Inside Automatic Microphone Mixer Systems

An automatic microphone system is a good option whenever multiple microphones (four or more) are being used, particularly if the sound system is intended to run without a live operator
shure mixer

Every time the number of open or active microphones in your church system increases, the system gain (or volume) also increases.

The effect of this is greater potential for feedback as more microphones are added, just as if the master volume control were being turned up.

In addition, unwanted background noise increases with the number of open microphones. Here, the effect is a loss of intelligibility as the background noise level rises closer to the level of the desired sound.

A good solution is to activate microphones only when they are addressed and to keep them attenuated (turned down) when not being addressed.

In addition, when more than one microphone is addressed at a time, the system volume must be reduced appropriately to prevent feedback and insure minimum noise pickup.

An automatic microphone mixing system can be a lot of help in this situation. Essentially, it’s comprised of a special mixer and an associated group of microphones, and it’s function is twofold: to automatically activate microphones as needed and to automatically adjust the system volume in a corresponding manner.

In some automatic microphone systems, ordinary microphones are used and all of the control is provided by the mixer. In others, special microphones are integrated with the mixer to provide enhanced control.

There are several techniques used to accomplish channel activation or (gating) in an automatic microphone system.

A look at a basic automatic microphone mixer setup. (click to enlarge)
In most systems, a microphone is gated on when the sound that it picks up is louder than some “threshold” or reference level.

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When the sound level falls below the threshold, the microphone is gated off. This threshold may be fixed, adjustable, or even automatically adjustable.

In any case, the threshold should be set so that the microphone is not activated by background noise but will be activated by normal sound levels.

Traditional threshold systems distinguish between background noise and the desired sound only by level.

However, if background noise becomes sufficiently loud, it may activate microphones unless the threshold is adjusted to a higher level.

Subsequently, if the background noise decreases, normal sounds may fail to gate the microphones on unless the threshold is lowered as well. Threshold adjustment is critical to automatic mic systems of this type.

Some recent automatic mixers incorporate noise adaptive threshold circuitry. These have the ability to distinguish steady signals such as background noise from rapidly changing signals like speech.

They can automatically and continuously adjust individual channel thresholds as ambient noise conditions change.

In addition, some designs can recognize that the same signal is being picked up by more than one microphone.

In that case, only the channel with the strongest signal is activated. This prevents both microphones from being activated when a talker is in between two microphones for example.

Certain other automatic systems, with integrated microphones, can actually sense the location of the sound source relative to the ambient noise and activate microphones only when the sound comes from the desired direction. These “directional gating” systems do not require any threshold adjustments.

There is another circuit within every automatic mixer that continuously senses the number of open microphones (NOM) and adjusts the gain of the mixer accordingly.

With a properly functioning automatic system, if each individual microphone is adjusted to a level below the feedback point, then any combination of microphones will also be below the feedback point.

Many automatic microphone mixers have additional control circuitry, often in the form of logic connections.

These are electrical terminals that can be used for a variety of functions, including: microphone status indicators, mute switches, loudspeaker attenuation, and the selection of “priority” channels.

Some automatic mixers have an adjustable “off attenuation” control: instead of gating the microphone completely off, it can be “attenuated” or turned down by some finite amount, to make the gating effect less noticeable in certain applications.

Another control included on some units is an adjustable “hold time”: when the desired sound stops, the channel is held on for a short time to avoid gating the microphone off between words or short pauses.

In addition, a function which locks on the last microphone activated insures that at least one microphone is on, even if no one is speaking.

Finally, most automatic mixing systems are able to be expanded by adding individual channels and/or by linking multiple mixers together to control large numbers of microphones simultaneously.

An automatic microphone system should be considered whenever multiple microphones (four or more) are being used, particularly if the sound system is intended to run hands-free, that is, without a live operator.

This is often the case not only in the worship facility itself but in fellowship halls, conference rooms, and auditorium systems.

Microphones should be selected and placed according to the normal guidelines (integrated systems require a microphone choice from the selection available for those systems).

It is recommended that the manufacturer or a qualified installed sound professional be consulted on the details of a particular automatic microphone system.


(Copyright Shure Incorporated, used by permission.)

__________________________________________
David McLain | The Mixer Guy | CCI SOLUTIONS
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How to Ring Out a PA System


This video demonstrates how to ring out a PA system to eliminate feedback. Releasing May '06, The Ultimate Church Sound Operator's Handbook by Bill Gibson, contains almost 500 pages of full color text, photos, illustrations, and a DVD with audio and video examples teaching everything you'll need to know to be a live sound engineer.

If you have difficulty seeing this training video, click on the title at the top of the article ("How to Ring Out a PA System").

