Electromagnets are the basis for microphones and loudspeakers and many other devices we use in audio. Excess electrons also escape power chords and end up in mic cables which is why you should try to run your power chords separate from mic cables and not side by side. Here is an interesting study of electromagnets.
In any Sound Reinforcement System there are many complicated ways to hook up a sound system and layers and layers of products you can learn about. The better you understand these things the better you will be able to set up a system to optimize the room. However, the basics are the basics. ALL the complicated systems in the world boil down to multiple connections of the basic things.
There are three categories of things to consider in a basic sound system.
Inputs
Devices that put sound signals INTO the system.
Ex: Microphones, DI Boxes, Pickups
Signal Processors
Devices that process the sound signals.
They may lower the amplitude, raise the amplitude, or add alterations to the signal itself.
Devices that deliver the sound back into the environment to the human ear.
Ex: Loudspeakers of various kinds.
Bonus Category - Delivery System
Things that carry sound signals.
Ex: Cables and Connectors
Transducersare devices that convert one kind of signal into another kind. In the case of Audio Engineering the signal is converted from audible sound pressure into electronic signal. It literally turns the sound pressure waves into their electrical voltage representation and then back into audible sound pressure again. Microphones and Pickups are transducers that turn sound pressure into electrical voltage. Loudspeakers are transducers that turn electrical voltage back into sound pressure.
The tool that gets the most attention on a day to day basis. Although there are many things you can do in the set up of your system, the Console is where you will spend most of your time. Microphones, DI boxes, and other inputs head into the board, sometimes assisted by a preamplifier to get the signal strong enough for the board to use, through a snake. That snake in analogue systems is comprised of a whole bunch of chords that are linked directly to each input. That snake in modern digital systems is often a simple Cat5(or6) cable.
After arriving at the Console the input signal strength can be adjusted up or down. Each input channel has adjustments that can be made to the incoming signal before it is passed through to the outgoing flow into the speakers. The MIXING CONSOLE allows adjustments to be made to the signal for either scientific or artistic reasons to make the sound pleasant to the hearers in the audience.
The first adjustment on the board channel is the GAIN (also known as Pot or Trim although these can mean different things too). If the signal is not strong enough it can be "gained" up and if it is too strong it can be "gained" down. This allows the operator to have a "usable signal", which is one that can be adjusted through other means and sent off to places it needs to go. Many times these adjustments can cause the outgoing signal to be stronger or weaker than the input signal so the gain must be adjusted to accommodate. Related to gain may be a small button that says "PAD" followed by a number (-15 or-20, etc). This button when activated will decrease the incoming signal strength by a dBu. PAD-15 will decrease the signal by 15 dBu. This is helpful when the incoming signal is so strong that the gain knob doesn't allow you to get a usable signal by simply turning it down.
The first EQ adjustment is often a High Pass and/or Low Pass filter. A HIGH Pass filter could also be called a LOW Cut filter. It prevents lower frequencies from coming through and allows higher ones to pass through. A LOW PASS filter could also be called a HIGH Cut filter. It allows signals lower than chosen to pass through but prevents anything higher. Much of the signal below or above a certain range is barely audible and may not even be musical. For example, especially with low frequencies, the frequency may be coming through as noise introduced to the system from electrical connections or nearby instruments or other inputs foreign to the actual things being mic'd. Electrical interference, from free electrons jumping wires into the Mic cables will produce a hum at 50-60 Hz, because the frequency of electricity in the USA operates at this frequency. So by putting a High Pass and setting it above 60 Hz you bypass this problem even if it is present in the originating signal. Often seen as simply HP/LP.
Other EQ functions will allow you to make adjustments to more specific frequencies, for example: Lowering a chosen frequency (say 250 Hz, 1 kHz, 8 kHz) while leaving those above and below it in tact. There could be up to three sets of adjusters. The following explanation is based on a decently expensive board that has all the right tools.
dBu: There is one adjuster that will turn up/down that frequency. This will be the one present even if the other two are not. In the event that it is the only selector possible, this will usually be a fixed point in the frequency range. (Which means that if you want to duck/cut down a little of 800 Hz and all you have is a knob that gives you 1000 Hz and 500 Hz you'll have to decide which one or a mix of both, works better).
Hz: There is another button that allows you to select the specific frequency to be adjusted. So if you need to duck that 800 Hz you can dial it over to 800 Hz and be more specific with your adjustments.
Bell: This is the lease likely to be present, but helpful when it is. The Bell allows you to choose how narrow or wide the selection is. Without the bell adjustment the chosen frequency, when adjusted, will impact the others around it as well (which you may want, or may not want). By narrowing the bell you are able to pick a specific problem without taking out things you do want. Or you can boost a specific thing you want to add flavor, say the attack of the beater on a kick drum or attack of the symbol, without boosting other things you didn't want inside.
Aux's
Another path is the auxiliary. The most layman way to say this is that Aux's create sub copies of signals and send them other places, other than the main outs to the speakers. You can often select whether this Aux will be sent prior to or after the EQ adjustments have been made and/or prior to or after the adjustments of the slider. The "Aux Send" may be sent to the hallway speakers, the mothers room, the band monitors, the recorder, or other places the signal is needed besides the main speakers. The other uses for Aux Sends might be to take a copy of the signal and change it somehow, compression, effects, etc, and then feed them back into the board through the same or different channel, often through an "Effects Return" or "Aux Return".
Similar but not the same is an "Insert". The insert actually takes the signal away from the board, adds some prepossessing to it, and then feeds it back into the board to complete it's journey at the same point you interrupted it. This signal, for example, but have effects or compression or other things added to it, BEFORE you add your EQ and adjustments. This may be what you need and may not, so think through what you want and then decide if an Aux or Insert works better for your application.
