Bluegrass music is (usually) based on fully acoustic instruments – i.e. those that don’t require electronics to produce a sound. As such, I’ve always believed that Bluegrass sounds the best when unamplified and in its native state. However, when performing for more than a handful of people, it is not practical to expect a balanced sound to reach the entire audience unless it is amplified in some fashion. Hence, sound reinforcement systems – mics, mixers, amps, and speakers – are a necessary evil in just about all Bluegrass performances.
Achieving a quality result using an electronic sound reinforcement system with Bluegrass music is difficult – much more so than with most other forms of music. There are good reasons for this, and we will cover them in this chapter. We will also cover some basics on gear, setting up sound and performing thru a sound system. First, however, we will cover some basics about sound reinforcement.
We will cover the rudiments of sound systems in this section. If you are even a little bit familiar with sound reinforcement, I’d suggest you skip this chapter and go right to the chapter on electronics and acoustic fundamentals.
From a physical point of view, sound is composed of compression waves that travel thru the air. The purpose of a sound reinforcement system is to amplify these waves and redistribute them in such a way that all the members of an audience can clearly hear the performance. Since no way has been found to amplify sound waves directly through mechanical or physical means, all sound reinforcement systems today use electronics to increase the energy of the sound.
Because electronic components will be used to provide amplification, it is necessary to convert the sound waves to electronic signals. These signals can then be processed and amplified electronically and then converted back into sound waves at higher power. Devices that do these conversions are called transducers and we will be concerned with two types: microphones and speakers.
Microphones convert acoustic energy into electrical energy and work by using a small diaphragm (enclosed in a capsule) that vibrates in response to sound waves. These vibrations cause a change in resistance, capacitance or produce a small, varying current in wires hooked up to the capsule, depending on the type of microphone. We will explore different kinds of microphones and provide more detail on how they work in a later chapter of this section.
Speakers work in the opposite way of a microphone – electrical energy is converted into acoustic energy (sound waves) – with the objective that the sound waves leaving the speaker have higher amplitude (i.e. are amplified) over the ones entering the microphone. Most speakers accomplish this objective by using a coil of wire wrapped loosely around a magnet. When electricity passes thru the coil, it exerts a force on the magnet and this in turn, moves a cone. Since the electrical current passing thru the coil is an exact representation (well, hopefully very close) of the sound that first entered the microphone, the speaker cone will move back and forth in a like manner. In this way, sound waves will propagate from the moving speaker cone that are identical but proportionally larger than the original sound waves picked up by the microphone.
You will undoubtedly encounter mixers and amplifiers in one form or another if you perform live Bluegrass music. Some systems have these two electronic systems as separate components and others integrate them into a single unit. Still others have amplifiers inside the speaker enclosure (and require an external mixer), and these are usually called powered speakers.
Briefly, the function of a mixer is to take multiple inputs (e.g. from microphones and pickups) and mix them together to form one set of signals to go to the amplifier. This is so you don’t have to have a separate amplifier and speaker for each and every mic or pickup on stage. The mixer has controls for setting volume, tone and perhaps other parameters for each input, so levels can be adjusted relative to each other. This is called “mixing” and one of the chief jobs of the sound reinforcement professional is to “mix” the sound during a performance so each instrument or vocal is at the right volume level. Larger venues will have a mixing board for FOH (Front of House, or the main audience) and a separate one for the monitor speakers that are usually at the performer’s feet.
An amplifier takes a line level signal from the mixer and boosts it in power so it can drive a speaker. As mentioned previously, amplifiers are often integrated with other components. For example, powered speakers are available these days at very reasonable costs – they contain a speaker and a power amp and accept a line-level input from the output of a mixer. Another example of an integrated amplifier is a powered mixer. This contains both a mixer and an amplifier. The output is going to be designed to drive speakers, so don’t hook it up to another amplifier or a powered speaker.
There are other electronic components that are frequently found in a sound reinforcement “rack” – these include equalizers, compressors, sound processors and the like. These are usually connected to the “send/receive connections” found on the outside of a mixing console and serve to add effects and processing to the sound. More and more processing is being done in the digital domain these days, and we will cover this in greater detail later in this section.
Interconnections are the interfaces between components in a sound system, and always involve a wire of some sort with a connector at both ends. There are three interconnection types that are generally present in a sound reinforcement system. They are source level, line level and speaker level connections. We will discuss each of these in detail later in this section but will provide a quick summary below:
Briefly, mic level interconnects are very low voltage and expect to connect into very high impedance inputs, typically into a mixer. Source level devices include mics (low and high impedance outputs) and instrument pickups. Instrument pickups (which typically are relatively high impedance outputs) can also be considered a mic level connection when there is no preamp or active electronics.
Line level interconnections are moderate in voltage and connect a low impedance output with a high impedance input. These interconnections run between electronic components such as a mixer and amplifier.
Speaker level interconnections are relatively high in voltage and connect a low impedance output with a very low impedance device (e.g. a speaker). The output stages of the output are designed to supply high current to a low impedance source.
Note that, generally speaking, that the three above types of interconnections are not interchangeable and should not be connected together.
Balanced and Unbalanced Interconnects
A standard unbalanced interconnect consists of one signal wire and a ground (return) wire. The problem with this kind of interconnect is that it is susceptible to noise. If stray electromagnetic radiation traverses the wire (which happens extremely frequently), it will induce unwanted currents in signal path. This will cause static, buzzing or other undesirable effects. The solution to this problem is to use a so-called balanced interconnect.
A balanced connection consists of three wires – one is a positive signal, the second is a negative signal (the output circuitry inverts the negative signal so that it is 180O out of phase with the positive one) and the third is a ground or neutral connection. The electronic circuitry that receives the signal only looks for and amplifies a difference in the + signal and the – signal. Since they are completely out of phase with each other, this difference represents (2X) of the signal.
This balanced interconnection approach yields an inherent advantage in its design called common mode noise rejection. As our unfriendly stray electromagnetic radiation transverses the wires, it induces an equal current in the + signal and the – signal. Since the receiver (on the input side of the connection) only pays attention to the difference in the + and the – signal, the noise will not be picked up because it will be (close to) identical on each lead.
It is good practice to always use balanced interconnects if they are available in your sound reinforcement system.
Many devices such as condenser microphones need power in order to operate. Luckily, in order to facilitate this without the use of batteries, there is a way to transfer power down a microphone or instrument cable. It needs to be a balanced connection, which, as discussed above, has three wires.
Phantom power is supplied usually by a mixer (or guitar amplifier) and it works by putting 48V of DC (direct current) potential from the neutral or ground connection to both + and – signal connections. Since the relative DC value of both signal connections is identical at 48V, the input circuitry (which as we learned above, only pays attention to differences in signal) will ignore the 48V and solely amplify the signal. Meanwhile, other circuitry back at the mic (or source device) can harvest this 48V to power its internal circuits (using the return or ground wire as the neutral to complete the circuit). The mixers and guitar amps that supply the phantom power frequently have switches to turn this 48V on and off for each channel. It is best to keep it off unless the source component needs phantom power in order to operate.
Two branches of physics and mathematics that are very important in understanding sound reinforcement are electronics and acoustics. There are many good books on these two disciplines, and we will not cover the topics completely in this section. We are only going to dip casually into the two subjects. The intent here is to provide a working knowledge of only the basics required to select, setup and run components that make up a sound reinforcement system and perhaps some additional background to make them a little more understandable. As it is presumed that the reader doesn’t have an operating knowledge of calculus or other advanced mathematics, I will do my best in simplifying these concepts and will express them to the degree possible in layman’s terms. Because of these simplifications, my descriptions will be necessarily imprecise, but they should serve to develop a basic working set of knowledge about electronics and acoustics.
Let’s start with some basic electrical concepts. Voltage is the electric potential that a circuit element has at any given point in time. It results from the accumulation of electrical charge, or electrons. Using the inevitable water supply analogy, voltage is equivalent to pressure in a pipe. High pressure within a given hose or pipe will cause water to flow more quickly than low pressure.
Current is the flow of electrons and is analogous to the amount of water flowing in a pipe or hose. Resistance is the measure of difficulty that electrons have in passing thru a given material and is similar to a pinch point in a hose.
Current, voltage and resistance are related thru Ohm’s Law which states that E=IR, where E is the Electrical Potential or voltage, I is the current and R is the resistance. High current (high water flow) happens when the voltage (water pressure) is high and the resistance (pinch points in a hose) is low.
Electrical power is the energy consumed or produced per unit time. It can be expressed as P=IE or power = current times voltage. Substituting in Ohm’s law, power can also be expressed as P = I2R or P=E2/R. Power is measured in Watts and Energy in Joules or Watt-hours.
Electricity can either be constant (Direct Current, or DC), or it can be oscillating with time in a regular (or irregular) fashion (Alternating Current, or AC). The specific format of Ohm’s law covered above applies only to Direct Current. For AC, a more general notion called electrical impedance is required and allows for a generalization of Ohm’s law to include circuits where electricity is varying with time. Since we will be dealing with audio, the electricity in the signal is usually oscillating in our components and we will need the more general concept of impedance in order to characterize our circuits.
Fully understanding electrical impedance is beyond the scope of this chapter, but for our purposes, impedance can be thought of as a form of electrical resistance that occurs as a result of varying or oscillating electrical fields. There are three types of electrical impedance that are important to us, the first of which is resistance (that we have already encountered above). The second form is called inductance and it is related to the way the current is changing in a circuit. The third is called capacitance and it is involved with the changing accumulation of charge (and voltage) in a circuit. Electrical impedance and “impedance matching” are important in the interfacing of electrical components – which sound reinforcement systems require – and will come up in later chapters in this section.
