Recording and Live Sound Tips and Tricks

If you ever have a really important show, tip the sound guy before the show and they will take much better care of you!

This is obviously not some unknown earth shattering revelation, but it may be something that is overlooked.  Many working sound professionals have 5 bands thrown at them per show in a club setting; many more in a festival setting.

The band that shows up with even a $20 and some kind words and appreciation for taking care of them will put that group right to the top of the priority list.  You may also find yourself getting more time to set up, more attention to detail, and maybe even a longer set.

On the flip side of making the sound guy happy, be sure not to piss him off before your set.  This is a relative stranger that will be part of the band that night.  At a minimum, introduce yourself before you unveil your list of demands.  Having an apathetic engineer is going to get your sound the bare absolute minimum of calories burned required to get through the set.

Hopefully this will help a few bands and sound engineers out there to get along just a little bit better.

Growing up in the late 70’s and early 80’s put me right in the middle of the pre and post digital age in my formative years.  As a young guitar player I quickly learned that the old tube technology in guitar amplifiers would give me a much better tone than the solid state technology.  The generation just before me did not have to even make a choice, there just was one option.

Fast forward to today.  The transistor and digital technology have progressed to a point where a $200 device can come pretty close to emulating every boutique and classic amp that you have ever hear of… let alone could afford.

The shift that we are going to see in the next generation should be nothing short of amazing.  Each new breakthrough leads to exponential growth.  I would imagine that in less than a decade even the most discerning guitar amp aficionados will no longer be able to identify the difference between a classic amp side by side against an emulated amp from an affordable device.

Where will that then take us?  Will future guitarists be unphased by seeing a Marshall Plexi in the pawn show window?  Will that bring down prices for us then “old timers”?

I see things already starting to change in recent years.  I have met many new players that have not ever used amps.  Just last week a friend came from out of town and I set him up with a Fender American Strat through a Fender Deluxe Reverb.  The look on his face was like a kid in a candy store (if you can find a candy store any where).

I believe that there is nothing like the viseral experience of a great amp cranked up with the tubes blazing and the speaker working harder than it should.  I guess you would not miss it if you never knew it in the first place.

I know that I will continue to use my amps as well as emulation to get the tones that I need for any given project.  It will be interesting to watch to see how thing progress.  We have already started to see the Line 6 amps out there that are bridging the gap between the two worlds.  Whatever the outcome, it should be an interesting ride.

I would have to say that this is one of the worst titles to an article, but it certainly is a question worth asking. I have recently been interested in reamping. What this means is that the original source material, like an electric guitar, would have its pickups recorded directly via a direct box. After this material is recorded and edited, it would be output through another type of DI box that is designed to convert the line level back to an instrument level as so you may plug a 1/4″ cable from it directly into an amp.

The first few time you do this it seems like the haunted mansion where the amp is seemingly playing itself. I was curious to see if there would be a significant loss in quality in using this method so I have made some examples and I would appreciate your feedback. Why even bother to go through the trouble? This can be a great way as a producer to dial in exactly the sound that you are looking for no matter what rig the session guy shows up with.

Lets say you have an important session and the guitarist shows up with the worst sounding little pawn shop special amp with a blown speaker and spent tubes. Instead of passing out and hitting your head on the console, just run the guitar’s output into a DI and connect the output to his amp so he can hear what he is used to hearing and you will have a raw file to work with through any amp of your choosing at a later date.

Another use of this technique is if you need to track a band all at once and you run out of amp closets. When you really need to keep any bleed out of the overheads because of impending edits, this could well save you butt in the session. Some purists will scoff at the suggestion, but most players will really be interested to see their gear crank out their performance while they are kicking back and relaxing.

Once you have a direct signal on tape or in your DAW, you now have the option of using plug-ins for emulation software as well. There are so many different varieties out there, but most of them will give you several presets that will mimic different classic amplifiers. Some are better than others, but I have been able to get useful sounds with some tweaking out of most that I have sampled.

Below are the test that I have completed to illustrate the different results of these methods. Please excuse the performance.

Original DI guitar signal (Countryman Type85)

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Fender Deluxe Amp (Shure SM57)

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Fender Deluxe Reamp via Radial Pro RMP (Shure SM57)

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Fender Emulation – SansAmp (Champ setting, drive 50%, level 75%)

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Fender Emulation – Amplitube (Warm Clean setting, slight overdrive)

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Marshall DSL 2000 Amp (Shure SM57)

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Marchall Reamp via Radial Pro RMP (Shure SM57)

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Marshall Emulation – SansAmp (JMP-1 Setting, no changes to preset)

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Marshall Emulation – Amplitube (Power Tube setting, added treble, presence, and volume)

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In conclusion, I found that the reamp of a good DI signal is remarkably similar to the original source data. The emulation plug-ins that I used, even with some tweaking, were not too close in getting similar sounds from the presets, but were certainly usable. I would highly recommend use of reamping in any situation that you may benefit from its use. Any loss in sound quality is negligible in relation to the trade off of its usefulness.

