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DaveMIX

Dave Muddiman


Last Updated: 12/14/2009

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Sign: Capricorn

City: NEWARK
State: Delaware

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Sunday, December 16, 2007 

Category: Music

----------------- Original Message -----------------

 


 

From:

ONE NIGHT STAND

Date: Dec 5, 2007 8:52 PM

 

 

Hi Dave.

Bob the bass player from ONS. I need professional help. Besides that I also need your professional opinion. Direct boxes for my bass; Any differences between passive and active that are of some significance? Any one better than the other? I'm not looking for any type of effects or any other type of bells and whistles, again unless there's an advantage. The outputs. Should they be XLR or will 1/4" suffice? Cables depending on outputs, but is a 1/4 to XLR sufficient for use ... anytime I'm just looking for something to run from my smaller amp's (Hartke B900) line out to a pa mixer when I'm too lazy to drag the head and 4x10 cabinet. I'm getting old.

Thanks for your help and really, don't feel obligated to respond. I'm sure you may just get enough requests for sound advice and can't get to everyone looking for it free.

Have a great holiday.

Bob



REPLY FROM DaveMIX

"What does a DI do, and how should I use it?" is a question I get a lot, and at first it was hard to answer (in English, especially).  And I'm too busy at a gig to explain this fully, so here is the big mother; everything I know about DI boxes for everyone to read - at their own pace.

DI BOX DEFINED

A "Direct Input" box, or "Direct Injection" box does three things at once:  (1) Impedance matching, (2) impedance conversion, and (3) balancing. 

Transmission of signal involves not only the input and output, but also the cable with its inherent losses.  As runs become longer, the approach to reducing losses changes.  In modern equipment this cable loss is managed, but if the inputs and outputs are out of range, you need a DI (or something) to address the mismatch. Were not talking volume matching (voltage, or power), but impedance matching (the amount of current required to transmit a voltage).

1) Impedance Matching

A DI box has inputs and outputs that match what they should be hooked up to.  Most instruments have a high impedance output (electric guitar, 500-3000 ohms), and should be plugged into a high impedance input (guitar amp input, 2000-10000 Ohms).  The mic input at a console is low impedance (400-800 Ohms) and should be feed with a low impedance source (a microphone, 20-300 Ohms). 

A DI has the appropriate impedance for those connections (input greater than 10,000 ohms, output less than 300 ohms).

2) Impedance Conversion

The DI box must change the impedance.  This is traditionally done with a transformer (passive) or an amplifier stage (active).  Active DIs were very popular when they were introduced in the 80s.

3) Balancing

The DI box must take the unbalanced signal from a guitar and make it appear as a balanced signal at the console.

The familiar unbalanced guitar cable has a single signal or "hot" wire and a wire braided "Shield" wire which completes the circuit (current return) and distributes a "signal ground" to refer the signal. Also the grounded shield completely surrounds the hot wire and blocks out noise (RF). 

A mic output has a shield, a hot (+), and a cold (-) where the cold is the exact opposite voltage as hot.  This reduces wire related problems in several ways.  One way is that the signal is referred between hot and cold, and the shield is not part of the audio signal but is just for isolating the hot and cold signal from noise.  Equal but opposite voltages leaves a sum of zero, reducing losses, and most importantly you get all the advantages of "twisted pair".

DI INPUT TYPES

The countryman Active DI has an input impedance of 1 million ohms.  Adding an active DI will, not alter this way the tone of the guitar/amp reaction.  Using a DI on an instrument not hooked to an amp, will also work, but you have barely completed the circuit, and could have some tonal changes or even background noise.

Imagine you are measuring the temperature of a glass of hot water with a thermometer.  If the thermometer is large, it will cool the water - the process of measuring would skew the results.  Likewise an active DI will skew the sound the least - 1 million Ohms is like a small thermometer.

Transformers, meanwhile, have improved.  Jenson transformers are quite good, and the new Lundahl brand of transformers is even better.  You can now have less distortion and less phase error with the new transformers than you will ever get with active.  The input impedance is lower than a Countryman Active, but it is well within reason with the new transformers.

Active DIs require power for the electronics.  This can be from a battery, phantom power, or both.  Batteries will wear out, phantom power makes a pop when engaged, and not every PA has phantom power. Also some if not all active DIs pop when turned on, which happens a lot since most active DIs turn on when plugged in.  Reliability of the power, and that popping sound are the downsides I see to active DIs.

Cheaper passive DIs have lesser transformers which have an audible effect on the sound of the instrument in both the PA and the instrument amplifier.

DI OUTPUT

The passive DI has a transformer which creates a "transformer balanced" output; a great method for balancing.  Active devices use amp stages to generate equal but opposite "active balanced" signals on the hot and cold output, this method and another electronic method called "servo-balanced" are also very good.

Some active DIs still have a transformer for balancing the output.  Cable Factory makes a DI that is switched either passive or active, but always has transformer balanced output.

A third method of balancing an output is not as good, but fortunately is never used for DIs, because avoiding this shortcoming is what DIs are for.  "Impedance balanced" outputs have an active amp for hot, a passive resistor to ground for cold, and a shield connected to ground - no longer the true definition of balanced.   Most mixers, Mackie for example, have impedance-balanced outputs for XLR outs.  Using a DI after the mixer would give you the improvement of a true balanced output.

GROUND AND SIGNAL GROUD ISOLATION

Before and after a DI, ground is used for a lot; grounding the shield for noise suppression, signal reference in unbalanced systems, connecting the chassis of audio gear together, and safety.  The way all DIs work, the unbalanced input ground doesn't have to be connected to the output ground if a simple "ground-lift" switch is included.  This allows an unbalanced signal shield to "float" while still having a signal reference. 