Sine waves are important in both electronics and acoustics. If you recall, a sine wave has the property of frequency (measured in cycles per second or Hertz, abbreviated Hz), wavelength (measured in inches or centimeters) and velocity (in meters per second) . Both sound waves and electrical signals can be characterized as sine waves (or sums of sine waves). Sound waves that are audible by the human ear range from 20Hz to about 20KHz (or 20,000Hz). The 20Hz wave measures about 55’ from crest to crest, whereas the 20KHz one spans only 0.03”. This is a tremendous range of wavelengths and will become very important when considering acoustics of the room you are playing in (among other things). Also implied is that the electrical form of those audio waves will consist of similar frequencies.
<diagram of a sine wave>
One property of sine waves that is important is the concept of phase. Simply put, phase can be thought of as the position (e.g. from left to right in our diagram) of the wave. The term phase shift represents how much a sine wave is moved to the left or right as compared to the original wave. Phase shift is measured in degrees, where 0O is no shift and 360 O is one full wavelength of shift. If I shift a sine wave to the right by (say) ½ of a wavelength, it looks the same as the original wave but it is 180O out of phase. If these two waves are added together, the sum will be zero – which is called phase cancellation. This can happen with electronic sine waves inside your mixer or with acoustic sine waves that come from two different speakers. Of course, if the phase difference in the two signals is zero degrees (e.g. they are in phase) they will add up arithmetically and the signal will be twice as “loud”.
Comb filtering is a surprisingly widespread phenomenon and it has some pretty important implications for sound reinforcement. It is one of the biggest factors responsible for sound coloration and unusual, hard to figure out problems with your sound system.
Comb filtering results when waves emerge from a single source or two coherent sources (such as two speakers fed by the same signal) and recombine in such a way that the phases are shifted. This can happen in several different scenarios:
- Comb filtering can occur when two or more microphones at different distances pick up audio from a single source. The phase of the signals entering these two microphones will be shifted because it takes more time for the audio to reach the more distant microphone.
- It can also occur when two speakers are at different distances from a listener (and the speakers are reproducing audio from the same acoustic source). Here again, the sound travels different path lengths before recombining in the listener’s ear, and one audio wave will be shifted in phase with respect to the other.
- Another common scenario where comb filtering occurs is when reflections of sounds from the walls of a venue reach the ears from a listener and combine with the direct sound from a loudspeaker. Once again, the reflected sound travels further and will be shifted in phase.
In another words, comb filtering happens all the time, to one degree or another, in any and all venue settings. Minimizing the effect will result in a much better listening experience for the audience.
Note that if the sound in each of the above scenarios were a pure tone, consisting of a single wavelength of sound, the comb-filtering phenomenon would not take place. Instead, the sound would be combined arithmetically, as described above in the paragraph on phase… i.e. if the two sources combined in such a way that the phases were one wavelength apart, they would add, and the sound would be 6dB louder and if the phases were separated by ½ of a wavelength, there would be complete cancellation and the listener would hear nothing. So, the result would only be a change in volume, depending on the difference in phase of the two signals, and not a change in the frequency makeup or coloration of sound.
However, sound from a Bluegrass band is not composed of a pure tone but of highly complex sound waves, consisting of many different tones (or frequencies) simultaneously. When these types of sound waves combine in the above fashion, the phase relationships actually depend not only on the distance difference of the sources, but also on the frequency of the sound. So, some frequencies within the sound will be shifted by a multiple of a single wavelength and add together. Other frequencies within that same sound will be shifted by ½ a wavelength and cancel totally. Frequencies in between will partially add and partially cancel. As you can imagine, this will strongly color the sound by emphasizing some frequencies and canceling others. The result is called a comb filter, because if you look at a plot of amplitude vs. frequency it looks somewhat like one of those old-style combs from back in the 50’s.
<Diagram of comb filtering effect>
So here we have a situation where we have spent thousands of dollars on expensive electronics and transducers to preserve a flat frequency response, and our resulting sound for the listeners is anything but flat. In fact, the frequency response curve is very lumpy, and in the more extreme cases will sound absolutely horrible. And this is what you hear from a typical venue.
We will discuss ways of mitigating the comb filtering effect later in this section.
The notion of a decibel, abbreviated dB is one of the more confusing things you will run into when dealing with sound reinforcement. You will encounter the dB term frequently when working with sound systems – it is probably written on the meters on your mixer for example. You will also hear the term dB when you talk to a sound engineer – these folks think in decibels – and actually for good reasons, as we will learn later in this section. It might be a bit painful, but it will help you quite a bit if you learn five things about the dB:
The dB is Logarithmic
The first thing to learn is that, when you measure things in dB, you are dealing with logarithms. Expressing things logarithmically is good for quantities that have huge possible ranges of values (such as the Richter scale for earthquakes), and sound clearly falls into that category. The threshold of human hearing is defined as 0 dB SPL, and the pain threshold is around 140 db. That is a ratio of 100,000,000,000,000 to 1 and in just about any linear way of representing these sound pressure levels, you would be writing either huge numbers or very small fractions. It is much better just to use a range of numbers between 0 and 140 to talk about sound levels.
Separate formulas for power quantities and pressure quantities
Secondly, there are two separate formulas for converting a “regular” number into its dB equivalent. If you are dealing with a quantity that involves power, the formula for calculating the dB value is as follows:
dB (power) = 10 times the log of the power ratio
Quantities expressing amplitudes, such as voltage and pressure scale differently, and the formula for calculating these quantities is:
dB (pressure) = 20 times the log of the pressure ratio
Sound pressure, or SPL, which we will be dealing with frequently, scales as a pressure ratio (=20 times the log of the ratio).
If you refer to the table below, you’ll see that to double something in power (which goes as 10 X log) is to increase it by roughly 3dB. If you are doubling a voltage, or a sound pressure level (which goes as 20 X log), its decibel value will increase by about 6dB. These two numbers: 3dB for doubling of power and 6dB for doubling of pressure… are important and you should remember them.
|dB||power ratio||amplitude ratio|
|100||10 000 000 000||100 000|
|90||1 000 000 000||31 623|
|80||100 000 000||10 000|
|70||10 000 000||3 162|
|60||1 000 000||1 000|
|−50||0||.000 01||0||.003 162|
|−70||0||.000 000 1||0||.000 316 2|
|−80||0||.000 000 01||0||.000 1|
|−90||0||.000 000 001||0||.000 031 62|
|−100||0||.000 000 000 1||0||.000 01|
|An example scale showing power ratios x and amplitude ratios √x and dB equivalents 10 log10 x. It is easier to grasp and compare 2- or 3-digit numbers than to compare up to 10 digits.|
From Wikipedia: https://en.wikipedia.org/wiki/Decibel
Note that microphones are sensitive to sound pressure and we will mostly be referring to that in Acoustics.
Third, Apparent loudness is a property somewhat different from sound power or sound pressure, although it is related. Apparent, or perceived loudness is defined as the volume as heard by the human ear and as processed by the human brain. This is an extremely complicated phenomenon and varies based on a number of things that impact the physical structures of the human ear as well as psychological effects that are hard to quantify. For example, doubling the sound pressure (+6dB) doesn’t necessarily double the apparent loudness. According to some recent research, most people agree on the following:
- 10dB increase in sound pressure is perceived as a doubling of loudness.
- 6 dB increase in sound pressure is a quitenoticeable increase in loudness
- 3 dB increase is generally noticeable
- 1dB increase is just barely noticeable
These are pretty good rules of thumb, and when you are adjusting volume levels at a mixing console; it is good to think in these terms.
An additional factor that has been extensively studied is how the human ear’s sensitivity varies with frequency. It turns out that the ear is most sensitive to sounds between 1KHz and 4KHz and this effect is most pronounced at low volumes. If you see sound measurements in dB (A) or dB(C) units, these have been weighted to account for this phenomenon (ascribing more weight to frequencies more sensitive to the human ear and vice versa).
dB is always a ratio
The fourth thing is that it the dB (written by itself, with no other appended letters or subscripts) is always used to express a ratio and not an absolute number. So when your sound engineer says he has 6 dB of headroom before feedback, that means that the ratio of the current sound level to the sound level where feedback will start is 6dB (as we learned above, 6dB is a ratio of 2).
dB-SPL, dBV and dBu
Finally, if the term decibel, or dB is appended with some letter(s) – for example, dBV, dBu, or dB-SPL – then that letter(s) implies the reference level. Since the reference level is specified, it becomes a real number and it is no longer a ratio.
In the case of SPL or sound pressure level, all dB-SPL values are referenced to the threshold of human hearing – which is defined as 0 dB-SPL. This is roughly the sound of a mosquito flying at a distance of 3 meters away from your ear. For example, a 6dB SPL would be double this threshold, and 20 dB-SPL would be 10 times this threshold.
dBV and dBu are used to specify voltages. dBV is referenced to 1 volt and dBu is referenced to 0.775 volts. An example: if the sound meter on your mixer is reading 0dBV that means that the voltage on the wire is 1 volt.
That concludes our primer on electronics and acoustics fundamentals. Next, we will turn to each device in the signal chain, starting with microphones.
Microphones are one of the most critically important elements in any sound reinforcement system, and it is essential that the correct microphone for the job be chosen with care. Placement of microphones is also quite critical in achieving optimal sound. In this section, we will develop a little bit of background on microphone technology and discuss the different kinds of microphones available for use. We will also go thru some techniques for proper microphone placement.
There are two important considerations when selecting microphones: 1) the directional characteristics, or pickup pattern and 2) the type of technology used by the mic (Dynamic, Condenser or Ribbon)
One of the most important aspects of any microphone is its sound pickup pattern, or directional characteristics. There are just a few common ones to know about, and it is especially important to choose the right pattern for your application.