PAN
PAN typically just means sending it to the right or left or a mix of both. If you turn the PAN knob to the left half way, but not all the way, you are sending MORE signal to the left then right by that much. On smaller boards this can be used to determine Busses or Subgroups as well.
PAN can be used creatively to create stereo imaging. Seeing PAN used by Buford Jones at an Audio Seminar I took recently opened my eyes to the possibilities of PAN. There is every reason to use the PAN selectively on nearly every channel. Nearly none of the PAN selectors ought to be straight up for most inputs. Moving sound around the room opens up room for other sound in different parts of your audible perception. Kick, Bass, Snare and Lead Vocal ought to be straight down the middle, nearly everything else can be PAN'd to it's appropriate place, typically as it stands visually.
At the same time you don't PAN everything hard right or left either. You choose degrees. If you are getting a stereo input from a keyboard, and the keyboard is on your far left. Try panning the left hard left and the right just left of center. If you have a variety of cymbal mics and tom mics, pan them each left to right, or right to left, from each other as they stand visually. This was when a tom walk down occurs the sound moves across your face.
You can also PAN things together.
Snare and Tambourine serve the same (similar) function musically. Pan them both center. Place the volume of the snare where you want it, and then bring up the Tambourine until you feel it adding to the snare without overpowering it.
Acoustic Guitar and Organs/Pianos can play well off each other. Pan them both opposite each other and similar volume, try panning them the same, see what hears/feels right.
Basically think of the entire piece musically and see what things work together
Side Note: Buford typically puts the BASS right next to the KICK on the board since they serve a symbiotic relationship to each other.
Outward Bound:
From this point you will find things are grouped, bussed, muted, fader-ed and that is all sent to the output. From the output they go to
Outboard Graphic Equalizer to set the room EQ. This EQ is often set up when the room is set up and never/hardly changed again, unless the room changes.
Limiters keep the signal from passing the safe zone and blowing the speakers or amplifiers.
Crossovers will break up this outgoing signal and send it to various specialized loudspeakers. The signal is most often broke up into two or three. Lows and High/Mids or Lowe, Mids, and Highs.
Amplifiers/Line Drivers (amplifiers that drive the signal to the place it needs to be for the loudspeakers to produce them.
Loudspeakers produce the signal back into an audible sound pressure wave and send that to the audience.
Transducer: refers to any device which changes one kind of energy into another. A Microphone
changes sound pressure waves (Acoustical Energy) into equivalent audio
signal (Electrical Energy). A Loudspeaker does the opposite.
Sources: Anything that supplies sound to an environment.
Amplifier: makes an electrical copy of an electronic signal (technically it need not
be a stronger copy to be termed an amplifier, but it usually is.) A Preamp amplifies
the signal before being sent down the road to the mixer or other devices. Some
input signals, like dynamic mics, may have such a small signal that they
require amplification to make the journey and be usable in the mixer.
Gain: is an increase in signal strength having sent an input signal through one
or more amplifiers. In some cases the original input signal could be as much as
1 Trillion or more times stronger as it reaches the loudspeakers.
Signal Processing: a sound is converted into an equivalent signal and manipulated. There are
numerous possible ways in which it can be mixed, reshaped, split apart, and
otherwise manipulated these accumulate into signal processing.
Equalizers: are tools used to adjust the frequency of an incoming-outgoing signal.
This may consist of a few basic knobs (like the treble/bass knobs in a car
stereo) or fully adjustable system across multiple frequencies. Sweepable EQ
is generally a knob that can be turned to a specific frequency in order to cut
or boost that frequency. Switchable EQ allows the operator to simply
choose between two or more preset frequencies, but not pick exactly what they
want. (Ex: You may see a preset at 100Hz and 250Hz but if you wanted 180Hz you
have to choose which of the two presets gets you closest) Parametric EQ
allows the operator to adjust the BELL of the frequency adjustment. A wider
bell will grab more frequencies surrounding the one you pick; a narrower bell
will aim to slice the exact point you need without taking out the others around
it. Graphic EQ is a series of up and down knobs at preset frequencies
that you can adjust; typically a wide range of EQ is present in Graphic EQ.
Monitors: signal copies are sent to stage speakers for performers/speakers to
monitor themselves. Many times in modern sound these are In Ear monitors so
that no stage speaker is present.
Microphones are a form of transducer. Types can include contact pickups which
are used for hollow bodied acoustic instruments, magnetic pickups as in
electric guitars, or in more traditional microphones: Dynamic Microphones have a moving coil
which creates magnetic electrical representations of the sound and are very
durable, Condenser Microphones use electricity to create a magnetic
field and can be quite sensitive which can be great for picking up sound but
more sensitive to feedback, and these require power to create the field.
Compressors: (also limiters and gates) adjust the levels of input signals are allow
only certain amounts through.
Sound Wave: Sound is created by compression and expansion. If you take a close look
at any speaker you will find a movement in and out repeatedly. As a speaker
moves out the air in front is compressed adding pressure on the air in front of
it compressing that air and so forth until the energy is dissipated. As the
speaker moves inward (further back from its starting place) this creates an
area of low pressure, vacuum, expansion. This sucks back on the air near it
into the vacuum and so forth. A series of compression and expansion happening
over and over creates pressure on the air. This pressure can be felt in the
body, it can create heat, it can cause windows to vibrate, wine glasses to
shatter, or it can be perceived by the ear. One movement of the speaker out, in
and return to its starting place would create a compression, then expansion, and
then return to normal. This would be ONE CYCLE of a sound wave.
