Recording and Live Sound Tips and Tricks

If you did not hear the Home Recording Show #67, check it out.  Here are all of the mix submissions.

Listen to the show

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Here are the mixes:

Ryan Canestro

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Jesse Zoller

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Jon Tidey

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Allen Wagner

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Anssi Tenhunen

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Dave King

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Dave Ondusko

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Dennis Zasnicoff

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Irving Isler

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Mark Bodah

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Mike

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Neil Gray

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William Rigout

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Dave Chick

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Paul Ward

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Steve Currington

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Scott Waldrep

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Craig Sowby – MyOnlineDrummer.com

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Andrea Beltrame

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Ian Rutherford

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Jay Walsh

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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.

Recently on The Home Recording Show I interviewed Kent Holmes.  We swapped out preamp tubes to get different tones out of different amps.  Check out HRS Show #34 to hear the interview for all the details.

Below are all 27 clips that we recorded.  Only 9 of them were able to make the show because of the length of the segment had we used them all.

- The Fender Deluxe Reverb ‘65 Reissue -

Epiphone Rivera (with mini humbuckers)

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Fender American Deluxe Stratocaster (DiMarzio Virtual Vintage pickups)

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Gibson Les Paul Studio

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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- Airline -

Epiphone Rivera (with mini humbuckers)

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Fender American Deluxe Stratocaster (DiMarzio Virtual Vintage pickups)

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Gibson Les Paul Studio

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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- Marshall DSL 2000 -

Epiphone Rivera (with mini humbuckers)

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Fender American Deluxe Stratocaster (DiMarzio Virtual Vintage pickups)

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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Gibson Les Paul Studio

Stock 12AX7 Tubes

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12AU7

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Vintage RCA 12AX7 (50’s or 60’s)

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As I am sure that most modern recording setups have a computer sitting at its core, keeping that computer happy should be a priority.

In last week’s Home Recording Show I outlined the problem areas, solutions, and some preventative maintenance to avoid catastrophe.    Among the tips are to backup often, keep off site backups, reformat recording drives every couple months, defragment system drives a couple times a year, and avoid Internet pollution.

Take a listen to the show.

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Download Show # 15

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.

(For the podcast segment I did on this topic at Home Recording Show, please click here.)

There has been much debate over how similar or how different the Shure SM57 and SM58 are against the GLS Audio ES-57 and ES-58 series of microphones. I have read several reviews and there seem to be two distinct camps forming. One group that believes that the GLS mics are every bit as good or better than the Shure counterparts and those that scoff at them as even being in the same league.

I have worked with both the Shure and GLS microphones at live sound venues. My initial impression was that the GLS mics were terrible. I was expecting to confirm this once I was able to test them side by side in the controlled environment of the studio. I was surprised to find that the GLS mics, while not a Shure replacement, are better than I had previously determined.

Here are a the tests that I conducted in the studio:

The first pair of audio tracks are the SM57 and ES-57 respectively on a clean guitar track:

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The second pair of audio tracks are the SM57 and ES-57 respectively on a distorted guitar track:

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The third group is my voice recorded simultaneously with the SM58, ES-58, and SM58a Beta respectively:

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What I noticed in the test was that the GLS Audio microphones are much more sensitive than their Shure counterparts. Less preamp gain was needed to achieve equal levels. This may be attributed to the base impedance of 300 ohms compared to that of the Shure at 150 ohms. I heard more bite in the higher frequencies and a less defined lower mid and bass response in the GLS mics.  Handling noise is also much less pronounced in the Shure models.

In the studio, if I were looking to get an electric guitar to stand out in the mix with a biting treble character, I would likely grab the GLS before the Shure. In most cases, even after these tests, I would be far more likely to use the Shure microphones for the majority of what I currently use them on.

In the live environment I would not recommend use the GLS microphones. Because of the bump in the sensitively and the higher frequency attack, feedback from the monitors is much more difficult to control. I have used them many times and some of the performers became quite difficult to accommodate. The venue that once used the GLS mics has since upgraded to the Sennheiser e835. My very first show after the switch was completely squeal free.