Omnidirectional microphones can pick up sounds equally from any direction. Figure 8 mics (sometimes known as shotgun mics) pick up directly from the front and the back but not at all from the sides. Cardioid mics are most sensitive at the front and taper off at the sides with almost total rejection of sound from the rear. Hyper cardioid and super cardioid mics have a much narrower pattern of pickup from the front but unlike cardioids, they typically have a lobe of sensitivity at the rear (with the null point off axis).
<Diagram of mic patterns>
You may be surprised to learn that all of these mic patterns are produced from just two different kinds of mic diaphragms. The first kind is a pressure sensitive diaphragm and it is very much like a barometer, simply measuring the air pressure at the front of the capsule. The front side of such a capsule is completely open to the air and the backside is completely enclosed, forming a small chamber. There is a very small hole at the rear of the chamber so that the diaphragm doesn’t become a true barometer and change with the weather (the small hole is invisible to audio frequencies). Pressure sensitive diaphragms, constructed in this way, will sense the pressure (sound) waves as they pass the mic equally from any and all directions. Therefore, a mic composed of a single pressure diaphragm that is (almost) sealed at the rear will be an omnidirectional one.
The second kind is a pressure gradient sensitive diaphragm. It is sensitive to changes in pressure but not the absolute pressure. The diaphragm is completely open on both sides and therefore will only deflect if there is a pressure difference across the membrane. Hence, this kind of capsule will only register a signal where there is a pressure gradient or difference in pressure from the front of the diaphragm to the back. The resulting sound pickup pattern shows equal sensitivity to sound entering the front and the back of the mic (although with opposite polarity in electrical signal). Since sounds coming from the sides of the capsule will cause equal pressure on both the front and the back of the mic, they are rejected completely and will result in zero signal. This is the classic “figure 8” pattern mic.
Combining the two types of diaphragms – the pressure and the pressure gradient diaphragms – results in an arithmetic sum of the two patterns. When you sum a sphere (omnidirectional mic) and a figure 8 pattern, you get a heart-shaped – or cardioid pattern. Remember the back part of the figure 8 is opposite in polarity (or out of phase) and will be subtracted from that part of the Omni pattern, resulting in phase cancellation or zero sum. The sides will simply have the same sensitivity as an Omni since the figure eight contributes zero at the sides. The front of the mic will have the two signals added together, resulting in more sensitivity.
There are two ways of constructing mics such that the patterns are arithmetically combined to form a cardioid– one is to have a mic capsule that contains two diaphragms (one of each) and the second is to alter the hole pattern of the enclosure behind a single diaphragm so that the capsule arithmetically sums both sensitivity patterns. Both types of cardioid mics can be found on the market but the typical ones that a Bluegrass player is likely to encounter is the single diaphragm variety.
Off axis response and the proximity effect
One thing that is important to understand about these various types of mic patterns is that they not only yield differences in sensitivity as you move from the front to the sides of the mic, but the frequency response characteristics also vary. This is the so-called off “axis response” and is especially critical when multiple singers gather around a single mic. Many mics – especially large diaphragm mics – have a significantly altered frequency response when you don’t address them right in front of the capsule. This mostly consists of a reduction of high frequencies and can color the resulting sound.
Another phenomenon that occurs with cardioid and figure 8 mic patterns is a significant boost in bass frequencies as the sound source gets very close to the mic capsule. This is called the proximity effect and is used with great impact when a singer wants his voice to sound deeper.
Neither of these phenomena occurs with omnidirectional mics, which is why they are typically used in a studio. Sound reflections, which are almost always off axis, are easily picked up by all mics. If you have a mic that distorts off-axis sound, such as a cardioid mic, reflections that enter the capsule from the sides can result in a strongly colored sound. This is especially audible if used in recording applications. Of course, omnidirectional mics are also very susceptible to feedback, which is why they are rarely used in sound reinforcement applications.
Next, we will explore the several different types of technologies that form the basis of microphones.
Dynamics are the most common types of microphones that you are likely to encounter in mic’ing Bluegrass bands. These microphones operate by the principle of induction and consist of a magnet and a moving coil. A very small diaphragm is attached to the coil, and because it is suspended in a magnetic field, it acts as a small electrical generator when presented with sound waves. A minute amount of electricity is produced by this effect. Further, the current generated mirrors the sound waves entering the mic.
Dynamic mics require no phantom power and are very robust (a huge advantage for your average Bluegrass fumble-fingered picker who, in my experience, is quite prone to dropping microphones). They also tend to have higher Gain Before Feedback (GBF) – another distinct advantage for supporting Bluegrass bands. And, they tend to be relatively inexpensive.
A couple of the downsides of dynamic mics are as follows: First, they produce an extraordinarily small amount of current. This needs to be highly amplified (in the pre-amp stages) and if you are not dealing with pristine preamps inside your mixer, this can and will introduce audible noise into your mix. Secondly, because the sound waves are doing actual work by moving the diaphragm, dynamic mics tend not to produce high frequencies as accurately as the other types of microphones. You can tell immediately if someone is using a dynamic mic, as the sibilants are not as clear. These types of mics also sound a bit muffled when picking up bluegrass instruments as some of the high frequency content is missing or slightly distorted. Third, the transient response of dynamic mics is not as good as condensers. We’ll discuss this more when we cover condenser mics.
I continue to use dynamic mics as my “go-to” microphone for lots of Bluegrass events, despite their disadvantages. In many of these venues, there are multiple, reflecting surfaces, low ceilings, and maybe even mirrors or glass behind the band. These are all indications that feedback is going to be a problem. In the absence of these problems, I’d certainly rather use condenser mics instead of dynamics for Bluegrass (see below – I love the pristine sound of a condenser for bluegrass instruments). However, in many typical Bluegrass venues, condensers can exacerbate high frequency feedback, especially in small bars or taverns. Feedback, to me, is the #1 killer of good sound and the most difficult to control. I’ll take every advantage I can when dealing with feedback, starting with microphone choice, and dynamic mics are a bit better in this area (probably because they muffle the high frequencies a bit).
A few of the better-known dynamic mics used in Bluegrass:
- Shure SM57, SM58
- Shure Beta 57, 58
- Sennheiser 835, 935, M421
Condenser microphones operate on the principle of changing capacitance in the microphone capsule as the result of the sound pressure variations in an acoustic wave. They are constructed in such a way that the diaphragm acts as one plate of a capacitor, with a fixed back-plate being the other plate. Sound waves acting on the diaphragm vary the distance between the plates, and this changes the capacitance of the capsule. This, in turn, causes a varying voltage that is subsequently amplified and placed on a low impedance output connection. Many condenser microphones need phantom power in order to provide bias voltages for the capacitor. In addition, since the mic capsule itself is very high impedance, active electronic circuitry is required to amplify the current in order to facilitate a low impedance interface to the cable and the mixer. Therefore, condenser mics need phantom power in order to operate.
Electret Condenser Mics
There is a sub category of condenser microphones that is worthy of note called electret condenser mics. An electret material is one that has acquired a permanent electrical charge and using such a material as part of the capsule eliminates the need for phantom power to bias the capacitor (although most electret mics use on-board preamps that do need phantom power). These mics are very robust, inexpensive and have been made in the billions for the cell phone industry. They can be made extraordinarily small and are typically used for lapel mics and for applications where the physical size of the device must be minimized. They have improved dramatically in quality in recent years; in fact, most condenser mics for pro audio are now electret.
Small Diaphragm Condensers
Small diaphragm condenser mics can be ideal for mic’ing Bluegrass instruments, especially in settings where feedback is not a major concern. The physical inertia of the diaphragm is minimal – about 1000 times less than a dynamic mic diaphragm. These mics are extremely responsive, which leads to excellent transient response. For Bluegrass applications, good transient response means that, for example, the very beginning sounds of a pick striking a string will be reproduced very accurately. You can really hear the difference between a good condenser mic and a dynamic for a picked string instrument.
Transient response is not the only consideration. High frequencies are also preserved better in condenser mics than in dynamic ones, which are important for preserving the overtones of an acoustical instrument. The small diaphragm also lends itself to flat off-axis frequency response, as phase cancellation (from comb filtering) becomes less of a factor, as the diaphragm gets smaller. All the characteristics of sound from a Bluegrass instrument will be well preserved, including the pick attack as well as most of the high frequency overtones.
Some of the more popular small diaphragm condensers include:
- Shure SM81, KSM137, KSM141, KSM 9
- AKG C1000
- Audio Technica ATM710 (for vocals)
- The Ear Trumpet family of mics (medium diaphragm)
Large Diaphragm Condensers
Large diaphragm condensers tend to make good vocal mics, especially in studio applications. In Bluegrass sound reinforcement situations, large diaphragm condenser mics like the AKG 3000 series are frequently used for “single mic” setups (see below). They tend to be very sensitive and have wide pickup patterns (and many of these mics have adjustable patterns). The large diaphragm will produce an excellent low noise signal. However, using a large diaphragm condenser on stage demands acoustic isolation of the mic from the stand (with a shock mount or a high pass electronic filter) as every vibration will be picked up and amplified. Note that sound entering the side of a large diaphragm mic will be altered in frequency response by having high frequency content attenuated. This is due to phase cancellation of sound that enters the capsule with wavelengths on the same order as the size of the diaphragm.
Here are a few of the more popular large diaphragm condenser mics:
- MXL 990
- AKG C3000
- Shure SM or KSM 42, 44
- Audio Technica AT4040 or 4033
- Neumann TLM103 or U87
Instrument Mounted Condenser Mics
Condenser mics are small enough such that they can be mounted on or in an acoustic instrument. This has the advantage of offering a much more natural sound compared to a pickup. Feedback can sometimes be a problem but no more so than with an external mic. Here are a couple of the more popular ones:
- Mini Flex 2
- K&K Sound Meridian
- Bartlett Audio Instrument mics
Ribbon mics rely on the same principle as dynamic mics – that of a moving coil suspended in a magnetic field. However, in the case of a ribbon mic, it is a very thin ribbon that is suspended instead of a coil. Ribbon mics are known for their excellent high frequency characteristics but are very fragile and must be handled with extreme care. Because of their fragility, they are rarely used in sound reinforcement applications and usually found in recording studios. I don’t recommend them for use in Bluegrass settings, they are simply too fragile.