Wavelength: The mathematical and graphical representation of the length of a wave is
from the beginning the curve up the return the curve down and return. That is
once complete cycle. This can be calculated. As Wavelength ( L ) = Speed of
Sound divided by Frequency (Hz).
Amplitude: the strength or intensity of a wave at a given instant in time is called
the amplitude. This is related to Volume, Loudness, and Sound Pressure Level.
Graphically the wave is larger vertically, while taking the same space
horizontally. In essence it's adding MORE pressure to the air around you
without hitting in more cycles per second.
Sound Wave Graphically: Visually on a graph the sound wave is represented as a curve up, a return
to normal, a curve down and ending as a return to normal. If X Hz is happens
exactly two times in the amount of time it takes Y Hz to happen once, X Hz
would be half the frequency. If you were to take the curve and pull it up/down
so that it is taking up more space vertically on the page, but it still hits
the save points the frequency hasn't changed, only the amplitude has.
Reverberation (Reflection): As sound waves leave a speaker in an enclosed room (or you clap really
loudly or yell or whistle...) the sound leaves its SOURCE and travels to
the walls, ceiling, and floor and bounces (is reflected) at a variety of
angles and some of it returns to the hearer. This is akin to the way water
waves move as you drop a pebble in water in an enclosed container, or how light
works in a room lined with mirrors.
Frequency Response: a components ability to produce audio output within a particular frequency range, that is to respond to certain frequencies. Ex: 20Hz to 20kHz +/_ 3dB
Inverse Square Law: each doubling of distance from the sound results in a fourfold reduction of sound power (equal to about 6 dB).
Hertz (Hz): We measure sound in Hertz. One hertz is one cycle of compression, expansion
and back to normal. The average human at birth with perfect hearing can hear,
perceive as audible, sounds as low as 20 Hz or as high as 20,000 Hz (20kHz).
This could be identified as PITCH.
Logarithmic Scales: are scales that are based on exponents. Without going into all the math behind this the basic idea, for our purposes, is that the scale is not linear. We do not HEAR linear, we hear logarithmically. Which is why audibly the half way point between 20 Hz and 20 kHz is 640 Hz (not 10 kHz).
Octaves: The audio spectrum (sound we can hear as humans) spans approximately ten
octaves, or ten doublings of frequency. The octave represents a portion (the
ratio 2:1) and it is the portions between different frequencies that the
hearing process recognizes, rather than the actual number-values between
frequencies. If you were to take 20,000 and divide it in half the number would
be 10,000. However 10 kHz is not half way through audibly the spectrum. We perceive
sound logarithmically. Half way from 20 Hz to 20,000 Hz is actually 640 Hz.
It's half the number of octaves.
Decades: the entire spectrum can be divided into three decades. Bass: 20 Hz
to 200 Hz, Mid: 200 Hz to 2,000 Hz, and High: 2,000 Hz to 20,000
Hz.
Decibel (dB): The science of logarithmic scale helped to develop the Decibel. Decibel is actually not a set unit of volume. In other words you cannot walk into a room and say that something it exactly 35 dB. This is because the decibel is more about ratios than actual set figures. If you turn something down by 6 dB you are controlling things in ratio to where it is now.
Sine Wave: is the simplest form of sound wave. It is a clear, distinct sound. Sound
is essentially caused by the vibrations of a sound source. Consider a tuning
fork. You strike the fork and the bars vibrate causing the air between them to
compress and expand at the same cycles/second (frequency/hertz) as the forks
bars. This constant production of one pitch over time is a sine wave. It is one
frequency. A piano turning fork set to A440 is 440Hz. A tuning fork produces a
sine wave. When the terms frequency or wavelength are used they are
generally assumed to refer to a given since wave component in a sound (The
Fundamental).
Complex Waveforms: when multiple sine waves combine they create complex waveforms. Different
types of combinations create different results. A perfect square wave is
the combination of odd numbered harmonics (3rd, 5th, 7th) going way past the
audible hearing range. Another complex wave is a single note played on a grand
piano is built from these harmonics. An "A" 220Hz, which is the
"A" below concert "A", will produce a fundamental of 880 Hz
(f)(1). It will also produce harmonics of the 2nd, 3rd, 4th, 5th, 6th, 7th, and
8th. This results in three octaves. (Octave doubles the 'f'). 880*2=1760Hz
which is f2, 1760*2 is 3520 Hz which is f4, and 3520*2 is 7040Hz which is f8.
This one note played on a piano produces one fundamental and at least 7
overtones totaling 8 sine waves of diminishing strengths hitting you at once.
The Simple Harmonic: is produces when other frequencies which are whole number multiples of the
original sine wave are produced simultaneously. The most common example of this
is a single guitar string. The string is struck and a note a frequency is
produced and this base note is called the fundamental (f). But other
sine waves are produced as well. The amount of them is dependent on the size
and type of string. The first overtone (2nd harmonic) would be equal to f X 2.
The 3rd harmonic (2nd overtone) is equal to f X 3.
Resonance: when a particular material has a natural tendency to react and vibrate at
a particular frequency (or more than one). Like a grand piano or acoustic
guitar.
(This one is such a good example that it's straight from the book!)
"The sound produced by a tuning fork itself is barely audible -
capable of being heard only when held very close to the human ear. In order to
be more readily heard it must be coupled to something more efficient at
radiating its particular frequency. The tines of a tuning fork have an
extremely slim surface in comparison to the wavelength they produce. A 440Hz
sound wave, for example, has a wavelength of 2.5 feet (0.75 meters) which is
overwhelming compared to the thickness of the tines. Consequently, air slips
around the sides of the vibrating tines with ease and very little of the
mechanical energy involved in the fork's motion is given to the air in the form
of acoustical energy (sound waves). When the stem of a vibrating tuning fork is
placed against an object with a larger surface capable of vibrating at the same
frequency, more air is set into motion, resulting in a louder sound. Some of
the energy imparted to the fork in striking it thus is used to power a more
efficient sound-source. This is an example of resonance.