_____________________________________________
David McLain | The Church Sound Guy | CCI SOLUTIONS
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PO Box 481 / 1247 85th Ave SE
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Voice: 800/426-8664 x255 / Fax: 800/399-8273
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Processing on-the-fly!

[Editor's note: Today's article is from a guest author, Jason Mageehan, from England. He's writing about a specific piece of equipment, the dbx DriveRackPA that, as he said in his cover note to me, "quite literally changed my life!" I enjoy the detail of how he uses it; even more, I love the passion with which he writes! Enjoy!]

My name is Jason, based just outside London in the UK. I discovered this website and found some of the articles really interesting and well written, so thought I could contribute something that I have found helpful in my life doing sound in church. Firstly a bit of background to me, I am a semi professional musician, a semi professional sound / recording engineer and a semi professional graphic designer. When I say semi-professional, that means that I get paid, but not often enough to make it a full time career!

So this piece of gear, the DBXDriveRack PA, as a sound guy who has worked in some interesting venues with some even more interesting PA systems, this little beauty has changed my life....oh yes, and when I say 'interesting,' that doesn't always mean 'interesting-good'; often it's 'interesting-bad'!

Let me tell you the story of how I first got hold of the device. Our church quite often will hold a large evangelistic meeting, about a year ago now, we rented the local municipal hall which seats about 1500 people to have Arthur Blessit, a guest speaker preach the word! We had a 2 hour window to set up the sound system, the band, set the stage, basically turn a dry venue into church for the morning. This was the PA system I was given to work with:

  • Meyer Active Tops
  • Logic Systems Twin 15" Passive Subs
  • QSC Amps
  • Allen and Heath GL2400 Mixer
  • Samson Compressors / Gates
  • Selection of In-Ear monitors and passive Foldback Wedges.

Stage monitoring kind of takes second place when you're on a tight limit with time and resources! As a musician myself, I know that if I can hear a basic mix with enough clarity that I can make everything out, I'm happy, and that's easy to do with the right ear, so let's forget about that and focus on the FOH. So, we have a mish-mash system, in a municipal hall with terrible acoustics. Having done sound there before, I knew getting the overall EQ Curve and crossover point would be essential, and to get that done really quick because I had to work with other peoples time constraints as well....I knew I needed some more time.

I rented the DBXDriveRack PA and a measurement mic (RTA Mic) as I had read good things about it, and thought at worst it would be a comprehensive crossover and look nice, and I could go back to the Behringer (yuk!) manual EQ unit the church owns, and at best it would do everything I wanted it to do (crossover and EQ) quickly and effectively.

I have to say, it did do everything I wanted it to do, this room, this PA system, the DBX literally brought it all together and it sounded like a fantastic install, not something threw together quickly for a service. It took about 90 seconds of white noise to be fed out of the speakers at performance levels for it to monitor the sound and make adjustments accordingly that raise and lower peaks in the sound spectrum to give you what is essentially a completely flat speaker response. Having a flat response is clever, but not particularly exciting to the ear, but obviously you can then overlay an EQ curve over the top of your corrected sound, which will then give you exactly the sound you are after.

Put simply, this box does exactly what it says it will do. I have since had my church purchase one. Normally it sits in the install rack running our FOH sound, but when we go out, (and when I have a gig!) I can take it out, use it and re-fit it, set it back to the church preset....easy!

One more story in my life with this unit, a recent fund raiser where I used my own Yamaha 03R digital desk and this unit, with a PA system that is truly a mish-mash of brands....

  • 18" Sound Lab Subs
  • 15" Ross Full Range Speakers
  • LD Systems Tops
  • Lyon Forge Amp
  • Behringer Amp

Now, the system is set up so the Sound Lab Subs, and the Ross Full Range speakers are actually both used as Subs, and the LD systems were the tops. The event was in an Old English church which have lovely acoustics for a choir, but not for a more contemporary set up. Again, I used this unit, and I honestly could not believe it. If I was blindfolded, I honestly would have believed that there was a D&B rig in there, or an EAW, basically I'd put it up against any big brand speaker company even with those speakers.

That may sound flippant, but those of us who've been around long enough, know that however good your system is, if you don't have the ears for it, it will sound bad. What this unit does is take a lot of the hard work out of the equation. I don't use any of the other features on the unit other than the crossover and the EQ facility, but here's a full run-down of the product: http://www.dbxpro.com/PA/PA.php [Editor’s note: the DriveRackPA has since been replaced by the DriveRack PA+: http://www.dbxpro.com/PA+/PA+.php].