Once microphones have been selected, microphone placement technique becomes the next step in achieving good sound for Bluegrass ensembles.
Mic’ing Bluegrass Instruments
There are two factors to consider when mic’ing bluegrass instruments. One is the distance from the mic to the instrument and the second is where the mic is pointed with respect to the instrument’s body.
When determining the distance between the mic and the instrument, there are two counter-opposing things to consider. First, as sound leaves an instrument, it takes a certain amount of distance for the sound to combine and become “natural sounding.” Mic’ing too close to the instrument doesn’t allow the sound to combine and the result will be a domination of sound from one particular area of the instrument. This is an important consideration since different areas of an acoustical instrument resonate with different frequencies, phases and amplitudes of sound. It is the combination of all the sound waves radiating off the entire instrument, from all the various places on the instrument radiating sound that is heard in a non-amplified fully acoustic setting – and that is what we are trying to reproduce. You simply can’t get this sound with close mic’ing, and this is why recording studios will usually mic at a distance of at least 18”.
The counter-opposing factor that argues in favor of close mic’ing is that it gives you much more headroom before feedback. As was noted earlier, feedback is almost always the primary consideration on a Bluegrass stage. We will discuss feedback in more detail later in this chapter, but it is worthy to note that, of course, feedback is unpleasant and will severely limit the maximum volume of your performance. Perhaps a less well-known effect of feedback is that it can and will strongly color the sound at volume levels well below the feedback point of the system.
Note that a sound source increases 6dB in sound pressure for each halving of distance to the mic. And since feedback depends on how loud the sound source is in comparison to the sound reaching back to the mics from the speaker(s), getting the instruments closer to the mic will increase the safety margin before feedback by 6dB for each halving of the distance. Close mic’ing can significantly mitigate feedback concerns in this regard. Further, close mic’ing can reduce or eliminate leakage of sound into the mic from room and stage reflections, which can also color sound pretty dramatically in a bad way.
My experience with a wide variety of Bluegrass stages would suggest that, in many cases, the arguments for close mic’ing would outweigh those for more distant mic’ing. The gains from lack of feedback and sound leakage seem to be bigger factors than the advantages to be had of a better, more natural tone from a more distant mic placement. I will say, though, if 1) the band is using in-ear monitors (eliminating the need for fold-back monitor wedges), 2) the main speakers are high quality and have a very well controlled radiation pattern and 3) the room doesn’t contain very many highly reflective surfaces, I would tend to increase the mic distance to capture more tonal characteristics from the instruments. The bottom line is that your mic’ing technique will depend strongly on these (and other) factors, and you will need to use considerable judgment in determining the proper distance.
Where do you point the mic?
The next question to be answered is where to point the mics with respect to the instrument’s body. This, again, is a matter of tradeoffs. For most instruments (except the fiddle), I have found that the best, naturally sounding place to point the mic is the body of the instrument near where the neck joins the body. This avoids any resonances that unavoidably occur near the sound holes (for guitar and mandolin) or any tubbiness (for a banjo’s head). However, mic’ing at this spot has the distinct disadvantage of not being the loudest place on the instrument. Once again, where feedback and sound leakage considerations dominate, I would tend to mic directly over the sound hole (or the banjo head) and use equalization to notch out any resonance effects.
Mic’ing a fiddle is more a question of how close you can get to the instrument and not interfere with the bowing. It does take a good distance for the sound of a fiddle to develop in any case, and the only practical place for a fiddle mic to be is above the instrument. Placement directly above the bridge will probably yield the most balanced sound and be in a spot that pretty much avoids the bow. Note that fold-back (wedge) monitors on stage can complicate matters for fiddlers, since they will be pointed almost directly into the down-pointing microphone and can cause feedback.
Mic’ing basses can be a bit tricky, and I have found that no matter where you mic the bass, you will need to compensate with equalization. Many basses seem to have a bad ‘wolf note’ right around 125Hz and this will need to be notched in order for it to sound good. But then again, sometimes when I mic up a bass, all I get is this muddy, deep, diffuse sound that has no body at all and needs a boost at 125Hz. Best practice is to try several places and use the one that sounds the most natural. Many people wedge a dynamic mic in the tailpiece with some foam rubber, but this seldom yields a natural sound. I strongly recommend the use of a pickup and an on-stage amplifier for basses, with a DI from the amp right into the mixer. This, in combination with a mic will yield the best sound but will usually sound fine with just the pickup.
Don’t forget to assess the bass EQ and mix from many points of the room – bass notes have wavelengths on the order of the size of a room and interference effects will almost always yield nodal points (complete sound cancellation) and reinforcement points (sound addition). You will need to get a good average balance point that sounds decent in most places of the room, and you’ll need to walk around to assess this.
One extremely important consideration is that, with a bass on stage, you are likely to experience bleeding of bass sound into all the other mics (especially if the bass uses an on-stage amplifier of any kind). This phenomenon is particularly noticeable if you are using condenser mics for the other instruments and will give a distinct hollow sound to the overall bass mix. Further, you won’t be able to control the bass sound using the bass fader. If this happens to you, the solution is to use a high pass filter for all the non-bass instrument condenser mics on stage.
I normally recommend a (very) close-mic’ing technique for Bluegrass vocals. This of course assumes that the band doesn’t use a single mic stage for vocals (see below for a discussion of single mic approaches). The proximity effect, which occurs on cardioid and figure 8 pattern mics, has a favorable impact for most vocalists, usually making their voice sound fuller. Further, feedback considerations and leakage (from reflections and other instruments) can be minimized with close mic’ing for the same reasons as with instrumental mics.
Novice vocalists are sometimes reluctant to get close to the mic. Further, they tend to move around a lot, causing dropouts in the sound. The only solution I have found is to use some amount of compression, which will also help with the overall voice quality. We will discuss compression in more detail later in this section. Note that compression can aggravate feedback and it needs to be used with care.
Back in the day, all Bluegrass bands used a single mic for live performances. This resulted in a choreography of sorts as band members moved in and out to take breaks. Vocals were “mixed” by all singers gathering around the (single) mic. This setup had an appeal beyond sound fidelity considerations: it made the show more entertaining by inducing motion and visual interest on stage.
A single mic setup (or hybrid approaches that feature a single vocal mic and multiple instrument mics) has made a comeback in recent years. The gathering of vocalists around a large diaphragm vocal mic can evoke a sense of nostalgia, which many traditional Bluegrass bands value. Further, the choreographed motion on stage can yield a more entertaining show when compared to five people standing stiffly upright in front of fixed mics for the entire performance.
I’ve seen the single mic technique used very effectively by bands. I’ve also seen it yield very hard to hear, trebly harsh renditions as well, with people tripping over each other. The key factor in my opinion is for the band to practice and develop proper mic distances as well as choreographed motions in advance of a performance.
The single mic discussion
There are some folks that say a better vocal blend can be achieved with a single mic setup, as the voices combine acoustically before they enter the sound system (instead of combining “electrically” after they enter the system as with multiple dedicated vocal mics). It could also very well be true that the vocalists themselves benefit by hearing the other voices and the blend naturally and close by, and not thru the sound system. These arguments have merit, no doubt, but I still believe that the stage show and choreographed motions (as described above) would still be perhaps the biggest reason for going to a single mic setup.
The argument against single mic is basically that, as outlined above in the discussion about close mic’ing vocals, sound fidelity can suffer as the sound source moves away from the mic. If three vocalists have to sing thru a single mic, they will necessarily need to be further away from the mic diaphragm. Furthermore, any advantage that the proximity effect would have on vocal quality is reduced or eliminated due to the increased distance of the sound source from the mic.
One final thought that argues in favor of using a single mic is that it will reduce the number of microphones on stage. Note that every microphone added to the stage reduces the GBF (gain before feedback) by 3dB-SPL – therefore getting rid of just two mics would give you 6dB more headroom. This will double the sound pressure level before feedback starts. That is a pretty significant advantage and can counter balance many of the disadvantages resulting from the vocalists being further from the mic.
As mentioned above, one of the major factors in selecting a single mic approach is that it benefits the stage show. Any motion on stage will (usually) add interest to a performance, and a band that has a well-oiled machine of an approach to movement around the single mic is, indeed, very entertaining to watch.
One system for using a single mic for instrumental breaks is to have the guitar and the bass on stage right (the bass being somewhat behind). The other three instruments rotate in and out as necessary as the guitar steps to the right to allow breaks. For example, let’s say the band is performing an instrumental tune. When it is time for a banjo break, the banjo player approaches the mic from stage left and slides to his right, in front of the mic, to take a break. The person taking the next break (say, the mandolin player) will move into the “on deck” position at stage left while the banjoist takes his break. He then backs up when his break is complete, allowing the mandolin player to approach the mic from stage left, sliding to her right, to be in front of the mic. And so forth. You can imagine a clockwise rotation of instrumentalists if viewed from above.
Vocals can be a bit tricky to set up also. Here, it is important for all three singers to gather as close as practical to the mic. Choreographing the vocalists might be done as follows: As the lead singer finishes a verse and prepares for three-part harmony in the chorus, he steps to his right, the tenor singer approaches from stage left and the baritone singer comes around back, just like JD Crowe does.
