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 sound engineers primer 
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Post sound engineers primer
I posted this to another forum and had good responses so I am posting here for those with some time and questions on running live sound...

This is the start of a primer on running sound. I started with the basics and will add to this as time permits. Any comments are of course welcome (hey it's a forum right!?!)

So you are walking into your first gig as a soundman. Because you were busy rolling blunts for the band, you did not really have time to check out the equipment before hand. Now they are setting up their equipment and you are standing there unprepared for the night. What to do? This is a little step by step guide to get you through the first set.

THE EQUIPMENT

The basic equipment you are going to be working with are as follows:
1. mixing board or desk known as Front of House (FOH)
2. amplifiers for the FOH speakers
3. crossover (possibly) in a two/three/four way system
4. speakers
5. effects processors (reverb, delay)
6. dynamics (gates, compressor, aural exciters (such as BBE sonic maximizer))
7. mic cords
8. mics
9. stage snake
10. insert cables

If you are at a venue with a house system the initial connections have been made between the stage and the mixer as well as the mixer and the effects and dynamics. I am also going to assume for now that the system is powered up. If it is not see the “More Power Scotty” section. Since this is the common walk in situation we will assume, for now, that this is the case. Your concern is going to be the mixer and the stage snake to start. Once we are finished going through setup, I will explain the use of effects and dynamics. We will get to setting up a full system from scratch after the basics are done.

SETTING UP THE BOARD

Before you get started you are going to preset the board channels to a starting position. We do this because you never know who was at the board last. The majority of FOH boards are laid out the same from top of the channel strip to the bottom:
1 gain knob
2 phantom power button
3 phase reverse button (on some boards but not all)
4 attenuation pad (some boards not all)
5 EQ section (these vary, there can be two/three/four or sometimes five EQ frequencies that you can adjust)
6 AUX sends (usually from two to 8 sends are available for each channel)
7 Pan knob (just like the balance control on your home system)
8 Channel Fader
9 Channel assignment knobs (more on this later)
10 PFL/solo button (this may be above the channel fader or below depending on design)

Each boards channel strip works the same way. The only difference is if the channel is mono or stereo. Mono strips are used for the majority of the instruments. Stereo inputs can be used for keyboards and the like that have two outputs for one instrument. The number of channels a board will have varies. Most clubs have somewhere between 16 and 48 channels depending on the size of the acts they have. If that seems like a lot, film production houses use upwards of 720 channels. Thank God we are not in film!

To preset the board turn all the gain knobs to zero, all aux sends to zero, and all eq’s to zero boost/cut (usually at 12 noon on the knob). Set the pan knob to 12 o’clock (this feeds both left and right equally and in the case of a mono system keeps you out of trouble).

NOTE: many of the systems you will run are mono not stereo. For a mono rig all the pan knobs for the channels should be at 12 o’clock or centered.

Bring the channel faders all the way down to zero also. This is called zeroing the board and gives you a starting point that is the same for every channel. In addition you are going to bring the master faders (these control the output of the board going to the crossover/amplifier/speakers) to zero also. Turn off all phantom power switches. Phantom power is used by condenser mics that require +48 volts to operate. If the board has only one phantom switch and you have a mix of dynamic and condenser mics, don’t worry. Phantom power does not hurt dynamic mics. Phantom power is applied AFTER all the mics are connected to the board but can be turned on and off to test a single mic when needed.

By doing these things ahead of every gig you will always know where the system is at.

Now take a piece of tape and run it across the bottom of the board just below the channel faders. You will use this to organize your channels and give yourself labels as to what is on what channel. A roll of white electrical tape is a must have for a gig. Black tape is very hard to read in the dark.


Now all the pretty lights are glowing and you have an operational system… maybe! The quickest way to find out is to pass signal through the system and see if you get sound. Remember that we are assuming that the whole system is up and ready to give you outputs through the main speakers.

Grab a mic cord and a mic. At FOH plug the cord into channel 1 and the mic into the cord. You will probably have to pull out the connector that is already there. This connector comes from the stage. REMEMBER TO PUT IT BACK ONCE YOU CHECK THE CHANNEL!! On the channel 1 strip press the PFL/solo button. PFL or solo lets you listen or monitor the channel selected. Note that if you press PFL/solo for channels 1 and 2, you will hear both. Many times in the heat of battle you will want to listen to a channels signal. Make sure that the channel you are going to listen to is the only one with the PFL/solo switch pushed in!