My final thoughts on this hotly contested topic is that the Shure and GLS Audio microphones are not interchangeable. They have different tonal characteristics, sensitivities, build materials, and specifications. At a price of about $30 each (and even less when purchased in bulk), the GLS mic is a fantastic deal. If you are just starting out in recording and your budget is minimal, this would be a great choice. I would strongly discourage anyone using them in live environments where feedback can be a problem. At the end of the day, I am sticking with my old standby microphones made by Shure.

Test notes:

The guitar tests were made by using a Fender Strat guitar signal recorded using a Countryman Type 85 DI directly into my DAW at 44.1kHz at 16 bit. The exact same signal was sent out via a Radial Pro RMP to a Marshall DSL 2000 into a vintage Orange 4×12 cabinet. The exact location on the speaker cone was marked with masking tape on the grill and the microphone was exactly two inches from the grill cloth. The same cables and signals were used for all tracks. The gain on the Focusrite Octopre was higher on the SM57 to match the output of the ES-57. No compression, filters, or EQ were used.

The voice test was made by setting up all mics very close together as seen in new coverage. I was at 3 inches away from all mics equally. All three were recorded simutaneusly with gain settings to match overall equal levels. The SM58 used the most gain, followed by the SM58a Beta, then the ES-58. No compression, filters, or EQ were used.

I always use compression when I am tracking. Be very careful to make sure that you are not using too much from the start because just like salt on food, you can not remove what you have already used and you can always add more later.

When you compress a signal, in simplest perceived terms, you are taking the loudest parts and reducing them and boosting the quietest sections of the source material. This can benefit you in many more technical ways beyond the scope of today’s show, but it will become obviously useful when it comes time for mix-down.

The basic signal chain that I use when tracking most everything is a microphone to a pre-amp to a compressor and then to my digital audio workstation. Once you have your pre-amp level dialed in the compressor’s settings are next. Depending on what you are recording and what you want to get out of your overall sound will determine what your settings will be. A good place to start is a 4:1 ratio with a medium to fast attack time and a medium to slow release. Make sure that the signal only engages the compressor at the louder passages. The portion of the signal under this threshold will remain unaffected. Use the make-up gain to get the overall signal back to the level of the where the loudest parts were originally.

Imagine the real world difference in the dynamic range between a whisper and a scream. With compression you can reduce that difference. How much you reduce that distance is how heavily you are to compress the signal. With the material you are about to record in mind, you can control the dynamics right from the start to have a dynamic range that will not only be usable on the track, but in some cases, will work at all.

Here are some examples to illustrate my point. The first track here is not compressed at all.

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This track is compressed lightly as I would normally do in tracking.

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This final track is heavily compressed. I would not recommend this at the tracking stage.

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Of coarse I am talking in extremes to make my point, but all sound sources have at least some dynamics. Certainly some have more than others. Human voices and acoustic instruments can vary wildly within a track. Light compression is almost always used in these cases in most studios around the world. If compression is not used in tracking, it will most certainly be used at mix-down, mastering, broadcast, or all of the above.

In conclusion, light to medium compression during tracking will give you a more dynamically balanced signal to work with from the start. You can always add more compression later, but you can not take it away once it is there.

With a geographically undesirable partner for your show, traditional methods will certainly limit your ability to get together unless you have both an ISDN line (service and expensive equipment), already own a private jet, or you like spending an obscene amount of time in the car. There is another way. And get ready for this… it is free! You can use Skype in a way that is similar to having your own personal free ISDN line. As I am sure that you already know, Skype is a free computer to computer VOIP system that is really easy to use and gaining quickly in popularity. I was first introduced to using Skype in this way by Leo Laport who has the largest podcasting network and uses this service on his nationally syndicated radio program.

There are several ways to capture the conversation either with software or external hardware. The most common would be to use a USB headset that comes equipped with a microphone like the Plantronics DPS 400. This device is great for most podcasters and is certainly be a quick and easy solution with great results. There are several software package that are out there that will capture the dialogue and give you an audio file that you can then use a free program like Audacity to add theme music, promos, commercials, announcements, and make edits. The computer you already have, a $50 headset for each participant, a free editing program, and you are there.