Timbre (Pronounced Tamber): The result of vibrating elements and resonating bodies combining to reinforce
certain frequencies. A guitar body has certain resonant characteristics and
naturally reinforces certain frequencies more than others. The string will
produce vibrations which cause frequencies (fundamental and harmonics). The
Timbre of each guitar will be different based on the strings used, material and
construction of the guitar body. Etc.
The Ear: inside the ear is the Cochlea which is a coiled sea shell shaped bone, resembling a snail. Inside that coil is a basilar membrane. As different frequencies interact with this membrane it perceives them as different. This is how the ear tells the brain to interpret sound. In other words, the ear is the very first Transducer! It turns audible sound pressure into electro-chemical signal to the brain.
Precedence Effect: has to do with the amount of time delay between the arrival of sound from two separate sources. If someone speaks to you on one side the sound of their voice reaches your far ear fractions of a second later than the closer ear. This doesn't appear to be two voices to you because your ear and mind know to interpret this as having been from the same source. You can tune speakers so that there are two up front and two closer to the middle of an audience, delay the speakers that are closer to hit the ear fractions of a second after the ones up front hit the ear and the sound will appear to have originated up front even though the closer speakers are in fact producing sound. This way you can add volume so that the hearer is into straining to hear but preserve the affect of the speaker/band originating the sound and not the nearby speaker.
Phase: (Read Here and Here) When two (or more) sine waves interact out of time, or affect each other. The most basic example is when the right speaker plays a sound and the left speaker plays the same sound but in exact opposite timing. Even partial timing issues can have the result of making a sound louder or quieter or totally disappearing.
Standing Waves: are when the sine wave fundamental (or it's harmonics) are equal to the distance from loudspeaker to wall and begin fold back in on themselves.
12 Tone Scale: The musical scale of A -F# that we use today in most modern music.
System Configuration: The set up of a sound PA system from input sources to signal processors to output. From Mic to Mixer to Loudspeaker.
Impedance: Electrical impedance is the measure of the opposition that a circuit presents to a current when a voltage is applied.
Ohm: Refers to resistance in a circuit. A speaker may be listed as requiring 8 Ohms. It must be set up correctly or risk damaging the system.
Current (AC/DC): Batteries,fuelcellsandsolarcellsall produce something calleddirect current(DC). The positive and negative terminals of a battery are always, respectively, positive and negative. Current always flows in the same direction between those two terminals.The power that comes from a power plant, on the other hand, is calledalternating current(AC). The direction of the current reverses, or alternates, 60 times per second (in the U.S.) or 50 times per second (inEurope, for example). The power that is available at a wall socket in theUnited Statesis 120-volt, 60-cycle AC power.
Voltage: Voltage is electric potential energy per unit charge, measured in joules per coulomb ( = volts)
Watt: Electrical power is measured in watts. In an electrical system power (P) is equal to the voltage multiplied by the current.
Circuit: An electrical circuit is a path in which electrons from a voltage or current source flow. Electric current flows in a closed path called an electric circuit. The point where those electrons enter an electrical circuit is called the "source" of electrons. The point where the electrons leave an electrical circuit is called the "return" or "earth ground". The exit point is called the "return" because electrons always end up at the source when they complete the path of an electrical circuit. The part of an electrical circuit that is between the electrons' starting point and the point where they return to the source is called an electrical circuit's "load".
Signal Types: An audio signal is a representation of sound, typically as an electrical voltage. Audio signals have frequencies in the audio frequency range of roughly 20 to 20,000 Hz (the limits of human hearing).
Balanced vs Unbalanced wiring connectors: Balanced means that the connectors and wires are built to handle three connections (or more). Load/Positive, Neutral/Negative, and Ground/Shield. Unbalanced will carry the Load-Positive and Neutral-Negative (these complete a DC Circuit) but not the Ground-Shield.
DC (Direct Current): electrical current that flows in ONE direction, positve to negative connections.Nine Battery.
Alernating Current (AC): Electrical current flowing with both phase and reverse phase current. Outlet.
Decibel (dB):
dB SPL is about sound pressure level. The minimum level is established at .0002 Dynes/Square Inch at 1kHz in a young child.
dBu is a measurement based on voltage used to quote normal operating levels and maximum capabilities of components. 0dBu is equal to 0.775 volts RMS.
dBm is based on computation of power. 0dBm is equal to 1 milliwat (0,001 watt) RMS.
Harmonic distortion is the addition of frequencies not present in the original waveform which bear harmonic relationship to the frequencies in the input waveform. Commonly associated with overloading circuits although it can occure below clipping range.
Transient distortion: is the inability of component of efficiently reproduce rapid changes in the intensity of a signal. This type of distortion results from a delay in the time the output waveform takes to accomplish an intensity change equivalent to that of the input waveform. A higher degree of transient response means a lower degree of transient distortion.
Intermodulation distortion occurs as a result widely different frequencies being produced simultaneously and usually occurs in the amplifiers and loudspeakers.
Phase distortion: is any alteration of the phase relationship of frequencies by a component. Phase distortion is possibly the least insidious of all the forms of distortion.