OK, you have the mic plugged in and the PFL/solo switch pushed, now grab the gain knob at the top of the channel strip and slowly start turning it up. On the boards meters you will see the signal for the soloed channel starts to come up. Most meters have green, then yellow, and finally red LEDs that indicate signal strength. There are some high-end boards that have meter for every channel but it all works the same way. Keep turning up the gain up as you talk into the mic until the level is just starting to go into the yellow. This is where you get to say the famous words “check one, two” and “hey!”. When you are doing this, give a good shout into the mic. The singer is going to be yelling at one time in the show and talking at other times. You want to know what the signals are going to be at each extreme end and set the gain to account for the yelling part. When the meter goes into the red, you are “clipping” the channel. This makes the sound distorted… not good. If you shout and the channel meter clips, back down the gain. DO NOT TAP ON THE HEAD OF THE MIC! That is a bad habit some guys/girls have and it is not good for the mic. If you need to, snap your fingers in front of the mic but don’t tap.

What if you don’t get any signal? First turn down the gain and try turning on phantom power, the mic could be a condenser. Next check to make sure you actually plugged into channel 1 (maybe you should wait next time to see if the weed is up to the bands standards  ). If that does not help grab a different mic followed by a different mic cord. I always carry a mic cord test box with me. The $20 bucks I spent on it has saved my butt numerous times. The last resort is to change channels and try again. Equipment failure is a fact of life and you need to be prepared for it.

Assuming you are seeing the meters lighting up (or moving if it is an older board with VU meters) it is time to bring the signal through the system.

Go to the master fader(s) and push them up about halfway, slowly.

NOTE: some boards have a left and right fader for the mains. The process is the same for these boards as it is for boards with a single master output fader.

Next to the channel 1 fader will be buttons that usually say something like “1-2, 3-4, main, mono”. These assign the channel and let you send the channel signal to different places. The 1-2, 3-4 refers to what are known as busses. Busses are a way to group a bunch of signals together, say all the vocals for instance, and use one fader to control them. Buss faders are usually located next to the master faders. As a new soundman, I recommend you do not use the busses until you are familiar with how the individual channels work. The button labeled “main” sends the signal from that channel to the main master fader(s). Press this in next to the channel 1 fader, grab the mic, do the “check 1, 2” thing and slowly raise the channel fader. If you start to hear yourself through the main speakers, you are about 75% of the way home. Getting signal out of the system tells you that if you can duplicate channel 1 for each channel you need to use, everything is going to be OK signal wise. Turn off the PFL, bring the channel fader down. You must be exhausted, grab a beer!

More Power Scotty!

Before you can get started you need to power up the system (if it is not up when you arrive). Easy right? Well there is an order that you should use each time you power up. The order is:
1. mixing board
2. effects/dynamics
3. crossover
4. amplifier with the volume controls set to 0dB or there lowest setting.

When powering down at the end of the night, reverse the order starting with turning down the amplifier. The reason you use this order is so that you do not cause a loud pop in the speakers by powering them up first and then the mixing board. Go to the stage or where ever the amplifier is and turn up the volume knobs. Some places have them preset and you can not change them. Some like to run their amps at halfway up, others at full. When in doubt ask first. I turn up to full. Do this slowly to make sure that you are not accidentally already driving the system with an unknown signal from a CD player or some other source.
Some venues have powered monitors for the stage. These are treated just like the mian speakers, power them on last and then turn up the volume on the back of the monitor. At the end of the night, turn down the volume and power them off first with the main speakers.


Tue Apr 22, 2008 2:22 am
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Post Re: sound engineers primer
Now that the system is up and running and we have proved that we can get signal through the board and out to the main speakers, it is time to start populating the channels for each of the instruments. The signals for each mic come from the stage via the stage snake. The snake has three parts, the head, the run, and the fan-out. The head of the snake sits on the stage and is usually a square or rectangular box. On it are a number of female XLR connections labeled 1 through X, were X is the total number of inputs that can be connected. A 24 channel snake has 24 XLR connections that take the signal from the stage to the back of the mixing board. It will also have a number of connections that run from the board back to the stage. These “sends” let you come out of the mixer and send signals to your amplifier/crossover/monitoring system without running more cables. The snake is no more than a bunch of mic cords bundled into one cable. Snakes are identified by the number of channels (mic), the number of sends and the length, for example a 100 foot 24x4 snake has 24 channels, 4 sends and is 100 feet long. About the longest snake you want is 150’. Once they get much longer they start to have issues that we don’t need to go into. So really snakes are like big extension cords for microphones.