Some people say it sounds “too digital.” In response to that, I can honestly say, when I use my digital desk and this together, yes, it sounds incredibly pure, and I like that. But what I tell people is this: this unit will very quickly and easily give you an solid foundation to build your soundscape. It has never failed me, and it has saved me on a number of occasions where the situations have been such that I've needed some help from a machine that has an ear that is better than mine.

Would I recommend it? Yes, go and rent one first to see if you like it though [Editor's note: or call CCI Solutions who has one in their "Try It Free" program in the US]. Also, before you make your next speaker purchase, think about this: if this unit costs you £400 [$499 in the US], and one speaker system costs £2000 and another costs £4000, have a serious play around here, because you might find that, like me, you can get a less able system to sound just as good as a top of the range system, and you'll have saved loads of money!

The unit has a tag line; "The cure for the common PA". In my opinion, it is.

Jason Mageehan
www.twitter.com/jasonmageehan
jason_dm@yahoo.com

How to Find Ground Loops and Prevent AC Hum

Note: This article is from Ebtech. It includes some advertising copy for the Ebtech Hum Eliminator along with the information about ground loops. I usually avoid articles with blatant advertising, but this is an exception because a) the information on ground loops is so good, and b) their product is really quite useful in the right application.

What is a Ground Loop?

When you hear hum in an audio system, it's almost always caused by a loop antenna effect between two or more pieces of gear, across signal lines. A loop antenna is formed by having a loop of wire where the beginning and end of the loop are connected - the loop can be any shape. The loop antenna(e) is basically a form of radio antenna and they tend to pick up the 60Hz AC signal being broadcast by a building's electrical wiring. They also pick up 120Hz, 180Hz, and all the other harmonics of 60Hz and, usually to a lesser degree, electrical noise being broadcast from all over such as radio/TV, hair dryers, etc. These loop antennae are closed circuits usually through the ground wires and hence are commonly called ground loops.

Examples of Ground Loops:

1. Going up the AC power cord ground from the electrical system wiring to a keyboard, going across a signal line ground from the keyboard to a mixer across the signal ground, down the mixer's power cord ground reconnecting to the electrical system wiring.

2. Going across the signal ground from a mixer to a reverb unit, going from the reverb unit across the signal ground back to the mixer and reconnecting inside the mixer.

3. Going up the AC power cord to the mixer, across the signal ground to the amplifier, down the amplifier's power cord ground and reconnecting to the electrical system wiring

4. Going up the AC power cord to a guitar amplifier, going across the input signal ground to an effects device left channel output, from the effects device right channel output to another guitar amplifier, down the second guitar amplifier's power cord ground and reconnecting to the electrical system wiring.

Which connection has the Ground Loop? (AKA Playing Audio Detective)

Identify the ground loop causing the trouble; not all ground loops cause noise or hum. For complex systems you may need to repeat these steps starting with a different piece of equipment in various combinations to locate the problem:

1. Strip the system down to one piece, such as the mixer, by disconnecting all interconnects and AC cords except for the mixer.

2. Add one piece of equipment at a time; hook up AC and interconnects (making sure all grounds are connected and in good condition) then listen for hum or noise.

3. Turn on and off the power each time you switch equipment to avoid pops and shorted outputs.

4. Proceed until you find the offending piece(s) causing the problem.

5. Plug the Hum Eliminator in all lines between the offending equipment and the rest of the system. For example ... insert the line outs of the keyboard into the inputs of the Hum Eliminator, then insert the line outs of the Hum Eliminator into the inputs of the mixer.

It is often helpful to listen through a pair of headphones. Quite often you will only hear hum coming from a particular input channel on a mixer and that is where the ground loop will be. Alternatively, if you hear hum coming out the speakers with all the mixer's channels turned down, it's likely that the problem is between the mixer and amplifier or other equipment that comes after the mixer.

Another common path for ground loops is through a chassis into the rack and then into another chassis. Test this by removing the chassis from the rack. The Hum Eliminator will help but you should also try isolating the chassis from the rack with electrical tape and insulating the rack screws with nylon washers.

Note: Never use the Hum Eliminator between an amplifier and speaker or the equipment may become damaged. Only use the Hum Eliminator on line level signals.

What about clipping or lifting the AC ground or signal ground?

While these methods may or may not remove your hum, they have some real drawbacks!

Removing or disabling the AC ground:

· Can cause electrocution

· Can cause distortion due to floating signal references

· Can cause some pieces of equipment to oscillate or become damaged

· Can cause current and noise meant for the AC ground to be dumped down the interconnect (line level) to another piece of equipment instead

Cutting the shield at one end of the interconnect cable:

· Can hinder the ability of the cable to serve as a signal return

· Can cause distortion and/or clipping of the signal since there is no voltage translation matching (shifting a signal to match ground and power supply).

· Can alter the cable's frequency response.