Definition VOLTAGE = (think the height of the water in a tank) the source pressure that pressures electrons to flow through a wire or other conductor Discussion We generally hear of the word ?voltage? used to refer to batteries such as the standard d-cell that provides 1.5 volts, or to power lines in homes that are 110 to 220 volt wires. In these cases we are referring to the voltage source which is either a battery or the voltage delivered to buildings through wires from the generator in a central power plant. We think of this type of voltage as the pressure applied that pushes electricity through the circuit. The word ?volts? is also seen on light bulbs, and in this case there is a different meaning implied. What this means is the voltage rating of the bulb, that is, the voltage needed to make the bulb light up. This will lead us to the next lesson, Ohm?s Law, which is defined as: Voltage = current x resistance, or V=I x R Definition CURRENT = (think flows like water) the flow of electrons through conductive materials such as a circuit Students should always think of current as a through quantity, that is, current flows through a wire. Voltage is an across quantity, that is, a voltage exists across a circuit. The formula for current is current equals voltage (or the measure of source pressure) divided by resistance (or things that offer resistance to the flow of electricity): Current = voltage/resistance, or I=V/R The unit used to measure current is the ampere.
Definitions INSULATOR = anything that electricity cannot move easily through; anything that offers a lot of resistance to electric flow CONDUCTOR = a thing electricity can easily pass through; metals and electrolytes (The word ?electrolytes? should be memorized in this context even if time doesn?t allow more than the defining of this word). ELECTROLYTE = a liquid or moist substance that can conduct electricity Lesson VII: WHAT IS OHM?S LAW? We have already mentioned Ohm?s Law in the above lessons in its three forms: V = I x R, I = V/ R, and R = V/ I. This means that there is a linear relationship between the voltage, V, and current, I, in a circuit, and the proportionality constant is the resistance, R. The first equation means that in any circuit, the current is equal to the voltage divided by the resistance: ________ voltage (volts) current (amperes) = resistance (ohms) This is the way in which Ohm?s Law is most often presented, and the definitions of voltage and resistance are derived from this one. One example of this law is if the voltage across a resistor is doubled, the current through it must also be doubled. Ohm?s Law can be used to derive the currents and voltages in series and parallel circuits, as will be seen below. Discussion of Circuit A circuit is a series of conductors, or things through which electricity can flow. It is a closed circuit when all of its parts are connected by conductive materials. It is an open circuit when there is either an opening in the pathway or there is a non-conductive material in the pathway such as plastic, air, or any electricity resistant material. A circuit must have the following parts: 1) a source of electric current such as a dry cell, a battery or electricity from a wall socket. This source of electricity is called the source. 2) a load = the thing that works because of the current such as a light, a bell, a motor, etc. 3) one wire that goes ____a) from the source to the load and ____b) a second wire that goes from the load to the source. A circuit may have more than one source and more than one load and any number of wires to complete the pathways between them, but every circuit must have one of each of these three elements. Definitions SOURCE = the source of electricity; the place where work is done to separate charge; a battery or current from the wall LOAD = the thing that works because of the flow of electricity; a light, a bulb, a buzzer, a motor, etc. CIRCUIT = (think ?circle? or ?circulatory system?); the pathway through which electric current flows; made up of conductive materials A) CLOSED CIRCUIT = a complete, circle-like pathway through which electricity moves B) OPEN CIRCUIT = a circuit with either a break in it (from an open switch or a loose wire etc.) or with a nonconductor interrupting the pathway through which electricity normally flows Demonstration of the fact that electricity flows through a simple circuit
CURRENT
(V) voltage / (R or Z) Resistance or Impedence in Ohms (O) = (I) Induced Current in Amperes.
Voltage/Ohms=Amperes
POWER
(V) Voltage X (I) Current in Amperes = (P) Power Watts
Voltage X Amperes = Watts
Power is related to the square of pressure.
SPEED OF SOUND: 1126 Feet per Second
WAVELENGTH The formula for calculating the wave distance, Wavelength, is:
Wave (λ) = Speed (v) of wave in feet (Feet per second) divided by the number of feet (f).
λ = v/f
,
So if you know the frequency (λ) and you know the Speed (v) you can always find out how long the wave is. For example: A4 is 440Hz.= (λ). Sound travels at 1126 f/s = (v).
Therefore:
440 = 1126 / f which turns into
1126/440 = f which turns into
2.56 feet.
A single cycle of a 440 Hz sound wave is 2.56 feet long
Wavelength = Speed of Sound (1126 F/S) / Frequency (Hz)
As a musician and general computer and science geek, I really geeked-out on this information, so forgive me if I think this is the coolest! Musicians should know some things about how keys work. Certain notes, represented by letters A-F, make up music. Middle C is in the middle of a Piano. But there are others "C" notes that are higher or lower in pitch than Middle C. You can continue to hit "C" notes over and over up and down until they are too high or low for human hearing. Many musician also know that there is math behind music. This math comes into play as an Audio Engineer in a BIG way. What's REALLY happening?
Audio Signals are calculated using the Hertz Standard. When a wave form completes one cycle (one compression (air particles tighter together than normal) and decompression (air particles further apart than normal) and returns to it's starting place), this is known as ONE Hertz, or 1 Hz.
Human hearing is generally considered to be from about 20 Hertz to 20,000 Hertz, otherwise notated as 20 Hz to 20 kHz.
In the Audio world the "A" note just above the Middle C on a piano is known as A440, or 440Hz. (Click Here for details). Now this is where my years of music came crashing into clarity. On an 88 key piano this A above Middle C is A4, and it is 440 Hz, also known as A440 by piano tuners. The A above A440, the next A up the keyboard is A5 or... A880. A5 is exactly double the frequency of A4. In other words there are twice as many cycles per second hitting you at A5 than A4. Following then A6, the next A up from A5 and two up from A4, is A1760 (880*2).