NOTE:
Some sends are on ¼” TRS (tip/ring/sleeve) connectors which might tempt you to use them to come out of the amp and into speakers like monitors or mains. Don’t. They are not made to handle power.

The fan-out is the end that plugs into the board. The fan-out has male XLR ends on it and each plugs into one channel on the board. Fan-outs have labels that correspond to each mic channel. So if you want the lead vocalist on channel 1, plug his mic cord into that channel on the snake head. I carry a sheet that has channels 1-48 written on it. As I plug in each channel I write down what I put on that channel (guitar – lead) on the sheet and then transfer that to the tape I put on the board. It helps to keep things organized for bigger shows. It also doubles as a reference if I am recording the night and need to know what is where. I keep all of these so that when and if I run sound for that band again I have a starting point. Some venues that have bigger stages have sub-snakes that plug into the main snake. These are usually 4 to 8 channel snakes with no sends. I carry one with me to plug all the drum mics into. This keeps the stage clean and makes troubleshooting a problem much easier. On a 40’ stage with the head of the snake at one side or the other, a sub snake can be very helpful and save you if you have 25’ mic cords!

We are now at the point where I think we need to create a band to work with. Our band, “The Cranky *******s” is a 6 piece band. The line-up is as follows;
4 vocals, 1 lead and 3 backup
2 guitars
1 keyboard
6 piece drum kit – snare, 2 racks, 2 floor, kick
Bass

Lead vocalist does not play an instrument. Back up vocals are keyboards and 2 guitarists.

So we can start by plugging in mics and assigning channels:

CHANNEL INSTRUMENT MIC
1 VOCAL 1 – LEAD KMS105
2 VOCAL 2 KMS105
3 VOCAL 3 KMS105
4 VOCAL 4 KMS105
5 GUITAR – LEAD SM57
6 GUITAR – RYTHMN SM57
7 KEYBOARD LEFT DIRECT BOX
8 KEYBOARD RIGHT DIRECT BOX
9 BASS DIRECT BOX
10 KICK DRUM 1 RE20
11 HAT AT 3031
12 SNARE SM57
13 RACK 1 D2
14 RACK 2 D2
15 FLOOR 1 MD421
16 FLOOR 2 MD421

Vocal channels are usually selected left to right with the exception of the lead vocalist who I like to have on channel 1. The whole band takes no more than 16 channels on the board.

Before I move you should know that there is no correct way to set up the channels on the board. You will find a number of FOH systems set up with the kick drum on channel 1. I feel that vocals, being the most important instrument, should be easy to find. Set the channels up as you please.

Connecting the mics to the snake is as easy as it sounds if you keep track. Once you have populated the snake, the next job would be to set up monitors.


Tue Apr 22, 2008 2:25 am
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Post Re: sound engineers primer
MONITORING, THE REDHEADED STEP CHILD

(no offense to any actual redhead step children in the crowd)

Monitoring is probably the hardest job out their. It is usually the thing that suffers the most and as a consequence, everyone gets pissy. I have always had enough monitor volume to rip the lips off the singer. How you ask? Behringer Feedback Eliminator. I don’t care what people say about this company, this is a great product and a must have for any band doing sound on their own or any sound engineer who wants to eliminate room resonance issues. Once you learn to set up this pup, you will never have to worry about monitoring again. I am fond of using it with a 2 channel 15 band eq. One feedback eliminator will handle two channels and with 2 channels of eq you have complete control of volume and feedback.

Not everyone will heed, or can spend the cash, to get the Behringer right now. So how do you do it without the unit? First, monitor placement is important. The monitor should NOT be directly in front of the mic. Every mic has a polar pattern, the pattern that determines what direction it picks up sound from. ALL mics have a lobe off of the back of them that picks up sound. The spot you what is at 30 degrees off center line. That would be 10 and 2 on a clock. This is the place where there is a null in the polar pattern of 99% of stage mics. When you put the monitor here it points into the ear. Put the monitor back from the place where the singer will stand most of the time about 1 ½ or 2 feet.

