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1.

INTRODUCTION

Ultrasonic directive speaker produces is a system that produces sound that travels in only one direction using a process called sound from ultrasound. Sound in natural form propagates in every direction irrespective of the orientation of the transducers. So when a sound is produced by a speaker in a room it can be heard in any corner or the room. This property of sound is quite different from light which can be directed in one direction using proper reflector arrangement for example in a torch. To make sound directional can be much tedious task but due to its diverse applications it has drawn the attention of scientist and engineers all over the world. Electronic industry too is giving this field a lot of attention and the primitive equipment for this system are even in the market. A unidirectional sound can be of great use in variety of fields like in places like museums for giving the visitor information about an article without disturbing other visitors, in commercial advertising for giving the buyer information about a product in supermarkets without disturbing other customers, in crowd control by police and of course for public addressing. With conventional sound systems this feature is impossible.

Figure 1.1 : Difference between audio spotlight and other speakers

This technology has been under development since early 1960s but has remained unusable for public due to unfeasibility of both production and operation. This technology
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was originally developed by the US Navy and Soviet Navy for underwater sonar in the mid1960s, and was briefly investigated by Japanese researchers in the early 1980s, but these efforts were abandoned due to extremely poor sound quality (high distortion) and substantial system cost. These problems went unsolved until a paper published by Dr. F. Joseph Pompei of the Massachusetts Institute of Technology in 1998 (105th AES Conv, Preprint 4853, 1998) fully described a working device that reduced audible distortion essentially to that of a traditional loudspeaker. But still the cost of the system was high enough to inhibit its public and commercial uses of this technology. However, recent development in technology and lowering in the cost of the component and not to mention huge interest of different sectors both commercial and noncommercial have led to great development in this field may research institutes and companies are working to improve this technology and make it more efficient. They are trying different modulation schemes, different types of parametric arrays and digital signal processing to improve both the quality and effectiveness of the sound produced. Producing a directional sound is not as simple as producing directional light. While a light can be made to form a beam just by using a reflector system around the source this would not be helpful for a speaker because of two reasons firstly, making an ideal sound reflector is nearly impossible, at most we can make use of a material that absorbs sound but it would reduce its effectiveness of the system greatly, and secondly even if we make a reflector successfully the scattering of the sound in air would make the reflector of no use. In all wave-producing sources, the directivity of any source, at maximum, corresponds to the size of the source compared to the wavelengths it is generating. The larger the source is compared to the wavelength of the sound waves, the more directional beam results. The specific transduction method has no impact on the directivity of the resulting sound field; the analysis relies only on the aperture function of the source, per the Huygens Fresnel principle. Sound waves have dimensions of inches to many feet, which roughly corresponds to the sizes of most loudspeaker systems. At high frequencies, however, the wavelengths are quite short, which can result in a narrow distribution of sound from the tweeters in a conventional loudspeaker system. By making the speaker larger, either through the use of a dimensionally large speaker panel, speaker array, or dome, higher directivity can be obtained

at lower frequencies. However, the maximum directivity of any reasonably-sized traditional loudspeaker is still quite weak. The ultrasonic devices bypass this physics limitation, as they create a "virtual" loudspeaker (out of ultrasound) that is physically very large - but it is invisible, made of ultrasound. For this reason, the resulting directivity of these devices is far higher than physically possible with any loudspeaker system. However, they are reported to have limited low-frequency reproduction abilities. The parametric array is a nonlinear transduction mechanism that generates narrow, nearly side lobe free beams of low frequency sound, through the mixing and interaction of high frequency sound waves and secondly an array of small loudspeakers, all driven together in-phase. This creates a larger source size compared to wavelength, and the resulting sound field is narrowed compared to a single small speaker. Parametric array generates of low frequency sound (modulating signal), through the mixing and interaction of high frequency sound waves. A sound of high frequency can be modulated by a low frequency sound just like electromagnet waves. This property of sound has been used for communication for long time. They is often used to carry messages underwater, in underwater diving communicators, and short-range (under five miles) communication with submarines; the received ultrasound signal is decoded into audible sound by a modulatedultrasound receiver. The high frequency wave used should be higher than that could be perceived by humans to make the communication inaudible to humans. Therefore high frequency sound waves or ultrasound is used. Ultrasound can be any wave having frequency above 20 kHz which is the upper limit of human hearing. But as given above this system requires a demodulator which can demodulate and extract the information in the ultrasound. So it is unsuitable for direct communication. But there is another property of ultrasound that is non-linear propagation. Because of their high amplitude to wavelength ratio, ultrasonic waves commonly display non-linear propagation. This property can lead to demodulation of ultrasonic wave to its modulating sound during its propagation through medium which is utilized in parametric array system. For producing sound from ultrasound, the ultrasound must be modulated, there are several modulating techniques amplitude modulation, dual side band suppressed carrier modulation, frequency modulation , pulse width modulation etc.. Amplitude modulation, dual side band suppressed carrier modulation is suited in the condition when the amplitude or loudness of the sound is preferred over its quality (fidelity). Frequency modulation and Pulse