So besides just being cool, why does this matter? Well, when many instruments are struck, like a guitar string for example, the "Fundamental" note struck, A440, is not the only sound being heard. The vibrations of that string are producing lots of 440, some of 880, a little 1760, etc. A note might sound fuller or emptier based on how many of these are present or missing. If you have EQ'd the higher octaves out of the sound the guitar might be sounding too thin.
That's not all that is happening. You can also add the fundamental to itself over and over. So take A2 which is A110 and add another 110 to that. You get 220 Hz, which happens to be A3. We already know that A4 is 440Hz. But what if you add 110 to 220? You get 330Hz. Which is (mathmatically rounded) E4. E is the fifth in an A-Chord. The third is C#, which as it turns out, after you add 110 to 330 to get A440, then add 110 again to 440 you get 550 Hz which is C#!
So as you play an A Chord you are playing sound waves that have very clear mathematical-physics-audio-wave relationships! People didn't JUST create music that sounded good, or stumble upon some ways to make sense of why certain sounds worked well together. There is very clear scientific-physics-math relationships going on that makes those relationships work.
A real trip is Phantom Fundamentals. These are when you are playing all the harmonics of a note but not the actual note itself. You ear knows what this means and will actually fill in the missing information. So you can go into a computer, produce many of the harmonics of a note but not the note itself and you will actually feel as though you are hearing that note! This is interesting from an EQ standpoint. You might need to cut out something that is causing trouble and you might be killing the fundamental note, but your ear will still hear it because of the harmonics you left in place. Very interesting.
These various harmonics play out in the frequencies we find ourselves working with. If we have phase canceling out certain portions of the audio realm we may find a production sounding too thing, heavy, thick, muddy, boomy, harsh, etc... and all of this may root back to some understanding of how these things are interacting in the harmonic nature of music. It's not the only factor, but it's A factor, and an interesting one. Forgive me for taking a little rabbit trail here with this side note.
Side Note: I find that exciting because it shows, yet again, scientifically that there HAS to be an intelligent creator behind the universe, because there is NO SUCH THING as design without a designer! What a cool thing to study Audio Frequencies and find God hiding there! If Satan was a musician, and many scholars believe he was, do you think he understands this better than we do, and how certain combinations of sound-pressure-waves can impact our body, mind, heart, spirit, etc in ways that we don't yet fully comprehend? Why does certain music make you feel happy or sad or angry or nervous or hopeful? Have you ever tried listening to a profound movie moment without the music? I think there is more to be said here, but that's another post. Here's an example of what I mean. (Click Here).
Waves:
There are many types of waves. Sine waves are pure notes being played without other notes compounding them, and without any kind of interference. A test signal on TV stations just before whether warning might be an example. Since waves could also be comprehended as the sound of a flute. Simple, Distinct, Pure.
However Pure Sine Waves are actually quite rare in the real world, except when produced electronically. Many times a complex wave is actually what we hear most of. Complex waves are many different types of waves, and lots and lots of sine waves, coming together to form new waves in combination. This new combined wave is what we are actually hearing most of the time. When two singers sing a duet and they we hear them together, they are creating a complex wave form.
Some other "types" of waves that are not simple sine waves are often, but not always, electronically produced:
Triangle Waves,
Square Waves, Square waves are going to sound VERY digital.
Sawtooth Waves.
(See a science and math explanation Click Here) (See and Hear Examples of Sine, Triangle, Square, and Sawtooth waves Click Here and Click Here, or see embedded videos)
Math of Sound
Sound travels at 768 Miles Per Hour in dry air at 68*F (or 20*C). Other ways of saying this are:
1236 KPH
1 Mile every 5 Seconds
343.2 Meters Per Second
1126 Feet Per Second
It's this last one we'll deal with in live sound reinforcement, especially indoors, most often. This is because sound waves have very predictable behavior. Complicated, but predictable.
Also sound is pressure. Air Pressure hitting your ears creates sound. If it hits your ears in certain patterns you will recognize that sound. The sound of glass breaking is hard to miss. If you hear a car screech it's tires and you hear the crunch of metal and glass breaking you know there has just been car accident. You know this without even seeing it. You know your significant others voice, (Mom, Dad, Wife, Husband, Kids, Etc).
Sound Pressure waves traveling in a cycle can begin to sound like music. If sound pressure waves hit your ear 440 complete cycles per second (440Hz) you have just heard A4!
With this understanding we can begin to calculate the distance of a single cycle of a single wave. This is useful when understanding how difference sized wave cycles will interact with the room you are engineering.
Myers Sound, among others, have developed software that will allow you to enter the exact dimensions of a room, which exact loudspeakers you will be using, (which gives you their dispersion patterns), and their placement and then you can see how the sound will function in that room before you ever buy a single piece of equipment, before you even build the building!
Sound Pressure Waves travel at 1126 feet/second. The symbol used for a sound wave is the Greek "Lambda" symbol which is"λ". The speed of the wave is represented by a "v"
The formula for calculating the wave distance, Wavelength, is:
Wave (λ) = Speed (v) of wave in feet (Feet per second) divided by the number of feet (f).
λ = v/f
,
So if you know the frequency (λ) and you know the Speed (v) you can always find out how long the wave is. For example: A4 is 440Hz.= (λ). Sound travels at 1126 f/s = (v).
Therefore:
440 = 1126 / f which turns into
1126/440 = f which turns into
2.56 feet.
A single cycle of a 440 Hz sound wave is 2.56 feet long
I've never really understood fractions in algebra. I enjoy math concepts, but not actually doing it. I'm still working on that. Seems to be a good tutorial on fractions at MathIsFun.Com. Also some GREAT infographics specifically on wavelength at Wikihow.