These files were created using a Les Paul to a Countryman Type 85 DI which fed directly to a Focusrite pre with no compression, effects, or filtering of any kind. The guitar cabinets were mic’d with a Shure SM57 and left in the exact position with the amplifier settings left untouched once the experiment was started. The SansAmp and Amplitube LE settings were slightly modified from the comparable preset of each actual amp used to closer emulate the direct mic sound as described above. The amps used were a Fender Deluxe Reissue and a Marshall DSL 2000 head connected to a 4×12 vintage Orange cabinet.

1. Think of each monitor as a separate mix than that of the house mix.

If you have four monitors and a house mix, you effectively have 5 separate mixes (4 mono and one stereo). If you are then going to make a recording with a “B” mix… You are then looking at 6 independent mixes. This can all get a bit confusing especially in a hectic environment of 5 bands a night all wanting a sound check and getting to play on time. One thing that should put your mind at ease is that your monitor mixes rarely need to be as complex as the house mix. Usually singers only want to hear themselves, so that is only one source sent to that one monitor. Simple enough, right? This is generally true with most vocalists. Guitars, especially acoustic, are likely to end up in the mix as well. If you have a monitor set up for the drummer, this is usually the most complex stage mix more like that of the house mix. If you think in terms of separate mixes, you are likely to keep this all straightened out in your head while you are in the moment.

2. Give each monitor mix it’s own EQ

This is a very important and overlooked tool that should not be discarded to cut costs. The main benefits to using independent discreet EQ control over each monitor mix is to greatly reduce feedback. This can be done by finding the offending frequencies per monitor location and reducing them. Deductive EQ should always be used; do not boost any frequencies. This is certainly a less is more situation. Also, rolling off the bass around 125kHz and below will make a large difference in clarity. These frequencies are of little help to the musicians on stage and create phase problems much more easily than their siblings in the higher range.

3. Each mix needs it’s own power amp channel

Once you have your own mix and EQ settings, a discreet amp is a must. To have a separate mix then combine them back together completely defeats the purpose. This may sound like a no brainer, but I have mentioned it because you would be surprised what I have seen attempted. There are several amp models that have 4 discreet channels and these would be more that adequate for 4 monitor mixes for a small to medium stage. Generally what you will see in larger settings is 2 channel amps that power 2 mono monitor mixes.

4. Location, location, location

Where you place the monitors will make them more or less effective. You need your performer to hear what they are doing, but you do not want to trap them on stage and you need to be able to give them what they ask for. Wedge floor monitors are the vast majority of what are used. I have see some ceiling mounted wedge speakers that have worked quite well in smaller fixed settings. Side fill speakers are sometimes used in larger stages and are very common for the drummer’s monitor. Speakers in the rear pointing forward are just asking for feedback problems. Make sure that no monitors are pointing anywhere close to directly at open microphones.

5. Treat your stage

Even minor acoustic treatments will make a large difference in what you can do with your monitors. Highly reflective surfaces will make things difficult in making your monitors effective. If your monitor output can bounce off the back wall then to the ceiling and directly back to the mic, you have yourself a pretty serious feedback loop. A club that I have worked at over the past couple years has a large window at the back of the stage. After putting up a furniture pad covering the glass, the monitors were able to be raised significantly for the performers. Basically, a $12 moving blanket solved a major problem. With all the professional acoustic tiles and treatments out there these days the sky is the limit.

I hope that this article will help you get a better mix to your performers resulting in a better overall performance. Also, if a performer can hear themselves clearly on stage, they are far more likely to want to come and play at your venue again.

Recently I was working at a club in Hollywood when I noticed some new labeling on the aux returns of the board. In a very amusing and equally effective conceptualization the reverb and delay were put into stylistic terms that most any music enthusiast would quickly and easily comprehend. The reverb was labeled “Elvis” and the delay was labeled “Lennon”. Anyone who has listened closely to Elvis or the Beatles over the years will know exactly what I what I am getting at. If you still are not quite know sure what I am talking about, check out Elvis singing Fools Rush and John Lennon singing A Day In the Life. They may both sound similar at first, but they are quite different.