If you connect a DI and get a hum from a "ground loop", then this ground-lift switch could be exactly what is needed. At the DI's output, the console is a main distributor of signal ground and should always be connected to the electrical earth ground.  The DI chassis and output sections are always signal-grounded to the console. At the DI's input you may have a grounded system (like a bass amp) that creates a loop with the signal ground.  As a rule of thumb, if the device doesn't have an electric plug with ground, then you cant have a ground loop. (do not disable a grounded outlet - if your devices were designed to have them, keep them grounded, and have a ground tester for mystery locations). 

You should always hook up the DI with ground on.  It's the right way and the safe way.  If you have a hum, try switching it off.  Sometimes the hum disappears completely, sometimes it gets louder.  If it gets louder, something is wrong beyond the scope of the DI.  If the hum goes away; that's what the switch is for.

Again, if it's an acoustic guitar and no amp, then you should always be grounded - you can't have get a ground loop from a battery.

SUPERFULOUS FLUFF

The manufactures make DIs with added features.  Ground is required.  But there is more, which I don't advise using.

Something that would destroy most DIs, would be hooking up to an amplifier output (speaker jacks) to get the tone of the amp driving a speaker.  But some DIs come with that option; you must select a switch first and then hook a speaker cable into the inputs - make sure the amp is off Sparky!  The Countryman Active DI does this, some like the "redbox" also had the beginnings of cabinet emulator technology also built into them.  Radial makes a red DI box that does what the red box did.

Some DIs offer bass cutoff switches and EQ (such as the Sans amp DI).  Having musicians set EQ for the house mix is can bring up some weird results, a better and more straightforward way is described next.

WHERE TO ADJUST WHAT

When you want to change your volume or EQ, it does make a difference where you do it.  If the DI is hooked up as standard to the bass amp input, then any changes to the bass amp will not be heard in the PA, while changes to the bass guitar or the effects will be heard in the PA as well as the bass amp. 

So if the adjustment is to meet the needs of changes in the music then do it on the instrument, and if you need to adjust because your perception isn't right on stage then do that at the bass amp.  This is true for any instrument with an amp and standard DI connection.

POST AMP DI

If the DI is after the amp because you have a DI with the ability to be connected to the speaker jacks, then the amp adjustments will be heard in the PA. This is a downside to this configuration.

Another way is to connect to a line out of an amp.  Again all amp adjustments will be heard in the PA, not good.  Still I'd connect the DI there for certain situations, but do you need a DI?  Read on.

NO DI, THE HARD-WIRED OPTION

Of the three jobs of a DI listed in the beginning, number three can be done for under $20.  With a cable and two connectors, you can convert unbalanced to impedance-balanced but you will not have corrected for the impedance mismatch.

Connect the unbalanced hot to the balanced hot, connect unbalanced shield to balanced shield, and connect the unbalanced shield to balanced cold.  You can not ground-lift.  The mismatched impedance will draw too much current, perhaps causing distortion or tonal changes - but the output impedance of today's new gear is dropping, and many instruments can take the load (current draw). Not recommended to long term use, but they are handy when someone asks for more DIs than you can spare.

YOUR CASE

You could run from lineout into the mixer without using a DI.  If line out is from an unbalanced 1/4" connector, then run an unbalanced cable with 1/4" connectors at both ends to your mixer.  All mixers have 1/4" inputs; some are more for line level and some are more for mic or instrument level, but it will work and be fine. 

Unbalanced instrument cable should not be too long, the upper limit is between 20 and 40 ft, but at line level you are okay.  Balanced line level is the output mixing consoles return from a mixing board down a 300 ft snake to the amps, so unbalanced line level to your mixer can go across a stage, or even down a snake.

BRING A SPEAKER

For Rock n Roll, a DI is not enough for bass or keyboards, the musician must also have an amp and speaker cabinet.  I see your Hartke B900 amp also has a speaker which is good, since otherwise you will only be in the main PA speakers.  You should not expect to be in stage monitors with most PAs since bass guitar makes notes much deeper that the voice.  Not only does it expose the vocalist's monitor to damage, it also reduces perceptibility for the singer. 

If you're doing a small acoustic-type show, you can get away with only being in the PA and maybe a bit in the monitors.  Reducing your volume will reduce how much you have to carry - but can you refuse the urge to rock?

DI RECOMMENDATIONS

I currently am buying only Cable Factory passive DIs with Lundahl transformers.  The basic model, Cable Factory DI-PRO-P1, cost under $200, sounds great, and always works.  I recommend it as the best overall DI.  They also make an active/passive DI, the DI-PRO-AP1.  They have other stuff too (they make my snakes) at CableFactory.com (1-888-383-4883), ask for Bob and tell him I sent you.  They only sell direct.

Radial also makes good DIs with Jenson transformers. They make many with feature options galore, many with features not conducive to reliable use in concert in my opinion, but useful in studios for example.  They make a good one for stereo laptop connections.  They also make one with cabinet emulation, and a very basic top-notch.  They make a few budget DIs, which I recommend against.

Countryman's "Type-85" active DI is very good sounding, and I still use them live for any instrument I want to be as pure and distortion-free as possible. They operate on both 9 volt battery and phantom power supplied for a mixing board. BSS also makes a great active DI, as well as Radial and Cable Factory.

I would avoid the cheaper DIs if at all possible.  With the best DIs, it's hard to measure any error even with a very clean a quite measurement machine.  With the cheaper ones it becomes easier to measure and even perceptible.

THERE IS ANOTHER WAY

A good mixing board has good balanced outputs, it also can take an unbalanced input- Hey it's a DI!  Sound engineers are picky about quality so, this is a big move up.  Focusrite, Neve, SSL, and other top live and studio console designers make "channel-strips" where all the features of a single channel are put in a rack-mount unit with transformer-balanced outputs.  You'd get a very good preamp, very effective EQ and maybe compression or limiting.  Some even let you adjust the input impedance to allow tonal variations based on the input circuit. You will lose out on the ground-lift switch, and $1000 - $3000 for the unit.