NOTE:
I discussing polar patterns of mics, may people know that the lobe that picks up sound from the back of the mic is not nearly as sensitive as the main pattern off the front of the mic. The question is then, if the back lobe is so small, why would it make a difference? The answer is that the big lobe in front is picking up a small output, your voice, and the back lobe is picking up a much hotter signal, the vocals coming out of the monitor. In the end, there is a good chance that the signal from the monitor can have enough strength to drive the mic and create a feedback loop.

Not many club systems have a separate monitor board. Most of the time you will run monitors from the FOH board. The way you do this is through the AUX sends on each channel. Depending on the board, most mixers have from 2 to 8 aux sends per channel. These are knobs on each channel strip labeled 1 through 4 for example. Aux sends come in two flavors, pre fader and post fader. The difference is were the signal that drives the aux comes from. The aux sends that are pre fader take the signal from a point before the fader so that moving the fader does not affect what the signal level is. Post fader auxs take there signal from the fader for that channel so as the channel fader is pushed up the signal to the aux send goes up. The pre fader sends are usually used for monitors and the post sends for effects. Each aux send has a corresponding output on the board. The sends that are on the snake connect to these outputs on the board. The other end at the snake head on stage connects to the monitoring system, be it a powered monitor or to an amp and monitor wedge or to the feedback eliminator an eq and then the amp and wedge. Most venues have some sort of eq available for each monitor channel. Some boards are equipped with a 2 or 3 band eq for this purpose, but not many in the under a two grand price range. You need to have one or the other. You need some type of frequency control of your monitor mix. Monitoring set up without some type of eq is on the edge of impossible. You can help by having the same types of vocal mics for everyone but that will only get you so far.

With the monitors placed correctly and the mic checked to make sure it is working correctly, the next step is to create some feedback. Yep, we want to make those monitors feedback. Here is the general idea:

With the monitoring system up and running (powered up and the volume on the amp or powered speaker turned up) go to the first vocal channel. Each aux send represents one monitor on stage. If you have two monitors, put them on aux send 1 and 2. For our band we have 4 vocals and 4 wedges on stage. At channel 1, start turning up the aux 1 send (the lead vocal channel). Turn the aux send up about a ¼ of the way and listen to see if there is any feedback. If not, go to the stage and talk into the mic. You should hear yourself through the monitor. It will be low but we know that it is working. Now go back to the board. Keep turning up the aux send VERY SLOWLY until the monitor starts to “ring” or feedback. That ringing is the first feedback frequency. At the eq you are going to want to start cutting that frequency by lowering the slider on the eq that is the closest to that frequency. Knowing what the frequency is may take some time to find until you start to recognize and are able to pick out frequencies. More on this in a bit. Once you find the frequency and lower it on the eq the ringing will stop. Continue to raise the aux send until the ringing starts again. It will now be at a new frequency. Continue going back and forth between raising the aux send output and cutting frequencies until the monitors ring at a number of frequencies at the same time. This is the absolute maximum that the volume can be before feedback occurs. You want to control the feedback without cutting every eq frequency. Only cut as much as you need to stop the feedback. Turn the aux send down just a bit below this level and write down or note the position of the aux send. Now go up to the mic and talk into it. if you have a frequency that is feeding back, cut it a bit and continue talking. You should be able to get a good robust signal out of the monitor this way. Also cover the head of the mic and make sure it does not feedback. Take the mic and VERY SLOWLY point it at the monitor face. VERY SLOWLY!!! As you turn the mic you are getting the meat of the polar pattern to pick up the monitor and you may have a holy crap amount of feedback. Again turn down the aux send a bit until there is no feedback. The reason you want to do this is that during the show the singer will almost always drop his or her hand as they pass a monitor and you want to have already dealt with the possible feed back that you can get. You wrap your hand around the head of the mic because only about half of the singers out there (pro’s included) have good mic technique on stage.

NOTE: GOOD MIC TECHNIQUE
A mic should never be cupped in a way that the hand is over the head of the mic. That includes putting your thumb over the screen, or and other body part. When you do this you change the polar pattern of that mic and it becomes more sensitive on the back and sides. The work that you just did to eliminate feedback is thrown out the window because the mic is now reacting differently. Good mic technique means your hand never goes above the windscreen. I have a number of singers who have a hard time remembering not to do this. I take the mic cord and put a loop in it and tape the loop to the mic. This creates a handle that they can slip their hand into and keeps their hand from going above the windscreen. This does make it harder to get the mic into the mic clip but teaches them proper technique. We wrap the mic during monitor checks intentionally so we can deal with the possibility before they actually do it live. Singers who play rarely have this problem.

