SCHOOL OF AUDIO ENGINEERING
Diploma in Audio Engineering

RA 101: INTRODUCTION TO STUDIO STUDIES

RA 101.1: INTRODUCTION TO AUDIO

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INTRODCUTION TO AUDIO Identifying the Characteristics of Sound Sound and music are parts of our everyday sensory experience. Just as humans have eyes for the detection of light and color, so we are equipped with ears for the detection of sound. We seldom take the time to ponder the characteristics and behaviors of sound and the mechanisms by which sounds are produced, propagated, and detected. The basis for the understanding of sound, music and hearing is the physics of waves. Sound is a wave which is created by vibrating objects and propagated through a medium from one location to another. In this subject, we will investigate the nature, properties and behaviors of sound waves and apply basic wave principles towards an understanding of music. The Elements of Communication Communication: transfer of information from a source or stimulus through a medium to a reception point. The medium through which the information travels can be air, water, space or solid objects. Information that is carried through all natural media takes the form of waves - repeating patterns that oscillate back and forth. E.g. light, sound, electricity radio and TV waves. Stimulus: A medium must be stimulated in order for waves of information to be generated in it. A stimulus produces energy, which radiates outwards from the source in all directions. The sun and an electric light bulb produce light energy. A speaker, a vibrating guitar string or tuning fork and the voice are sound sources, which produce sound energy waves. Medium: A medium is something intermediate or in the middle. In an exchange of communication the medium lies between the stimulus and the receptor. The medium transmits the waves generated by the stimulus and delivers these waves to the receptor. In acoustic sound transmission, the primary medium is air. In electronic sound transmission the medium is an electric circuit, Sound waves will not travel through space although light will. In space no-one can hear you scream. Reception/Perception: A receptor must be capable of responding to the waves being transmitted through the medium in order for information School of Audio Engineering Chennai 1

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to be perceived. The receptor must be physically configured to sympathetically tune in to the types of waves it receives. An ear or a microphone is tuned in to sound waves. An eye or a camera is tuned in to light waves. Our senses respond to the properties or characteristics of waves such as frequency, amplitude and type of waveform. What is Sound? Sound is a disturbance of the atmosphere that human beings can sense with their hearing systems. Such disturbances are produced by practically everything that moves, especially if it moves quickly or in a rapid and repetitive manner. The movement could be initiated by: Hammering (rods), plucking (string), Bowing (strings), Forced air flow (vibration of air column - Organ, voice). Vibrations from any of these sources cause a series of pressure fluctuations of the medium surrounding the object to travel outwards through the air from the source. You should be aware that the air is made up of molecules. All matter is made out of particles and air is no exception. As you sit there in the room, you are surrounded by molecules of oxygen, nitrogen, carbon dioxide and some pollutants like carbon monoxide. Normally these particles are all moving around the room, randomly dispersed and all roughly the same distance from adjacent particles as all of the others. The distance between these particles is determined by the air pressure in the room. This air pressure is mostly a result of the barometric pressure of the particular day. If it’s raining outside, you’re likely to be in a low pressure system, so the air particles are further apart from each other than usual. On sunny days you’re in a high pressure system so the particles are squeezed together. Most of the characteristics we expect of air are a result of the fact that these particular molecules are very light and are in extremely rapid but disorganized motion. This motion spreads the molecules out evenly, so that any part of an enclosed space has just as many molecules as any other. If a little extra volume were to be suddenly added to the enclosed space (say by moving a piston into a box), the molecules nearest the new volume would move into the recently created void, and all the others would move a little farther apart to keep the distribution even or in ‘equilibrium’.

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Because the motion of the molecules is so disorganized, this filling of the void takes more time than you might think, and the redistribution of the rest of the air molecules in the room takes even longer. If the room were ten feet across, the whole process might take 1/100 of a second or so. If the piston were to move out suddenly, the volume of the room would be reduced and the reverse process would take place, again taking a hundredth of a second until everything was settled down. No matter how far or how quickly the piston is moved, it always takes the same time for the molecules to even out. In other words, the disturbance caused by the piston moves at a constant rate through the air. If you could make the disturbance visible somehow, you would see it spreading spherically from the piston, like an expanding balloon. Because the process is so similar to what happens when you drop an apple into a bucket, we call the disturbance line the wavefront. It is important to note that the particles of the medium in this case molecules of air, do not travel from the source to the receiver, but vibrate in a direction parallel to the direction of travel of the sound wave. The transmission of sound energy via molecular collision is termed propagation. If the piston were to move in and out repetitively at a rate between 20 and 20,000 times a second, a series of evenly spaced wave fronts would be produced, and we would hear a steady tone. The distance between wave fronts is called wavelength. Wave Theory Waves are everywhere. Whether we recognize or not, we encounter waves on a daily basis. Sound waves, visible light waves, radio waves, microwaves, water waves, sine waves, waves on a string, and slinky waves and are just a few of the examples of our daily encounters with waves. In addition to waves, there is a variety of phenomenon in our physical world which resembles waves so closely that we can describe such phenomenon as being wavelike. The motion of a pendulum, the School of Audio Engineering Chennai 3

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motion of a mass suspended by a spring, the motion of a child on a swing, and the “Hello, Good Morning!” wave of the hand can be thought of as wavelike phenomena. We study the physics of waves because it provides a rich glimpse into the physical world which we seek to understand and describe as physicists. Before beginning a formal discussion of the nature of waves, it is often useful to ponder the various encounters and exposures which we have of waves. Where do we see waves or examples of wavelike motion? What experiences do we already have which will help us in understanding the physics of waves? Waves, as we will learn, carry energy from one location to another. And if the frequency of those waves can be changed, then we can also carry a complex signal which is capable of transmitting an idea or thought from one location to another. Perhaps this is one of the most important aspects of waves and will become a focus of our study in later units. Waves are everywhere in nature. Our understanding of the physical world is not complete until we understand the nature, properties and behaviors of waves. The goal of this unit is to develop mental models of waves and ultimately apply those models in understanding one of the most common types of waves - sound waves. A wave can be described as a repeating and periodic disturbance that travels through a medium from one location to another location. But what is meant by the word medium? A medium is a substance or material which carries the wave. A wave medium is the substance which carries a wave (or disturbance) from one location to another. The wave medium is not the wave and it doesn’t make the wave; it merely carries or transports the wave from its source to other locations. In the case of a water wave in the ocean, the medium through which the wave travels is the ocean water. In the case of a sound wave moving from the orchestral choir to the seats in the house, the medium through which the sound wave travels is the air in the room. To fully understand the nature of a wave, it is important to consider the medium as a series of interconnected or merely interacting particles. In other words, the medium is composed of parts which are capable of

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interacting with each other. The interactions of one particle of the medium with the next adjacent particle allow the disturbance to travel through the medium. In the case of the slinky wave, the particles or interacting parts of the medium are the individual coils of the slinky. In the case of a sound wave in air, the particles or interacting parts of the medium are the individual molecules of air. And in the case of a stadium wave, the particles or interacting parts of the medium are the fans in the stadium.

Consider the presence of a wave in a slinky. The first coil becomes disturbed and begins to push or pull on the second coil; this push or pull on the second coil will displace the second coil from its equilibrium position. As the second coil becomes displaced, it begins to push or pull on the third coil; the push or pull on the third coil displaces it from its equilibrium position. As the third coil becomes displaced, it begins to push or pull on the fourth coil. This process continues in consecutive fashion, each individual particle acting to displace the adjacent particle; subsequently the disturbance travels through the medium. The medium can be pictured as a series of particles connected by springs. As one particle moves, the spring connecting it to the next particle begins to stretch and apply a force to its adjacent neighbor. As this neighbor begins to move, the spring attaching the neighbor to its neighbor begins to stretch and apply a force on its adjacent neighbor. Sound propagates through air as a longitudinal wave. The speed of sound is determined by the properties of the air (or a medium to be more specific), and not by the frequency or amplitude of the sound. Sound waves, as well as most other types of waves, can be described in terms of the following basic wave phenomena: (i) Transverse waves (ii) Longitudinal waves

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Transverse Waves For transverse waves the displacement of the medium is perpendicular to the direction of propagation of the wave. A ripple on a pond is an easily visualized transverse wave.

Elasticity and a source of energy are the preconditions for periodic motion. A pond has an equilibrium level, and gravity serves as a restoring force. When work is done on the surface to disturb its level, a transverse wave is produced. Waves on a Pond A pebble thrown into a pond will produce concentric circular ripples which move outward from the point of impact. If a fishing float is in the water, the float will bob up and down as the wave moves by. This is a characteristic of transverse waves.