width modulation can give better frequency response or fidelity. Therefore when the purpose of system is to transfer speech, FM or PWM will be preferred. We have created this system using pulse width modulation. The whole system can be expressed as following block diagram.

Audio Signals

Pulse Width Modulator

A Half Bridge Driver

Array of Piezoelectric Transducers

Driver Figure 1.2 : Block Diagram of Directional Speaker System

The audio signals are first pulse width modulated then by using a half bridge driver they drive an array of piezoelectric transducers. The array consists of a large number of piezoelectric transducer placed in same plane and are being driven by same output to generate coherent ultra- high frequency sound waves. The audio signal is given through an audio amplifier which amplifies the sound that can be used in succeeding stages. In the following text we will try to understand the theoretical and practical aspect of this technology which is essential for development of such system.

2. REVIEW OF LITERATURES

For designing and making this project we reviewed various books, Journals, Research papers and other literature in order to grasp the scientific principles and laws involved in this field. Also via various means we tried to find the best technology that can lead us to the best implementation of the scientific laws. Main literature reviewed may include The audio spotlight: An application of nonlinear interaction by Masahide Yoneyama and Junichiroh Fujimoto which explained the theoretical aspects of this technology. Also we reviewed various literature that explains the basic building blocks of this technology like Ultrasound, Modulation techniques, Human Hearing Mechanism, Heterodyning, working of piezoelectric transducer etc.. The following are the major key issues to be reviewed very carefully before we go for a working of this system 1. Technology Overview Review of development history and scientific outlines of the technology 2. Human Hearing Hearing mechanism of humans and its range. 3. Ultrasound the carrier wave for this system 4. Heterodyning an important concept on which this whole system 5. Modulation a basic overview of various kind of modulation that can be used for making this system 6. Pulse Width Modulation the kind of modulation we used 7. Piezoelectric Transducer- an overview of speaker that are capable of producing ultrasonic sound

2.1 Technology Overview

The regular loudspeakers produce audible sound by directly moving the air molecules. The audible portions of sound tend to spread out in all directions from the point of origin. They do not travel as narrow beams. In fact the beam angle of audible sound is very wide, just about 360 degrees. This effectively means that the sound you hear will be propagated through the air equally in all directions. Conventional loudspeakers suffer from amplitude distortions, harmonic distortion, inter-modulation distortion, phase distortion, crossover distortion, cone resonance etc. Some aspects of their mechanical aspects are mass, magnetic structure, enclosure design and cone construction. In order to focus sound into a narrow beam, we need to maintain a low beam angle that is dictated by wavelength. The smaller the wavelength, less the beam angle and hence, the more focused the sound. The beam angle also depends on the aperture size of the speaker. A large loudspeaker will focus the sound over a smaller area. If the source loudspeaker can be made several times bigger than the wavelength of the sound transmitted, then a finely focused beam can be created. The problem here is that this is not a very practical solution, thus the low beam angle can be achieved only by making the wavelength smaller and this can be achieved by making use of ultrasonic sound. The technique of using a nonlinear interaction of high - frequency waves to generate low frequency waves was originally pioneered by researchers developing underwater sonar techniques in 1960's. In 1975, an article cited the nonlinear effects occurring in air. Over the next two decades, several large companies including Panasonic and Ricoh attempted to develop a loudspeaker using this principle. They were successful in producing some sort of sound but with higher level of distortion (>50%).In 1990s, Woody Norris a Radar Technician solved the parametric problems of this technology. Audio spotlighting works by emitting harmless high frequency ultrasonic tones that human hear cannot hear. It uses ultrasonic energy to create extremely narrow beams of sound that behave like beams of light. Ultrasonic sound is that sound which have very small wavelength in the millimeter range. These tones make use of non-linearity property of air to produce new tones that are within the range of human hearing which results in audible sound. The sound is created indirectly in air by down converting the ultrasonic energy into the frequency spectrum we can hear.
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2.2 Human Hearing