By following the above you will get the most volume out of the monitors with the least chance of feedback. The frequencies on the eq that you did not touch are the same sliders you want to use to make the monitor sound “right”. Remember that if you raise any of the frequencies you just cut, you have undone your hard work. Those frequencies that are left can help you make the vocals stand out. Here are some helpful frequencies:

300 – 400 Hz, adds chestiness to the vocal
1-3 KHz, helps to cut through the mix, but also a feedback frequency
8 KHz, adds sizzle and helps to cut through stage volume (also the frequency that is associated with sibilance, the “s” sound)

FIND THAT FREQUENCY

How do you find the frequency that the mic is feeding back at? Here are some helpful starting frequencies from low to high:
160 - 200 Hz
900 Hz – 2.5 KHz
4 - 6 KHz

These are good starting points to look for. I also eq out anything below 100 Hz and everything above 10 KHz before I ever start “ringing out the monitors”. You can train your ear by boosting the individual frequencies on the eq until the monitor feeds back and memorizing the frequency. Do this on your own time or before the doors open to avoid a beating by the patrons.

Repeat this for each monitor and the associate mic. You may already see the benefit of having all the vocal mics the same. Once you set one, you can duplicate the eq settings for each monitor and have a close start for ringing the monitor out. Matched monitors and mics is the absolute best scenario along with a … say it together… a Behringer. I will not get into set up for this unit unless someone has one and is having problems or there is a need to know out there in forumland.

Monitors are tough to get right for the fledgling engineer. Monitors are tough for any soundman especially running them from the FOH. Practice and listen. I check monitors between every set, making small adjustments. Talk with the band and ask them what they like and don’t like. Try to get them involved. Once you have rung them out, have the singers try them and give you an idea of what sounds right to them. The better the monitoring, the more comfortable the band is. Singers will not have to scream to hear themselves and they will be momentarily satisfied.


Tue Apr 22, 2008 2:26 am
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Post Re: sound engineers primer
While on the topic of monitoring, for those who are just getting started setting up a system to gig out with, think ahead about purchases. If you are considering buying a small system, look at puchasing speakers that work as monitors too. The idea is to have the ability to move your main speakers to monitoring when and if you upgrade. Monitor speakers should ideally be all the same for each band member. I would suggest something with a 15" bass driver and a high frequency horn. This type of speaker can work as a main or monitor (vocal or drum monitor). There are plenty of good systems out there by Yamaha, Peavey, JBL and the like that will do the job. I would encourage you to try to get a speaker with a 2" high frequency driver rather than a 1" unit. The 1" units never seem to have enough gas if the stage volume gets high. The price difference is worth it in the long run.


Tue Apr 22, 2008 2:27 am
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Post Re: sound engineers primer
So with the monitors up and running lets look at setting up specific instruments. The things that fight you are mic selection, room resonance, and monitor bleed (preamps too in reality). To combat these things we have the eq, dynamics and effects. I would like to start with equalization.
Equalization is probably to second most abused corrective measure in live sound, following compression. Sound Engineers tend to take what they have for tone in an instrument or vocal and boost what they hear until it sticks out of the mix. The problem with this is that they are not addressing what we don’t hear. With eq’ing you are trying to fill in the areas that are weak to make it sound natural. You should add to those weak areas to give a full sound. The second thing you are doing is to put the voice or instrument into the overall frequency spectrum that we have to work with. (HUH?) Let me explain a bit.