Longitudinal Waves In longitudinal waves the displacement of the medium is parallel to the propagation of the wave. A wave in a “slinky” is a good visualization. Sound waves in air are longitudinal waves.

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A sound wave is a classic example of a longitudinal wave. As a sound wave moves from the lips of a speaker to the ear of a listener, particles of air vibrate back and forth in the same direction and the opposite direction of energy transport. Each individual particle pushes on its neighboring particle so as to push it forward. The collision of particle No.1 with its neighbor serves to restore particle No.1 to its original position and displace particle No.2 in a forwards direction. This back and forth motion of particles in the direction of energy transport creates regions within the medium where the particles are pressed together and other regions where the particles are spread apart. Longitudinal waves can always be quickly identified by the presence of such regions. This process continues along the chain of particles until the sound wave reaches the ear of the listener.

Sound is a Mechanical Wave Mechanical waves are waves which require a medium in order to transport their energy from one location to another. Because mechanical waves rely on particle interaction in order to transport their energy, they cannot travel through regions of space which are devoid of particles. That is, mechanical waves—like sound waves, cannot travel through a vacuum. Sound is a mechanical wave which results from the longitudinal motion of the particles of the medium through which the sound wave is moving.

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Explanation First, there is a medium which carries the disturbance from one location to another. Typically, this medium is air; though it could be any material such as water or steel. The medium is simply a series of interconnected and interacting particles. Second, there is an original source of the wave, some vibrating object capable of disturbing the first particle of the medium. The vibrating object which creates the disturbance could be the vocal chords of a person, the vibrating string and sound board of a guitar or violin, the vibrating tines of a tuning fork, or the vibrating diaphragm of a radio speaker. Third, the sound wave is transported from one location to another by means of the particle interaction. If the sound wave is moving through air, then as one air particle is displaced from its equilibrium position, it exerts a push or pull on its nearest neighbors, causing them to be displaced from their equilibrium position. This particle interaction continues throughout the entire medium, with each particle interacting and causing a disturbance of its nearest neighbors. Since a sound wave is a disturbance which is transported through a medium via the mechanism of particle interaction, a sound wave is characterized as a mechanical wave.

Sound is a Pressure Wave Sound is a mechanical wave which results from the longitudinal motion of the particles of the medium through which the sound wave is moving. If a sound wave is moving from left to right through air, then particles of air will be displaced both rightward and leftward as the energy of the sound wave passes through it. The motion of the particles parallel (and anti-parallel) to the direction of the energy transport is what characterizes sound as a longitudinal wave. A vibrating tuning fork is capable of creating such a longitudinal wave. As the tines of the fork vibrate back and forth, they push on School of Audio Engineering Chennai 8

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neighboring air particles. The forward motion of a tine pushes air molecules horizontally to the right and the backward retraction of the tine creates a low pressure area allowing the air particles to move back to the left. Because of the longitudinal motion of the air particles, there

are regions in the air where the air particles are compressed together and other regions where the air particles are spread apart. These regions are known as compressions and rarefactions respectively. The compressions are regions of high air pressure while the rarefactions are regions of low air pressure. The diagram below depicts a sound wave created by a tuning fork and propagated through the air in an open tube. The compressions and rarefactions are labeled. Since a sound wave consists of a repeating pattern of high pressure and low pressure regions moving through a medium, it is sometimes referred to as a pressure wave. The crests of the sine curve correspond to compressions; the troughs correspond to rarefactions; and the “zero point” corresponds to the pressure which the air would have if there were no disturbance moving through it. The diagram above depicts the correspondence between the longitudinal nature of a sound wave and the pressure-time fluctuations which it creates. The above diagram can be somewhat misleading if you are not careful. The representation of sound by a sine wave is merely an attempt to illustrate the sinusoidal nature of the pressure-time fluctuations. Do not conclude that sound is a transverse wave which has crests and troughs. Sound is indeed a longitudinal wave with compressions and rarefactions. As sound passes through a medium, the particles of that medium do not vibrate in a transverse manner. Do not be misled sound is a longitudinal wave. Simple Harmonic Motion When a mass is acted upon by an elastic force which tends to bring it back to its equilibrium

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configuration, and when that force is proportional to the distance from equilibrium, then the object will undergo simple harmonic motion when released. A mass on a spring is the standard example of such periodic motion. If the displacement of the mass is plotted as a function of time, it will trace out a pure sine wave. It turns out that the motion of the medium in a traveling wave is also simple harmonic motion as the wave passes a given point in the medium.

Wave Graphs Waves may be graphed as a function of time or distance. A single frequency wave will appear as a sine wave in either case. From the distance graph the wavelength may be determined. From the time graph, the period and frequency can be obtained. From both together, the wave speed can be determined. Amplitude/Loudness/ Volume/Gain

When describing the energy of a sound wave the term amplitude is used. It is the distance above or below the centre line of a waveform (such as a pure sine wave). The greater the displacement of the molecule from its centre position, the more intense the pressure variation or physical displacement, of the particles, within the medium. In the case of air medium it represents the pressure change in the air as it deviates from the normal state at each instant. School of Audio Engineering Chennai 10

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Amplitude of a sound wave in air is measured in Pascal which is a unit of air pressure. However for audio purposes air pressure differences are more meaningful and these are expressed by the logarithmic power ratio called the Bel or Decibel (dB). Waveform amplitudes are measured using various standards. Peak Amplitude refers to the positive and negative maximums of the wave. Root Means Squared Amplitude (RMS) gives a meaningful average of the peak values and more closely approximates the signal level perceived by our ears. RMS amplitude is equal to 0.707 times the peak value of the wave. Our perception of loudness is not proportional to the energy of the sound wave this means that the human ear does not perceive all the frequencies at the same intensity. We are most sensitive to tones in the middle frequencies (3 kHz to 4 kHz) with decreasing sensitivity to those having relatively lower or higher frequencies. This phenomenon will be discussed in detail later in this text using Fletcher and Munson Curves. Loudness and Volume are not the same: Hi-fi systems have both a loudness switch and a volume control. A volume control is used to adjust the overall sound level over the entire frequency range of the audio spectrum (20Hz to 20 kHz). A volume control is not frequency or tone sensitive, when you advance the volume control; all tones are increased in level. A loudness switch increases the low frequency and high frequency range of the spectrum while not -affecting the mid range tortes. Frequency The rate of repetition of the cycles of a periodic quantity, such as a sound wave is called frequency. Frequency can be defined as the number of cycles that a periodic waveform completes in the time of 1 (one) second.
Fig. 1

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Frequency is denoted by the symbol ƒ, and is measured in Hertz (Hz) - formerly called cycles per second (cps or c/s) - kilohertz (kHz), or megahertz (MHz). The three diagrams on the right Fig. 2 illustrate this. The first diagram shows a complete cycle of a sine wave completing 1 cycle in 1 second. Its frequency is therefore 1Hz. The second diagram shows a sine wave completing 5 cycles in 1 second. Its frequency is thus 5Hz. Fig. 3 The last diagram shows the same sine wave completing 10 cycles in the time frame of 1 second. Its frequency is 10Hz. Likewise a 1000Hz tone will complete 1000 cycles in a second and a 20,000Hz tone complete 20,000 cycles in 1 second. The only sound which consists of a single frequency is the pure sine tone such as produced by+a sine wave oscillator or approximated by a tuning fork. All other sounds are complex, consisting of a number of frequencies of greater or lesser intensity. The frequency content of a sound is commonly referred to as its spectrum or spectra. The subjective sense of frequency is called pitch. That is, frequency is an acoustic variable, whereas pitch is a psychoacoustic one. Wavelength The wavelength of a wave is merely the distance (in meters) which a disturbance travels along the medium in one complete wave cycle. Since a wave repeats its pattern once every wave cycle, the wavelength School of Audio Engineering Chennai 12

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is referred to as the length of one complete wave. For a transverse wave, this length is commonly measured from one wave crest to the next adjacent wave crest, or from one wave trough to the next adjacent wave trough. Since a longitudinal wave does not contain crests and troughs, its wavelength must be measured differently. A longitudinal wave consists of a repeating pattern of compressions and rarefactions. Thus, the wavelength is commonly measured as the distance from one compression to the next adjacent compression or the distance from one rarefaction to the next adjacent rarefaction. The Greek lambda (λ) is used as an abbreviation for wavelength

Below are some examples of various frequencies and their corresponding wavelengths by using the formula λ = c÷ f 20H =17.2m 100Hz =3.44m 500Hz =0.688m (68.8 cm) 1000Hz =0.344m (34.4cm) 10000Hz =0.0344m (3.44cm) 20000Hz =0.0127m (1.72cm)

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Traveling Wave Relationship A single frequency traveling wave will take the form of a sine wave. A snapshot of the wave in space at an instant of time can be used to show the relationship of the wave properties frequency, wavelength and propagation speed. The motion relationship “distance = speed of sound x time” is the key to the basic wave relationship λ = c x T and using this can be expressed in the standard form:

c
Speed of Sound We have seen that in an elastic medium such as air, the pressure wave of alternating condensations and rarefactions moves away from the source of the disturbance. Since the movement of sound wave is not confined to a specific direction, it should be referred to in terms of speed. The lowercase letter c is usually used as the specific abbreviation for speed of sound in air. The speed of sound in dry air is given approximately by;

feet/ sec

The speed of sound at 0ºC in air at sea level is 331.4 m/s. The speed of sound increases 0.6 m/s for every 1ºC increase in the temperature. Likewise, the speed of sound decreases 0.6m/s for every 1ºC decrease in the temperature.