· Can defeat the shielding effect.

Why using the Hum Eliminator is the safer and better solution:

The Hum Eliminator is completely transparent; its audiophile quality components don't change your sound. With a flat frequency response from 20Hz to 70kHz (way beyond the range of human hearing) the Hum Eliminator is the answer.

The Hum Eliminator breaks the ground loop, keeping all AC grounds intact. It provides isolated signal returns and performs automatic voltage translation matching.

The Hum Eliminator automatically converts from unbalanced to balanced without signal loss. With the Hum Eliminator you can run a signal across a room from a pre-amp, effects unit or keyboard without picking up AC hum from power cords and without the signal loss you get from a DI box. Balanced outputs from the Hum Eliminator benefit from true common mode rejection (CMR), canceling out noise from AC power cords and other sources.

The Hum Eliminator will match any ground potential difference between two pieces of equipment. If the ground of your keyboard is 6 volts higher than the ground of your mixer, the Hum Eliminator will shift the entire signal of the keyboard down by 6 volts to compensate without affecting the keyboard at all

Courtesy Ebtech Audio. Used by permission.


See also this article on ground loops.


__________________________________________
David McLain | The Humming Guy | CCI SOLUTIONS
Be seen. Be heard.
PO Box 481 / 1247 85th Ave SE
Olympia, WA 98507-0481
Voice: 800/426-8664 x255 / Fax: 800/399-8273
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An Introduction to Lapel Mics

By Marty McCann

Question from: Ian Stott of Architectural Audio Singapore

"What EQ Settings/curve would you recommend for a lavaliere mic in a difficult environment such as a reverberant church or hall!"

I would recommend a Cardioid pattern lapel mic, even though the pattern is altered when placed on the chest. Some put the mic too far down on their chest. The user should drop their chin to their chest and put the mic directly below that point. The farther down from this point results in even more mid-range chest cavity resonance that colors the sound. This chest cavity resonance and the gain before feedback is a constant battle with lapels (Wireless or hard-wired).

Too often the church customer (who is often rather technically challenged in the first place), tries to resolve the EQ'ing of the lapel mic with the overall main system EQ. This of course is a mistake, because of how it will adversely affect the overall systems performance, i.e., normal vocal microphones and instruments taken direct are negatively affected by the hacking away of the main equalizer in order to chase away frequencies that are problems for the wireless Lav mic only. A dedicated EQ (inserted into the wireless Mic's channel) is the only way to begin to even get any kind of a handle on this application. Even then due to the drastic amount of Mid-range cut necessary to get intelligibility out of the lapel Mic system (this is before feedback suppression), there is often not enough cut remaining in this region for further control of feedback.

Over the years, I have addressed this problem in high visibility, high $'s installations by either using a parametric along with a 1/3 octave to tweak the system, or more recently (since we no longer manufacturer a parametric), I specify ½ of a 2/3 Octave EQ and a 1/3 Octave EQ to process the lapel Mic's channel.

Now here is where the problem is further complicated. In many installations, the lapel Mic is used by more than one individual (sometimes several). Due to the individual nature or timbre of peoples speaking voices, along with the fundamental resonance's of each voice (that is determined both by the vocal chords and the size of the chest cavity), one size does NOT, fit all. The pastor or CEO doesn't understand this at all. At times when it can be determined that certain designated people will be using the lapel system, I have specified 2 (yes 2) CEQ-280a programmable Equalizers, with stored setting for various presenters or speakers.

Now down to the EQ process. Too often the less experienced system integrator or operator will just ring out the mic for feed back. This results in less than desirable tone and intelligibility. EQ for intelligibility first then go for feedback suppression (once again you probably need more than 1 EQ to accomplish both effectively). On the average the required mid-range notch is centered somewhere from 315 to 630 Hz (this is the individual variable) depending on the person speaking. This notch can be two to three octaves wide at the -3dB down points (depending on the lapel mic and user). Because of the small Electret diaphragm and its proximity to the users mouth, there is often more energy above say 8 kHz than is necessary. A variable high cut is a good tool here (that's why I prefer the EQ-31FX over the Q-31FX, the variable low cut is also handy here). Some people's voices exhibit a strong sibilance in the annunciation of Ssss sounds. This of course is mainly at 6.3 kHz on a 1/3 Octave EQ.

When the budget won't allow for two equalizers, one technique is to first start with all of the EQ sliders at the top (this is not a good idea with some cheaper filter designs due to the ripple or poor summing of the filters), then to EQ for tone, followed by appropriate cuts for feedback suppression. With some cheap EQ's this technique would also result in a poor S/N ratio.