Standing Waves
wikimedia
What if I wanted to know the frequency of a room that is most likely to be the same length wave as the room, or some duplicate of that length? Why would I want to know that?
Because if a wave is the same length as the room, or two cycles of that wave is equal to the length of the room or 3 cycles, etc, the wave may back in on itself and create a "Standing Wave". For the science of standing waves see: PhysicsClassroom or Wiki. The concept is that if you have a wave hitting the back wall of a room at exactly the the length of the wave, or some duplicate of it, you will have the wave reflect back in on itself.
This essentially doubles the strength of the wave. But as you send more original sound that direction and more reflections back in on itself they start working into each other in a unique form of constructive interference and the waves do not get any longer, but the amplitude gets higher and higher, in other words they get louder!
So you can have parts of a room, or even specific frequencies that actually begin to get louder and louder even though you haven't turned up anything on the board!
Perception of Sound
medicinenet
The Ear is an amazing tool, created by God, to perceive sound around us. The Cochlea is the tool inside the inner ear that perceives sound. There is a great deal of science that has gone into studying the ear both for Sound Reinforcement, because there is a lot of money in it, and for the deaf or partial deaf, because there's a lot of money in that science too.
At birth most humans have a hearing range from approximately 20 Hz to 20 kHz. But we do not hear equally at all frequencies.
The Fletcher Munson curve shows us how human hearing picks up certain frequencies much more easily than other frequencies. So a speaker producing 50Hz and 1.5kHz at the same pressure level will "sound" like different volumes to the human ear. Therefore when a room is tuned this curve must be taken into account. One way of seeing this is Pink Noise vs White Noise. Humans hear from approximately 1.5kHz to 4.5 kHz more sensitively than they do other ranges above and below.
Interestingly, sound actually travels 4.5 times faster in water, and 15 times faster in Iron. The Denser a material is the more particles there are to vibrate and interact with their neighbors. You see, much to my surprise, when you CLAP the air does not move from your hand to my ear. What actually happens in air (as in water and iron) is that the air particles compress and decompress causing the air particles around them to compress and decompress until finally they react to the air around your ears and that pressure hits the air in your ear canals and reacts to your ear.
mrescience
The best way I know to imagine this, since you cannot see air, is to see this happen in 2D on the surface of water. I challenge you to try this every so often. Go to a sink, bathtub, pond, pool, or all of the above, and drop something into it, or tap it with your finger. You'll see waves in the water. In a container the waves impact the walls and move back in and interact with new original waves and begin partial cancellations until eventually the waves all die down. In an Audio Environment this bouncing sound and interaction with the original source sound is called reflections and reverberations and echo.
A reverberation is perceived when the reflected sound wave reaches your ear in less than 0.1 second after the original sound wave. Since the original sound wave is still held in memory, there is no time delay between the perception of the reflected sound wave and the original sound wave. The two sound waves tend to combine as one very prolonged sound wave. If you have ever sung in the shower (and we know that you have), then you have probably experienced a reverberation. The Pavarotti-like sound which you hear is the result of the reflection of the sounds you create combining with the original sounds. Because the shower walls are typically less than 17 meters away, these reflected sound waves combine with your original sound waves to create a prolonged sound - a reverberation. (see the website link for a more detailed description.)
Low frequencies will tend:
Be less directional (they are perceived behind the speaker and in front equally)
Have more energy
To travel further
Medium Frequencies will tend:
To be more directional
Have less energy
Feel very PRESENT (due to the Fletcher Munson curve).
High Frequencies will tend:
To be VERY directional (you stand behind the speaker and they disappear)
Feel very CRISP up close.
Fall away quickly. At the back of the room you may not hear them but you'll still hear the mids and bass.
The directional nature of Mids and Highs is why you can stand outside of a booming worship service or concert, like I did this morning before going into a rocking church service at Gateway, and all I can hear is the booming of the drums and bass and some undertones of male voices, but nearly nothing else.
Why can I hear the bass, and a few mid range moments, but not the rest? This is due to this directionality feature of different frequency ranges, and that is due in part to wavelength. Some Sub-Bass frequencies can be 10-15 feet long. The just penetrate right through everything. Some very high frequencies can be mere inches long.
Well, suddenly I walk past two sets of double doors (inner and outer) and when I walk into the auditorium I am struck by how clear everything sounds now.
External Factors
Atmospheric Pressure, Humidity (water in air means thicker air), Temperature, Altitude (thiner or thicker air), Wind, and other things can all play a part in altering the color of your sound. Indoors this won't matter as much, but outside on a cold windy humid day in the mountains the same band playing the same song through the same PA and PA Settings could sound different than they would outside in a hot dry desert at sea level. These things all play a factor in how sound travels. Different frequencies have different wave lengths so they will be affected differently by these conditions.
Quiz:
If I play A2 on the piano the fundamental is 110 Hz. List 6 overtones present in this complex wave.
If I play the following sine waves simultaneously, what frequency will appear to be the fundamental of the perceived complex wave? In other words, what is the missing fundamental?
196 Hz, 392 Hz, 784 Hz , 1568 Hz , 3136 Hz
If I am in a room that is 8 foot by 8 foot. What is the wavelength that will create standing waves most in this room? Hint: λ = v/f
What is the frequency of Middle C?
What note is 195.998 Hz?