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The speed of sound is slower at higher altitudes since the air is much thinner. In the vacuum of space, sound cannot propagate. Factors like pollution and humidity in air increase the speed of sound slightly. The speed of sound is faster in denser mediums. The table to your right compares the speed of sound in different mediums

Example 1 What would be the speed of sound be if the temperature was 20ºC? 331.4 + (0.6 x 20) = 331.4 + 12 = 343.4 m/s Example 2 How far can a 100Hz sine wave travel in 2 seconds at 20ºC, sea level? From d = t x c, where t = 2 sec and speed of sound = 343.4 d = 2 x 343.4 = 686.8 meters Particle velocity The term particle velocity refers to the velocity at which a particle in the path of a sound wave is moved (displaced) by the wave as it passes. It should not be confused with the velocity at which the sound wave travels through the medium, which is constant unless the sound wave encounters a different medium in which case the sound wave will refract. If the sound wave is sinusoidal (sine wave shape), particle velocity will be zero at the peaks of displacement and will reach a maximum when passing through its normal rest position. Phase If a mass on a rod is rotated at constant speed and the resulting circular path illuminated from the edge, its shadow will trace out simple harmonic motion. If the shadow vertical position is traced as a function of time, it will trace out a sine wave. A full period of the sine wave will correspond to a complete circle or 360

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degrees. The idea of phase follows this parallel, with any fraction of a period related to the corresponding fraction of a circle in degrees. The phase of any point on a sine wave measures that point’s distance from the most recent positive-going zero crossing of the waveform. The phase shift between two points on the same sine wave is simply the difference between their two phases.

However, phase shift is more often used to describe the instantaneous relationship between two measured sine waves in relation to time. The repetitive nature of sound waves allows them to be divided into equal intervals and measured in degrees, with 360º in one full cycle. If another sine wave of identical frequency is delayed 90º, its time relationship to the first sine wave is said to be a quarter wave late. A half-wave late would be 180º, and so on. Waves that are of the same frequency which commence at the same time are said to be in phase, phase coherent or phase correlated. In such a case, the phase angle between the two sine waves would be 0º Those commencing at different times are said to be out of phase, phase incoherent or uncorrelated. We know that when two waves of identical frequency and amplitude cancel out each other when they are 180º out of phase. So when, there is a phase difference and amplitude difference between them which is less than 180° and more than 180°, you will have a certain amount of cancellation and also addition in the amplitude of the fundamental of the waveform. When more waveforms with different phase and amplitude levels are added on, the waveform in slowly transformed.

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This is one of the factors which play a part in how complex waveforms are shaped. Most phase problems result in the loss of low frequency (bass) and coloration of high frequencies. You can try this at home by listening to a song you know well on your home hi-fi system. Then reverse the positive and negative leads on one speaker and listen to the song again. You will find that the sound seem ‘wider’ than before but has less bass content and lacks high frequency definition. The formula to calculate phase angle between waveforms is given by; Phase (ø) = Time (in seconds) x Frequency x 360° Example 1 For instance, a 250Hz wave, delayed by 1ms will be shifted in phase by 90°. How do we get this value? Phase (ø) = Time (in seconds) x Frequency x 360° Phase (ø) = 0.001s x 250x 360°= 90° Example 2 What would be the phase shift if the same wave was delayed by 0.25ms? Phase (ø) = Time (in seconds) x Frequency x 360° Phase (ø) = 0.00025s x 250 x 360°= 22.5° Example 3 If you want to shift two identical 500Hz waveforms with a phase relationship of 180°, how much delay would you apply to either waveform to be phase-shifted by that amount? Phase (ø) = Time (in seconds) x Frequency x 360° 180° = (X sec) x 500 x 360° X sec = 180°÷ 500 x 360°= 180÷ 180 000= 0.001 seconds or 1ms Comb filter or Combing Alteration of the frequency response of a system as a result of constructive and destructive interference between a signal and a delayed version of the signal is called Comb filtering. This is often created acoustically in small rooms because of the interaction between the direct sound and the reflected resulting in a series of peaks and nulls in the frequency response. Plotted on a linear frequency scale, such a comb-filter response looks like the teeth of a comb.

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The Behavior of Waves
Although it is convenient to discuss sound transmission theory in terms of free space (or also known as free field) environment, in practice the sound pressure waves encounters all sorts of obstacles in its path. In addition to regular room surfaces – walls, floor, and ceiling – there may be any number of additional objects that act as acoustic barriers. Each of these surfaces has an effect on the sound that is eventually heard and the effects usually vary considerably over the audio frequency range. Reflection of Sound

The reflection of sound follows the law “angle of incidence equals angle of reflection” as does light and other waves, or the bounce of a billiard ball off the bank of a table. The main item of note about sound reflections off of hard surfaces is the fact that they undergo a 180 degree phase change upon reflection. This can lead to resonances such as standing waves in rooms. It also means that the sound intensity near a hard surface is enhanced because the reflected wave adds to the incident wave, giving pressure amplitude that is twice as great in a thin “pressure zone” near the surface. This is used in pressure zone

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microphones to increase sensitivity. The doubling of pressure gives a 6 decibel increase in the signal picked up by the microphone. Reflection of waves in strings and air columns are essential to the production of resonant standing waves in those systems. Phase Change upon Reflection An important aspect of the reflection of sound waves from hard surfaces and the reflection of strings from their ends is the turning over of the wave when it reflects. This reversal (180 º change in phase) is an important part of producing resonance in strings. Since the reflected wave and the incident wave add to each other while moving in opposite directions, the appearance of propagation is lost and the resulting vibration is called a standing wave. Plane Wave Reflection “The angle of incidence (θi) is equal to the angle of reflection(θr)” is one way of stating the law of reflection for light in a plane mirror. Sound obeys the same law of reflection.

Point source reflecting from a plane surface When sound waves from a point source strike a plane wall, they produce reflected circular wavefronts as if there were an “image” of the sound source at the same distance on the other side of the wall. If something obstructs the direct sound from the source from reaching your ear, then it may sound as if the entire sound is coming from the position of the “image” behind the wall. This kind

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of sound imaging follows the same laws of reflection as your image in a plane mirror. Reflection from Concave Surface Any concave surface will tend to focus the sound waves which reflect from it. This is generally undesirable in auditorium acoustics because it produces a “hot spot” and takes sound energy away from surrounding areas. Even dispersion of sound is desirable in auditorium design, and a surface which spreads sound is preferable to one which focuses it. Absorption of Sound Different materials absorb specific frequencies, with high frequencies being the most susceptible to absorption. Concrete, glass, wood and plywood reflect sound waves, while draperies, carpet and acoustical tiles absorb more sound waves, especially the higher frequencies. The more humid air is, the greater it’s potential for high frequency absorption. That is why the humidity levels must be controlled in a concert hall. When the air is extremely cold, there is little humidity. As a result sounds from a jet plane sound closer and more piercing on a cold brisk day as compared to the identical sound on a warm summer day when more high frequencies are absorbed in the humid air. Refraction of Sound Refraction is the bending of waves when they enter a medium where their speed is different. Refraction is not so important a phenomena with sound as it is with light where it is responsible for image formation by lenses, the eye, cameras, etc. But bending of sound waves does occur and is an interesting phenomenon in sound.

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If the air above the earth is warmer than that at the surface, sound will be bent back downward toward the surface by refraction.

Sound propagates in all directions from a point source. Normally, only that which is initially directed toward the listener can be heard, but refraction can bend sound downward. Normally, only the direct sound is received. But refraction can add some additional sound, effectively

amplifying the sound. Natural amplifiers can occur over cool lakes. Diffraction of Sound Diffraction is term used to describe the bending of waves around obstacles (i.e. walls, barriers, etc) and the spreading out of waves beyond small openings whose size is smaller relative to the wavelength of the frequency. Diffraction is exclusively displayed by low frequencies

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(bass). The long wavelength sounds of the bass drum will diffract around the corner more efficiently than the more directional, short wavelength sounds of the higher pitched instruments.

