FUNDAMENTALS OF SOUND//VOL. I “What is Sound and How Do We Hear It?” written by MADELEINE CAMPBELL

illustrated by POY BORN

“there is in souls a sympathy with sounds: and as the mind is pitch’d the ear is pleased with melting airs, or martial, brisk or grave; some chord in unison with what we hear is touch’d within us, and the heart replies.” ­william cowper “sound is of no use to human evolution. in fact, it gets in the way.” ­haruki murakami

I) WHAT IS SOUND? the word “sound” describes how our brains receive and interpret an auditory stimulus. sound arrives at our ears in the form of sound­pressure waves that act as repetitive disturbances of air particles. these waves are caused by variations in the pressure of the medium in which they travel, such as air or water. for the purpose of this book, we will stick with air as the chosen medium. think of an inflated balloon about to pop. the moment you stick a needle through an inflated balloon, air molecules in the center exert pressure on the surrounding molecules in an attempt to get to an area of lower pressure. like dominos, these molecules will exert a force on the molecules on their surrounding molecules and the trend continues. these continued disturbances create waves.

II) CHARACTERISTICS OF SOUND sound is measured in decibels ­ a logarithmic unit that describes the loudness and intensity of a sound. a waveform is a graphic representation of sound­pressure waves traveling through air over time. wave forms have multiple different characteristics that will affect how they sound.

i) wavelength

wavelength describes the distance from the beginning to end of the waveform as it completes one 360 degree cycle. an easy way to identify a complete wavelength when looking at a waveform is to remember “crest to crest or trough to trough.” find the highest or lowest point of one wave and the highest or lowest point of the next wave ­ that is one complete 360 cycle. the time it takes to complete one cycle is called the period.

ii) frequency frequency is the number of cycles the waveform travels in one second. a cycle can begin at any degree point on the wave, however, it must travel the full 360 degrees to be considered a full cycle. sounds with higher frequencies have higher pitches. sounds with lower frequencies have lower pitches. frequency is measured in hertz (hz). the standard concert “a” to which most professional orchestras tune is usually 440hz, give or take one or two hertz in either direction. when a frequency is doubled, the pitch of a note increases one octave. when a frequency is cut in half, the pitch will decrease an octave. this means that 220hz will be the “a” below the standard concert “a” and 880z will be the “a” above. frequency and wavelength are inversely proportional. this means as one increases, the other decreases. if the wavelength increases, the frequency decreases and vice versa.

iii) amplitude

amplitude is the distance above or below the centerline of a waveform. it measures the amount of change in air pressure caused by sound waves. a larger displacement from the centerline equates to a greater pressure variation and when listening back to a recorded waveform, a louder volume.

root mean square (rms) is a method developed to attempt to find an appropriate average of the amplitudes of a waveform over time. amplitudes at different points along the waveform are squared and subsequently averaged. the rms value will always be positive because the square of any number, positive or negative, is positive. in a perfect sine wave, the rms value will always be .707 multiplied by the peak amplitude ­ the measurement of the greatest positive or negative value. a frequency response chart is a good way to measure the relationship between frequency and amplitude. frequency is found on the x­axis and amplitude on the y­axis. the graph will show what happens to the amplitude when frequency increases or decreases from a certain point and vice versa.

iv) phase sound wave forms occur in repetitive cycles. the phase of a wave is the angular point in the cycle the given wave is in at a specific time. when two or more identical waves are in phase with one another, meaning they are at the same point in the cycle at the same time, their amplitudes will increase.

when two or more identical waves are out of phase with one another, meaning they are at different points in the cycle, there will be resulting cancellation. when playing back out­of­phase recorded sound waves, they will sound hollow in comparison to in­phase waves. when two identical waves are out of phase, the amount that one wave is ahead or behind another is called the phase shift. when two identical waves are completely out of phase with each other meaning they are 180 degrees out of phase, they will cancel each other out.

v) shape waves are divided into two categories: simple and complex. simple waves are have identical harmonic structure and look the same as one another. they are repetitive. examples of these waves include sine waves (like you saw before), triangle waves, sawtooth waves and square waves.

most audio waves are complex waves. they are created by naturally occurring sound and are rarely exactly symmetrical. they can be broken down into various combinations of individual sine waves.

vi) timbre

the characteristics mentioned up to this point focus on the individual sine wave itself ­ which is made of a singular frequency which produces a singular pitch. if musical instruments produced perfect sine waves, they would sound no different from one another. timbre (pronounced tam­ber) is what gives different instruments their distinctive sounds ­ what makes a 440hz concert a sound different on a trumpet versus a flute.

vii) harmonic envelope

a waveform’s harmonic envelope describes the variations in level over the duration of a note. it is broken up into four distinct sections:

• the attack describes how long it takes for the waveform to reach its full volume immediately after it happens • the decay describes how long it will take the volume to reach a level of sustain after the initial attack • the sustain describes the volume at which the note is held after the decay period • the release describes how quickly the note takes to cease after it has been released

III) THE HUMAN EAR now the bare basics of sound waveforms are covered, but how do we perceive them? sound waves travel through the air and arrive at our ear. the human ear is a prime example of a transducer ­ something that converts energy from one form to another. the human ear is divided into three main sections ­ the outer ear, the middle ear and the inner ear. sound waves travel through the air and arrive at the ear. they are then collected at the pinna ­ part of the outer ear ­ and travel through to the ear drum. here the waves are transformed into mechanical vibrations which are then carried to the inner ear through three small bones called the hammer, the anvil and the stirrup. the hammer, anvil and stirrup serve two main purposes. first, they amplify the sound by increasing the vibrations that are coming from the ear drum. then, they act as a protective limiter by reducing extremely loud, harsh sounds ­ like explosions and fireworks. the vibrations then travel to the cochlea ­ a small tube­like organ in the inner ear that holds two fluid filled sacks. these hold tiny hairs that span the length of the cochlea. these hairs respond to the various frequencies of the vibrations depending on where they are placed along the cochlea. these vibrations cause an electric signal which the auditory nerve ­ a.k.a the hearing nerve ­ then transmits to the brain. these small electric signals are what the brain interprets as sound. our ears can hear a wide range of sound waves – extremely low sounds from about 20 hz to extremely high sounds around 20,000 hz. as humans get older, it is common for the high­end range to deteriorate, making it difficult to hear higher sounds ­ around the 15k hz range and up. this condition is called presbycusis.

IV) MICROPHONES

microphones are another kind of transducer that serve a critical part in hearing amplified sound. generally speaking, a microphone is an acoustic­to­electric transducer that takes sound waves and converts them into an electric signal. this signal is then able to be recorded and played back. most microphones are categorized into three smaller groups based on their transducer principle: dynamic

microphones, condenser microphones and ribbon microphones. other types of microphones exist – but we will start with these three.

all microphones have one major similarity – a diaphragm. this is a thin piece of material (different in various microphones) usually located in the top part that vibrates when it is struck by a sound wave. these vibrations cause other parts of the microphone to vibrate which induces an electrical signal.

i) dynamic

dynamic microphones, sometimes called moving coil microphones, are especially good for loud sounds. they can handle a high amount of audio signal like screaming vocals or loud drums sounds. dynamic microphones work by the principle of electromagnetic induction. in a dynamic microphone, the diaphragm is attached to a small wire coil that is fixed around a permanently placed magnet. as sound waves enter the microphone, the diaphragm moves causing the attached coil to move around the magnet. this produces current (signal) within the coil.

ii) ribbon a ribbon microphone, like the dynamic, works by electromagnetic induction. its diaphragm is made of extremely thin ribbon ­ so delicate that it can disintegrate if you breathe on it too heavily. the small ribbon is corrugated and suspended into a magnetic field. as sound waves enter the microphone, they cut across the ribbon’s slits and cause it to move back and forth. this creates an electric signal within the magnetic field. ribbon microphones are far more sensitive than dynamic microphones and their output is much lower.

iii) condenser condenser microphones, also known as capacitor microphones, work differently than dynamic and ribbon microphones. a capacitor is simply an electronic component that stores electric charge. they function upon the electrostatic principle as opposed to the electromagnetic principle. the diaphragm acts as one plate of a capacitor which is closely situated next to the other, a metal plate. the diaphragm is a membrane and is coated with a gold metallic material to allow it to induce charge. when a sound wave hits the diaphragm, it vibrates, causing variations in distance between the two plates and in the output voltage. more specifically, when the two plates are close together, capacitance increases and a charge current occurs. when the two plates are farther apart, capacitance decreases. the fluctuations in capacitance produce a small but audible signal.

iv) phantom power another major difference between condenser microphones versus dynamic and ribbon microphones is phantom power. the inside of condenser microphones contain active electronics. all devices with active electronics have their own power supply and the potential to add to or amplify a signal in some way. (this is the opposite of passive electronics ­ which don’t use amplifiers and only take away from an

audio signal) for a condenser microphone, the necessary power supply is phantom power ­ a 48 volt flow of direct current that is transmitted through microphone cables to provide power.