 

Now, that was a lot, let me know what you're thinking.

-DaveMIX

Currently reading:
Glory Days at Delaware: The Completely Unofficial Modern History of College Life in Newark, DE UD 1987 - 2007
By Darren Kane
Release date: 26 October, 2007
Sunday, March 25, 2007 

Current mood:  relaxed
Category: Music
 Brian McKinney

Brian asks:
My name is Brian McKinney from Dr Longwood -- a band in northern DE (www.drlongwoodband.com)

Anyway, I am in need of your audio expertise. I run a crown xls602d after my tuner and sansamp rack to a basson 4x10 cab. The amp is run in bridged mono and is rated at 1200 watts rms and my cab is rated at 1000 watts rms -- Now my question revolves around clipping. I am running my drive and level on the sansamp at around 6 o'clock and the amp is all the way up as have been told to run it. The other night my amp was clipping -- How is this happening when it has more power than what the cab will handle. The cab doesnt even seem to be breaking a sweat and I am wooried about damaging the cab as I've read clipping will damage the voice coils on the speakers. So do I need an amp with more power or do you have any info on this matter? I've seen you have helped others and would appreciate any insight you could give me. Thanks , Brian


My Response:
You have a good question. A bit difficult to simply answer. I'll begin with the obvious and move towards the stuff that you through time will have to figure out yourself.

#1 Clipping
The clip indicators on your amp illuminate when the amp's output has a different waveform than the input. This can occur for several reasons, but it always indicated that the amp is being run beyond its capacity.

If your equipment is in great shape, the clip light means that the entire amp is being driven beyond its capacity. It is the output stage, the last set of transistors, which will clip first, the rest of the amp should still run clean. This is not bad for the amp, but like you have already been told, clipping waveforms include a huge amount of high and low frequencies far beyond our hearing range, the abruptness of the change really does harm once the speaker has momentum. The worst part is, clipping happens at the amp's maximum output.

So if a power amp ever clips, the speaker needs to be able to handle that condition. For a 1000 watt speaker, if the amp were clipping all the time, I recommend the amp only be rated at 500 watts, or maybe less depending on how the watts are rated.

Clipping could also occur because your equipment is in trouble. If a speaker is subject to excessive peak power (or other power excesses), the speaker's coils will become partially blown - it will work but might draw too much current from the amp's output stage. The excessive current would deplete the available power reserves and cause clipping

#2 Acoustics
I notice your cabinet is new and you used to have two cabinets totaling about twice as high. This means you switched to a set up with speakers only at your feet.

Mid and high frequencies will not be properly reproduced in the area where your ears are. In fact the bass will sound wrong too. Any burst of deep bass requires a very wide frequency response to be accurately reproduced. You may not hear it, but for every burst of deep bass there is a great deal of high frequency content both electronically and acoustically. With only the bass portion reproduced in your listening area, you will have trouble perceiving the bass and it will seem sluggish and slurred.

With perception reduced, you could have tried to increase the power only to get the same poor results. This could explain how you could have not been satisfied even though your amp was clipping power.

Another thing about perception, the ears are dulled by music with excess bass and mid-bass compared to other sounds. We can perceive far more detail in a song with a "flat" response (flat for human hearing as discovered by Fletcher-Munsen in the 1950s). Listen to the radio, you'll notice, nobody is airing any songs with lots of bass, all the sounds appear equal and fresh. So if your area has excessive bass you will have trouble understanding your own playing and the vocals and guitars as well.

#3 Preamp Input and Output Levels
If your electronics are to be perfectly clean, you want no distortion in any stage of your signal chain. You'd get that by having all your "Master Level" and power amp knobs at or near to maximum, and setting the input gain as low as possible while still getting the overall volume you need.

Now, if you wanted some "overdrive" you would set the input gain knob higher until the onset of distortion is reached, but that's going to introduce a problem - with the preamp out and power amp in set at max, and the input stage being driven beyond the max, wont the result still be "beyond max?"

Yes it will. If you overdrive the input, every single wave peak would put your amp and your speaker into their designed maximum. Now, speakers and amps aren't supposed to go into maximum very often, maybe a few times a minute would be pushing it, but wave crests are occurring about 2000 times a second, and if every single one of them is an overload condition, you will be quickly damage your gear starting with the speaker coils.

So, if you choose to get distortion in your electronics, you must do that with your input to the preamp, and you must set your preamp out (or your power amp in) to a reduced level. You must avoid that clip light. Your amplifier is already a bit too powerful for your speakers, so the overload condition must be avoided because your amp is very powerful.

#4 The Speaker Cab
Four 10" speakers and a tweeter. This means there are some passive electronics connected to the tweeter which could fail. The 10" drivers can fail outright, or work partially as I described before, or work a couple of ways that sound horrible. Obviously you speaker sounds okay!

Also the cabinet has no port. The 10" are protected from damage by the airtight seal in the cabinet. Deep bass is hindered from damaging the drivers because that would require a slow increase and decrease in air pressure inside the box. The cabinet would keep the drivers from moving too far and tearing themselves up, but your drivers would still be being pushed by the amps. So work is being done that you can't hear to well. Also the airtight cabinet makes the drivers "springy" and provides more comparative mid-bass tones or "punch" as they market it. Without the airtight seal, your drivers would not be protected as designs require and driver failure will result.

Sometimes a speaker is overused, and has a reduced output at the end of the night, but then will work fine the next night. Some damage has still been done, but your day of reckoning (or re-conning) is yet to come, so just keep using them.