Not only does this distinction help you you remember which effect is which, but also the ingrains the style and emotion behind the feel of each. These effects can also be used together. Less is certainly more unless you are going for an obviously over the top effect. If you know individually which each one is doing, then combining them can be quite useful if the song or performance call for it. Those that just crank them both up because they are there usually murder an otherwise good performance or track.

Wikipedia describes the terms as follows:

Reverberation is the persistence of sound in a particular space after the original sound is removed[citation needed]. When sound is produced in a space, a large number of echoes build up and then slowly decay as the sound is absorbed by the walls and air, creating reverberation, or reverb. This is most noticeable when the sound source stops but the reflections continue, decreasing in amplitude, until they can no longer be heard. Large chambers, especially such as cathedrals, gymnasia, indoor swimming pools, large caves, etc., are examples of spaces where the reverberation time is long and can clearly be heard. Different types of music tend to sound best with reverberation times appropriate to their characteristics

In audio signal processing and acoustics, an delay or echo (plural echoes) is a reflection of sound, arriving at the listener some time after the direct sound. Typical examples are the echo produced by the bottom of a well, by a building, or by the walls of an enclosed room. A true echo is a single reflection of the sound source. The time delay is the extra distance divided by the speed of sound.

There are a great multitude of different effects, filters, and processes that can be used on vocals, but these are the two that are used the vast majority of the time. Of coarse there are infinite resources about these subjects on the Internet that you can research until the end of time. I hope with my simplified explanation, you can spend more time experimenting in a live sound environment or the studio and less time reading web pages.

A few great units that will give you these effects are TC Electronic D-Two Multi-Tap Delay and the Lexicon MX300 Reverb. Check them out.

This is a question that seems to have a simple answer at first. The logical conclusion is that there is one microphone for the performer… how hard can it be? If you have ever heard a comedian without an audience, it is a much different experience. In fact, it is almost creepy. You can see now how quickly that conclusion is about to change.

The first thing that we need to do is capture the performer’s audio to a track that is not mixed with any other audio. This will obviously be the focal point of the recording. Secondly and almost as important is the audience. A good crowd can make or break a show. I recently engineered a comedy night at a local club and the audience did not laugh or participate at all. The acts that night were leaving the stage so quickly the night ended almost an hour early. To get a good audience properly mixed into this project you need to mic them and give them their own track as well. The more mics and separate tracks the better, but one signal path will do as the minimum.

The quality of microphones and recording equipment will certainly impact the recording, but getting the mics in the right places will make a more noticeable difference. Take a look at the room before you make any concrete decisions about your mic placements. Make sure that you are getting as little direct sound from the stage PA as possible. The tighter you can get in on groups of people with multiple mics the less bleed from the PA speakers you will be likely to get. If you only have one microphone to capture the entire crowd, get as creative as possible to maximize them and minimize any audio coming from the performer. The more isolation you can create the more you will have to work with when you get it home.

Now comes the mixing. If you are still with me on you will have a minimum of two discreet tracks to work with. Depending on the quality of the audio, there are several different things you can do at this point to give it a little something extra. Compression is a must in the mix stage as well as the final master. Spoken word will take much more compression than modern music. If you have ever listened to talk radio you know what I mean. Even though the setting get a little extreme, that is what most people are used to hearing. Don’t go too far overboard, but don’t skimp. Another thing to think about are some gentle EQ curves to balance things out. A little EQ will go a long way. As a general rule, do not boost any frequency more than 6 dB. If there is a lot of low end rumbling on either track, consider rolling off the bass starting somewhere below 100 Hz.

Now that you have your mix sounding pretty good, if you used more than two mics on the audience to separate tracks, now is the time to do some stereo imaging. Make sure that you check for phase problems. Remember that you are always going to have some bleed from the stage and sometimes they will hit mics at different times and can cause some weird things to happen. Let your ears be the judge and check the signals at different times at different settings.

A final thought about what to do if you are having problems with your mix at this point. If there is just too much bleed and it all sounds like you recorded the thing in a cave, you may be able to us gating to smooth things out. What a gate does, as its name implies, is opens and shuts the mic audio at a predetermined level. For example, this can be used to turn on the audience mics only when there is laughter present. Be careful because you can cut out things that are meant to be there. This technique can also be done manually line by line in your recording software. You just need to decide how much time you want to spend. Just remember when you are setting up, the more you get right on location, the less that you will have to fix later in the mix.