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Loudspeaker Sound Contours One consequence of diffraction is that sound from a loudspeaker will spread out rather than just going straight ahead. Since the bass frequencies have longer wavelengths compared to the size of the loudspeaker, they will spread out more than the high frequencies. The curves at left represent equal intensity contours at 90 decibels for sound produced by a small enclosed loudspeaker. It is evident that the high frequency sound spreads out less than the low frequency sound.

Frequency perception and the Human Ear Waveforms Types Waveforms are the building blocks of sound. By combining raw waveforms one can simulate any acoustic instrument. This is how synthesizers make sound. They have waveform generators, which create all types of waves, which they combine to form composite waveforms, which approximate real instruments. Generally waveforms can be divided into two categories (i) Simple tone (ii) Complex Simple Tone A simple tone is a sound having a single frequency. A sine wave is an example of a pure tone, in that there is only a single frequency and no harmonics. The sine tone is the fundamental or ‘backbone’ of every complex tone there is in existence.

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The French scientist Jean Baptiste Joseph Fourier (1768-1830) made a statement that any acoustic event, no matter how complex, can be broken down into individual sine waves. He decided that any periodic waveform can be simplified into its constituent harmonic content built of a combination of sine waves with specific frequencies, amplitudes and relative phases. On the other hand, it would be theoretically possible to artificially or synthetically re-generate any complex sound by layering multiples sine waves together in the right proportion. Therefore, we can plot the square wave (shown above as an amplitudetime graph) in a ‘harmonic analysis’ or ‘Fourier analysis’ shown below. Complex Tone A tone having more than a single frequency component is called a complex. For instance, a tone consisting of a fundamental and overtones or harmonics, may be said to be complex. Two waveforms combine in a manner which simply adds their respective amplitudes linearly at every point in time. Thus, a complex tone or waveform can be built by mixing together different sine waves of various amplitudes. The following is a summary of the resultant complex waveforms that can be synthesized by the addition and subtraction of harmonics to a sine wave. Square Wave An audio waveform theoretically comprised of an infinite set of odd harmonic sine waves.

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Saw tooth Wave An audio waveform theoretically comprised of an infinite set of harmonically related sine waves.

Triangle Wave An audio WAVEFORM theoretically comprised of an infinite set of odd harmonic SINE WAVE. It is often used in sound synthesis because it is less harsh than the square wave. The amplitude of its upper harmonics falls off more rapidly.

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Difference between musical sound and noise Sound carries an implication that it is something we can hear or that is audible. However sound exists above 20 kHz called ultrasound and below our hearing range 20Hz called infrasound. Regular vibrations produce musical tones. The series of tones/frequencies are vibrating in tune with one another. Because of their orderliness, we find their tones pleasing. Noise produces irregular vibrations their air pressure vibrations are random and we perceive them as unpleasant tones. A musical note consists of a Fundamental wave and a number of overtones called Harmonies. These harmonies are known as the harmonic series. Fundamental Frequency and Harmonics The lowest resonant frequency of a vibrating object is called its fundamental frequency. Most vibrating objects have more than one resonant frequency and those used in musical instruments typically vibrate at harmonics of the fundamental. The term harmonic has a precise meaning that of an integer (whole number) multiple of the fundamental frequency of a vibrating object. When the frequency of an overtone is not an integer multiple of the fundamental, the overtone is said to be inharmonic and is called a partial or overtone; its waveform is aperiodic. Instruments of the percussion group, bells, and gongs in particular, have overtones which are largely inharmonic. As a result, such sounds may have indefinite or multiple pitches associated with them. The term overtones or partials are used to refer to any resonant frequency above the fundamental frequency - an overtone may or may not be a harmonic. To obtain the frequency of the harmonic number, the formula below is used.

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Overtones and Harmonics Many of the instruments of the orchestra, those utilizing strings or air columns, produce the fundamental frequency and harmonics. Other sound sources such as the membranes or other percussive sources may have resonant frequencies. They are said to have non-harmonic overtones.

Cylinders with one end closed will vibrate with only odd harmonics of the fundamental. Vibrating membranes typically produce vibrations at harmonics, but also have some resonant frequencies which are not whole number multiples of their fundamental frequencies. It is for this class of vibrators that the term overtone becomes useful - they are said to have some non-harmonic overtones. Timbre (Quality of Tone) Timbre or tone quality is determined by the behavior in time of the frequency content or spectrum of a sound, including its transients which are extremely important for the identification of timbre. The presence and distribution of these frequency components, whether harmonic or inharmonic, and their onset, growth and decay in time together with phase relations between them, combine to give every sound its distinctive tonal quality or timbre. Often qualities of timbre are described by analogy to color or texture (e.g. bright, dark, rough, smooth), since timbre is perceived and

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understood as a ‘subjective’ impression reflective of the entire sound, seldom as a function of its analytic components. With musical instruments, timbre is a function of the range in which the sound has its pitch as well as its loudness, duration, and manner of articulation and performance. The same applies with speech, where timbre is the basic quality which allows one to distinguish between different voices, just as between different instruments or other sounds. Pitch In the same sense that loudness is the subjective sense of the objective parameters of intensity or amplitude of a sound, the subjective impression of frequency is pitch. As such, pitch is a psychoacoustic variable, and the degree of sensitivity shown to it varies widely with people. Some individuals have a sense of remembered pitch, that is, a pitch once heard can be remembered and compared to others for some length of time; others have a sense of absolute pitch called perfect pitch. The term ‘pitch’ is used to describe the frequency of sound. For example, middle C in equal temperament = 261.6 Hz Sounds may be generally characterized by pitch, loudness, and quality. The perceived pitch of a sound is just the ear’s response to frequency, i.e., for most practical purposes the pitch is just the frequency. The pitch perception of the human ear is understood to operate basically by the Place Theory, with some sharpening mechanism necessary to explain the remarkably high resolution of human pitch perception.

The Place Theory and its refinements provide plausible models for the perception of the relative pitch of two tones, but do not explain the phenomenon of perfect pitch. The just noticeable difference in pitch is School of Audio Engineering Chennai 28

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conveniently expressed in cents, and the standard figure for the human ear is 5 cents. Cents Musical intervals are often expressed in cents, a unit of pitch based upon the equal tempered octave such that one equal tempered semitone is equal to 100 cents. An octave is then 1200¢ and the other equal tempered intervals can be obtained by adding semitones Effect of Loudness Changes on Perceived Pitch A high pitch (>2 kHz) will be perceived to be getting higher if its loudness is increased, whereas a low pitch ( 85dBSPL) will sound weaker in bass and high end later on when played back at lower levels. Conversely, a program mixed at low listening level will seem to have more bass and treble when played back at a louder level. Why does this happen? Well, human beings have an inherent tendency to compensate for the level differences in bass and treble content and most always want more of it. That’s the reason why more and more home consumer and car audio systems incorporate frequency dependant amplifiers (or graphic equalizers) to cater to individual consumer preferences. From the perspective of audio engineering, mixing audio with a misconception that ‘more bass or treble is better’ can be detrimental to the overall integrity of the mix. Given this facts of auditory life, there’s something to be said for keeping listening levels within reason during a mixdown session. If that level gets too loud (as often happens), the final product will probably suffer as a result, when heard by the listener at a normal listening level. Furthermore, listening to loud volumes subjects the hearing system to fatigue, causing the threshold of hearing and pain to shift somewhat and misinterpret what’s being reproduced by the speakers. It is also worthwhile to mention that studio grade professional speakers can reproduce larger quantities of bass and treble without distortion or other erroneous sonic artifacts. However, the same mix if played back on a normal system with average speakers may distort, sound muddy (clouding of the frequency spectrum by bass frequencies) or thin (excessive high frequency content).