Hearing or audition is the ability to perceive sound by detecting vibrations through an organ such as the ear. It is one of the traditional five senses. In humans and other vertebrates, hearing is performed primarily by the auditory system: vibrations are detected by the ear and transduced into nerve impulses that are perceived by the brain (primarily in the temporal lobe). Like touch, audition requires sensitivity to the movement of molecules in the world outside the organism. Both hearing and touch are types of mechanosensation The eardrum of an ear simplifies incoming air pressure waves to a single channel of amplitude. In the inner ear, the distribution of vibrations along the length of the basilar membrane is detected by hair cells. The spacetime pattern of vibrations in the basilar membrane is converted to a spatialtemporal pattern of firings on the auditory nerve, which transmits information about the sound to the brainstem. The basilar membrane of the inner ear separates out different frequencies: high frequencies produce a large vibration at the end near the middle ear, and low frequencies a large vibration at the distant end. Thus the ear performs a frequency analysis, roughly similar to a Fourier transform. However, the nerve pulses delivered to the brain contain both place and rate information, so the similarity is not strong. Hearing range describes the range of frequencies that can be heard by human, though it can also refer to the range of levels. In humans the audible range of frequencies is usually said to be 20 Hz (cycles per second) to 20 kHz (20,000 Hz), although there is considerable variation between individuals, especially at the high frequency end, where a gradual decline with age is considered normal. Sensitivity also varies a lot with frequency, as shown by equal-loudness contours, which are normally only measured for research purposes, or detailed investigation. Routine investigation for hearing loss usually involves an audiogram which shows threshold levels relative to a standardized norm.

2.2 Ultrasound

Ultrasound is cyclic sound pressure with a frequency greater than the upper limit of human hearing. Ultrasound is thus not separated from normal(audible) sound based on differences in physical properties, only the fact that humans cannot hear it. Although this limit varies from person to person, it is approximately 20 kilohertz in healthy, young adults. The production of ultrasound is used in many different fields, typically to penetrate a medium and measure the reflection signature or supply focused energy. The reflection signature can reveal details about the inner structure of the medium, a property also used by animals such as bats for hunting. The most well-known application of ultrasound is its use in sonography to produce pictures of fetuses in the human womb. There are a vast number of other applications as well.

Figure 2.1: Uses of Ultrasound

The upper frequency limit in humans (approximately 20 kHz) is due to limitations of the middle ear, which acts as a low-pass filter. Ultrasonic hearing can occur if ultrasound is fed directly into the skull bone and reaches the cochlea through bone conduction without passing through the middle ear. It is a fact in psychoacoustics that children can hear some high-pitched sounds that older adults cannot hear, because in humans the upper limit pitch of hearing tends to become lower with age. A cell phone company has used this to create ring signals supposedly only

able to be heard by younger humans; but many older people can hear it, which may be due to the considerable variation of age-related deterioration in the upper hearing threshold. Ultrasound can be modulated to carry an audio signal (like radio signals are modulated). This is often used to carry messages underwater, in underwater diving communicators, and short-range (under five miles) communication with submarines; the received ultrasound signal is decoded into audible sound by a modulated-ultrasound receiver. However due to the absorption characteristics of seawater, ultrasound is not used for longrange underwater communications. The higher the frequency, the faster the sound is absorbed by the seawater, and the more quickly the signal fades. For this reason, most underwater telephones either operate in baseband mode i.e. at the same frequency as the voice and is basically a loudspeaker. Because of their high amplitude to wavelength ratio, ultrasonic waves commonly display nonlinear propagation. The non-linear characteristic is due to the fact that it takes more time for air molecules to be restored to their original density than to be compressed (Figure 2.2). When the sound pressure is high and frequency to a shock wave may be produced by returning air molecules colliding with the ones being compressed. In fact, an audible sound is produced by any molecule not completely returning. When the frequency of the vibration rises, the non-linear characteristic tends to become noticeable by an effect best described as air viscosity.