Do you have a band and want to record your band’s shows but do not have a way to do it?  Do you work at a venue but the management will not spring for a recording system.  Are you just a regular person, but you just have lots of source material that you want to archive?  Do you want to do a podcast?  Whatever your needs or expertise, if you want to record just about anything and you do not want to spend much or any money to do it, I may be able to help.

If you have found your way to my site, you are obviously using a computer and are likely interested in music.  As technology seems to progress at a rate that not even the geekiest of us can keep up with, it is pretty safe to say that you have an old computer in the closet collecting dust or know somebody that does.  If you can get your hands on this old beast, most of the battle will have already been faught.   We do not need the fastest computer because it only will need to do one major task.   It will become a dedicated recording machine.  I have a computer that I bought 8 years ago that does the job without a complaint.

At this point, we are going to get a little geeky.  If you are running Windows and the machine seems to be reasonably healthy, there is only one step to take and the fun begins.  The program that we are going to use for this project is called Audacity.  This is a free open source program that is full of options and quite reliable.  I have been using it my laptop for recording live shows without fail… so far.  Click on Audacity to learn about the software and to download it.

Once you are up and running with Audacity run a microphone or 1/8″ line in into your sound card, send in some source material, hit record on Audacity, and you capturing audio.  Make sure that your levels are nice and hot without going over.  After you have a recorded file you can even save them in MP3 format without having to use any other applications.  It is all right there for you.

If the computer that you want to use is not in the best of health, I would suggest that you install Ubuntu as your operating system.  I would highly recommend this for anything on that machine that came before  Windows XP.  With Ubuntu you will have a free reliable open source operating system that will run like a champ even on older hardware.  In most cases it will run faster and more reliably, but there will be a slight learning curve if you are accustomed to using Windows.  The support documents and user groups for Ubuntu are fantastic if you do decide to switch over.

If you are lucky enough to acquire a laptop for this purpose, you will certainly increase the portability of the recording station. The main difference with the laptop is that there is a low likelihood that there is a good sound input built in. This can easily be remedied with a USB sound card that will accept RCA inputs. There are a couple good ones out there, the cheapest of which that I have found to be useful is the Behringer U-CONTROL.

Now that we have a recording machine, we need to know what we are recording.  All the above scenarios will accept a line level input.  This is the output of any standard CD player, a mixer, or even a headphone out jack in a pinch.  If you want to record a microphone, you will need to get a mixer to amplify that signal to a line level.  That being said… go out there and record something.

Equalization is always an interesting topic. I have found it to be completely overused in many cases, completely overlooked in other cases, and completely misunderstood in most cases. To illustrate my point, check the settings in a couple different car radios. The controls are either untouched or the treble and bass are completely cranked. Chances are that somewhere in between, the truth lies. Without getting into the finer points of EQ that may well be debated until the end humanity, I will explain what EQ is, give you a few guidelines on how to use it, and explain the four basic types of equalization that are currently standard issu.

The human ear can hear frequencies from 20-20,000 Hz. Within this range are the changes that we make with our EQ devices. Some devices will effect large chunks of the spectrum centered on a specific frequency while others can be quite surgical and exacting. In simplest terms, if you picture all audible sound as a straight dotted line where each dot represents a frequency, equalization is the ability to change the relative volume of the dots individually without affection the others. Below is a chart showing what ranges of the spectrum instruments generally fall into.

The main types of equalization are Fixed, Sweepable Mid, Parametric and Graphic.

Fixed is the most basic in the list. You have seen this on on most every home stereo system in the past several decades. These are the Treble, Midrange, and Bass controls. Each control is anchored on a specific frequency and when active effects the frequencies on either side of it in a bell shape of a specified range. A large group of frequencies are affected, but it is very easy to use and adequate for most consumers.

Sweepable Mid EQ is usually found on higher end mixers and recording consoles. This is similar to the Fixed EQ with one major addition. The midrange has an extra settings knob. This frequency knob lets you pick the precise frequency range that you want. It is no longer anchored like in the fixed configuration. This is especially useful when you really know the frequency that you want to get at.

Parametric EQ is the big boy on the list. This is for complete control of your EQing universe. These days you are very likely to see it as a plug in used by your recording software as opposed to outboard gear, but the functionality is all the same. Taking the sweepable EQ to the next step, you get a Q control. The “Q” is the bandwidth. You can grab a large amount of frequencies with a “wide Q” or just a handful with a “narrow Q” setting. So dial in the frequency, select the Q, and reduce or boost your favorite part of the signal.