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The Ear’s Protective Mechanism In response to sustained loud sounds, muscle tension tightens the tympanic membrane and, acting through the tendon connecting the hammer and anvil, repositions the ossicles to pull the stirrup back, lessening the transfer of force to the oval window of the inner ear. This contributes to the ear’s wide dynamic range. In short dynamic range refers to the softest and loudest sound a person can hear without any sort of discomfort

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Effect of Extreme Noise and Frequencies on the Ear On exposure to noise (which may as well be very loud music levels), the ear’s sensitivity level will decrease as a measure of protection. This process is referred to as a shift in the threshold of hearing, meaning that only sounds louder than a certain level will be heard. The shift may be temporary, chronic or permanent. Threshold shifts can be categorized in three primary divisions:Temporary Threshold Shift (TTS) During short exposure to noise, most people experience a rise in the auditory threshold which normally disappears in 24 hours, but may last as long as a week. Permanent Threshold Shift (PTS) or Noise Induced Permanent Threshold Shift (NIPTS) After prolonged exposure to noise, permanent hearing damage may result in the inner ear. Chronic Threshold Shift or Compound Threshold Shift If exposure to noise occurs repeatedly without sufficient time between exposures to allow recovery of normal hearing, TS may become chronic, and eventually permanent. This is a particular danger when people who work in noisy environments are exposed to further noise afterwards in driving, at home and at places of entertainment. Susceptibility to threshold shifts (TS) varies greatly from person to person, men generally being more sensitive to low frequency sounds, and women more susceptible to high frequencies. Sounds in the 2 - 6 kHz range seem to induce greater temporary threshold shift (TTS) than other frequencies. This is also called aural fatigue. One of the body’s reactions to loud sounds is a constriction of the blood vessels (vasoconstriction) which reduces the blood supply reaching the hair cells of the Organ of Corti. The outer rows of hair cells respond mainly to low intensity sound levels and thus are easily saturated by loud sounds, particularly when their source of blood is diminished. This leaves only the inner rows of hair cells working since they need a higher intensity for stimulation.

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Thus, TTS implies a temporary hearing loss for low level sounds (somewhat analogously to the protective closing of the iris in bright light and the resulting temporary desensitization to low light levels). If the outer hair cells are not allowed to recover through periods of quiet, they gradually lose their ability to respond and eventually die. TTS may also be accompanied by tinnitus, a ringing in the ears. So if you seriously want to be an audio engineer, think twice before you go into a club which is playing back music at extremely high intensities. As you intoxicate yourself with all those wild cocktail concoctions, the brain’s sensitivity is gradually diminished and the physical hearing organ – your ear – undergoes damage. Take care of your ears. The graph below shows the duration of high intensity sound exposure in minutes, hours, days, and weeks and how it affects the threshold of hearing.

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Important things to remember about the ear Ear’s response to frequency is logarithmic Non-flat frequency response Dynamic range is about 120 dB (at 3-4 kHz) Frequency discrimination 2 Hz (at 1 kHz) Intensity change of 3dB to 1 dB can be detected depending on the frequency range. Small sensitivity to phase Structure of the ear The outer ear (pinna) The middle ear (ossicles) The inner ear (cochlea) Pinna Gathers the sound D ifferentiates (to some extent) sounds from front and rear Sound localization due to the interference on the pinna The auditory canal 3 cm long, 0.7 cm diameter protects thin eardrum from foreign objects acts as a resonator for specific frequencies in the range of 2Khz to 4kHz Auditory canal resonance (at the frequency where the canal is quarter wavelength long) causes acoustical amplification at the eardrum (about 10 dB in the 2-4 kHz region). Further amplification due to the diffraction of sound waves around the head (total amplification 20 dB) The middle ear Eardrum vibrates as a result of acoustic variations in pressure Converts acoustical energy into mechanical energy stops acoustical energy The mechanical action of the bones transmits the vibrations of the air to the liquid in the inner ear. Protects the delicate inner ear against excessively intense sounds The inner ear About 4 cm long Filled with fluid and sensory membranes Vibrations of the oval window cause traveling waves in the fluid of cochlea.

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The traveling waves stimulate the hair cells in the membranes that transmit impulses to the brain via the auditory nerve. Cochlea is a sound-analyzing mechanism capable of amazing pitch and frequency discrimination. Ear can differentiate between a tone of 1000 Hz and 1003 Hz (discrimination of 0.3 %). Noise Spectra If we have all frequencies with random relative phase, the result is noise in its various incarnations, the two most common of which are white and pink noise. White Noise White noise is a type of noise that is produced by combining sounds of all different frequencies together. If you took all of the imaginable tones that a human can hear and combined them together, you would have white noise. The adjective “white” is used to describe this type of noise because of the way white light works. White light is light that is made up of all of the different colors (frequencies) of light combined together (a prism or a rainbow separates white light back into its component colors). In the same way, white noise is a combination of all of the different frequencies of sound. You can think of white noise as 20,000 tones all playing at the same time. White noise is defined as a noise that has equal amount of energy per frequency. This means that if you could measure the amount of energy between 100 Hz and 200 Hz it would equal the amount of energy between 1000 Hz and 1100 Hz. This sounds “bright” (hence “white”) to us because we hear pitch in octaves. 1 octave is a doubling of frequency, therefore 100 Hz - 200 Hz is an octave, but 1000 Hz - 2000 Hz is also an

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octave. Since white noise contains equal energy per Hz, there is ten times as much energy in the 1 kHz octave than in the 100 Hz octave. Due to this fact, the level of white noise signal voltage raises 6dB per octave. Because white noise contains all frequencies, it is frequently used to mask other sounds. If you are in a hotel and voices from the room nextdoor are leaking into your room, you might turn on a fan to drown out the voices. The fan produces a good approximation of white noise. Why does that work? Why does white noise drown out voices? Here is one way to think about it. Let’s say two people are talking at the same time. Your brain can normally “pick out” one of the two voices and actually listen to it and understand it. If three people are talking simultaneously, your brain can probably still pick out one voice. However, if 1,000 people are talking simultaneously, there is no way that your brain can pick out one voice. It turns out that 1,000 people talking together sounds a lot like white noise. So when you turn on a fan to create white noise, you are essentially creating a source of 1,000 voices. The voice next-door makes it 1,001 voices, and your brain can’t pick it out any more. Pink Noise Pink noise is noise that has an equal amount of energy per octave. This means that there is less energy per Hz as you go up in frequency. Pink noise is achieved by running white noise through a pinking filter. A pinking filter is no more that a 6dB per octave roll off filter. Since white noise rises up 6dB per octave, there is a drop of 6dB of the energy each time you go up an octave. To say it in another way, the 6dB roll-off filter negates and equalizes the amount of energy per octave –

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neutralizing the rising response. So, if you were to listen to pink noise, it would sound relatively equal in all frequencies compared to white noise which will be a bit brighter. Pink noise is commonly used for evaluating and calibrating sound reproduction systems like speakers. Psychoacoustics Psychoacoustics may be broadly defined as a study of the complex relationship between physical sound and the brain’s reaction and interpretation of them in a sound field. Until recently, psychoacoustics has devoted more attention to the behavior of the peripheral auditory system than to the details of cognitive processing. The discipline is a branch of psychophysics in that it is interested in the relation between sensory input stimuli and the behavioral or psychological response that they provoke. Because of individual variations in observed responses, statistical results are most often achieved. Some of the traditional psychoacoustic concerns involve the perception of pitch, loudness, volume and timbre. Response The response of a device or system is the motion (or other output) resulting from excitation (by a stimulus) under specified conditions. A qualifying adjective is usually prefixed to the term (e.g., frequency response, amplitude response, transient response, etc.) to indicate the type of response under consideration.

Frequency response curves for (a) two violin strings, showing characteristic resonance regions, and (b) a loudspeaker which reproduces frequencies

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Binaural Localization Humans, like most vertebrates, have two ears that are positioned at about equal height at the two sides of the head. Physically, the two ears and the head form an antenna system, mounted on a mobile base. This antenna system receives acoustic waves of the medium in which it is immersed, usually air. The two waves received and transmitted by the two ears are the physiologically adequate input to a specific sensory system, the auditory system. The ears-and-head array is an antenna system with complex and specific transmission characteristics. Since it is a physical structure and sound propagation is a linear process, the array can be considered to be a linear system. By taking an incoming sound wave as the input and the sound pressure signals at the two eardrums as the output, it is correct to describe the system as a set of two self-adjusting filters connected to the same input. Self-adjusting, in the sense used here, means that the filters automatically provide transfer functions that are specific with regard to the geometrical orientation of the wave front relative to the ears-and-head array. Physically, this behavior is explained by resonances in the open cavity formed from pinna, ear canal and eardrum, and by diffraction and reflection by head and torso. These various phenomena are excited differently when a sound wave impinges from different directions and/or with different curvatures of the wave front. The resulting transfer functions are generally different for the two filters, thus causing ‘interaural’ differences of the sound-pressure signals at the two eardrums. Since the linear distortions superimposed upon the sound wave by the two ‘ear filters’ are very specific with respect to the geometric parameters of the sound wave, it is not far from the mark to say that the ears-and-head system encodes information about the position of sound sources in space, relative to this antenna system, into temporal and spectral attributes of the signals at the eardrums and into their interaural differences. All manipulations applied to the sound signals by the ears-and-head array are purely physical and linear. It is obvious, therefore, that they can be simulated. Although humans can hear with one ear only - so called monaural hearing - hearing with two functioning ears is clearly superior. This fact

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can best be appreciated by considering the biological role of hearing. Specifically, it is the biological role of hearing to gather information about the environment, particularly about the spatial positions and trajectories of sound sources and about their state of activity. Further, it should be recalled in this context that inter individual communication is predominantly performed acoustically, with brains deciphering meanings as encoded into acoustic signals by other brains. In regard of this generic role of hearing, the advantage of binaural as compared to monaural hearing stands out clearly in terms of performance, particularly in the following areas : (i) Localization of single or multiple sound sources and, consequently, formation of an auditory perspective and/or an auditory room impression. Separation of signals coming from multiple incoherent sound sources spread out spatially or, with some restrictions, coherent ones. Enhancement of the signals from a chosen source with respect to further signals from incoherent sources, as well as enhancement of the direct (unreflected) signals from sources in a reverberant environment.