It is very important to achieve a balanced sound in the trio, which depends strongly on the distance from each singer to the mic (as well as how loud each of the singers are). All non-omnidirectional mics have significant drop off in sensitivity to the sides, so the singers to the left and the right of the mic probably need to be significantly closer than the person directly in front of the mic. It is important for someone in the audience (hopefully during a practice) to give feedback on distances, and for the singers to memorize and repeat these for each trio performance. The distance is also a function on how loud and projecting each voice would be. You can imagine that Don Rigsby would have to stand pretty far back in a trio if he performed as an invited guest (by some magical feat) in my band as his voice would be much more penetrating than any of our singers. Note that in-ear monitors, which will be discussed in a later chapter, will go a long way to helping vocalists achieve the needed balance if they are available. One other technique to tune these distances can be to use a headphone distribution box (as you might find in a studio) during practice and use isolating headphones to get an idea of what the best vocal blend will be.
Many bands augment the single (vocal) mic with one or more satellite mics for instruments. This hybrid approach has the advantage of allowing the blending of vocalists into a single mic while not having to solve the problem of rotating instrumentalists in and out to take breaks. It also gives a more consistent volume level from instruments that are doing backup.
Speakers are the final transducer in the signal chain and serve to convert a relatively high-powered electrical signal into audio waves. There are several different kinds of speakers, and it is beyond the scope of this book to describe them all. The major speaker technologies that you will encounter in Bluegrass sound reinforcement settings are 1) moving coil / paper cone and 2) piezo-electric.
As you might imagine, a moving coil / paper cone speaker involves attaching a paper conical cone to a coil that is suspended in a magnetic field. As (changing) current passes through the coil, the speaker cone moves either forward or backward, which in turn, produces a sound wave that emerges from the surface of the cone. In fact, it is a kind of linear motor, converting electricity into motion. It is the exact opposite of a dynamic mic, which converts sound into motion and then into electricity. Large moving cone speakers are almost always used for woofers, or speakers that respond to low frequency sounds. Smaller ones can be used for midrange or high frequency speakers as well.
Piezo-electric speakers, on the other hand, rely on a piezo-electric crystal, which will shrink or expand in response to voltage variations. The shrinking/expanding crystal is attached to a (usually Mylar) membrane that produces a sound wave. These are often used for tweeters, or speakers that respond to high frequency sounds. These tweeters can either be mounted directly on a speaker cabinet or be attached to a horn, which acts as a mechanical amplifier.
Typical sound reinforcement speakers contain one woofer and one tweeter – these are known as two-way speakers. A three-way speaker would have a woofer, a midrange and a tweeter.
If one were to hook up an amplifier directly to the woofer/ tweeter combination without any additional circuitry, it might work but would be very inefficient because the tweeter would have to deal with low frequency content as well as the high frequencies intended for it. Similarly, the woofer would need to handle high frequencies in addition to the low frequencies. Lots of amplification would be required to produce even modest sound levels and lots of heat would be generated at higher volumes – enough to probably blow out the speakers.
The solution to this problem is to use something called a crossover. A crossover is a frequency divider – it takes a signal and divides it into high frequencies intended for the tweeter and low frequencies for the woofer. A passive crossover is installed after the amplifier (most always within the speaker cabinet itself) and consists of inductors, capacitors and resistors. An active, or electronic crossover is installed before the amps and divides a line-level signal into high and low frequencies. You’ll need separate amplifiers for the treble (tweeter) and the bass (woofer) if you are using an active crossover. This approach is sometimes called bi-amping or bi-amplification because of the use of two amps per channel instead of just one. If the signal is divided in three, it is a tri-amp situation.
Unpowered speakers will typically contain a passive crossover inside the enclosure. Some may have separate terminals that bypass the crossover, intended for use with an electronic crossover and two amps, but these will be clearly marked. Powered speakers can either have passive crossovers (and a single amp inside the cabinet) or be bi-amplified and contain an active or electronic crossover (as well as at least two amps inside the cabinet). You’ll hook these powered speakers up the same way, whether or not they are bi-amped or not.
Crossovers are also used for subwoofers. The intent is to take the low frequencies (usually less than 80 or 100Hz) completely out of the woofer/tweeter speakers and direct them to a subwoofer, which is designed specifically for those very low frequencies. This considerably cleans up the sound from the main speakers, as they don’t have to deal with these power-hungry low frequencies. In addition, this configuration eliminates certain types of distortion in the mid-bass due to the woofer cone moving back and forth with very low frequencies.
Most subwoofers are actually powered subwoofers, containing an amplifier in the subwoofer enclosure, and may or may not have an electronic crossover or a low pass filter on the input. Sometimes your subwoofer hooks up to your (powered) main speaker systems – many main speakers have an internal crossover and an output for the sub, peeling off the low frequencies and directing them to a separate terminal that hooks to the subwoofer. In these cases, you would hook the subwoofer to this terminal of the powered main speaker, not the output of the mixer.
Unless you have a kick-drum in your band, you are unlikely to benefit much from a subwoofer in a Bluegrass ensemble. This is because the lion’s share of frequencies coming off of the bass are above the cut-off point for a typical subwoofer. However, if you are using cheaper or smaller speakers as mains, a subwoofer might help – some of these main speakers are unable to produce the fundamental frequencies of a bass (you are really hearing the first overtone from these kinds of speakers). A subwoofer, with the crossover frequency set appropriately, can reproduce the fundamental and will add considerable depth to such a setup.
We discussed the directional characteristics, or pickup patterns of microphones in the previous chapter. In a similar manner, speakers have varying radiation patterns, depending on the design of the speaker. Here are several of the more common types:
Point source speakers
This geeky term is being used more frequently to describe what a layman would call a normal speaker. The term “point source” comes from physics, where a sound (or other) source has an intensity that falls off proportional to the distance from the source squared. Normal speakers are far from looking like a point, but do, in fact, behave roughly as a point source – and this means that the sound drops off pretty dramatically with distance. Better quality point-source speakers will carefully control the dispersion angle of sound and this will be in the spec sheet. It is always important to install these speakers (safely) on tall speaker stands, with a slight angle downward if the speaker cabinet socket supports it.
If you have been to a rock concert recently, the stage probably had line arrays installed on either side of the stage. Line arrays are made up of a vertical column of speakers, and in most large venue installations, they are mounted up high and form a bit of an arc.
Going back to the underlying physics, a vertical line of speakers behaves differently than the point source speaker. Theoretically, the intensity drops off linearly with distance, which is a lot slower than the distance-squared drop off of the point source speaker. This means you can reach the back of the audience easier, without producing deafening volume at the front.
Line arrays are composed of multiple, independent speaker units, which are stacked on top of one another to form a vertical column. Each of the individual speaker cabinets will be designed to have a very distinct radiation pattern, and they will stack in such a way to get a continuous field of sound from the column of multiple speakers.
I have found that you don’t need the whole column of speakers to benefit from these line array-type speakers. A single line array speaker, used in unison, makes a great main speaker in more modest size rooms. A pair of these can make ideal speakers for Bluegrass sound reinforcement applications. The vertical dispersion pattern of these individual elements is engineered to be restricted to 15 or 20 degrees and seems to be perfect as this constrains the energy towards the audience and not up towards the ceiling. Not only is this much more efficient, reflections from the ceiling can color the sound in quite a bad way thru effects such as comb filtering. Minimizing the sound going upward seems to be a very good idea in most all situations. You’ll want to mount these high on speaker stands though – make sure you get help putting them up there as they can be very heavy.
A mixing console (or the equivalent functionality packaged as part of an all-in-one head unit or powered mixer) is used for taking multiple inputs, balancing them in volume, adjusting EQ and filters, adding effects such as compression and reverb and then distributing to mains and monitors.
Amplifiers are used to boost the signal from the mixing console and present it to the speakers in such a way that it can be used to drive high current through a low impedance load. Amplifiers usually have a single control to set the gain, or volume of the output.
Mixers and amplifiers can be combined into a “powered mixer.” Speakers and amplifiers can be combined into “powered speakers.”
There is a plethora of mixers in today’s marketplace, many of which are reasonably priced. Here is a list of features that I would recommend, at a minimum, for supporting a Bluegrass sound reinforcement application:
- At least 10 channels with balanced inputs (preferable to have 12+) – An eight-channel mixer is probably the absolute minimum you’d need for a typical Bluegrass band, supporting five pickers and three vocalists. And you can get by with even fewer channels if the band uses a “single mic” approach. However, if you have a full Bluegrass band comprising five people, and they all sing, you will need 10 inputs. Furthermore, it seems as if there is always a need for more – for things like DI from instruments or preamps and for that guest star who is always called up to sit in with the band. In addition, several of the inputs should support both mic and line level sources. This will allow support of DIs, which always seems to be needed.
- Low noise preamps – The preamp is the very first stage within the mixer that the signal encounters when coming in from the mics and is arguably the most critical. Dynamic mics especially will present a very low voltage signal, which needs to be boosted a great deal before the mixer can process it – and if the preamp introduces noise at this stage; it will be amplified tremendously along with the signal from the mic. Introducing noise right at the beginning of the signal chain will color the entire mix and, in addition to being audible and unpleasant, it can be made even worse if effects such as compression are used. Make sure that any mixer you are considering has the lowest noise preamps that you can afford. Either consult with the specification or audition the mixer in the showroom by attaching a very low voltage signal such as a dynamic mic and turn the volume up. Buy the mixer with the least amount of static background noise.
- Auxiliary busses / outputs – A stereo mixer will always have two output “busses” (left and right channel) that connect to the main outs. In addition to the two main busses, I would recommend as many “aux” busses as possible (my mixer has six), with the ability to flexibly route audio on to these busses from each channel. The option to route audio onto aux busses pre- or post- fader can also be important. You will use these busses, which connect to “aux outs” in order to drive individual monitor wedges or separate in-ear monitors for band members.