Finally we come to the graphic EQ. This is basically a fixed EQ with 31 separate knobs… but since that many knobs on a single rack space would be inoperable for humans, sliders are used. The frequency spectrum is divided equally against the number of sliders. The most common use of this type of EQ will be at the end of the chain of a live mixing board to deal with the acoustics of the room. Far more often than not, deductive EQ is employed in this situation.

Now that we have the types of EQ out of the way and we know more clearly what EQ is… how do we use it? That is a never ending topic that I do not have time to write. I am not immortal. At least I don’t think so. Generally I would say that a little bit will go a long way. Extreme settings will get you extreme results that can be fun for an effect, but likely useless for much else. Also, you can get amazing results not in boosting the frequency that you want, but you reducing the ones that you do not want.

As a final thought, just do what sounds good. Do some experiments. Have fun learning. There are many that will tell you that you always need to add 4dB of 6k to a kick drum, but maybe you do not want that click of a beater in your kick. Just pay attention and listen to what the song needs and you will do what is right in the end and not what you think that you should be doing.

I am very often asked what is the most important componant of making a good recording. Most people are usually quick to tell you that all you need a quality microphone. I agree in part, but the question is only half answered. Just as important as your microphone, if not more in some cases, is your microphone preamp. The best mic in the world through a poor preamp would have trouble competing with a microphone of much lesser quality through a great preamp.

Wikipedia defines a preamp as: A preamplifier (preamp) is an electronic amplifier which precedes another amplifier to prepare an electronic signal for further amplification or processing.

What the preamplifier does is take the very faint signal from the microphone and amplifies it to “line level”. This is the point where it is ready to be recorded or in the case of live sound, it will be sent to the power amplifier where the signal will be boosted for a second time to power the speakers. Inside of every home stereo unit there are preamps and power amps all included inside the same box. This may be one reason why the concept of separate components is usually misunderstood as an all-in-one process when we first start out exploring electronics and recording.

When I first started out in recording, I had little concept of preamplification. I was using what was attached to tape machines and mixers with little thought of the actual signal path. At that time when I asked guys in the know how to make better recordings I never got the full answer. After years of trial and error and much research, I finally figured it out. The quality of the mic plus the quality of the preamp plus how well they work together will get you your optimal signal.

There are so many options out there for you these days and many are quite good. Generally speaking, it is safe to say that you get what you pay for. The market is pretty good at dictating the actual worth of these devices. Sometimes if I am looking to purchase a particular unit, I will check out eBay to see if people are getting rid of them in mass or if you can’t find them anywhere because people are not willing to let them go. Also, great unbiased user generated reviews can be found at Harmony Central.

As a final thought to add another level of complexity to what can be a confusing issue, adding a quality compressor after the preamp before your recording medium will ensure that your levels are within the proper range, strong, and clean. Compression is somewhat of a black art and there are several great articles online that will get into all the miunte details. A quick Google search should get you what you are looking for. As a simple guideline for using compression while tracking: A little bit goes a long way. You can always add more later, but you cannot take away what is already there.

I hope that this article will help many of you who are still only getting half the answer as I had for so many years. Please let me know if this was a help to you.

If you are using a computer to record sound, you are like the majority of us these days. With the great quality, endless editing possibilities, readily available plug ins, all at an increasingly lower price… it is a no brainer. We are certainly living in the digital age.

Many of us started tinkering with the sounds of our early personal computers. For the first time we could load up sound effects, simple synth programs started popping up, and privative manipulation of sound files were given to the masses.

Now that we have far more sophisticated systems these years later, all of those old toys have been moved to the digital basements of our hard drives. Why not break them out and see if they have some life?

I recently used a USB keyboard to trigger a simple synth program, ran it out of the computer’s sound card, and then into Pro Tools. It was exactly what the song needed. We ended up leaving the Roland keyboard in the box.

Another great use of the old with the new could be utilized in podcast production. With the recent explosion of this new material on the Internet, some of those old folders full of sound effects that you collected and was not sure what you would ever do with… set up clips to play on the fly during a broadcast. I am sure that you have heard the cleverly placed sound effects on many radio talk shows. Why not incorporate them into you own show? You have the technology!

If you have an old computer laying around that you could dedicate to simple sound output, you would have your ideal situation. Many do not, but you can push your computer to work double duty. The computers that are coming out today are really powerful enough to pull the load. Remember I am talking about the programs of several years ago here. If you try to get your system to run a CPU heavy sequence on top of your DAW software… prepare to lose some performances. I am not saying that it can not be done, but I would not recommend it.