Having your cabinet on a table, or tilted against the back wall will sound much better, but make sure it doesn't slide or fall! The best position, however, is positioned like a vocal monitor at the front edge of the stage pointing right at you. Hard to do with a small stage, I admit, but it is the best.

No matter what you do, don't forget a speaker always sounds best pointed right at your head. If it now has too much treble for you, then turn it down, because it was obviously sending too much treble before where you didn't need it (and that increases reverb and reduces perception).

#5 Preamp XLR outputs to PA System
You should use the isolated output XLR connector for connection to a PA system, and not the other effected XLR connector, which you are probably connecting your power amp with.

This way the preamp's eq controls are set by you to enhance the sound of your bass cabinet onstage (to please you and the band). The eq controls on your actual bass guitar(s) will control the tone of *both* the PA and your bass cabinet (to please needs of the song).

Once again:
Bass Guitar = Controls Bass amp and PA together
Bass Amp = Controls the bass amp only

#6 EQ and Overdrive Revisited

Imagine, we'll assume all frequencies have a perfectly flat response through you whole system (clean or distorted) and the speakers are working at their absolute maximum.

Let's say you are overdriving your preamp's input stage, and every wave is being softly clipped. Now if you turn up the bass EQ *after* the overdriven stage, there will be too much bass going to the speakers, even if you don't play a bassy note because the waves a clipped at one level, but the bass is being increased afterwards - making every waveform have more bass.

In contrast if you were to turn up the bass EQ *before* the overdriven stage the result would be much different. The bassy notes would be more distorted, and the distortion would reduce the bass and increase the treble relative to each other. This would not cause damaging levels to proceed to you speakers if everything is set up right. But notice the EQ before the distortion is pretty much useless. This is one major reason why sound engineers want distortion-free reproduction - so the EQs will work like they should.


Other Issues
Make sure your speaker cables are in great condition, and all connections are solid. You definitely don't want any shorting. Power amps in bridge configuration are more likely to fail due to shorts or damaged speakers. In fact, if one side to the amp fails it will destroy the other very very quickly! (mutual assured destruction?) If your amp goes into "fault" condition you have a failure of your speakers or amp - fortunately that hasn't happened. Make sure you amp can ventilate air - excessive heat in the amp also causes a fault.

Conclusion
Continue using your amp rig as it is, but be careful not to set the power amp in clip. Know in advance if you want to be clean or overdriven and take the steps needed to secure safe operation.

Right now your amp has more RMS power than your speakers and I suspect you amp has much more peak power than the cabinet's 2000 Watt-peak rating, so do not move to another amp without more speakers to handle it first.

Position your cabinet as best as possible - you only play as good as you hear (and rarely does just turning it up let you hear it better).

Use Eq carefully; bass pulses must have mid-range and treble to reproduce the pulse accurately. Avoid turning up the bass and scooping out the mids - professionals never do the scoop eq.

Also try to run you entire system as cleanly as possible, do not add distortion unless you need to for the song because you give up control over your tone when you use distortion. If you do need to use distortion for the song, use the least amount you can get away with.

Best wishes
David Muddiman
Starground Audio
Let me know how it works
And let me know if you have any more questions about this - that could happen!



Brian's Reply:
Wow what a thorough answer. I have read 2 or 3 times and see all of the possibilities -- It will take me time to work this out but my main question was answered -- I don't need to buy an amp with more power.

Thanks so much for the answer I really appreciate it.
Currently listening:
Tentacle Dreams
By Marc Group Klock
Release date: 24 August, 2004
Wednesday, January 31, 2007 

Current mood:  rejuvenated
Category: Music

Lefty Groove, Mad Sweet Pangs and Lefty Groove together onstage

In the last ten years there has been a big drive for cheaper sound and to find less experienced operators to run sound a clubs and restaurants for less money than ever. This is being driven solely by club/restaurant owners who want to retire with a boat and house near the shore, and is directly opposite to the desires and requirements of the best area musicians.

But one thing will always work in the favor of great sound systems and the bands and production people who work with them: Not everyone understands good sound technically, but they always remember their favorite show.


Audience in pleasure-mode


They know what it sounded like when everything came together. Was it the band or the soundman? Well, both must have been perfect for it to sound so good, right?

So in a competitive world, a better sound can make one band better than another, just like far better songs can make one band better than another.

How is sound made better? Well it isn't by creating a new and possibly currently popular sound and forcing it upon the artists. Sound operators must instead know how the music should work and recognize what the band is doing, and make sure that experience works as the distance from the stage is increased.

When the sound becomes more honest, talented artists will improve leaps and bounds once they realize everyone can hear everything, and that their volume war needs a truce.

To do this the operator must understand the system design, and how the designer used the loudspeakers to achieve good sound throughout. But design is often shortcut if anyone not interested in music becomes involved - like a restaurant owner, or future boat owner.

And here's where I come in. Most sound systems in Delaware, as well as everywhere, aren't as good as they shoud be. A serious live band should not accept this, and the touring acts will not. They begin with a technical rider in their contract requiring a very good sound system that meets the bands needs or else they bring in one that meet their needs. This rider to the contract is developed with full input from the live engineer(s) and band.


A Meyer MICA line array - a rider pleaser!


Then the tour's engineer tunes the system. Some are already fine, but most are horribly configured and need lots of help. Sometimes there's something wrong and the tour opens their truck and gets their own gear after all. There is an audio-industry standard for how loudspeakers should be built by manufacturers (phase-tuned for an anechoic chamber), which is not well respected or understood in most houses because this audio technique is counter to intuition.

Since the 1950's live audio scientists have worked on improving this model for great sound. In the last 25 years there have been no detractors, no one who can find any science that phase-tuning for anechoic chambers is wrong - no one!