While on this subject, we have had tremendous results with the performance of our new PVM-2 wireless headset mic. Because of it's positioning away from the chest and close to the mouth, it needs VERY LITTLE EQ, and can often suffice on the channel strip EQ on a decent mixer. The problem is a lot of people think they look like a Dork with the headset on. In my case, I have overcome the dorky feeling because the end result is soooo much better performance.

Also, some theater productions tape the lapel Mic over the actor's ear or into their hairline (using flesh colored surgical tape). This works well with some of today's smaller Mic elements, such as the PVM-1 Lavaliere microphone.

©2009 Peavey Electronics. Used by permission All rights reserved. Terms/Privacy
__________________________________________

David McLain | The Wireless Guy! | CCI SOLUTIONS
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PO Box 481 / 1247 85th Ave SE
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Voice: 800/426-8664 x255 / Fax: 800/399-8273

email: dmclain@ccisolutions.com
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Why Not Wye? When Combining Two Signals Into One Is Not A Good Idea

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right

image

Wye-connectors (or “Y"-connectors, if you prefer) should never have been created.

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.

A wye-connector used to split a signal into two lines is being used properly; a wye-connector used to mix two signals into one is being abused and may even damage the equipment involved.

Here is the rule: Outputs are low impedance and must only be connected to high impedance inputs—never, never tie two outputs directly together—never.

If you do, then each output tries to drive the very low impedance of the other, forcing both outputs into current-limit and possible damage. As a minimum, severe signal loss results.

“Monoing" Low End
One of the most common examples of tying two outputs together is in “monoing” the low end of multiway active crossover systems. This combined signal is then used to drive a subwoofer system.

Since low frequencies below about 100 Hz have such long wavelengths (several feet), it is very difficult to tell where they are coming from (like some of your friends). They are just there—everywhere.

Due to this phenomenon, a single subwoofer system is a popular cost-effective way to add low frequency energy to small systems.

So the question arises as how best to do the monoing, or summing, of the two signals? It is done very easily by tying the two low frequency outputs of your crossovers together using the resistive networks described below.

You do not do it with a wye-cord.

Summing Boxes
Figure 1 shows the required network for sources with unbalanced outputs. Two resistors tie each input together to the junction of a third resistor, which connects to signal common. This is routed to the single output jack.

Figure 1. Unbalanced Summing Box

The resistor values can vary about those shown over a wide range and not change things much. As designed, the input impedance is about 1k ohms and the line driving output impedance is around 250 ohms.

The output impedance is small enough that long lines may still be driven, even though this is a passive box. The input impedance is really quite low and requires 600 ohm line-driving capability from the crossover, but this should not create problems for modern active crossover units.

The rings are tied to each other, as are the sleeves; however, the rings and sleeves are not tied together. Floating the output in this manner makes the box compatible with either balanced or unbalanced systems.

It also makes the box ambidextrous: It is now compatible with either unbalanced (mono, 1-wire) or balanced (stereo, 2-wire) 1/4-inch cables.

Using mono cables shorts the ring to the sleeve and the box acts as a normal unbalanced system; while using stereo cables takes full advantage of the floating benefits.

Stereo-to-Mono Summing Box
Figure 2 shows a network for combining a stereo input to a mono output. The input and output are either a 1/4-inch TRS, or a mini 1/8-inch TRS jack. The comments regarding values for Figure 1 apply equally here.

Figure 2. Stereo-to-Mono Summing Box

Balanced Summing Boxes
Figures 3 and 4 show wiring and parts for creating a balanced summing box. The design is a natural extension of that appearing in Figure 1.

Figure 3. Balanced summing box using XLR connectors

Figure 4. Balanced summing box using 1/4-inch TRS connectors

Here both the tip (pin 2, positive) and the ring (pin 3, negative) tie together through the resistive networks shown.

Use at least 1 percent matched resistors. Any mismatch between like-valued resistors degrades the common-mode rejection capability of the system.

Termites In The Woodpile
Life is wonderful and then you stub your toe. The corner of the dresser lurking in the night of this Note has to do with applications where you want to sum two outputs together and you want to continue to use each of these outputs separately.

In other words, if all you want to do is sum two outputs together and use only the summed results (the usual application), skip this section.

The problem arising from using all three outputs (the two original and the new summed output) is one of channel separation, or crosstalk. If the driving unit truly has zero output impedance, then channel separation is not degraded by using this summing box.

However, when dealing with real-world units you deal with finite output impedances (ranging from a low of 47 ohms to a high of 600 ohms).

Even a low output impedance of 47 ohms produces a startling channel separation spec of only 27 dB, i.e., the unwanted channel is only 27 dB below the desired signal. (Technical details: the unwanted channel, driving through the summing network, looks like 1011.3 ohms driving the 47 ohms output impedance of the desired channel, producing 27 dB of crosstalk.)