I really just like to push things as far as possible or find a new use for what I already have laying around. There are so many possibilities for items that may have seemingly long outlived their usefulness. Look around, you may find that your next favorite toy is a freebie!

Have you ever seen an acoustic guitar connected directly to a mixer? Have you seen a 100 foot guitar cable? Not usually. Most likely that situation would create an antenna broadcasting more AM radio signals than guitar signal. What is one to do to make the reach all the way to the mixer while maintaining pristine acoustic tone?!

This is where the DI box steps in. Years ago when I discovered this black art, I couldn’t believe that nobody had mentioned it to me earlier. What this device does, in the simplest possible terms, is it takes a high impedance unbalanced signal (guitar, bass, keyboards, etc) and converts it into a balanced low impedance signal typical of that of a microphone. Why does that make any difference?

I glad that I asked myself that question. Let me explain. The real magic happens with the switch off to a balanced cable. If you look at a microphone XLR cable, you will see that there are 3 prongs. The advantage in the 3 conductor setup, as opposed to the 2 conductor, is that it can send two lead signals and one ground. Through some rather simple but extremely clever engineering, the early pioneers of sound engineering figured out that that any noise introduced in the cable could be almost completely removed when that noise is flipped out of phase with itself. Rarely will you see and unbalanced guitar cable at a length of 20 feet, but mic cables connected through a snake can easily run 150 plus feet with very little undesirable added noise.

There are many different DIs on the market of varying features and quality. The heart of this device is the transformer. The better the quality of the transformer and connectors the better the quality of sound you should expect. I myself and a purest when it comes to DIs on stage. I prefer the path of least resistance found in passive DIs with no controls beyond a ground loop lifter and input selector as sound on the Countryman Type 80 Direct Box as pictured above. There are many great active direct boxes with all kinds of imaginable features that should work great in desired situations. It all really depends on what you want to get done. There are some really high end expensive active DIs that I have used in the studio, but I would never put them in a live stage environment!

In conclusion, if you are running acoustic guitars, keyboards, or bass guitars into a PA or recording rig, to get the best signal, find yourself a nice DI.

Surprisingly, most people seem to think that a PA system is really simple to set up. All you have to do is plug a microphone into an amp and hook up the speakers, right? If you are primarily concerned about speech in an auditorium you would be correct. I have seen this setup at countless clubs that claim to be music venues. At this level, a few relatively inexpensive pieces of gear will really make all the difference in the world.

Without getting too much into of the minute details of effects, compression, EQ, crossovers, manufacturers, and speaker designs, I will give you a basic overview of the components of a “standard” music venue setup. (Minute details to follow in future articles.)

Starting from the bottom, lets work our way up. The heart of the operation is the mixer. Make sure that if you are building a system to get all the features that you need from the beginning. It is quite easy to make upgrades later if you do not have to replace your mixer every time to grow. Some examples of quality club mixers are the Mackie Onyx series and the Venice Midas series mixers. Both of these examples have quality preamps for the microphone inserts, excellent routing options, and powerful EQs.

Next on the list is the compression. This is an extremely overlooked part of the puzzle that is very important. This is likely due to the fact that it is widely misunderstood. In simplest terms, compression limits the dynamic range of the signal. What this means (and why it is important) is that you can automatically set the range of the quietest and loudest signals. An overly dramatic example, with an extreme setting, would render a whisper and a scream equal in relative volume. This extreme setting would be defined by sound engineer lingo as squashed. With a more realistic compression setting on individual signals, they all sit better together in the mix and retain some of their original dynamics. Also, using these methods you don’t have to run the system as hot for everything to be clearly enjoyed.

Many compressor units come with built in gates. These are particularly useful if you are using several microphones on a drum set and other percussion. What they do is turn off the mic until there is enough signal that is set to your specification. The benefit is that the fewer “open” mics you have on stage at any given time the less opportunity there is for feedback and other sounds bleeding into unintended microphones. I would be weary using gating when it comes to vocals because it can often chop off the beginning and/or end of words. There is very little more annoying than this to an audience.

A limiter is a more extreme form of compression that will should be positioned at the end of the mix in the signal chain. This will create a brick wall for the entire mix. Any signal that goes past the set threshold will be stopped completely. The setting on this unit should just kiss the slight intermittent peaks. Be forewarned, heavy limiting will completely suck the life out of any mix.

Reverb and delay do not need much explanation. If you are attempting a system of this size you will most likely be familiar with these “echo” and “repeating” effects. Make sure that when you set up these effects and outboard EQs that you are using the board’s inserts and returns. If you are running short on returns, you can use the stereo channels on the board for the effect’s return to the mix.