The audio spectrum is approximately 20Hz to 20KHz. Our hearing is usually a bit less than this, about 80Hz to 12-13KHz for the average person, less for some, more for others. The difference is almost always in the upper frequency limit. High frequency hearing goes as we get older and is dependant on how we listen to music (volume) and other factors like playing guitar (deaf people) or work (jackhammer operator). Within that audio spectrum we want to place the different instruments and vocals. We try not to overlap one to the other if possible. This gives everyone their own little space. Bass drum, bass guitar and lower end of the keyboard take up the lows (30Hz- 300Hz), guitars and male vocals (300-1.5KHz), female vocals (400-2KHz) snare crack and sibilance (4-6KHz) and cymbals and upper end of keyboards (5-8KHz ). These are not set in stone but approximations. Each instrument has harmonics that cover a wider range, but the meat of the sound is usually around these frequencies. For vocals to cut through the mix, we may eq out certain areas of other instruments to give the vocals space. The higher the frequency, the easier it is to bring through the mix but, unfortunely, the more fatiguing the sound can become. Hot cymbal hits hurt as does the crack of a snare at 5Khz. Bass guitar however does not have the high end but can get muddy with to much in the 120-400Hz range. Any one frequency the sticks out too much becomes the focus of the sound and can cover up everything else in that range.

In addition to frequency there is the volume of the individual instrument. Vocals are the King! They need to sit high in the mix (volume wise). They should sit even with the snare drum and the click of the kick drum. Below is how I like to set volumes from high to low:

Vocal, snare and click of kick
Lead guitar during solos and cymbals
Guitars, keys, and other instruments

All of the above refer not only to the volume but represent the higher frequencies. The lower frequencies in order of volume are:

Kick drum
Bass
Lower end of keyboards/synths.

These are my starting points and will vary depending on the tone of the instrument. If the kick drum is tuned high, I will let the bass guitar hold the pocket down and use the click to pronounce the kick drum. If the bass is playing higher up the fretboard I switch this around. I will make these adjustments song to song. This adds variety to the music and keeps it interesting. You should never feel you have to eq nor should you be afraid to crank up a specific frequency if it sounds right. Always start with the eq set flat, no boost, no cuts. You should try to get your eq’s set before the show and only make small adjustments during the gig until you are comfortable with how it affects the mix. Every eq change has the possibility of creating feedback either through the monitors or the FOH. Experience will tell you how these affects can be avoided. Changes should be done slowly and evaluated before adding or subtracting more. To complicate things even more, every system has its limitations. This will also affect your decisions. A good set of closed back headphones are a must have at every gig. In a loud room it is the only way to hear minute changes to timber and tone. May times I find just a small change is all that is needed to make something stand out.

As I have said before, most times when you can’t hear a specific instrument, and this is especially true of rock, there is a better than 50% chance everything else is too loud. This comes from mixing hundreds of bands and working with dozens of other sound engineers. Most engineers take the approach that if you can’t hear it during the course of the show, you need to turn up. Once you turn up that instrument, there is now a new instrument that can’t be heard. They turn that up and this continues until the mix goes to crap. The only instrument I will usually turn up without thinking is the vocals. They are King after all. Before I grab that fader to turn up the bass, I may listen to the kick drum and see if it is overwhelming the lower end of the frequency spectrum, not leaving enough volume “space” for the bass.

This does not mean that you are not going to turn things up, you will. But think first. A good mix does not have to be loud to be good.

Drums are a separate topic all to themselves. Drums cover a very wide frequency range but the signal is very short in duration, with the exception of the cymbals. Drum volumes from the snare through the kick drum should be even in volume. Cymbals should be a bit below these levels. As the player rotates through the kit, each drum should be heard with the same power and volume. This usually means that as the instrument goes down in frequency, the relative level of the fader goes up. It is much harder to get low end to come through the system than the higher frequencies. I usually need to have the floor toms and thump of the kick a bit higher than the snare to get the same level out. Look at the link below. It has the Fletcher-Munson curves that show that lower frequencies need to be higher to achieve the same loudness as a signal in the higher frequencies. It also takes more power to create lower frequencies. This is way your average subwoofers require a lot more power amp wattage than the high frequency horns in a system.

http://en.wikipedia.org/wiki/Fletcher-Munson_curves


Reverb and compression can easily change the way the signals can sound. I will get to compression next along with gating and address reverb/delay/chorus following dynamics.


Tue Apr 22, 2008 2:27 am
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Post Re: sound engineers primer
Dynamics and Processing:

Most live rigs and studios have a number of possible dynamic processors that are used to control signals. These include gates, compressors, limiters (a form of compression) and aural exciters (BBE, Aphex, and the likes), reverb, delays and pitch correction. All of these and a few more can come in very handy to tame individual instruments, vocals or the main outputs. Here are the primary tools that shape your sound.