(ii)

(iii)

Definitions A few frequently encountered terms are given brief definitions here, for the sake of the discussion which follows:-

Image localization. - The term localization refers to the perception of the point at which a sound source, or image, seems to be situated with respect to the listener’s own position.

Arrival angle. - The angle from which an original sound source arrives, with zero degrees understood to indicate a source that is directly in front of the listener.

Interaural. - Refers to any comparison between an audio signal measured at one ear, and the same signal measured at the other ear. Reproduced source - Any sound recorded earlier and played back over one or two speakers.

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Localization Localization refers to our ability to make judgments as to the direction and distance of a sound source in the real world environment. We use various cues to help us localize the direction of a sound. When considering the sound source and the environment, the fact that sound waves travel in all directions from a particular sound forces the listener to cope with direct and indirect sounds. Direct sound is the most direct path that sounds takes, that is from the object creating the sound to the actual perceiver of the sound. Indirect sound incorporates all of the reflections of the sound that the perceiver hears at a delayed interval from the direct sound and provides the listener with information as to the space, location and distance of the sound within the environment. Is the sound emitting from a large or small room? Is it outside or inside? Is it in a reverberant space or a non-reverberant space? These are a few of the multitude of factors that influence our judgment of a sound’s location. Sound enters the ear canal through direct paths, and indirect paths that reflect from the complex folds of the pinna. When the reflections of the indirect sounds combine in the ear with the direct sounds, pinna filtering occurs, changing the received sound’s frequency response. The ear/brain duo interprets this equalization, producing cues (assisting zenith localization, for example) from the filtering effect. To provide still more directional cues, small head movements allow the ear/brain to judge relative differences in the sound field perspective. With our marvelous acuity, we can hear sounds coming from all around us, whether they are naturally created or coming from the speakers of a stereo or surround sound system.

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We can perceive where sounds come from - above, below, behind, to the side—this is called spatial localization. In particular, our stereophonic ears can discern azimuth or horizontal (left-right) directionality, and zenith or vertical (up-down) directionality. We perceive directionality using localization cues such as Interaural Time Difference (ITD), Interaural Intensity Difference (IID), and pinna filtering (Comb filtering). The median plane is the region where the sound sources are equidistant from the two ears. The horizontal plane is level with the listener’s ears. The frontal or lateral plane divides the head vertically between the front and the back. The position of the sound source relative to the center of the listener’s head is expressed in terms of azimuth (0-360 degrees, from in front of the head all the way around the head), elevation (angle between the horizontal plane up 90 degrees or below -90 degrees) and distance.

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Localization Parameter Using the listener’s own position as a reference point, the localization of a sound source may be conveniently described by two parameters: distance and arrival angle. Distance cues The perception of the distance from which a sound arrives is itself a function of four (4) variables, each which is discussed below. The variables are (i) Loudness (ii) Ratio of direct to reflected sound (iii) Frequency response (high frequency attenuation) (iv) Time delay Loudness All else being equal, it is obvious that the closer a listener is to the sound source, the louder it will be. However, all else is rarely ever equal, and loudness by itself is relatively uninformative, For example, turning a volume control up or down does nothing to vary the impression of distance, unless the level change is accompanies by one or more other important distance cues. Direct – to – Reflected Sound A far for important distance cue is the ratio of direct to reflected sound that reaches the listener. As an obvious example, consider a sound very close to the listener, and another at a great distance. In either case there is one direct path to the listener and many reflected paths, as the sound bounces of various surfaces in the listening area. The difference between these two examples is that one that is closer to the listener and hence reaches the listener via a shorter path. Its intensity is somewhat preserved with minimal energy loss as governed by the Inverse Square Law and the listener hears the direct sound almost entirely with little or no reflected sound. By contrast, sound arriving from the distant source

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is accompanied by many reflections of itself, some which arrive just after the direct sound. Again the Inverse Square Law is at work: with little difference in path length, there is less distinguishing difference between the amplitude of the direct and reflected sounds. By distinguishing the direct to reflected ratio in a sound source, the listener is able to perceive the distance of the sound source itself. High Frequency Attenuation As a pressure wave travels through the surrounding air, there is gradual loss of high frequency information due to atmospheric absorption. For example, at a temperature of 20°C, a 10 kHz signal is attenuated some 0.15dB to 0.30dB per meter, depending on the relative humidity. This high frequency attenuation may help convey a feeling of distance, provided the listener already has some frame of reference; that is the same source has been heard close-up, or is so familiar that the listener knows from experience what it is supposed to sound like when nearby. Time Delay As a final distance cue, it takes a certain amount of time for any sound to reach the listener. For example, the sound produced by a musician in the last row of a large ensemble may arrive some 20 or more milliseconds later than the sound of a front and center placed soloist. With the earlier sound serving as a frame of reference, the later arrival of a more distant source becomes a subtle yet powerful distance cue. When a sound reaches our ears, the ipsilateral ear, which is closest to the sound, perceives it first and the sound is louder than at the contralateral ear, which is further from the sound source. Most sounds are not equidistant from each ear and, as mentioned earlier, some frequencies bounce off the body while others curve around the body. Variations in sound reaching each ear created by these effects may be measured as an (IID) interaural intensity difference and an (ITD) interaural time difference.

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The interaural time difference is frequency independent, relying on the fact that lower frequency wave forms bend around the head. There is a difference in the time it takes the sound to reach the ipsilateral ear and the time it takes the sound to bend around the body and reach the contralateral ear. Distinctions can be made with time lags between the ears as small as 800 microseconds, or less that one millisecond. IID and ITD cues only enable us to identify the sound images as left, right or inside the head. The term lateralization is used to describe the apparent location of sound within the head, as occurs when sound is perceived through headphones. Haas Effect Also called the Precedence Effect, or Law of the First Wave front, describes the human psychoacoustic phenomena of correctly identifying the direction of a sound source heard in both ears but arriving at different times. Due to the head’s geometry (two ears spaced apart, separated by a barrier) the direct sound from any source first enters the ear closest to the source, then the ear farthest away. The Haas Effect tells us that humans localize a sound source based upon the first arriving sound, if the subsequent arrivals are within 25-30 milliseconds. If the later arrivals are longer than this, then two distinct sounds are heard. The Haas Effect is true even when the second arrival is louder than the first (even by as much as 10 dB!). In essence we do not “hear” the delayed sound. This is the hearing example of human sensory inhibition that applies to all our senses. Sensory inhibition describes the phenomena where the response to a first stimulus causes the response to a second stimulus to be inhibited, i.e., sound first entering one ear cause us to “not hear” the delayed sound entering into the other ear (within the 35 milliseconds time window). Sound arriving at both ears simultaneously is heard as coming from straight ahead, or behind, or within the head.

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The listener’s perception of angle from which a sound arrives is determined by subtle differences between the ways each ear hears the same signal. These are:relative loudness between the two ears or Interaural Intensity Differences (IID) time of arrival difference between the two ears or Interaural Time Difference (ITD) frequency response differences between the two ears or Interaural Spectral Difference (ISD) Interaural Intensity Differences (IID) In theory, there will be an interaural intensity difference when an original sound source arrives from an off-center location. In practice however, that difference is sometimes so slight as to be imperceptible. For example, consider a sound source originating 90º off-axis, but located 2 meters away. Given an ear spacing of say 21 cm, the ratio of the distances to each ear is 2 / 2.21 = 0.90. These means the interaural difference attributable to distance alone is 20 log (0.90), or only 0.87dB. (For the moment, we ignore the effect of the head itself as a barrier between the source and the distant ear.)