- Wireless console capability – Many mixers these days will support a Wi-Fi connection or an Ethernet connection where a Wi-Fi router can be attached. In addition, these vendors will supply a mixing console app that can be installed on a tablet or PC. This is a relatively new feature and is extremely valuable as it allows the use of the tablet or PC as your main mixing console. The mixing engineer can then wander around the venue with the console in hand, checking each area for sound quality. These applications, sometimes known as “virtual’ mixers, have buttons and sliders are rendered on the LCD screen of the tablet or PC. They typically support a much higher range of features and settings than would be possible with physical buttons. For example some “virtual” mixers have as many as 50 different effects that can be called up without plugging in any external equipment. This and other advantages come along with the transition to digital processing inside the mixer and will be discussed in detail in a later chapter. Examples of mixers supporting this functionality is the XAir series from Behringer and the Mackie 1608L.
My recommendation on amplifiers has changed in recent years. In fact, I would not recommend buying a separate amplifier, or a powered mixer at all at this point in time. I’d suggest purchasing some powered speakers instead. With the advent of class D amplifier circuits and digital approaches to amplification, amplifiers have been made much more efficient and considerably lighter than in years past. This has made the construction of a powered speaker much more practical and lightweight. Further, when an amplifier “knows” the loading characteristics that a given speaker presents, the circuitry can be designed accordingly, making for a much better product. The manufacturer doesn’t have to build the amp circuits to anticipate any speaker on the market – but instead design only for a single speaker.
Even the smallest sound reinforcement system can be assembled with 8” powered speakers. And these systems can scale up to handle the largest audiences with powered line arrays and subwoofers. Additional powered speakers can be added (with appropriate delay circuitry if they are to be placed back away from the stage) to handle dead spots in the audience.
Powered speakers are available from all the major manufacturers. JBL has a competitive line of speakers, as does Mackie and others. I prefer offerings from QSC – their K2 series is just about perfect for most bluegrass situations. Look for class D amplification, as this will make the speaker considerably lighter and powerful. It is not unusual to find 2000 watts of class D amplification in a typical small (8-12”) powered speaker.
Operating a mixer and amplifier is pretty intuitive, and I won’t provide step-by-step instructions to do so in this chapter. One important thing that might not be intuitive, though, is how to set up your gain structure to optimize the capabilities of your system.
There are several proper ways to set the gain structure for your system, some require pre-fader listens and multi-meters, and others require a bit of math. Back in the day, it used to make sense to take the time necessary to optimize gain structure very carefully for each stage of your sound system, but with the advent of low noise preamps and digital internals to your electronics, this has become less important. What I present here is a very pragmatic approach that probably won’t be recognized by professional sound reinforcement people as the best one but seems to work well for me and is quick and easy to do. Further, it results in a gain structure that gets the most critical things aligned correctly (such as trim settings for mixer channel inputs).
You will have at least three opportunities inside the mixer’s signal chain to set volume levels and one or more volume adjustment on the amplifier(s). There might also be opportunities to set input levels of DI sources thru volume controls on external preamps or perhaps on the instrument itself. Setting all of these volume controls at the correct relative levels is called setting your gain structure. Improperly setting your gain structure can lead to over-modulation in one part of the signal chain (if, say, a particular setting is too high); and high noise introduction in another part (due to a particular volume setting being too low and a following stage needing to boost the signal too much as a result).
My standard practice for setting gain structure assumes you have meters of some kind on each channel – these meters typically read green if the signal is normal, become yellow as the signal approaches full strength and then turn red when the signal level is strong enough to over-modulate the channel. I will also assume that each channel strip on your mixer has a trim, or input setting knob that sets the gain level of the preamps.
My procedure is to first set all the faders to the 0dB level (this would be near the top of the slider, as most sliders have the zero point about ¾ or more of full volume for that channel) and then have a trusty assistant speak at full volume into each mic (at the same distance and consistent volume for each mic) and/or play any instruments that have DIs. Using a combination of the trim / input control setting on each channel and any external controls (such as stomp boxes or preamps), I will get all the signal levels to start bumping up into the “yellow” (referring to the meters on each channel). I’ll usually then pull the faders back a couple of dBs to define a starting point for balancing the overall mix.
I then set the gain of the master faders on the mixer such the main meter is in the green, maybe just bumping into the yellow. I’ll then adjust the level on the amp such that the volume in front of house is acceptable. If the amp volume control is down close to its minimum, I’ll usually turn down the faders on the mixer and turn up the amp’s volume control up and average the gain between the two a bit.
Once all this is done, I’ll go on to set up the monitors and ring out the system. We’ll cover this in detail in the “Controlling feedback” chapter later in this section.
One final point: As the band does its sound check, and even during the performance itself, I find I need to continue to tweak the gain structure to keep things optimized. It is very easy to fall into the trap of continually pushing faders up as you hear the need for more volume from (say) the mandolin, then the fiddle, then the bass, and so forth – and end up with all the faders at or near the top. If you find yourself with most or all your faders set at or near the top, then it will be important, perhaps during a break, to rebalance the gain structure so there is headroom in each channel. Some folks maintain best practice of initially setting 40-60% gain on faders to allow for this kind of a scenario, but you will add a bit of noise into the mix due to the non-optimal gain structure.
It is important for the performers to clearly hear what’s happening on stage. The usual solution for this is to install monitors that direct sound back into the performer’s ears. The two different kinds of monitors we will discuss are 1) fold back, or wedge monitors and 2) in-ear monitors
Many small or medium size speakers, in addition to being used as main speakers for smaller venues, are frequently designed be placed on the floor in front of the stage. These speakers typically have tilted (flat) sides that enable horizontal placement on the floor, with an upward angle of 45 to 60 degrees – perfect for monitor wedges. Other speakers are especially designed to be stage monitors and are wedge shaped. In either case, the usual installation for these types of speakers is directly in front of each musician, between the mic and the audience.
If your mixer supports multiple aux busses with independent mixes, and if you have a sufficient number of amplifiers (or powered monitor speakers), custom mixes can be provided for each musician. Otherwise, these speakers can be daisy-chained from a single mix (if unpowered speakers, pay close attention to impedance and power considerations – speakers hooked up in series add impedances directly, whereas speakers hooked up in parallel add with a reciprocal formula 1/Rt=1/R1+1/R2+ … + 1/Rn – be careful not to exceed the recommended maximum power and/or minimum impedances supported by your amp(s)). For powered monitor speakers, daisy chaining is easy as most of these speakers have a pass-thru jack where the next speaker in the chain can be attached and you don’t have to worry about impedance.
Feedback is a major concern with fold-back monitors, and, assuming you are using cardioid mics, it is best to have the back of the mic pointed directly at the monitor speaker (if possible). It is also important to have some kind of dedicated equalization on the monitor mix, because the frequencies that cause feedback with monitors are usually quite different than the ones causing feedback in the FOH. More about controlling feedback will be discussed later in this chapter.
In-ear monitors are a relatively recent innovation that many Bluegrass bands now use. They have several distinct advantages: 1) they eliminate any feedback concerns due to monitors 2) they allow for complete isolation 3) they enable a completely custom, personalized mix with no bleed through 4) today’s wireless technology enables a very high-quality connection 5) they are considerably lighter to transport than monitor wedges.
In addition, I have found that in-ear monitors are very useful for singers. For many people, they allow the hearing of intonation of one’s voice much easier and therefore the vocal parts can be more in-tune. They also allow one to balance the blend better in a three-part harmony by moving closer or further from the mic. It takes a bit of time before they feel natural when singing: because the ear canal is blocked, your voice will sound different to you (similar to singing with your finger blocking your ear canal). But in the long run, for most people it will improve singing technique and intonation as well as the ability to blend with harmonies.
Some people don’t like to feel isolated from the rest of the band or the crowd, and the in-ear headphones can make you feel like your head is wrapped up in a big, clear piece of cellophane. I usually use only one earphone, leaving the other one dangling – this gives me a better sense of connection with the band and the crowd.
It is possible to configure multiple in-ear receivers onto a single channel, although this is not an optimal use for this technology. Most bands have one receiver / transmitter pair dedicated to each musician and use aux busses to push a customized mix to each one.
I’ve used both the Carvin and the Sennheiser G3 transmitter/receiver pairs, and I highly recommend the Sennheiser even though it is considerably more expensive.
If you are in a touring band, one of your biggest fears is probably what kind of monitor mix you will experience during your next gig. Indeed, perhaps one of the hardest-to-control variables for a traveling band is continually switching between all the various kinds of in-house sound systems from night to night. Every venue will have a different sound engineer with a different approach to mixing the monitors. Some of them do a great job whereas others, not so much. Bad monitor sound can turn a great gig into a horrible experience for a band.
To address this concern, many bands will hire their own sound person, who travels with them to all the gigs. Some will even bring their own console, mics and other equipment along with their own soundman. This solution can be very effective and of course, it also controls the quality of your FOH sound, but it is expensive. Not everyone has the room to carry a Midas Pro2 mixing console in the band’s 1993 Ford Econoline van along with the sound guy, a full-size bass and five sweaty band members.
The new in-ear monitor technology combined with the recently available extremely lightweight digital wireless mixers enables a different, perhaps more practical solution to the monitor mix problem, and allows any band to have complete control, leaving only the FOH sound to the venue personnel. Further, with a few quick connections, a relatively small and portable rack containing all the necessary equipment can be spliced into just about any venue in just a couple of quick minutes. Lots of touring Bluegrass bands are doing this, or something similar, to control their own destiny with monitor sound.
Referring to the diagram above, all the components can easily be packed into two 3U or one 6U rack, which will weigh less than 10 lbs. In addition, each member of the band can control his/her own mix using an iPhone or an Android phone. Here is the way it works:
- The XLR signal splitter separates and isolates the band’s monitoring mixer from the in-house sound console (using isolation transformers). This allows the band direct access to the mic level connection, with no disruption of the in-house mix. Things like phantom power would continue to be supplied and controlled by the in-house mixer.