And I can measure this. I own a Meyer SIM3 room response analyzer - the best in Delaware any many miles beyond. I can measure the response of the room, measure the equalizer(s) attempting to fix it, and measure the result at the same time. Sometimes equalizers don't fix rooms, and the theory for why this is has been around since the 1800's, this machine shows why it doesn't work, and can predict how well the eq will work in that frequency range before you even try it.


Mythbusters tests can amplified human voice break glass using a Meyer UPA-1P speaker and Meyer Sim3 analyzer

Maybe this seems like forward thinking, but it's simple really ... Most people are too cheap to work with the best. But there is another simple truth - No body ever got famous playing live without working with great engineers to overcome this ancient physical dilemma.

The first live show to use computers to further this technology was The Grateful Dead at Red Rocks in 1983. Alignment was performed by Bob MacCarthy, the developer of Meyer SIM and John Meyer the owner of Meyer Sound of Berkeley CA and the designer of most loudspeakers used by the Dead (and all since 1982).


The Dead's Outdoor System 1980'era. I now own one of the speakers in this picture


Other acts/venues to follow include:
Rush
Joni Mitchell
Primus
Les Claypool
Barenaked Ladies
Les Missarables
Frank Sanatra
Velvet revolver
Most Broadway shows since 1988
Dave Mathews Band
Carnegie Hall
The Kimmel Center
Most Broadway-class theatres on the Eastern rim
The San Francisco Symphony
Umphrey's Mcgee


Umphrey's Mcgee with the Meyer system at Bonnaroo

The list is small but all these people are hugely famous for their sound quality. I have worked closely with Bob MacCarthy and John Meyer about the issue of small venues such as those in Delaware, and they continue to seek my input as I am also learning a great deal more from them after my original induction into their solution.

And this is what I do. I make bands sound better. What is better is hard to define for some. You pick up the phone, you expect the voice you hear to sound like a voice on the telephone ... what if it sounded like a voice standing right next to you - shocking huh? Likewise a rare few who listen carefully expect a sound system to sound like they always have heard one - and a few are disappointed when it is different, but they are about 5%, the other 95% says "This was the best show I ever heard - I must see this band again." Because everything instrumental and vocal is in place and "fleshed out."

And that is what I do.

Music that people remember.

As Don Pearson, owner of Ultrasound, the sound company for the Grateful Dead and Santana since 1982 told me "The sound of every show must be perfect" I hesitated and told him live sound was difficult and restaurant owners were cheap - he snapped back at me "Live sound must be perfect! Perfect is the goal! If you don't reach it you failed, your friends, the fans, the band everyone - you must be perfect !!"

David Muddiman
Starground Audio
Newark Delaware
302-593-2621 cell
302-239-7278 home

http://davemix.tripod.com

Anyway I bring a lot to the table much of which I didn't discover, but perhaps the most important thing is friendship.

-dave





Velvet Revolver's 1/2 Million Dollar Meyer MILO Line Array
Currently listening:
The Way Up
By Pat Metheny Group
Release date: 25 January, 2005
Thursday, July 20, 2006 

Category: Music

----------------- Original Message -----------------
From: Chester River Runoff
Date: Jul 19, 2006 7:43 PM

Do you know anything about "feedback destroyers" or the likes of that? You plug them into the effects loop of your PA, or maybe between the mic and PA, and then crank it up and watch as the feedback disappears. Some sort of black magic like that. We want one, but don't know much about them.

sam
Chester River Runoff


----------------- Reply Message -----------------
From: Chester River Runoff
Date: Jul 19, 2006 10:18 PM

I wondered about feedback eliminators a few years ago. After some studying with experts on the subject, and some experimentation with my gear I found they were not needed. Eliminating feedback is just one of many reasons for corrective equalization.

For example, a feedback eliminator will try to correct for when the room makes a frequency louder, but does it correct for when the room makes the frequency quieter? No.

It will fail to reduce frequencies which may not be equalizable with the current system, but there is no way the feeback eliminator looks for this.

A single system (like one monitor speaker) will have many anomilies of both cancellaion and addition of the room with the speaker. Add many systems (mains and monitors of all types) and even if the PA speakers were perfect there would still be many many anomalies. Many closely spaced together and hard to predict.

So the real trick is to have lots of equalization you can control, have a great ear, and know what is the best way to fix each problem.

The way to get better results than "knowing how to wing it" is computerized prediction models based on the loudspeakers used and the room dimensions and equalization applied to each component. Then the follow-through is to measure the system and room response before and after equalization, thereby proving that the system tuning will fix the anaomalies which are most serious for that condition.

Notice that feeback eliminators do nothing to tune or otherwise align the components in a soundsystem - it could even keep you from fixing an obvious problem. And what if your mix is just wrong, the vocals feedback because they are way too loud compared to everything else?  That would be using the wrong tool for the job.

So it's like using a processor or computer to automatically mix a band. You need a human to say "wait, maybe the guitar player WANTS to be soft, I'll wait to see what they do next before I just turn it up" Now, there is automatic mixers for conferences and the like, but not for bands. Same with the feedback eliminators, I'd recomend them for conferences, when they can't afford an engineer on staff and he'd probably be too bored to pay any attention anyhow. Also it's just speach and half assed speakers with little power or presentaion requirements. But no one uses feedback eliminators in professional concerts.

One main problem is in order for feedback eliminators to work they must detect feedback first, which is a physical thing which will be audable. So events with such devices will have occasional feedback; so much for "Eliminating" or "Destroying" the feedback. Anyway you still need a soundcheck and you still need lots of EQ, so once you have the right tools, you shouldn't need a feedback eliminator. If you have feedback with the right tools, you aren't using them well, and that's the underlying problem causing the feedback, not the lack of a magical device.