In the more normal listening environment shown in the second diagram, the arrival angle is 30º, and the source distance is 4 meters. Under these conditions, the additional path length to the distant ear is only 0.1 meter, so the interaural difference is reduced to 20 log (4/4.41) = - 0.2dB. From these examples, it can be seen that under normal listening conditions the interaural level differences from a slight additional path length to the more distant (or contralateral) ear is not much use as a localization cue.

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Interaural Time Differences (ITD) Within a certain frequency band, time becomes a very important localization cue. For example, consider a plane pressure waveform arriving from some off-center location. As noted above, if the sound pressure were measured at each ear, there would be almost no difference in level. However, the sound would arrive a bit later at the contralateral ear relative to the ipsilateral ear. In the following example, a distance of 21 cm is used as the spacing between the ears. To determine the additional time taken for a sound to reach the contralateral ear, the calculation below can be used.

Interaural Spectral Difference (ISD) When a signal arrives from some off-axis location, the effect of the listeners own head cannot be ignored; it becomes an acoustic obstacle in the path of the contralateral ear. At low frequencies, the resultant attenuation is minimal, due to the diffraction of low frequencies around the head. However as the frequency increases, its wavelength decreases, and the head become more of a barrier to the arriving sound. Therefore there is considerable high frequency attenuation at the distant ear as the sound source moves farther away off centre.

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Head Related Transfer Functions Each sound source creates a frequency spectrum that includes the range of frequencies that make up a particular sound. When wave forms approach the human body, they are affected by interaction with the listener’s torso, head, pinnae (outer ears) and ear canals. The total of these properties are captured as HRTFs. They are dependent on the direction of the sound. For example, before a sound wave gets to the eardrum it first passes through the outer ear structure, called the pinna. The pinna acts as a variable filter; accentuating or suppressing mid and high frequency energy of a sound wave to various degrees, depending on the angle at which the sound wave hits the pinna. Sounds that are above 1500 Hz are reflected off the body, while waveforms of lower frequency actually bend around the body. The 1500 Hz wave form is significant because one cycle is roughly the same size as the diameter of our head. HRTFs can be thought of a two audio filters, one for each of the ears, that capture the listening cues that are applied to the sound as it travels through the environment to the eardrum. The filters will change depending on the direction of the sound source. Early Reflections & Reverberation Reverberation is the multiple repetition of an audio signal that becomes more closely spaced with time. Direct sound represents sounds reaching the listener by a direct path, while indirect sound results from the reflection of audio signals reaching the listener at a delayed time. Reverberation is the sum

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total of all of the reflections of a particular sound, even though they arrive at different times, to a given point. Our perception of sound reflections helps us to determine the distance of the sound source. Moreover, the interaction of the sound with the room has a large effect on how close the sound is to the ears. When a person is close to a sound source the reflections take an acute angle, creating a greater relative distance between the arrival time of the direct sound and the indirect sound.
Diagram shows how a listener sitting in an auditorium listens to not only the direct sound, but multiple reflections of boundary surfaces.

At farther distances, reflected sounds make an obtuse angle, consequently there is less of a difference in the arrival time between direct and indirect sounds from the sound source to the listener. It is easy to localize a sound that is less than two meters from the listener. Beyond two meters it is difficult to determine distance without receiving reinforcement from reverberation. At distances greater than two meters the intensity of direct sound decreases exponentially with increasing distance, while the intensity of the reverberation level stays constant. The human ear compares the ratio of direct sound to the reverberation level in determining its distance from the sound source. Reverberation and reflections of the sound can be described graphically. The graph below shows the direct sound, followed by reflections of one wall called the 1st order reflections, followed by two walls called the 2nd order reflections and so on. The reverberant sound in an enclosed space, like an auditorium dies away with time as the sound energy is absorbed by multiple interactions with the surfaces of the room. In a more reflective room, it will take longer for the sound to die away and the room is said to be ‘live’. However, if it is excessive, it makes the sounds run together with loss of articulation - the sound becomes muddy, garbled. In a very absorbent room, the

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sound will die away quickly and the room will be described as acoustically ‘dead’. But the time for reverberation to completely die away will depend upon how loud the sound was to begin with, and will also depend upon the acuity of the hearing of the observer. To quantitatively characterize the reverberation, the parameter called the reverberation time is used. A standard reverberation time has been defined as the time for the sound to die away to a level 60 decibels below its original level. The reverberation time can be modeled to permit an approximate calculation. Early Reflections Those reflections reaching a listener after the arrival of the direct sound, but before the arrival of reverberation sound resulting from late reflections. The early reflections give rise to a feeling of spaciousness in the music hall, but in the typical listening room they tend to confuse the stereo image, giving rise to coloration of sound due to combing. Rationale for 60dB Reverberation Time The reverberation time is perceived as the time for the sound to die away after the sound source ceases, but that of course depends upon the intensity of the sound. To have a reproducible parameter to characterize an auditorium which is independent of the intensity of the test sound, it is necessary to define a standard reverberation time in terms of the drop in intensity from the original level, i.e., to define it in terms of relative intensity. The choice of the relative intensity to use is of course arbitrary, but there is a good rationale for using 60 dB since the loudest crescendo for

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most orchestral music is about 100 dB and a typical room background level for a good music-making area is about 40 dB. Thus the standard reverberation time is seen to be about the time for the loudest crescendo of the orchestra to die away to the level of the room background. The 60 dB range is about the range of dynamic levels for orchestral music.

Psychoacoustic Effects
Cocktail Party Effect Imagine yourself listening to a conversation at a cocktail party; you hear the sound which travels directly from the speaker’s mouth to your ear, and also the combination of an enormous number of echoes from different surfaces. The ability of the human hearing system to selectively attend to a single talker or stream of audio from a background cacophony of ambient noise heard at the same time is referred to as the ‘cocktail party effect’. Someone with normal hearing can, remarkably, concentrate on the original sound despite the presence of unwanted echoes and background music. The separation of auditory targets from a similar background, required when listening to a speaker on a cocktail-party, is a cognitive task involving binding of acoustic components that belong to the same object, segregation of simultaneously present components that belong to non-target objects, and tracking of homologue objects (same speaker) over time.

Figure 1: During conversation, we hear a combination of the original sound, a series of echoes, and interfering noises such as other conversations.

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Masking When we listen to music, it is very rare that it consists of just a single pure tone. Whilst it is possible and relatively simple to arrange to listen to a pure tone of a particular frequency in a laboratory, such a sound would not sustain any prolonged musical interest. Almost every sound that hears in music consists of at least two frequency components. When two or more pure tones are heard together, an effect known as ‘masking’ can occur, where each individual tone can become more difficult or impossible to perceive, or is partially or completely ‘masked’, due to the presence of another relatively louder tone. In such a case, the tone which causes the masking is known as the ‘masker’ and the tone which is masked is called ‘maskee’. Given the rarity of pure tone being heard in the music that we hear everyday, these tones are more likely to be individual frequency component of a note played on one instrument which either masks other components in that note, or frequency components of another note. The extent to which masking occurs depends on the frequencies of the masker and masker and their relative amplitudes.

The example above shows a 1 kHz tone played by both a trumpet (a loud instrument) and a flute (a soft instrument). The resultant shows that the loud trumpet has masked the flute completely due to their respective frequency components being somewhat alike

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Adding Loudness & Critical Band

When two sounds of equal loudness when sounded separately are close together in pitch, their combined loudness when sounded together will be only slightly louder than one of them alone. They may be said to be in the same critical band where they are competing for the same nerve endings on the basilar membrane of the inner ear. According the place theory of pitch perception, sounds of a given frequency will excite the nerve cells of the organ of Corti only at a specific place. The available receptors show saturation effects which lead to the general rule of thumb for loudness by limiting the increase in neural response. If the two sounds are widely separated in pitch, the perceived loudness of the combined tones will be considerably greater because they do not overlap on the basilar membrane and compete for the same hair cells. The phenomenon of the critical band has been widely investigated. It has been found that this critical band is about 90 Hz wide for sounds below 200 Hz and increases to about 900 Hz for frequencies around 5000 Hertz. It is suggested that this corresponds to a roughly constant length on the basilar membrane of length about 1.2 mm and involving some 1300 hair cells. If the tones are far apart in frequency, not within a critical band, the combined sound may be perceived as twice as loud as one alone.