- The Behringer XR18 or equivalent mixer provides a maximum of seven independent mixes that can be controlled with Wi-Fi connections (the XR18 has a Wi-Fi access point on board). You can have as many as four connections simultaneously (more if you use an external Wi-Fi access point instead of the internal one). Band members can download the free XAir software from the App Store and independently control up to seven mixes.
- I strongly recommend using the Sennheiser G3 wireless units; they are much cleaner and more robust than the equivalent from other vendors (although they are considerably more expensive). An external antenna is also an option for larger stages.
The final pieces of gear we will be discussing are signal-processing units. They take an incoming audio stream and process it mathematically, altering the frequency response, amplitude or phase or perhaps even introducing new sounds into the stream. For some odd reason, the unprocessed signal is referred to as a dry one, and correspondingly, the processed stream is wet.
Signal processors can be spliced into the sound system in fundamentally two different ways:
- In Parallel – The signal is forked – the wet fork gets the processing and the dry fork doesn’t – and then recombined downstream, resulting in a sum of wet and dry (should we call this moist?). A mixing “knob” can alter the amount of processing, or “effect” between 100% (totally wet) and 0% (totally dry). Reverb is a good example of a parallel process.
- In series – Here the signal is not forked, but instead the entire stream receives processing. A digital delay (e.g. used to make remote speakers sound good) is an example of a series process.
Signal processing can be done on an individual channel, on a combination of channels or on the entire (main) mix.
Signal processors can be housed in individual rack-mounted units or be packaged as part of a mixing console. Today’s digital mixers feature lots of signal processing effects – once the stream is encoded in digital format, it can easily be processed by DSPs (digital signal processor) with no additional circuitry – just the magic of software. Beware, there may be some latency introduced as a result of this processing – but most of the better mixing consoles use technology to carefully align wet and dry signals to avoid phase effects such as comb filtering.
An equalizer is a piece of equipment (or could be implemented by some software running in a digital console) that takes a stream of audio and alters or shapes its frequency content. There are two major types:
This should be a familiar piece of electronics to most people. A graphic EQ will take a stream and divide it into multiple frequency bands, each band can be independently boosted or faded with a slider. A so called 1/3 octave graphic equalizer will divide an incoming sound into a total of 31 bands, or 3 bands per octave for 10 1/3 octaves. It is called a graphic EQ because a pattern is formed from the 31-slider knobs that look like a rough frequency response graph when viewed from the front.
Graphic EQs are very useful for notching out frequencies that are prone to feedback in a procedure called “ringing out” the system. We will describe this further in the “controlling feedback” chapter. Graphic EQs can also be used as a tone control on steroids, although each of the individual faders may color the sound in undesirable ways by slightly shifting the phase – I much prefer using parametric EQs for this purpose, which will be described next.
A parametric EQ is an equalizer that gives one almost total control over altering the frequency response of part or all of a sound stream. It uses adjustable filters, and is called “parametric” because it allows one to independently alter three different parameters:
- Center frequency – this is the middle frequency of the filter
- Q, or bandwidth – this is the width of the filter. Wide bandwidth (small Q) filtering will take in a broad sweep of frequencies around the center whereas narrow bandwidth (high Q) will only capture a small portion of the frequency spectrum.
- Amplitude – this is a control that alters the gain of the signal within the frequency limits of the filter. Amplitude can either be boosted or faded.
A fully parametric EQ has all three controls, whereas a partial or semi-parametric EQ will just allow adjustment of two out of the three.
Compressors can be very useful tools for sound reinforcement, but only if the effect is used sparingly. These units, which also can be implemented in a digital console, reduce the dynamic range of an audio stream – i.e. make the loud sounds softer and make the soft sounds louder. Most compressors that we encounter in sound reinforcement only act to make the louder sounds softer and leave the softer sounds alone.
<insert figure of compression envelope>
There are a couple of parameters that can be set on a compressor:
- Compression Ratio – this is the magnitude of the reduction in dynamic range. 2:1 is a small compression ratio; 5:1 is very big. My advice is to keep it modest.
- Threshold – Compressors do not act on the entire signal, but only on audio that exceeds the threshold setting. This setting will depend strongly on what you want to compress. Most compression units will show a real-time dB meter along with a schematic of the compression envelope – you should increase this setting (i.e. decreasing the threshold) until it “eats” into the signal and you start hearing the compressor working. Again, use of discretion with both the Threshold and Compression Ration settings is important, as too much compression can ruin the sound and increase susceptibility to feedback.
Envelope – A compressor doesn’t apply compression the instant the audio
exceeds the threshold. Instead, it acts
during what is called a Gain Envelope.
There are three parameters that can be set that control the shape of the
- Attack – this is how quickly the compressor ramps down the gain once the threshold has been reached. You typically don’t want the compressor to start its action immediately, as that would dull the transient sounds that can be important in proper reproduction of percussive sounds (such as a pick striking a string). So, the solution is to set an attack at a few (say 10) milliseconds, which will allow transients to pass uncompressed.
- Hold – Once the signal has fallen below the threshold level, the hold setting (if your compressor has one) will specify how long the compression will continue at the preset compression ratio.
- Release – the release setting specifies how long it takes for the compressor to ramps up the gain (or the compression down to 1:1). You don’t want a compressor to release immediately as it can distort very low frequency sounds. If it is set too long, it can cause “pumping” or “breathing” artifacts – you’ll know these when you hear them. Release times are quite a bit longer than attack times, 150 milliseconds would be a typical one.
I will use compression in a couple of situations for Bluegrass bands. Sometimes, compression can tighten up the bass if it seems muddy and not well defined. Secondly, compression on vocals (especially with novice vocalists that continually and inadvertently change their distance from the mic) can help to smooth out the voice quite a bit.
Note that you may have to readjust the volume of the signal after it is compressed. Some compressors have a separate adjustment called Make Up Gain to help control the compressed signal, so it is similar in overall level to the original.
Everybody probably knows what a reverb is – it is the echo chamber effect that you can apply to vocals and perhaps selective instruments to make them sound “real.” Reverbs were first used in studios, in order to make it seem like the vocalist was sitting in a real room as opposed to an anechoic studio. Back in the day, a reverb was composed of a huge steel plate and weighted 600 lbs. Spring reverbs became a popular add to a guitar amp in the ‘50s and ‘60s.
For Bluegrass sound reinforcement, I use just a touch of reverb for vocals and sometimes for the fiddle. There are dozens of different reverb types these days; some of them even simulate the old plate reverb sound. Just about any reverb will do for bluegrass vocals, though.
A set of delay lines is very useful for larger venues. A common approach for a large outdoor concert or a long indoor room is to place a set of speakers ½ way into the audience. This allows the rear of the audience to clearly hear the stage without turning up the volume too high for the people in the front.
If one were to hook the speakers directly, with no delays, the audience would hear a very echo-y confused sound. This is because electrons travel close to the speed of light and sound travels much slower, at about 340M/S or about 1000 feet / sec. If the speakers were placed 100 feet into the audience, the electrical signal would arrive to those speakers about 1/10 of a second before the sound wave from the stage reached that position. Therefore, people seated next to or beyond those speakers would hear sound from the stage about 1/10 sec after the sound from the speakers placed in the middle of the audience. This would be clearly audible and very disturbing. The solution is to use a delayed signal to drive the rear speakers.
<block diagram of a sound reinforcement system>
The figure above consists of a generalized block diagram for a sound reinforcement system. It is important to understand the interconnections between each of the components. Each type of interconnection features specific kinds of connectors as well as varying levels of impedance and signal voltages.
This is probably the most sensitive connection in the system. Mics operate on extremely low voltages and these wires are especially prone to problems with noise and intermittent connections. Most microphone connections are balanced with low impedance outputs and high impedance inputs on the mixer (or preamp) side. Connectors are almost always XLR jacks on both ends of the cable although some high impedance mics use ¼” connectors. The cables themselves consist of balanced, shielded, controlled impedance wires.
Low impedance (e.g. dynamic) microphones can support relatively long cable runs until meeting the first preamp stage (typically in the mixer). Snake cables of 50-100’ can easily be used with low impedance source devices. High impedance mics and pickups with no active electronics are more susceptible to noise and runs with these mics should be kept as short as possible, preferably under 20’ before entering the first pre-amp stage.
Many sound reinforcement systems need some sort of interconnection between various components. As noted in the previous chapter, these interconnects are all line level connections and are relatively robust. The cables themselves are always shielded, impedance-controlled wires that feature either XLR jacks on both ends; or a combination of XLR and TRS ¼” connectors. It is highly desirable to use balanced cables whenever they are supported by your electronic components.
Some example interconnections in this category include the wire between the mixer and an amplifier and between the mixer and various sound processers. Note that the connection between a mixer and a powered speaker is also of this type.
The connection between an amplifier (or an integrated head unit) and the speakers features relatively high voltages and currents. One common termination for speaker wires is a male TS ¼” connector on both ends. It is very easy to mix up one of these speaker wires with a line level interconnect cable though. This is a mistake because the internal impedances of the two types of wire are far different – the speaker cable is a twisted pair, whereas the line-level wire needs to be a coaxial-shielded cable. Do not substitute one for the other, even though it is physically possible to do so as the connectors do fit.
A relatively new jack technology designed specifically for speakers is the so-called “Speak-on” connectors. These are usually blue in color and feature a twist-to-connect hookup. They are designed to be high power and low impedance and can’t be mixed up with line level interconnect cables.
Cables are the weak link in a sound system. Every time they are used, they are stretched, walked on, spooled and then pulled in various directions. Further, as we learned in the previous chapter, mic cables transfer signals at extremely low voltages and even balanced cables are sensitive and can become intermittent very quickly.