Number of Feedback eliminators on Broadway Tours: 0
Number of Feedback emilinators in the Kimmel Center: 0

Eliminators or "Destroyers" (a new word to me) are making their money by being put into packages that do other things but "It also has a feedback eliminator" helps get the product sold to musicians at music stores.

My next beef will be the BBE sonic maximizer - it seems nobody knows what it was really made for and people use it on everything, mostly for fixing poorly choosen eq and amp settings, which is not what the device is for at all.

PS don't use a feedback eliminator in an effects loop, theory of proper EQing aside, it's still the wrong way to patch it, 100% wrong. Stay "In Line" - 100 percent of the signal must go through the device, not just a mixed portion.

PPS feedback occurs much more often when there is one person in the band who wants to be louder than the others. Aside from not being able to blend their music correctly anymore, their are many other direct and indirect consequences to the "more me than him" mentality.

dave


Dave Muddiman Owner Starground Audio of Newark DE with Bob McCarthy of St Louis MI, Alignment Engineer of the Kimmel Center, together in Orlando FL in front of Bob's measurement tools many which he developed with John Meyer of Meyer Sound Labs of Berkeley CA

PEACE!!!

Thursday, June 01, 2006 

Current mood:  creative
Category: Music
----------------- Original Message -----------------
From: Evan
Date: May 31, 2006 2:06 PM

> hey i have a band and i'm look for a good pa system
> do u know of any good ones to buy?


------------------- Reply Message -------------------

Oh, you may not want to get me started, I don't know if your question is a trap for me or if it will be one for you, but the answer is very multi-layered.

It should be design for the specific situation with enough allowances to tolerate for extreme physical stresses which live music may present to each component. There are so many parts in an effective system, that knowing what amount of power (how it's measured) can and can't be allowed to go where is very hard to judge and later make fool-proof.

In addition to not blowing up while operating, the systems must have enough acoustical output for your band. Your monitor system should be capable of being 6dB louder than your band when they are playing their loudest. The main system should be capable of being at least 6dB louder than the monitor system. Do the math and you find you need lots of very loud speakers and amps, or else you could consider everyone learning to turn down together, no?

As power is resolved, the tone problem should be resolved at the same time. In order for a speaker to be equalizable it must first have a natural tone. To acheive this all frequencies must propagate from the speaker so that all frequencies arrive at the listener's position at the same time. Otherwise the time slur will cause a frozen-flanger-like sound to alter the tone no matter how hard you try to eq it.

The manufacturers that are recognized world wide for making temporally accurate loudspeakers are Meyer Sound Labs, L'Acoustics, Nexo, EAW and Martin. Some make traditional speaker products for music store retail sales as well. Meyer Sound, considered the best in the world, and top in world-wide sales for concert tours and Broadway makes only top of the line, very accurate and temporally flat speakers.

Since about 97f all distortion heard by the audience is due to anomallies in the speaker/room interreaction, it is designing a system which reacts properly with the room which becomes most important. Also important is durablity, which over time can predictably make a better engineered speaker/amp interreaction save money if the device is used for a significant amount of time.

This was all documented and proven true in the early 1900's. Despite recent restaurant and bar owner's attempts to make us think cheap, the most modern math and measurements continue to prove that cheaply designed speakers are tuned to not destroy themselves at the cost of being recognizably unnatural in sound.

Other important elements are proper wiring and grounding and signal routing, enough inputs for the signals that need to be reinforced, and the completeness and wholeness of the package conceptually. The system's grounds must be properly connected and the whole system must tap to ground at one un-point.

Maybe most important, is the operator must be competent and a very honest listener. And not buying something you don't need - soooo many people mess that one up.

When in doubt, sub-contract the job to a sound professional. Guys like me cut right through the long math and the hype and deliver a service no single band starting could ever afford.

-dave
http://davemix.tripod.com/services/


It may seem like a big PA for an acoustic band, but actually the math shows this is one of few occasions the power ratios are truely right.


Drummers can be very loud so any speaker the drummer listens to must be capable of being louder than him. A tall order.

Evan's First Response
----------------- Original Message -----------------
From: Evan
Date: Jun 1, 2006 3:00 PM

hey thanks right now i have a behringer power amp 2400 watts and i run that through a peavey mixer powered but dont use to power. and my mains i have 2 are peavey tls seris 5x and my monitors i have 2 are peavey tls seris 2x so i think its 600 watts per speaker if thats correct do u think thats enough power. because we play rock.

thanks again,

-Evan

----------------- Original Message -----------------
From: DaveMIX
Date: Jun 2, 2006 8:48 AM

Evan,

Well I can't tell from your quick description, but maybe you can tell by just listening.

Quick answer: You list Watts of your amps, but this does little to help anyone find the acoustical output. Watts are measured in many different ways. For example: 2400 watts could mean 1200 watts peak from two channels into the greatest load the amp can handle, which is about 175 watts RMS per channel into 8 ohms, or 2400 watts could mean 2400 watts RMS into each channel which would be 4800 watts RMS from both and about 19,000 watts peak when adding both channels together. We also need to know the speaker's RMS and Peak capabilities, load, and most importantly efficiency. Thirdly, we should know the cable's length, and cauge to calculate the losses the wire contributes to the power transmission.

Long answer: Give me the model numbers of everything important (you have listed the speakers already, I think), the guage and length of the speaker wire, and I'll look up numbers on the internet and do the math.

Don't forget: If you can't hear everything, it is much easier to turn the loudest stuff down, than buy more gear and try to be louder still.