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Missing Fundamental Effect The subjective tones which are produced by the beating of the various harmonics of the sound of a musical instrument help to reinforce the pitch the fundamental frequency. Most musical instruments produce a fundamental frequency plus several higher tones which are whole-number multiples of the fundamental. The beat frequencies between the successive harmonics constitute subjective tones which are at the same frequency as the fundamental and therefore reinforce sense of pitch of the fundamental note being played.

of

the

If the lower harmonics are not produced because of the poor fidelity or filtering of the sound reproduction equipment, you still hear the tone as having the pitch of the non-existent fundamental because of the presence of these beat frequencies. This is called the missing fundamental effect. It plays an important role in sound reproduction by preserving the sense of pitch (including the perception of melody), when reproduced sound loses some of its lower frequencies especially in radio and television broadcast. The presence of the beat frequencies between the harmonics gives a strong sense of pitch for instruments such as the brass and woodwind instruments. For percussion instruments such as the cymbal, the sense of pitch is less definite because there are non-harmonic overtones present in the sound. Doppler Effect You may have heard the changed pitch of a train whistle or a car horn as the train or car approached and then receded from you. As the train approaches the pitch of the blast is higher and it becomes lower as the train recedes from you. This implies that the frequency of the sound waves changes depending on the velocity of the source with respect to you, as the train approaches the pitch is higher indicating a higher frequency and smaller wavelength, as the train recedes from you the

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pitch is lower corresponding to a smaller frequency and a correspondingly larger wavelength.

Whenever relative motion exists between a source of sound and a listener, the frequency of the sound as heard by the listener is different compared to the frequency when there is no relative motion. This phenomenon is known as the Doppler Effect. The formula for Doppler Effect is f ’= f [V+ (-VD)] ÷ (V + (-VS) Where f’ is the Doppler Effect frequency. f is the original frequency of the sound wave, C is the speed of sound, VD us the speed of the detector or listener, and VS is the speed of the source. Ear Training The basic requirement of a creative sound engineer is to be able to listen well and analyze what they hear. There are no golden ears, just educated ears. A person develops his or her awareness of sound through years of education and practice. We have to constantly work at training our ears by developing good listening habits. As an engineer, we can concentrate our ear training around three basic practices music, microphones and mixing. Listening to Music Try and dedicate at least half an hour per day to listening to well recorded and mixed acoustic and electric music. Listen to direct-to-two track mixes and compare with heavily produced mixes. Listen to different styles of music, including complex musical forms. Note the basic ensembles used production trends and mix set-ups. Also attend live music concerts. The engineer must learn the true timbral sound of an instrument and its timbral balances. The engineer must be able to School of Audio Engineering Chennai 71

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identify the timbral nuances and the characteristic of particular instruments. Learn the structuring of orchestral balance. There can be an ensemble chord created by the string section, the reeds and the brass all working together. Listen to an orchestra live, stand in front of each section and hear its overall balance and how it layers with other sections. For small ensemble work, listen to how a rhythm section works together and how bass, drums, percussion, guitar and piano interlock. Learn the structure of various song forms such as verse, chorus, break etc. Learn how lead instrument and lead vocals interact with this song structure. Notice how instrumentals differ from vocal tracks. Listen to sound design in a movie or TV show. Notice how the music underscores the action and the choice of sound effects builds a mood and a soundscape. Notice how tension is built up and how different characters are supported by the sound design. Notice the conventions for scoring for different genres of film and different types of TV. For heavily produced music, listen for production tricks. Identify the use of different signal processing FX. Listen for panning tricks, doubling of instruments and voices. Analyze a musical mix into the various components of the sound stage. Notice the spread of instruments from left to right, front to back up and down. Notice how different stereo systems and listening rooms influence the sound of the same piece of music. Listening with Microphones Mic placement relative to the instrument can provide totally different timbral color. e.g. proximity boost on closely placed cardiod mics. A mic can be positioned to capture just a portion of the frequency spectrum of an instrument to be conducive with a particular “sound” or genre. E.g. rock acoustic piano may favor the piano’s high end and require close miking near the hammers to accent percussive attack, a sax may be miked near the top to accent higher notes or an acoustic guitar across the sound hole for more bass. The way an engineer mics an instrument is influenced by: (i) (ii) Type of music Type of instrument

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(iii) (iv) (v) (vi)

Creative Production Acoustics of the hall or studio Type of mic Leakage considerations

Always make A/B comparisons between mics different and different positions. The ear can only make good judgments by making comparisons. In the studio reflections from stands, baffles, walls, floor and ceiling can affect the timbre of instruments. This can cause timbre changes, which can be problematic or used to capture an “artistic” modified spectrum. When miking sections, improper miking can cause timbre changes due to instrument leakage. The minimum 3:1 mic spacing rule helps control cross-leakage. Diffuser walls placed around acoustic instruments can provide openness and a blend of the direct /reflected sound field. Mic placement and the number of diffusers and their placement can greatly enhance the “air” of the instrument. Listening in Foldback and Mixdown A balanced cue mix captures the natural blend of the musicians. Good foldback makes musicians play with each other instead of fighting to be heard. If reed and brass players can’t hear themselves and the rest of the group they tend to overplay. A singer will back off from the mic if their voice in the headphone mix is too loud, or swallow the mic if the mix is too soft. They will not stay in tune if they cannot hear backing instruments. Musicians will aim their instruments at music stands or walls for added reflection to help them overcome a hearing problem.

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Frequency Balance An Engineer should be able to recognize characteristics of the main frequency bands with their ears. Hz 16 – 160 Band Characteristics Positive Warmth Negative Muddiness

Extreme lows Felt more than heard Bass No stereo information Harmonics start to occur

160-250

Fatness

Boominess, Boxiness Hornlike(500-1000 Hz) Ear fatigue (1kHz-2kHz) Tinny, thin

250-2000

LowMidrange

Body

2000-4000

High Midrange Presence

Vocal intelligibility

Gives definition

4000-6000

6000-20000

Highs

Loudness and Definition, closeness. Spatial energy, information closeness Depth of field Air. Crispness. Boosting/cu tting helps create senses /distance

Brash

Noise

Dimensional Mixing: The final 10% of a mix picture is spatial placement and layering of instruments or sounds. Dimensional mixing encompasses timbral balancing and layering of spectral content and effects with the basic instrumentation. For this, always think sound in dimensional space: left/right, front/back, up/down. Think of a mix in Three Levels: Level A 0 to 1 meter Level B 1 to 6 meters Level C 6 meters and further School of Audio Engineering Chennai 74

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Instruments which are tracked in the studio are all recorded at roughly the same level (SOL) and are often close miked. If an instrument is to stand further back in the mix it has to change in volume and frequency. Most instruments remain on level B so you can hear them all the time. Their dynamics must be kept relatively stable so their position does not change. Level A instruments will be lead and solo instruments. Level C can be background instruments, loud instruments drifting in the background, sounds, which are felt, rather than heard and Reverb. Summary

Stereophonic sound (3D sonic view) is based on our amazing process of encoding sounds with our bodies and decoding it with our brains. Sound localization is based on time/intensity cues or changes in the spectrum of the sound. Listening is highly personal experience, unique; No two of us hear a given sound in exactly the same way. Our Hearing system is very effective. It has ; (i) Wide dynamics (ii) Frequency and timing (phase) coded. We do not perceive physical terms, we perceive subjective terms. (i) Frequency pitch (ii) Magnitude loudness (iii) Spectrum timbre (iv) Time subjective duration (v) Frequency scale hearing equivalent scale (criticalband-rate scale), masking Human auditory system is the final receiver judging the sound quality of the audio system. Audio systems should be adapted to the characteristics of the auditory system.

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References http://hyperphysics.phy-astr.gsu.edu http://arts.ucsc.edu/ems/music/tech_background/tech_background.html http://www.glenbrook.k12.il.us/gbssci/phys/Class/waves/u10l2d.html “Am I Too Loud?” Jour Audio Engineering Soc, V25, p126, Mar 1977. Beranek, Music, Acoustics and Architecture, Wiley, 1962 Acoustics and Perception of Speech, 2nd Ed, Williams and Wilkins, 1984 Cohen, Abraham B, Hi-Fi Loudspeakers and Enclosures, Hayden, 1968 Halliday & Resnick, Fundamentals of Physics, 3E, Wiley 1988 Huber, David and Runstein, Robert, Modern Recording Techniques, 4th Ed., Boston: Focal Press, 1997. Terhardt, E., “Calculating Virtual Pitch”, Hearing Research 1, 155 (1979) Tipler, Paul, Elementary Modern Physics, Worth, 1992, Tipler, Paul, Physics for Scientists and Engineers, 2nd Ed Ext, Worth, 1982. White, Harvey E. and White, Donald H., Physics and Music, Saunders College, 1980 E. Zwicker, G. Flottorp, S S Stevens, Critical Bandwidth in Loudness Summation, JASA, 29 (1957) pp548-57.

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