Recording and Live Sound Tips and Tricks

(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 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.

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.

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.

After spending thousands of dollars on “industry standard” condenser vocal mics looking for that raw rock vocal sound, I was finally turned on to the secret weapon… the Shure SM7. This is a large diaphragm dynamic microphone that is usually seen in radio stations, podcast studios, and used in many voice over applications. Other great mics in this category are the Electrovoice RE20, the Sennheiser MD421, and the Heil PR-40 but more on them in future articles.

Would you believe that a microphone only costing $350 dollars was used to record the vocals for Michael Jackson’s Thriller?! Its true. You know that if he wanted to splurge on a more expensive mic, the budget for the project would have certainly allowed for it! Bottom line, this is a fantastic mic at an unbelievable price when compared to the results of other professional vocal mics.

The beauty of the Shure SM7 is that it is a dynamic and will not require phantom power and it is highly directional so it will sound great even in an untreated room. Also, it is pretty far from delicate. If you happen to drop it, it will work after you pick it up off the floor unlike ribbon and condenser mics. It does, however, need a good amount of amplification from your preamps so make sure that you have a clean preamp with a decent amount of gain.

Now lets get down to it. I usuall take off the windscreen that comes on the mic (actually they give you two windscreens) and used a standard pop filter. This setup will give you a solid direct signal with a surprising dynamic and frequency range. Check it out, you will be glad you did… especially if you have your eye on mics costing upwards of 5 times the price.