-dave

Evan's Second Response
----------------- Original Message -----------------
From: DaveMIX
Date: Jun 3, 2006 7:08 PM

ok the mains are http://peavey.com/products/browse.cfm/action/detail/item/108381/number/00571060/cat/110/begin/1/TLS? 5X.cfm
The moniters are http://peavey.com/products/browse.cfm/action/detail/item/116196/number/00574020/cat/112/begin/1/TLM? 2X Floor Monitor.cfm
the power amp is http://www.behringer.com/EP2500/index.cfm?lang=ENG
the mixer is http://www.peavey.com/products/browse.cfm/action/detail/item/111151/number/00510600/cat/88/begin/1/XR® 600G.cfm

But on the mixer it is a powered mixer i dont use the power i just go from the monitor and main outputs on the mixer.

Thanks again,

-Evan




----------------- Original Message -----------------
From: DaveMIX
Date: Jun 11, 2006 4:55 PM

Evan,

Here are the results of my study:

PART 1 AMP

Your amplifier is rated 1200 peak watts max for each amplifier in the maximum load while amplifying a single note. In this way the manufacturer has secured the highest possible number for "Watts" when marketing their product.

Your RMS power would be less for several reasons. First the signal you will be usiing is music, not a single pure tone. Secondly, you probably have two mains and two monitors, so you will be running the amps at 4 ohms, not 2 ohms. And most importantly, the amp is rated for peak power, but you will be using the device dynamicly and peaks will distort if you operate the device with peaks greater than the peak rating.

AES and SMPTE (Audio Engineering Society and Society of Motion Picture and Television Engineers) both recommend a minimum of 6dB difference between Peak and RMS performance or else the listener will suffer fatigue from the unnatural sound. You can hear it too, when vocals sound harsh when important notes are hit, for example. So you're going to learn to use it in a way it won't distort - this means RMS usage will be 1/4 the power of the Peak rating to achieve acceptable listening conditions.

Your amps are rated:
1200 Watts Peak into 2 ohms with one pure note
750 Watts Peak into 4 ohms with one pure note
650 Watts Peak into 4 ohms with music (20 to 20K)

1/4 of that rating is 165 Watts RMS (with 650 Watts peak) from each amp.

Each amp delivers to two speakers which will be driven with 82 Watts RMS (with 325 watts Peak)

PART 2 SPEAKERS

Your mains, the TLS-5X is an 8 ohm speaker with 1/4 inch jacks, rated at 1000 Watts peak and 500 Watts RMS with vocal music (66 to 20k). Same as the math above the speaker's useful RMS output is 250 Watts each. At greater than 1000 Watts Peak, your speaker may become quickly damaged, or the tweeter protection will turn off the highs, or both. A bad place to be.

Your monitors, the TLSM-2X is very similiar but with a 12" woofer. An 8 ohm speaker with 1/4 inch jacks, rated at 1000 Watts peak and 500 Watts RMS with vocal music (68 to 17k). It has the same crossover with protection provided for the tweeters only. The efficiency is not listed but assume the smaller driver, and box make it -2dB the performance of the mains.

The frequency response is the speaker's output, but reflects the desired input as well. Since the speakers are not rated to create deep bass, it would be very unwise to put such sounds into the mix going to such speakers. Turning down the lowest Graphic EQ slider on the XR600G mixer would increase the life of the speaker with the downside being the artifacts of the equalization effecting perceptability somewhat.

Extreme highs can also be harmful. Since the speaker can't produce above 16k, there's no point trying to put a correction in the mix - most rooms fall off into 100
istortion above 14k anyway for many complex reasons beyond this report.

There is no protection for exessive use of the woofers. The tweeters are always ( 99rated for much less power than your woofer, so if you amp goes full power reproducing only a Flute for example, the tweeters will experience far more than their rating. This is the reason for the tweeter protection circuit, just realize it is not a foolproof protection, in fact the protection itself can fail sometimes.

PART 3 ACOUSTIC OUTPUT

Your mains are rated to produce 98 dB at 1 meter with 1 watt.

It's a logarithmic equation, you may be familiar with such math, but here's the easy way to remember it:
10x the watts = 10db

So for your speaker:
1 watt = 98dB
10 watts = 108dB
100 watts = 118dB
1000 watts = 128dB

So your speaker can withstand peaks of 1000 watts, your highest possible resulting acoustic output is 122dB RMS/128dB Peak at one meter from the ideal amplifier.

Your amp delivers 82 Watts RMS (with 325 watts Peak) to each speaker. Which is about 116dB RMS/123dB peak in your situation.

PART 4 ANALYSIS

116 dB RMS is a bit low for rock and roll, there are many other uses which demand less of a speaker than rock. Exercise classes and church sermons for example, so you are in the corner of the industry that really needs all the output speakers can produce. I would recomend a system which produces 130 dB RMS/ 136 dB Peak per speaker to amplify a rock band.

Looks like you should stick to amplifying vocals only and definitely keep bass notes and bass drums completely out of the mix.

Still, the Amplifier's specifications are especially misleading to the point I don't trust them at all. The 650 Watts into 8 ohms has few descriptions of what that measurement means. Lets imagine that it means 650 Watts RMS - your going to have a lot more volume and a lot of failures due to excessive power. 650 watts RMS includes 2600 watt peaks sometimes and that will take your speakers out especially if someone makes a high note all by itself. But I consider it highly unlikely that a company like Behringer would have specs that were generous, far from it. I can assume from how they rate distortion - they are trying to be misleading to new consumers, in my estimation.

If you control levels carefully you can obtain a great sound performance, but don't be assuming your set-up would make every band happy, because some don't like to compromise. Not having the band in the PA can make for some strange issues, like hearing the vocals well at a distance, but not the guitar or just the opposite. Be on the look out for this issue - get a wireless guitar set-up or a long cord and walk around and listen. That's reason 101 for having a soundcheck.

To increase your sound levels to my minimum recomendation would require a new speaker design (or about 14 more same speakers - got room for that?), and a perfect-fit amount of amplification. It would also need to be at least as reliable as you are now, you should have better protection especially with nothing protecting woofers.

