager, professional products
George Riley, marketing manager
Don Kirkendall, manager of advertising and promotion
Frank Spain, national service manager
Lloyd Loring, sales promotion manager

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Tech Tip Of The Day: Line Level Into Mic Only Inputs

Can I connect a line-level signal to a microphone input?

 
Q: At our church we’ve got a really old mixer kicking around that gets used for portable PA.

It has a few inputs, all XLR, that are each labeled microphone input.

Trouble is, we’d like to use one of these inputs with an iPod but are really afraid of messing something up.

Can I connect a line-level signal to a microphone input? Will this work?

A: You’re right to be cautious, as mic levels are much lower than (typical) line-level signals.

This means that any equipment designed to work only with a microphone as an input may be overloaded when you hit it with a line-level signal. The likelihood of this varies widely so it’s always best to try it first and see what your results are.

However, just so you know, when we say overloaded we’re really just talking about distortion, not actual damage. So, when trying these steps, your equipment is never in any real danger.

If after plugging the device you do find the input distorts and there is no way to change the input level of the mic-level device (via dip switches, knobs, or other means) then your only alternative is to lower the output of the sending device. However, first I’d absolutely try to change the input level of the channel you’re using, as that may eliminate the problem all together.

As I mentioned, there are a couple of ways to lower the output of the sending device:

1. If possible, simply turn down the output of the sending device until the receiving device stops distorting. This has the advantage of being easy (and free), but it can compromise the signal-to-noise ratio of your signal., so it should be used judiciously.

2. You may be able to assemble various level-matching devices to make the whole thing work — for example, a line-to-instrument level re-amp-type device feeding a direct box, feeding the mic input.

3. If neither of these does the trick, then plan 3 is to buy (or build) a pad to put in the signal path to lower the voltage of the signal coming in.

At the end of the day, you could end up spending more to “make it work” than it’s really worth, but I’d absolutely try some of the free solutions first and see of they don’t solve your problem.

 
For more tech tips go to Sweetwater.com

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Thursday, June 24, 2010

Audio / Video Microphones Explained

The most common microphone types used for video and when they're appropriate.

In audio, it’s not unusual to have multiple responsibilities semi-related (or completely unrelated) to audio.

Though you may be the Front of House guy, it’s not out of the question to be asked to handle some audio/video sound either in your free time or on a job.

No matter if you’re a seasoned pro or a volunteer still learning the ropes, here are some good application hints for when to use the following are some good hints on choosing the right mics for some common audio/video applications.

In most situations, there is no “right way” to do it, but by understanding the different options, you’ll be better equipped for every eventuality.

Though obvious to some, it’s worth stating that the first step in getting the sound of someone’s voice onto tape is the microphone. Microphones serve a very basic purpose: to change acoustic energy to electrical energy.

They convert sound waves into an electrical signal which can be modified, amplified, or recorded. Since the microphone’s function is so basic, you might well ask why there are so many different kinds of microphones.

It’s simply because some types of microphones are better suited to certain uses than others, just as pickup trucks are better than small sports cars for carrying large, heavy loads.

If you are familiar with the different types of microphones, and how and when to use them, your productions will start sounding less like a home video and more like the nightly news

Physical Design
In choosing a microphone for a specific application, the first thing that must be considered is how it will be used.

Will it be held by the person talking? Will it be clipped to the user’s clothing? Will it be located a few feet away from the subject, so that it remains out of the frame?

Handheld: The most common kind of microphone is the handheld type. This style is the most flexible, because it can be held by the user, mounted on a floor or desk stand, or attached to a flexible “gooseneck” on a lectern.

A good quality handheld mic should have an internal shock mount which will minimize handling noise (thumping sounds transmitted through the handle and picked up by the microphone cartridge), and it should be ruggedly constructed to withstand physical abuse.

If you can have only one microphone in your kit of audio gear, it should be a handheld mic. Models at the upper end of the price scale will usually offer clearer, wider-range sound, better shock mounting, and more durable construction.

Tips on Using Handheld Mics: Whether held in the hand or mounted on a stand, the microphone should be positioned about 6”-12” from the talker’s mouth, pointing up at about a 45-degree angle.

With some types of microphones, holding the microphone very close (3”-6”) will cause additional emphasis of the lower frequencies (known as proximity effect), resulting in a “warmer”, bass-heavy sound.

Lavalier: Another popular mic for video use is the lavalier type.

Historically, the word “lavalier” refers to microphones which are hung on a cord around the wearer’s neck, but the term has grown to include almost any small microphone that attaches to the user’s clothing.

Lavalier microphones leave the talker’s hands free to gesture, hold notes, or demonstrate a product.

In addition, they are usually very small and therefore tend to disappear on camera. Also, using a lavalier will keep the distance from the microphone to the talker’s mouth fairly constant, reducing the need for frequent mixer adjustment once the levels have been set.

A disadvantage of lavalier mics is the fact that they tend to be single-purpose microphones — they rarely sound good if handheld or used away from the body.