Now 27 dB isn’t as bad as first imagined. To put this into perspective, remember that even the best of the old phono cartridges had channel separation specs of about this same magnitude.

Therefore stereo separation is maintained at about the same level as a high-quality hi-fi home system of the 1970s.

For professional systems this may not be enough. If a trade-off is acceptable, things can be improved.

If you scale all the resistors up by a factor of 10, then channel separation improves from 27 dB to 46 dB.

As always though, this improvement is not free. The price is paid in reduced line driving capability.

The box now has high output impedance, which prevents driving long lines. Driving a maximum of 3000 pF capacitance is the realistic limit. This amounts to only 60 feet of 50 pF/foot cable, a reasonable figure.

So if your system can stand a limitation of driving less than 60 feet, scaling the resistors is an option for increased channel separation.



Dennis Bohn is vice president of research and development for Rane Corp., and is a noted technical writer. For more articles of this nature, see the Rane Pro Audio Reference at www.rane.com, and note that these materials are also available for purchase in book form. Courtesy ProSoundWeb and Rane. Used by permission.

__________________________________________
David McLain | The Connector Guy | CCI SOLUTIONS
Be seen. Be heard.
PO Box 481 / 1247 85th Ave SE
Olympia, WA 98507-0481
Voice: 800/426-8664 x255 / Fax: 800/399-8273
email: dmclain@ccisolutions.com
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Tweaking DSP for Stage Monitors: Tips & Tricks To Maximize Potential

Some problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized

November 17, 2009, by Bruce Main

bruce dsp

In the days when digital signal processing (DSP) first stalked the arena, it was the guys at front of house that had all the fun. They would ignore their comm light for long periods of time while staring intently at asymmetrical crossovers on their laptop displays.

But now that DSP is ubquitous, the guys at the other end of the snake are beginning to experience the joys of audio in the digital domain.

The real beauty of DSP in monitorland is in optimizing wedges and side fills so that little or no equalization is needed to minimize feed back.

You might have a rack full of graphics, but the show will start out with all of them set flat. These units will only be used for on-the-fly feedback reduction or to ”personalize” a mix.

And with DSP, a preset can be designed for each model in your inventory, particularly useful if you have multiple wedge types. So let’s get on to the tweaking part.

Very few monitor cabinets are inherently “flat”. There are compromises made due to factors like box size versus low frequency driver selection, box size versus horn selection, low frequency driver dispersion versus. crossover frequency, etc.

Your favorite microphones also have frequency anomalies, especially in off axis response. Combine the two and things can get quite complicated.

Many of these problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized.

The first step is to achieve flat on-axis response from your monitors. I would highly recommend that if you don’t already own (Gold-Line) TEF, (Rational Acoustics) Smaart, or at least a real-time analyzer (RTA) then beg, borrow or steal some audio analysis equipment for this portion of the procedure.

Measured with TEF, a high-frequency device before equalization. (click to enlarge)

At an AES convention a few years ago, an equipment manufacturer that I have a relationship with set up a loudspeaker, an EQ and a pink noise source in their booth. Passersby were invited to try to equalize the pink noise by ear to achieve a flat response from the loudspeaker.

One person was spot on, but most of the results were pretty scary, even though the participants were all audio professionals. We need accuracy of +/-1 or 2 dB to give us a truly flat baseline to work from.

Some may have the ears to accomplish this, but most of us don’t. If you’re using an RTA, be sure to get one that displays in increments as small as 1 dB.

Again with TEF, the data of a high-frequency device after equalization. (click to enlarge)

Measuring the low frequency response of a loudspeaker minus the room’s acoustic contribution is a somewhat tricky proposition. A 40 Hz wavelength is approximately 28 feet long.

In order to measure those frequencies properly with a time windowed measurement system the window has to be long enough to contain at least one full wavelength.

Unfortunately that means that it is long enough to contain room reflections that contaminate the measurement. At higher frequencies this is not a problem because the wavelengths become short enough for the windowing to provide anechoic measurements.

Because real-time analyzers are time blind they include room reflections at all frequencies.

Testing Methods
There are a couple of ways to deal with this. One is to do your testing outdoors far enough from any buildings to minimize reflected energy.

My favorite approach comes from Don Keele, one of the really smart guys in our industry. He places the measurement microphone about one inch from the center dome of the woofer, takes a measurement, then places the mic one inch from the port, takes a measurement, and then sums the two responses.

The signal-to-noise ratio of the measurement is improved greatly because of inverse square law gains that result from being so close to the source.

A word of caution here: if the device under test has a maximum output of 120 dB at one meter, it’s output at one inch will approach 136 dB. This may be enough to do bad things to your expensive measurement microphone.