PART 5 ADDITIONAL ISSUES

I made a point before that the mains need to be 6dB louder than the same signals in the monitors. Your two amps and two speaker types are rated for the same power. Even if you mixed the mains 6dB louder than the monitors, this relationship would fail once the main amplifier was in overload, and the monitor amp was not. Unfortunately this will be a critical part of the music which should deliver the vocals as clearly as possible so the audience can percieve everything without additional loudness being required.

To do this:
Use 1amp channel for each main
Use 1mixer amp channel for each monitor

This requires additional Anaysis
Your amp
450 watts peak into 8 ohms
115 watts RMS into 8 ohms

Speakers as you use them now:
116 dB RMS/123 dB Peak Mains
114 dB RMS/121 dB Peak Monitors

Your speakers with my new idea:
118 dB RMS/124 dB peak Mains
112 dB RMS/119 dB Peak Monitors


This way distortion doesn't also alter the relationship between mains and monitors. You almost get the right ratio, and you have the effect of adding two more main speakers without the downfall of the destructive interference and amp and transport costs.

If the mains turn off or protect themselves, go back to the other configuation immediately - this is all at a bit of risk because you have little protection at the drivers.

PART 6 YOUR FEE

Rate $30/hr
Hours 8 (minimum hours)

$240 Sub-Total
-$240 MySpace Friend Discount

$0.00 Total



Stay in touch, I'm curious how it turns out.
-DaveMIX
Starground Audio
Newark Delaware


Evans' Third Response

----------------- Original Message -----------------
From: Evan
Date: Jun 11, 2006 5:10 PM

so what ur saying is take the mains and put them into the power one per chanel and use the powered mixer for the monitors. also if i bought another power amp would that help? last what ur saying for the rock band is dont mike the drums or guitar and dont di a bass just use the pa for singing?





----------------- Original Message -----------------
From: DaveMIX
Date: Jun 12, 2006 2:36 AM

1 Yes, that was my final suggestion, based on what you had. Also, since the amp is almost capable of 1000 watts peak, but not into 8 ohms, you get a bit more power but should still be safe. The idea of actually trying to run the maximum every moment is going to end up with enough strain to reduce reliability, you're best to have a safety zone.

2 No, I don't recommend more power without more power-acceptable speakers, and only if the end result will actually be louder which sometimes it isn't. Without changing speakers you get 2dB closer to the max acceptable limits while still having about 3 dB more that the speaker can take compared to what the amp can do. Going any closer to the speaker max isn't not worth doing.

3 Yea, putting it all in the mix, you aren't able to do, barring extremely low stage volumes throughout your show. Plus it's a lot more gear and channels than you may have now. It can add up fast and cost a lot.

My fix, may help, and requires maybe a few wires, or maybe no gear at all. Putting the Behringer Amp in parallel mode is easy. The math was rushed at the end, you will have even less power in the monitors than I stated before, so check it out with an open mind, you may want more power in the monitors, especially when the band practices and you don't use the mains.

Also keep your eyes on overload indicators - Red on the Behringer, and Yellow on the far right of the Mixer, and avoid those conditions.

Tell me how it works.

-dave



Evan's Fourth Response

----------------- Original Message -----------------
From: ROCKSTAR EVAN
Date: Jun 16, 2006 3:43 PM

hi dave,

What do i need to mike drums guitar and bass? and shouls i get the tls serius a a subwofer?




----------------- Original Message -----------------
From: DaveMIX
Date: Jun 16, 2006 3:21 PM

The latest questions have already been answered. The best manufacturers of concert speakers are Meyer Sound Labs, L'Coustics, EAW, Nexo, Martin, and D&B, I forgot D&B they're quite good too. Notice Peavey, Fender, Community, Behringer, Cerwin Vega, EV, JBL are not mentioned because their products are not always good enough for concert use.

It is very hard to recomend a specific product set without knowing the artist's show and the budget involved, it would likely be bad advice. Right now you've done just about all the "impulse buying" you can do (even if you tried real hard to be careful, I call it impulse buying because few music stores can help you, they just sell stuff). Your big choice is to take a big plunge and have a system designed that is complete and meets all your needs at a cost tens of times more than you've ever spent on sound, or just stick with what you've got and reherse and do small shows with it. This way you save a ton of money, especially if your band breaks up. You band might grow faster than you can afford a great system, and hiring sound is a perfect solution for that because you hire companies with enough gear for each type of venue. So you play an arena, you don't have to deplete your funds to buy ten more power amps. Makes sense, yes?

Your current set up is far from ready to handle amplifying drums. You don't have enough equipment (crossover, amp, cables, sub-woofers, mikes, mike cables, mike stands, truck space). You also don't have enough acoustic power from the current equipment and no clear way to upgrade what you have.

I've made a few assumptions about cables, mikes, and your budget. If you're very rich, that dream rig could be assembed in a month. But that's no too likely.

-dave


Evan's Fourth Response



----------------- Original Message -----------------
From: DaveMIX
Date: Jun 12, 2006 1:53 PM

Dave,

Thanks that did help but what do u suggest for a rock band pa system speakers and monitor, subs, ect? also if i did get new equipment whats good stuff to buy? and last for what i have right now i shouldnt mike drums right?

Thanks again,

-EVAN



Evan,

The answer to your question has already been given to you.

Reread what I've already said.

In addition, your system must produce 110dB in the listening area, which will vary, right? Your speakers produce 118dB at one meter, which means after at 2.2 meters you're down to 110dB. 30 feet into the room, the two speakers are below 100dB. So you are way off on the amount of acoustic output needed in the mains system. Why add subs to that? You shouldn't, if that's the case, should you still be wondering.

You, like everyone else, needs the system shown below. But how many of us will do the kind of work to afford it?