Headphones are also often missing from the bigger setups. I am still not sure why. The primary benefit in having them is that you can “solo” individual tracks that are only sent to the headphones mix letting you hear what is going on with those particular tracks. This can really help you fine tune certain signals. Remember, it is not what the tracks sound like on their own, but how they all sit together in the mix!

After the limiter I like to use an exciter/enhancer. This will add some EQ and desirable “noise” with a propriety algorithm. Nobody but the designers really know exactly what they do, but your ears will certainly notice the difference. They really do what their name implies. A couple great examples of these devices are the BBE Sonic Maximizer and the Aphex Aural Exciter.

You may have noticed that there are outboard EQs between the mixer on both the main speaker system as well as the monitor system. There are separate reasons for this, but both very important for your overall sound and control. The main speaker EQ is used to compensate for the acoustics of the room. Generally once you make these precise adjustments, you want to leave them alone and lock up the gear so nobody can ruin your hard work. The monitor system requires the EQ to adjust for feedback in the monitors. The preformers on stage need to hear themselves and with the microphones so close to the speakers, there are a few frequencies that may pose a problem. Once you isolate these offending frequencies (different for each room and position), notch them out with the EQ. I highly recommend that you do not boost any of the frequencies; deductive EQ is the way to go in both of these situations.

A subwoofer is an optional addition to a system, but it will put out that low frequencies that many concert goers today are accustomed to hearing in live sound venues. With this kind of bottom end power, people tend to go a little power mad; less is certainly more. Get a good thump going, but do not drown out everything else! Too much bass in a room can cover a vocal and guitar mix more than you would think.

If you do go with the subwoofer option, you will need a good crossover. This sends the low frequencies to the subs and gives the mains a break and puts more sonic energy to the them for the highs and midrange frequencies. You can manually set the cutoff of the frequency split. This decision will depend on the size of your mains and the acoustics of the room.

Finally we come to the snake. This is crucial for getting your sound to and from the booth away from the stage so you can hear what the audience is hearing. You can’t make a good mix if you are sitting with the band. All of your connections will go into the snake and come quite a distance to connect with your mixer. No powered signals should be sent through the snake. You can, however, use the sends on the snake to send the outputs of the mains and monitors from the board to the power amps that should be located near the speakers that they are powering. Make sure to put the amps out of reach of the bands and patrons. Also make sure that your snake will give your enough connections to facilitate everything that you are need to accomplish.

Simple right?! Setting up a live PA system can be quite involved. This is a basic club setup at that. From here we still have microphones, recording setups, and endless effects possibilities. I think that this may be what is exciting about the process. The possibilities are limitless. I hope this may help you if you are attempting to start this project from the beginning or will use this information to upgrade your current systems. Please leave me comments if you find this information useful or have any thought to add that I may have missed.

The worst gear imaginable is not what you want to build your recording setup on, but a few select pieces can add that something special that may have been otherwise missing. I have to admit that I love quirky strange mics, amps, and miscellaneous gear. Pawn shops have been a weakness of mine for quite some time now. I have found things that nobody knew existed that I instantly had to have. Make sure that you understand where I am going with this… for this junk (treasure) to work for you, you will need to have plenty of quality gear as well. I am just talking about adding a little flavor to an already prepared meal.

At this top of this pile, my favorites hands down are tube amps. I am not just about the vintage amps of the 50’s and 60’s, but any tube amp. I had a friend find an old PA amp from an elementary school in the 50’s that was used for radio, PA, and records. He made a modification to the mic input, ran the output to a guitar cabinet, and at full volume with a Les Paul the tone was magic. It didn’t sound like a Fender or a Marshall at all. It didn’t sound like anything that I have ever heard. It was just cool.

Next on my list are cheap and old mics. I got a microphone from the 99¢ store. This thing is horrible. It is not usable in any conceivable traditional manner. It is great through in a two mic setup on a guitar cabinet. It seems to only get that mud frequency right, but with a little EQ and blending with the real mic it can add that little something extra. I also have an old Sony mic from the 70’s that came with an 8 track I believe. I don’t know what possessed me to use it as a kick drum mic, but the upper midrange thump on it was amazing. In retrospect, I would have never done that today, but now thinking about it, I may just use it again in the two mic setup going forward.