So start out at a fairly low volume level and work your way up. This test will give you a good idea of what is going on from 200 Hz on down.

The first parameter I set is a high-pass filter to prevent signals that the box is not capable of reproducing from wasting power and potentially damaging components.

Then correct any large EQ anomalies with parametric filters. The microphone can be moved to normal listening distance for this and all subsequent tests.

Make sure that the distance is at least three times the longest dimension of the box under test. This puts us in the far field. I do this test with no crossover engaged for the low-frequency device.

If you test from 200 Hz to, say, 5 kHz, you get a good idea of the total low frequency response curve of the box.

When you’re done, a configured system should look something like this on the DSP software. (click to enlarge)

Next, look at the upper range of the device’s response curve. There will be an obvious point where the amplitude drops off or the speaker gets into breakup modes represented on the test display by narrow notches and/or peaks.

Use parametric filters to flatten the response as much as possible within the useable range of the device. Choose an upper crossover point for the woofer that filters out the nasty modes and only utilizes the relatively flat part of the speaker response.

The high-frequency test is next. I prefer doing this test with no crossover engaged, however, the sweep frequency must be started at a high enough frequency to avoid damaging the driver.

Check the manufacturer’s recommendation for the lowest suggested crossover point and start your sweep there. If full range pink noise is being used as the test signal, start with the crossover engaged to protect the driver.

Using the parametric filters in your DSP of choice, correct for frequency response anomalies.

If the box is using a constant directivity horn you may need to use a shelving filter to increase the high frequency output above 2 to 3 kHz. Get the response as flat as you can across the full frequency spectrum.

Remember, if you leave a 3 dB peak in the response and it happens to coincide with a 3 dB peak in the vocal mic response, it will cost you 6 dB of headroom.

You should discover an overlapping frequency range where the low- and high-frequency devices are behaving in a fairly linear fashion. The crossover can be set anywhere within that region. As a general rule, if the horn is small, set the crossover towards the high end of the overlap zone.

Larger horn mouths provide pattern control down to lower frequencies, so if the horn is larger, you can set the crossover point lower while maintaining good directivity from the device.

Next, the levels and time alignment between the low- and high -requency sections should be set. With the mic on-axis and centered between the horn and woofer, do a full range sweep. Set the crossover outputs so the average volume level is the same across the entire frequency spectrum.

Then look at the frequency and phase response at the crossover frequency. Pretty ugly, eh?

Using an impulse response or ETC measurement, look at the arrival times for the two devices. Set the alignment delay on the DSP to eliminate the time arrival offset.

Now look at the frequency and phase again. Better? You will need to do a little fine tuning to get the flattest possible phase line.

If you’re using an RTA, this part is harder. Try inverting the polarity of the high output on the crossover. You should see a notch at the crossover frequency. Adjust the delay until the notch is at its deepest. Reset the high polarity back to normal.

If the time offset is correct the notch should disappear. If your RTA has a 1/12th octave mode, it will be easier to see.

Some real time analyzers have a loudspeaker timing analysis feature as well. Using asymmetrical crossover slopes can produce better (or worse) off-axis response.

Experiment with this if you have time, but this magazine isn’t long enough to cover all the possible permutations. Program in some brick wall limiters just before the little red lights on the amps start to dance.

Less From Two Than One?
Save these settings as a preset in your DSP of choice and repeat the process for each type of monitor wedge in your inventory as well as side fills and drum monitor rigs. You can also use these settings as a basis for multiple wedge setups.

But remember that when you use multiple cabinets of any sort, comb filtering will occur because of the time arrival differences. These peaks and notches are non-minimum phase. That means that they are not “EQ-able”. (Is that a word? It is now!)

Because of this, sometimes it’s possible to get less output with two wedges than with one.

But riders being what they are, go ahead and do a preset for dual wedge setups. The crossover and time alignment settings will remain the same, but you may get some summing in the low frequencies.

Use the RTA to check the frequency response because a TEF sweep will be too frightening to look at. If you need a preset for absolute maximum output with a particular vocal mic, try putting it on a stand exactly as if you were setting up for a show. Plug the mic into the test microphone input on the test rig.

The response will be a combination of the speaker under test and the off axis response of the microphone. EQ the response to be as flat as possible. It may not sound pretty, but it will get loud. (At least until the singer cups the microphone, sealing off the back of the cartridge, turning it into an omni).

Voila! A look at the final measurement result, courtesy of TEF. (click to enlarge)

You may also want to do presets for full range response or one with a higher frequency on the low-cut filter for vocal only. Sometimes it is beneficial to attenuate the lows for an acoustic set to avoid exciting acoustic guitars or pianos. Save a preset and switch back and forth at the appropriate times.