v) electret another type of capacitor microphone is the electret condenser. inside the microphone is a permanently charged plate. when sound waves hit the diaphragm, it moves in response creating variations in capacitance with the plate. the most significant different between these microphones and “normal” condensers is that electrets themselves do not require external power supplies, however these microphones frequently contain internal preamplifiers that are phantom powered. because of their high quality and low cost of production, they are used in most cell phones, small hand­held recording devices and headphone sets.

vi) polar patterns in addition to being sorted by their transducer principle, microphones are also classified by their polar pattern ­ their sensitivity to sound in regards to the direction from which the sound travels and arrives at the microphone. ++ an omnidirectional microphone picks up sound from all directions equally. an advantage of omni microphones is that they do not need to be aimed in a certain direction towards the sound source. however, they are unable to reject any noise you may not want, like crowd or room noise. ++ a cardioid microphone has high sensitivity to sound from the front and rejects sound from the back. this is helpful when miking multiple instruments at once. for example, when miking a drum kit, use a cardioid microphone on the snare drum so that the cymbal next to it will be rejected and not bleed into the snare’s microphone track.

++supercardioid microphones have good sound pickup from the front, however, it is slightly narrower than cardioid microphone and thus will reject more ambient sound. it also has narrow sound pickup at its rear. this is especially useful when a single sound source needs to be captured in a loud environment.

++figure 8 microphones (also called bidirectional) pick up sound from the front and the rear, but not the sides. most ribbon microphones have the figure 8 pattern.

V) SIGNAL FLOW so if sound goes into a microphone, then how do we hear it amplified? this is where the basic ideas of signal flow come in.

as previously stated, the internal parts of a microphone generate a small electric signal. at this point in time, it is at mic level. it is so small (roughly .1 millivolt) that it must be raised before it can power a larger amplifier. to do this, that signal travels through the microphone cables to a pre­amplifier which accepts the signal at its input and increases it to what is called line level. essentially, the pre­amplifier boosts the signal enough to take on a much bigger amplifier. the preamplifier might be located in a recording and mixing console ­ those large sound boards you may have seen if you’ve ever walked into a recording studio or attended a live concert. often, professional recording studios have various outboard preamps that are used for specific sounds or recording situations. once a pre­amp brings the signal to line level, it is able to undergo signal processing ­ a series of methods of manipulating audio signal in a way other than simply amplifying the signal. this includes equalization ­ the ability to raise or lower the amplitude of a sound’s specific frequency ranges. for example, you can lower the 100­300hz range a few decibels to make a woofy kick drum sound a little less muddy. signal processing may also deal with the dynamics of a signal. through a compressor, the dynamic range ­ the ratio between the loudest part of the signal to the quietest part of the signal ­ is reduced. if the dynamic range of a band’s song is 50 db, that means the loudest parts are 50 db stronger than the softest. compression is frequently used in live sound to ensure that the signal will stay in a certain range,

even if the band plays too loudly for the recording devices to handle. compressors are also frequently used to boost the overall dynamic level of certain signals. signal processing may also affect the timing of a signal. delay processors hold a signal for a designated amount of time then release it so it is delayed in relation to other signals. this creates often creates an echo effect.

reverb units may also be employed. reverberation describes the sound remaining in a room after the source of the sound has stopped. for example, if you sing a note in a cathedral, it will reverberate, depending on its size, for a few seconds after you stop singing. reverb units can simulate all different kinds of reverb, depending on the size and sound of what you are looking for after a signal has been processed, it is sent out of the mixing console or audio interface through the line outputs (remember, it went in through the line inputs) to another amplifier ­ frequently called the power amplifier ­ that boosts the signal up to speaker level. this is usually somewhere around 10v. out of the amplifier, it travels through a speaker cable to the last part of the audio chain: the loudspeaker.

a loudspeaker is also a transducer, however, it performs the reverse of what the microphone does, taking the amplified electric signal and converting it back to acoustical energy ­ now an audible sound. dynamic speakers, also called moving coil speakers, work in exactly the opposite way that a dynamic microphone works. they contain a metal coil, a diaphragm, a magnet and a cone. the electrical current is transmitted out of the amplifier and into the speaker where it flows through the coil, creating an electromagnetic field. this reacts with a permanently placed magnet behind it. the voice coil is attached to the diaphragm which is connected to the cone in front. the magnetic fluctuations cause the diaphragm, and in turn the cone, to move back and forth. this moves the surrounding air all around and the sound is projected forwards. in addition to dynamic/moving coil speakers, ribbon speakers are also used. ribbon speakers, like ribbon microphones, have a thin ribbon suspended in a magnetic field. when the electrical signal hits the ribbon, it moves back and forth to create sound that is projected outwards for you to hear. ~~~~

these are some of the fundamentals of sound.

copyright 2013//madeleine campbell and poy born/don’t steal our work madeleine@treelady.com