Another fun thing is using older consumer electronics in the signal path of your current recording gear. You may have to get all kinds of different adapters, but if you like to tinker a bit and you are looking for something different and fun, this stuff is great. I had a 70’s EQ for a stereo system that had some really amazing tone. I wish I knew where it is now. I have a friend that still swears by his 80’s EQ unit from his stereo from when he was a kid. He still uses it on his masters to this day. We are talking about a piece of gear that was maybe $50 then and most likely about $5 today on eBay.

You never know until you try some of this gear. Not all experiments will come out well and some will even come out completely different than you thought and you may think of a whole new use for it at that time. If you have any great stories about oddball gear, please let me know. I would love to hear them.

We all know what an iPod is at this point. Most of us know that it can be hooked into our home music systems as well as our in our cars. Why is there such a price difference between the professional and/or factory configurations and the “do it yourself” cable out of the headphone jack setups? Is there really a difference? If so, does it really matter to me?

Today I was in one of the large warehouse consumer electronics stores picking up some parts to help out my Father. I overheard a larger man that I believed to be a biker talking on his cell phone about iPod connectors. Once he hung up, I answered his questions and without my asking confirmed that he rides a rather large bike. At one point a store clerk approached us. After asking a simple question the resulting blank face spoke volumes.

What he was trying to do was to output sound from his iPod to his car stereo through a cable and charge the device from (what once was called) the cigarette lighter. His stereo came equipped with an 1/8″ input on the face and the power source was located right below that. Couldn’t be more simple right? There are two issues that may be a factor in this setup which include ground loops and the line out audio vs the headphone out audio (there is a difference). The ground loop occurs when you power the device and the radio from the same source without a proper ground which may ad noise from 60 cycle hum to the engine revving sounds that can make any sane person cuckoo. This can be overcome by a ground loop lifter cable from Radio Shack. The next issue is where you sound source is coming from. The plug from the bottom of the iPod is the ideal source. This is a line level out that has no volume control, but is much cleaner do to the fact that the headphone out has an added amp that will color the sound slightly. Think of it this way… the iPod is amplifying the sound and then the car radio is amplifying the signal once again. Not the best if it can be avoided. The FM transmitter will resolve both issues, but if you are in any medium to large city there will be so much noise on the air, that they are virtually unusable in terms of great quality.

If you want to push the limits to get the ultimate in sound quality out of your iPod, there is a way. Now that the new models are out and you can have 160 GB of data, you have more than enough room for full quality audio files. The device will let you upload full quality (not mp3 format) files. You can have your CDs ripped directly at full quality. You will get a 10th of the amount of songs, but if full quality is your bag… then you are not too worried about that. Now it is time to take the headphones that can with your iPod and throw them in the trash. There are several other options out there, but the ones that consistently come in at the top of the pile are the Shure E4c-n Sound Isolating Earphones. These two tips will make a difference that you will really be able to hear. It will make this popular little device reach beyond what you ever though possible.

Who would win in a fight to the death for sonic superiority when it comes to vacuum tubes vs. transistors? That is a hotly debated and clouded issue. The only way I can give a simple answer to such a complicated answer is this: you get what you pay for. High end gear is high end gear no matter what its guts are. The most common misconception that I have seen is that if there are tubes in a product that it will sound warm and full and in solid state that it will sound cold and brittle. I will attempt to simply explain both technologies and the subtle differences.

Let’s start with tubes. This is the technology that up until the 70’s was in most consumer electronics products. Not much more complicated that a light bulb, almost as bright and hot, is used to control electric currents. They work by creating signals, strengthening them, combining them, or separating them from one another. This big open design will add artifacts to amplified signals that with quality tubes in well designed gear is quite favorable and desirable. As you can see in the picture below that there is a considerable difference in size. From a manufacturing standpoint, you can see why tubes have been pushed to the past and in niche products.

Transistors work in much the same way as tubes but in an extremely compact form factor on a silicon chip. The accuracy of design and lack of artifacts that are inherent in the tube technologies lead the early users of this new gear to believe that the sound was sterile. In all actuality, it was likely too accurate. This new level of clarity without tube coloration not been heard previously. Also keep in mind that the first generation of any new technology does not achieve perfection in its first time at bat. Eventually we got used to the new sound and were started to incorporate the new technologies with old to make an interesting new hybrid.

What does this all mean? If want that tube tone, it will usually cost you. When you see an inexpensive product with a tube built in, it is most likely a marketing ploy to sell an inferior product simply by incorporating the tube in a weak design. Tubes will have more inherent noise in general, but will offer some great warmth and natural compression in audio signals. Solid state devices will give you a better signal to noise ratio and a more accurate signal at a lower price point. Remember that with either technology, you get what you pay for!