Answers *I used WHITE font on WHITE background. Simply Highlight the area below this to see the answers. You may have to copy them and past them to a Word Doc or Notepad and recolor them:
110(F), 220,330,440,550,660,770
98 Hz
(Each is half the other. 3136/2=1568... 1568/2=784...196/2=98)
140.75 Hz
Room is 8 Feet (f). Sound travels at 1126 f/s (v). 1126/8=f.
Which is somewhere between Db and D below Middle C.
The sine wave is the most basic component of understanding the science of sound. Sine wave images are a graphical representation of mathematical understanding of how sound pressure waves interact in an environment.
As it is graphed you find three points. From start to mid point is half a cycle and from the midpoint to end is half a cycle. In the real world picture a loudspeaker. The cone moves out and in and that completes one cycle. As the cone pushes air out it compresses the air in front of the speaker which in turn compresses the air in front of that air and so on. The cone then pulls back and sucks in the air behind the compression and this is called decompression/expansion phase. Another example of this in nature is a pendulum.
Take a YoYo completely uncoiled, hanging down, still holding onto the end of the string. In your other hand take the YoYo itself and hold it out to your side. Let go. As the YoYo swings down it doesn't stop at the middle by your feet, it swings past and up again, level to the other side from where you were holding it out.
The energy released by dropping the YoYo from one end has to fully expend, then it returns down and back toward where you let it go in the first place. But it doesn't quite come all the way back. Some energy from raising up to the opposite side and dropping again caused it swing back. With each swing it does not raise as high as it did before. Each swing has less energy than it did before. Until eventually it stops swinging as the energy is expended completely.
In the case of a speaker cone the initial push compresses the air, similar to releasing the YoYo. The cone returns back to normal but the speaker cone is pulled back as far in as far as it was pushed out, this pulls the air back in. The series of compression and de-compression creates sound. The push of the cone, and the drop of the YoYo, is represented as the first half of the sine wave on the graph. This is the curve going up to the right on the graph ending at a point even with the start. This is half of one cycle. The retraction of the cone, and the return of the YoYo, is the other half of the cycle.
Cycles
A series of these sound cycles together represent the sounds that you and I can hear and recognize. They are measured in what is known as Hertz. Hertz are a measurement of cycles per second. One Hertz is equal to One of these cycles per second. If the speaker cone moved out, back, and returned to normal sitting position once per second, this would be equal to 1-Hertz, also seen as 1Hz.
Audible sound, for most humans, and the sound that we deal with as Audio Engineers on sound boards, is measured as 20 Hz to 20,000 Hz. 1,000 Hz is also known as 1kHz. So most often you will see the measurement represented as 20 Hz to 20kHz.
Phase
If you were to invert the sine wave, meaning that it went down first then went up, or the speaker cone pulled back first, then pushed out, it would have NO effect on what you hear. The speaker would produce the same sound. It doesn't matter if compression hits you first or decompression, the sound is audible due to its pattern, and the pattern would be the same.
However, phase creates interesting effects in an environment when it interacts with other sounds. If a speaker is creating a sound and a paired speaker creates the same sound inverted, it will have a canceling effect. This is because the compression from one is hitting as the other speaker is sending decompression, then they both switch. Mathematically these cause total cancellation.
In the real world there is seldom any such things as total cancellation because of the many variables and factors involved in real world environments. First, in a 3D environment air can simply move up or down or around itself so some cancellation will be overcome. Second, Speaker A may have started compression micro seconds before Speaker B started decompression and therefore it is not exactly the inverse timing.
This type of cancellation though can have profound effects. You could be working speakers to the point of blowing them, sending WAY too much energy through and still not getting noticeable increase in volume that you expected. If one of your speakers is out of phase this could be the reason and your "turning it up" could blow the system and not solve the issue.
Reflections
Phase in the real world can also happen in a room, even if both speakers are set up right. There is a phenomena known as reflection. The sound wave bounces off of a wall, ceiling, floor, maybe hits another surface, wall to wall, ceiling to wall to floor and back up. As the sound waves begin to bounce and reflect they hit the new originating waves from the speakers. This can induce partial cancellations around different parts of the room. Due to the nature of sound waves being of different length, the cancellations could occur only at certain frequencies, and not simply across the entire sound. So you could be getting highs and lows but no mids, for example.
Phase Shift Alterations
Complete Phase Cancellation happens when a sine wave is confronted in time with it's exact inverse in perfect timing. This is actually rare. What is more likely is that the partial inverse confrontations will happen. Due, for example, to a reflection having to bounce off of the wall and another wall and back again it will not come back at the same point in time as the originating sound. They may meet micro seconds or longer. This creates what is known as constructive or destructive combination.
As waves combine milliseconds apart from each other, in and out of phase they can combine to create portions of a wave that is stronger and portions that are weaker. This then begins to make up the environment in which we are to start mixing sound. Some of this can be seen on the diagram above found on Wiki's website.
When you have a sound input coming in through your system there may be times that you want to reverse the waves, invert them. This is known as reversing the polarity. Again, there is no difference in how you hear a sound when it's inverted unless it's acting on a sound equal to itself but inverted from itself. This can be used to our benefit.
Example: Two singers, each holding a microphone a few feet apart from each other. Singer A holding Mic A will be loudest in Mic A. But Singer B will be picked up by Mic A too, just at a lower volume. As Mic B comes into the system it will have picked up Singer B louder in Mic B than Mic A naturally because Singer B is closer to Mic B. Now, if you invert Mic B (reverse the polarity) when the two Mics (A Plus B) are combined the reverse phase of the lower sound of Singer B in Mic A will electronically be canceled out by the louder Singer B in Mic B. The result is less system noise and a clearer Singer A and Singer B. (Remember the same thing is happening with Singer A's voice in Mic B and it's naturally reverse from Mic A too.)