Gates:

Gating is most often used to control signals from drums. They allow you to determine how long a signal is heard. The controls let you determine at what level the gate opens, say -6dB, how long it stays open (usually in milli- seconds or seconds) and at what level the gate closes. They work just like a gate on a fence. If you think of a gate in this manner, you can visualize how they work. If the gate opens at too low of a level, any drum hit will open the gate (bass or heavy guitars could too). If it opens to late, the initial attack of the drum can be lost. If it takes too long to close then you are picking up the rest of the kit until it closes. Attack and release determine these parameters. You can also determine how long the gate stays open after the signal has gone below the point you want it to close (the threshold) and whether it closes completely or not. This is called HOLD. There is usually no setting that will work for every situation. For most drums you want a fast attack and release time. This gives you the pop you want. I usually start with a preset (if I am mixing in the box (ITB) and tweek it from there. For hardware gates I start with a fast attack and adjust the release until I don’t have bleed-thru from the rest of the kit. For live work, I use a shorter release time, studio a slightly longer release.

Compression and Limiting:

Most guitar players use some type of compression in their rigs. The same goes for live sound. First it should be understood that the difference between a compressor and a limiter is the ratio. Limiters have ratios over 20:1, compression has ratios of 1:1 up to 20:1. The two main controls to consider are threshold and ratio. The threshold is set so that any signal above this point is compressed, the ratio tells us how much the signal is compressed. It is a way to control the peaks or RMS value of the signal, pushing them down to be closer to the level of the rest of the signal. The two types of compression, peak and RMS act differently on the signals. Peak is only looking at signals that are short transients, fast spikes and the like. RMS looks at the signal average as a whole and compresses everything, including the peaks. Most times, a peak detection is used to keep channels from clipping due to big transients. With larger ratios, you can imagine that the overall level of the signal is now smaller (and quieter) than the original signal but more under control. To bring the overall level back up, there is a makeup gain stage at the output of the compressor to add over all volume. This is how a limiter is used to make the mix louder, more like what you hear on the radio. Speaking of radio, all radio stations have very heavy duty limiters on the main outputs going to the RF amplification stages. This is so that all the songs come through at the same volume relative to every other song and it helps them to control the dynamics of the signal to the amplification stage. They are known as brick wall limiters as very little will go above the threshold that is set. In live sound, a limiter is a good tool to put between the mixing board and the amplifiers to keep control of the output and saves the speakers from receiving any big signals that could damage them. Compressors can be used for vocals to control them, helping a soft singer or a screamer from distorting or being hard to hear. I use them in this way to keep the average output from a vocal channel constant. How quick a compressor works is determined by the attack, which can be set like the gate for faster or slower sources. Also like the gate there is a release that determines how quickly after the signal drops below the threshold a compressor stops compressing the signal. Setting the attack to slow may not react quick enough to the signal. Setting the release too fast can cause something known as pumping, where the processing of the signal can be heard turning on and off. Compression works very well for monitoring as it will give you a more steady output and allow you to get more gain in the monitors.

Aural Exciters and other processing:

There are a number of processors that fall into this category including BBE Sonic Maximizers and Aphex Big Bottom processors. These units work in various ways to enhance the sound by adding harmonics or adjusting phase/amplitude relationships of the signal. There are those in the biz that swear by them and those that want nothing to do with them. My take is that used in small amounts on certain channels/instruments they can be beneficial. I do not like to use them on the overall mix as I find that they impart an artificial sound when used in this way. I do like using the Aphex in small amounts on bass or kick drum to add to the lower end of the frequency range. They can be helpful in digging out the tone of an instrument if you find yourself with a signal that the mic did not give you’re instrument the correct sound. I would rather track the instrument again than try to fix it in the mix, but that is not always possible. I would suggest that before handing the signal off to this type of processing, try working more with eq and dynamics to shape the sound.
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Post Re: sound engineers primer
Nice! I will get these added to our mixing area. I guess I have some reading to do :)

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Post Re: sound engineers primer
cool, I'm gonna go thru one or two a day, theres some great reading there.

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Tue Apr 22, 2008 3:37 am
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Post Re: sound engineers primer
thanks ya'll. Anything you would like to see or any clarification needed, let me know and I will try to answer. I have been mixing live for quite some time so I have seen it all! I'm no Big Mick but have my share of nationals under my belt.. Enjoy! As with everything else, each situation is slightly different but these are the basics, YMMV.

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Post Re: sound engineers primer
Great Read. :clap:

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