That being said, what if you want to step it up a notch? What if you already have a good amount of audio gear and you want to get the most out of it using Skype? This is where the fun begins. Lets say there is a situation where there are three people in three different cities that want to do a high quality podcast. There are two ways this can be accomplished.

The first is that one of the three participants will be in charge of making the recording. This should usually be the one that doing the most talking or is the host of the show. This is because the mic that this person is talking on will go directly to the recording gear at the highest possible quality. Next is to route the two Skype calls to a channel and record them. The two people that are on Skype will be in a conference call and will be able to hear one another just fine. The real trick is to send them your voice without sending their voices back to them creating an endless loop. This kind of mix is called mix-minus. Mix-minus is a particular setup of a mixing console, such that the output to a certain device contains everything except the input from that device. It sounds confusing, and it is… but once you get it straightened out in your head, it really makes perfect sense. In addition to all of that setup, the two people that are not at the base station can really up their quality by using good preamps and microphones that are fed into Skype. This will produce a dramatic improvement over the USB headset mics. I generally recommend a large diaphragm mic like a Shure SM-7, EV RE-20, Heil PR-30 or PR-40, or Sennheiser MD421. I usually do not like the use the condensers in this situation because they pick up a lot of background and room noise. Unless you have a treated room, you will be better off with a  Shure SM57 over a more expensive condenser mic.

Did I mention that there is another way? Indeed I did. If the quality is still not up to par for you with your friends on the Skype end of the line, there is still hope. Have them record their audio from the mics that they are using with no other source material to a track. That track can then be exported and sent to the member that is doing the recording and lined up with the original conversation. This method requires much more time and effort. Some times the files will not add up exactly especially the longer the program. For short shows, there should be much less of a chance for disaster. In this scenario, we have both people from the Skype end sending in full fidelity audio and the mix down would then sound as if all three parties were actually in one room.

Here is a little more to think about. Your Podcast shows will never be send out to the public is high quality wav or aiff files. You will most likely be sending out an MP3 files at 128kb or less. That being said, it will mask some of the quality loss of a Skype call. It really all depends on what your priorities are and how much time that you have to dedicate to your show.

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.

What is this guy talking about?! I promise that it will make sense after I explain. Lately I have been doing a lot of recordings for singer/songwriters. Many of my clients come in with an acoustic guitar and lyrics. From there it is then my job to then build a band around them. I almost always start with a click track, a scratch acoustic track, and a scratch vocal track.

What does any of this have to do with the title of the article? Nothing yet, but hang in there.

Next up is drums and bass. If I have two people to do the job for each, I will record them together so they can lock it down as a team. This is my preferred method, but sometimes it is just me and I have to really plan it out before I get started. This usually requires a bit more editing to make sure everything lines up and swings.

From here, the final acoustic is laid down. Now after all of that, we can now get to my premise. The foundation is laid, now it is time for some flavor. Unless the client has exact specifications what is to happen next, this is where the session gets more interesting.

I am sure that most people who are doing some recording know several musicians in the area that love to play anywhere and everywhere. Give one of these guys a call and have them over. Play them the song once or twice, set up the recording rig, and let them jam out the song two or three times. They are undoubtedly going to miss the transitions of the song because they really do not know it, but you are not looking for a contiguous performance.

This is where you will sharpen your skills as an editor. Take a listen to the phrases that catch your ear and then move them to the part of the song that will be best suited by it. Make sure that you are flexable and work with your client until they are happy. This is not about you! I usually do a full edit my way and then alter it until it suits my client’s tastes.

Using this “hunt and peck” method of jamming and editing, I have come up with some of the most interesting combinations of sounds and sonic flavors than most ideas that were completely clever and intentional. To me it seems to be a simple blend of uncolored unintentional human feel combined with overproducing to yeild a hybrid that resembles neither giving way to something fresh and new that really works.