Setting up a sound system before a gig can be a very stressful experience when things aren’t working right, and an intermittent cable is usually the culprit of many problems. It is especially bad when a cable starts to become intermittent in the middle of a show. If you want your cables to work reliably, I’d suggest you do the following:
- Only buy premium quality cables (especially microphone cables). I use only Mogami cables – they tend to be more expensive, but they will last considerably longer and, in my experience, won’t suffer intermittent connections.
- Be very careful with cables when putting your sound system away. I roll my cables carefully, making sure there are no residual stresses in the casings. The way to do this is to create a loop around 8-12” in diameter and hold it with a cable tie. As you roll the cable onto this loop, make sure you twist the cable to the right (clockwise) with your thumb and forefinger (with your right hand) as you spool it onto the loop (also clockwise, held by your left hand). If you do this just right, the cable will lay into a nice circle with no kinks. If you see any bends or kinks, you probably are not winding it correctly. This technique will eliminate any residual stress in the casing of the cable, and they will last much longer.
- Periodically test your cables (especially mic cables) for intermittent connections. One way to do this is to set up your sound system someplace quiet and then manipulate or twist each cable slightly with the volume turned up. If you hear any static while doing this, the cable is probably going to become intermittent or fail totally very soon and needs to be replaced.
- When (for example) a mic fails to operate, or sounds fuzzy, first try replacing the cable. It is especially important that you have spare cables that you know function well packed away in your sound system for this reason.
I have found that feedback is probably the most limiting factor in doing sound reinforcement for Bluegrass music. There are a couple of reasons for this, most especially that Bluegrass is (usually) an all-acoustic music and needs lots of mics. The theoretical gain before feedback (GBF) is cut by 3dB for each and every mic added to the stage. And given the reflective surfaces and lack of close mic’ing, it can be far worse than that.
Further, mic’ing Bluegrass instruments can induce resonances in the system, causing feedback at a particular frequency. Finally, with a stage full of monitors and sound bleeding off-axis into the 10 or so mics on stage, there are many sources of possible feedback.
Here are some best practices that I have learned that help with controlling feedback.
- Use dynamic mics (not condensers) if you are in a venue with lots of reflective surfaces and/or low reflective ceilings.
- High quality mics are a must. Less than high quality mics can be prone to having non-flat responses and will exacerbate feedback.
- Any damaged mics will usually accentuate certain frequencies and must be avoided.
- High quality cables will help by not introducing noise or buzzes into the system. Noise doesn’t directly affect feedback but a noisy system close to its feedback point seems to me to be much less stable than a quiet one.
- A carpet on the floor of the stage can help reduce sound reflections, which will enter mics off-axis (and be subject to sound coloration due to both the reflection and the off-axis mic response).
- Angle the FOH (main) speakers such that their radiation pattern doesn’t include any part of the stage where the mics are. Look out for and avoid any reflections near the main speakers that might direct sound back to the stage.
- It is always better for feedback to use in-ear monitors. If using wedge monitors, point the back of the (cardioid) mic directly at the speaker (if possible). For hyper and super cardioid mics, the back of the mic should be 20 or 30 degrees off axis from the monitor.
- Be sure to ring out the system (see below) first before the audience enters.
- If you can get away with it, do a short (and hopefully not too noisy) second system ring out when the room is at least 2/3 full because people in the room will significantly change the acoustics. I have been frequently able to do a “secondary” ring-out in a full house without anyone noticing by very carefully controlling the faders and just barely getting the system to start to ring. Since a preliminary ring-out has already been done, it is easier to be much subtler at this stage.
This is a procedure that characterizes the acoustic environment that the sound system is placed into and compensates for any frequency peaks that cause feedback with equalization. It is somewhat controversial – I have heard some sound engineers say that ringing out a system is pure hooey. However, my experience has been that if you use the right kind of equalizers, you can gain considerable headroom with a ringing out procedure.
I have found that multi-band fully parametric EQs work best if you have access to them – you can usually get a very narrow filter and move it to where the feedback is. If you don’t have a multi-band parametric EQ (with at least four bands), a 31-band graphic EQ will work just fine.
A real time analyzer is a big help to have – it will display a graph of sound amplitude vs. frequency, and when you hear feedback, you can easily identify and isolate the frequency of the squeal with the RTA. Some of the latest digital mixers feature both four band fully parametric EQ and an RTA in addition to a 31-band graphic EQ and these are a joy to use.
Note that if you have one of those powered mixer systems that has a six or seven band graphic EQ on board; it will be of very limited use in equalizing feedback frequencies out of the system. The bands are far too wide to usefully notch problematic frequencies and will probably just end up coloring the sound. It is far better in my experience to try to instead, get the frequency response to be as flat as possible out of that kind of system and to use that on-board EQ to alter tonal characteristics. Optimally, if using a graphic EQ to control feedback, it should be a 31-band 1/3-octave unit.
Here is the procedure that I use for ringing out the system:
- First, set your gain structure to at least approximately the final settings (see earlier in this section for instructions on how to do that). It will help at this point to roughly set all the mix faders for all of the mics too.
- Set your main faders on the mixer all the way down. Turn all the monitors completely off. Turn the amplifier volume controls (if they have one) up to the normal setting (see “gain structure” earlier in this chapter to determine that setting).
- Gradually increase the volume using the mixer main faders until you start to hear the first squeal. Identify that frequency and notch it out either using a graphic EQ or a fully parametric EQ. I would start with a setting for that frequency band of -3dB or so below where the feedback stops, and then increase the volume again using the main faders. If that same frequency continues to feedback, continue to lower the setting for that band until it stops.
- Repeat the above step until a second frequency starts to feedback and then identify and notch out that frequency.
- Repeat until you have multiple frequencies feeding back.
- Turn the volume controls for the amplifiers back to all the way off (or mute or turn off the amps)
- Set the main mixer faders back up to about -3dB
- With someone playing or talking into the mic, turn the amplifiers up until you find the volume that is appropriate for that audience. Increase the mixer faders until the system starts to feed back then return them to the -3dB setting. The difference in these two settings is your headroom. If the two settings are too close together, you will need to take steps to increase the GBF of the system by moving mics and speakers around, introducing non-reflective surfaces, etc. I like to have at least 6dB of headroom.
- Reduce main amplifier volumes back to zero or turn off the main speakers
- Repeat the above procedure for the stage (wedge) monitors with the main speakers turned off. If you have the ability to add EQ onto each of the monitors individually, then repeat the procedure for each one of the monitor speakers.
Section VI: Sound Reinforcement for Bluegrass Bands (you are here)
 Coil is wrapped in a loose enough fashion that it can move relative to the magnet
 Wireless in-ear monitors are becoming much more popular now and have many advantages. They will be covered later in this section.
 Send/receives refer to connections to and from the mixing board, which takes the signal from within the board and routes it to an external connector. The sound can be processed by an external component and then returned to the mixer via the “receive” connection.
 There may or may not be a preamp or some sound processing devices between the source and the mixer. If there is, that device would convert the source (mic) level into a line level interconnect.
 It is really the voltage difference that we are going to be using which is the difference in electrical potential between two points. Voltage can be thought of as the difference in potential due to an accumulation of electrons at one point in a circuit as compared to another point.
 Audio signals generally vary at rates between 20 cycles per second and 20,000 or 20K cycles per second. There are also modulation and other electrical techniques, which take the frequency into the megahertz (millions of cycles per second) range. These frequencies all constitute “alternating current” in our sense of the word and cause us to need to consider the more general notion of impedance rather than just resistance.
 The equation that relates wavelength, frequency and velocity is ln=V where l is the wavelength, n is the frequency and V is the velocity.
 Wavelengths will be different since electrons move much faster than the speed of sound thru copper wires.
 An excellent reference describing the decibel (and acoustics in general) can be found at the website operated by the UNSW School of Physics in Sydney, Australia http://www.animations.physics.unsw.edu.au/jw/dB.htm#radiation
 Human hearing also behaves logarithmically so it makes doubly good sense to use it as a system to talk about sound levels.
 These numbers are always ratios, as we will learn next.
 When expressed by an absolute measure of physical pressure, this reference level is 20 uPa where uPa are units called “micro pascals.” 1 uPa = 1/100,000 of a pascal. 1 Pa = 1 N/M2 where N is force in newtons. Atmospheric pressure is roughly 100,000 pascals.
 As we will learn below, dynamic and condenser mics are constructed using a small diaphragm that translates sound waves to electrical signals. Ribbon mics are similar but consist of a ribbon instead of a diaphragm.
 Since the associated electronics in the mic are low impedance, this electricity can be transferred efficiently thru long cables to the mixer’s preamp stages.
 The complete opposite will work as well – where the on deck picker stands behind the person taking the break. Once the current player finishes his break he slides to the left and the on-deck picker moves forward to be in front of the mic – this would constitute a counter clockwise rotation.
 These cables feature a male connector plugging into a female jack at the mixer and a male connection on the end of the microphone plugging into a female connector on the cable. An XLR jack has three pins inside what looks like a circle and has strain relief interlocks to prevent the accidental unplugging of a connector.
 Another reason to keep cable runs short with high impedance mics is that the capacitance of the cable forms a low-pass filter because it is in series with a high impedance source. Increasing cable length lowers the cutoff frequency, causing high frequencies to be lost.
 An exception is the all-in-one head unit that has both an amp and a mixer on board – this component only needs mic level and speaker level connections.
 TRS stands for Tip-Ring-Sleeve and this looks like a standard ¼” headphone male connector with three bands of copper on the connector (the three bands being the tip, the ring and the sleeve). Instead of being used for a stereo connection in the case of the headphone jack, the three bands are used for the three connections needed for a balanced connection (see the previous chapter).
 TS=Tip, Sleeve – looks like a ¼ headphone jack with only two bands of metal instead of three
 Do exactly the opposite if you are left-handed – hold the cable in your right hand and spool with your left, twisting to the left.