While the lavalier mic’s small size makes it easy to conceal behind lamps or other objects, an equalizer is usually necessary to make the mic sound natural when it is not attached to the person talking.

Tips on Using Lavalier Mics: For best results, lavalier mics should be placed on the outside of clothing, about six to eight inches below the chin.

They are generally clipped to a pocket, lapel, or necktie. If none of these options are available, the mic can also be clipped to the collar of a shirt or blouse.

Sound quality in this position tends to be somewhat muffled, however, because some high frequencies (which contain consonants) do not fully wrap around to the area under the chin.

Concealing a lavalier microphone: In some productions, it is necessary to conceal the microphone.

It is important to prevent both the microphone and the first few inches of cable from rubbing against either the body or clothing, which will cause noise. Here are some options:

- Under the shirt collar. The mic is lightly taped to the inside of a dress shirt collar, near the opening in front. The cable can be routed around to the back of the neck, over the collar and under the shirt.

- On eyeglasses, on the inside of the temple. The cable is routed over the ear and down the back.

- On the forehead or cheek, secured with medical tape or gum. A disadvantage of this method is that the microphone is directly exposed to perspiration and makeup.

- On the chest, secured with double-sided tape to both the skin and the inside of the shirt. Try to avoid placing the mic behind any material having more than one layer. This reduces pickup of high frequencies, which results in a flat, “muddy” sound.

Double-miking: In some cases, even a remote chance that the microphone might fail during a live event constitutes an intolerable risk.

For this reason, a news anchor or key presenter may wear two lavalier microphones for redundancy. Only one mic is used at a time; if the primary mic fails, the backup mic channel can be turned up immediately.

Double-miking with lavalier microphones is usually achieved with a special tie clip or bar that holds two microphones. When wireless microphones are used, each lavalier mic must be connected to its own body-pack transmitter.

These two transmitters must be on different operating frequencies, and their signals must be picked up by two different receivers, as discussed later.

Surface Mount: These microphones are designed to work on a flat surface. They are usually physically contoured to look less intrusive on a conference table or desktop.

The microphone element is located very close to (but not touching) the surface, so that sound waves reflected from the surface arrive at the mic element at the same time as the direct sound.

This effectively doubles the sensitivity of the microphone compared to a free-standing handheld type at the same distance. (This sensitivity boost assumes that the surface is sufficiently large to reflect even low-frequency sound waves.)

Tips on Using Surface Mount Mics: Surface mount microphones work best when positioned on a smooth, flat surface, such as a table or desk.

If table vibrations are a problem, try putting a very thin piece of soft foam rubber underneath the mic. (A computer mouse pad with a hard top surface often works well.) In some situations, surface mount mics can even work well when mounted on a wall.

Keep in mind that the sound quality of this type of microphone is affected by the size of the surface on which it is placed. For best results, use a surface at least 3 feet square; using a smaller surface will tend to reduce pickup of low frequencies.

The effect on speech frequencies is usually mild, and may actually improve intelligibility of very low voices by reducing boominess.

Shotgun: The shotgun microphone is so named because the long, slotted tube in front of the microphone cartridge makes it resemble a shotgun.

This “interference tube” helps reject sounds coming from more than about 30 degrees off to the sides, while still picking up sounds from the front.

This extremely directional pickup pattern (called a line/gradient pattern) makes shotgun mics popular for TV news and movie sets.

Shotgun microphones are not telephoto lenses for sound. They do not allow you to zoom in on a conversation from 100 feet away. Here’s a much more accurate analogy: imagine looking through a long tube at a person standing 20 feet away.

The person’s image does not appear to be any larger or closer, but is somewhat easier to see, because the eye is not distracted by things happening off to either side. This is exactly what shotgun mics do best - screen out sounds coming from the sides.

In practice, a shotgun microphone can typically be placed at four to five times the acceptable distance for a standard omnidirectional microphone. Keep in mind that the shotgun mic will also pick up sounds coming from behind the subject.

Tips on Using Shotgun Mics: Shotgun mics can be positioned either slightly above, below, or to the side of the sound source, so that the mic does not appear in the camera frame. Try to avoid aiming the mic at a hard surface, such as a tile floor, brick wall, or hard ceiling.

These surfaces reflect sound waves, and may reflect background noise into the microphone or cause the sound to be slightly hollow. A heavy blanket can be placed on a reflective surface to provide some temporary sound absorption.

Shotgun mics are more sensitive to wind noise than standard microphones, so try to avoid moving the mic rapidly and use a foam windscreen if possible. Larger “zeppelin” or “blimp” type windscreens are usually necessary outdoors. Also, it’s a good idea to use a rubber-isolated shock mount to control handling noise that may be transmitted through a stand or boom.

Hopefully, no matter your skill level, you’ve found some new information that will be useful when the next time you find yourself working with video. Remember, when it comes to microphones, the number one goal is clean audio, but a little bit of experimentation never hurt anyone.

Supplied by Shure Incorporated. For more information visit Shure.com.

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