With the current crop of DSP devices, it’s not uncommon to find configurations like four input, eight output that work perfectly for either four two-way mixes or two three-way side fills and a cue wedge. Look for routing flexibility and the ability to store lots of presets.

But most of all, listen to the units. Audio quality varies as much with digital equipment as with analog.

It’s too easy to buy this type of device based on a laundry list of features and functions when the most important thing is great sound.

Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems. Courtesy ProSoundWeb.

__________________________________________
David McLain | The Monitor Guy | CCI SOLUTIONS
Be seen. Be heard.
PO Box 481 / 1247 85th Ave SE
Olympia, WA 98507-0481
Voice: 800/426-8664 x255 / Fax: 800/399-8273
email: dmclain@ccisolutions.com
online: www.ccisolutions.com
blog: www.churchsoundguy.com
facebook: www.facebook.com/churchsoundguy
linkedin: http://www.linkedin.com/in/davidmclain
podcast: http://www.ccisolutions.com/podcast
Clearance Bin:http://www.ccisolutions.com/clearance

700 MHz Wireless Changeover: If not now, WHEN?

In August 2008, the FCC proposed a rule change prohibiting all Part 74 devices (this includes wireless microphones, intercom, and monitors) in the 700MHz band shown above as early as February 18, 2009. We are still waiting for the FCC to issue the final rule, and thus a specific date. When the DTV transition was delayed until June 12, 2009 we thought maybe that would be the date. Now that has come and gone and we still have no guidance from the FCC for a ‘vacate by’ date. The new owners of the spectrum are urging the FCC to require Part 74 devices migrate out of the spectrum no later than February 18, 2010. It is not a question of IF a ban will be imposed but WHEN. Technical presentations to the FCC have indicated there is a strong possibility of interference between Part 74 devices and the new commercial and public safety uses. Public safety has already deployed networks in 40+ markets.

If you have equipment in this spectrum, there are several reasons you should be migrating out of the 700Hz band.

  1. You do not want to risk interfering with public safety life saving operations.
  2. Most 700MHz gear for the US market is outdated and you will benefit from new features and bandwidth.
  3. Replacement parts are already becoming obsolete as reputable manufacturers curtailed or eliminated US models operating in this range two years or more ago.
  4. The financial impact will be immediate if you wait till you receive interference or a “violation” notice.

Planning, budgeting and replacing over the next few months will mitigate the burden on your finances. Plus several manufacturers are offering trade-in rebates that expire in December 2009. Links are also shown for manufacturers that can change the frequency range or re-band certain models.

We often get asked, “Will the FCC really come looking for devices operating in the 700MHz band?” The short answer is YES. Unlike other countries, the FCC does not have a fleet of detector vans searching cross-country for violators. Enforcement will mostly be based on complaints received from the new owners of the spectrum. The FCC already has a presence at major events so it is reasonable to assume they will be monitoring the 700MHz band for unauthorized use. The FCC can also utilize selective detection/enforcement localized around a geographic or high profile area. It is extremely important to understand that FCC fines are steep, typically starting at $10,000. (Sending a junk fax or telemarketing to someone on the ‘Do Not Call’ list is $4500 per fax/call.) Should you ever find the Enforcement Bureau ‘knocking on your door’ you are well advised to cease doing whatever brought them there and avoid a repeat visit.

An example of the interference problems you may encounter in the 700MHz band already exists. Qualcomm bought TV channel 55 (716-722MHx) nationwide in a previous auction and have already deployed a video streaming service call Media Flo or Flo TV in 40+ cities/areas. That number is likely to double by the end of 2009 and significantly increase coverage in 2010. Other users of the spectrum are likely to deploy similar broadband transmissions like the Qualcomm signal shown on the left. If is close enough and strong enough you will not find a usable frequency in that channel.

Most users with 700MHz band equipment have/will upgrade to equipment that operates in a portion of the remaining UHF TV channels 14-51 working around DTV signals. Planning is critical to getting the right frequency range for your location.

© Professional Wireless Systems

__________________________________________
David McLain | The Mixer Guy | CCI SOLUTIONS
Be seen. Be heard.
PO Box 481 / 1247 85th Ave SE
Olympia, WA 98507-0481
Voice: 800/426-8664 x255 / Fax: 800/399-8273
email: dmclain@ccisolutions.com
online: www.ccisolutions.com
blog: www.churchsoundguy.com
facebook: www.facebook.com/churchsoundguy
linkedin: http://www.linkedin.com/in/davidmclain
podcast: http://www.ccisolutions.com/podcast
Clearance Bin:http://www.ccisolutions.com/clearance