Figure 2.2 : Non Linear Property of Ultrasound 9

The non-linear property of the air for ultrasound is used to demodulate the modulated ultrasound via heterodyning. When two finite amplitude sound waves (primary waves), having different frequencies, interact with one another in a fluid, new sound waves

(secondary waves) whose frequencies correspond to the summand the difference of the primary waves may be produced as the result. This phenomenon was first analyzed by Westervelt and is well known as "nonlinear interaction of sound waves," or the "scattering of sound by sound." Based on Lighthill's arbitrary fluid motion equation as shown in Eq. (2.1), Westervelt derived an inhomogeneous wave equation which is satisfied by the sound pressure of secondary waves produced by the nonlinear interaction [Eq. (2.2)].

2.1 : density of fluid, Tij: stress tensor, 2.2

...2.3

In Eq. (2.2), ps is the secondary wave sound pressure,p1 is the primary wave sound pressure, is the nonlinear fluid parameter, and c0 is the small signal sound velocity. The solution for Eq. (2.2) may be expressed by the superposition integral of the Green's function and the virtual second source [right side of Eq. (2.2)] as shown in Eq. (2.4).

.2.4

Where r is the observation point position vector, r' is the source position vector and v is the nonlinear interaction space.

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When the primary wave consists of two continuous sinusoidal waves and both are planar and well collimated. A new type of loudspeaker has been developed on the basis of the nonlinear interaction of sound waves mentioned above. In this type of loudspeaker, ultrasound is amplitude modulated by an audio signal and radiated from a transducer array as finite amplitude waves. When the amplitude-modulated ultrasound wave interacts is a nonlinear fashion in air, the modulated signal (the audio signal) can be demodulated in the air. When two sinusoidal sound waves are radiated in the air, two new waves with angular frequencies of ariseby nonlinear interaction of the two original

sinusoidalwaves,whoseangularfrequencies are

.Thereforeone might expect the

secondary wave which corresponds to the modulation signal, to appear in the air as a result of the nonlinear interaction between the carrier ultra sound and the lower and upper sideband waves, provided that a finite amplitude AM ultrasound wave is radiated into the air. That is, the AM ultrasound is self-demodulated by the nonlinear interaction. In this case, since the modulation wave is reproduced in the air, a new type of loudspeaker can be devised if the modulation signal is selected as the program audio signal. If a finite amplitude ultrasound beam, modulated by an audio signal g(t), is radiated into the air from a transducer array, the sound pressure p1of the primary wave (AM wave) at a distance from the array on axismay be representedbyEq. (2.5)

)]

.2.5

Wherep0is the initial sound pressure of the ultrasound, m is the parameter indicating modulation index, and a is the absorption coefficient of carrier sound. A virtual audio signal source occurs in the primary sound beam because of the nonlinearity of the acoustic interaction in air. This sound source may be represented by Eq. (2.6) using Eq. (2.2) and Eq. (2.5)

)
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)+

.2.6

In the above equation, the second term on the right side implies a harmonic distortion component arising from the interaction between the lower and upper side band waves. If the primary sound beam cross section is assumed to be circular with radius a, then the demodulated audio sound pressure ps at the point from the array, on axis, can be calculated analytically using Eqs. (2.4) and (2.6) in the form ( ) .2.7

On the other hand, the sound pressure of a harmonic distortion component may be expressed as ( ) .2.8

Fourier transform of Eq.(2.7) can be expressed as ( ) ( ) * ( ) + ( ) .2.9

Where Ps( ) is the Fourier transform of ps(t), and Gs ( )is theFourier transformof g(t). As evident from Eq. (2.9),Ps( )isproportional to
2

and thus the frequency

characteristics of the reproduced sound show a 12dB/oct dependence. Consequently, the audio signal (modulation signal) must be processed by an equalizer having -12 dB/oct frequency characteristics before the audio signal is introduced into the AM modulator.

2.3 Heterodyning

Heterodyning is a radio signal processing technique invented in 1901 by Canadian inventor-engineer Reginald Fessenden, in which new frequencies are created by combining or mixing two frequencies. Heterodyning is useful for frequency shifting signals into a new frequency range, and is involved in the processes of modulation and demodulation. The two frequencies are combined in a nonlinear signal-processing device such as a vacuum tube,
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transistor, or diode, usually called a mixer. Heterodyning creates two new frequencies, one is the sum of the two frequencies mixed, and the other is their difference. These new frequencies are called heterodynes. Typically only one of the new frequencies is desired, and the other signal is filtered out of the output of the mixer. Heterodynes are closely related to the phenomenon of beats in music. Heterodyning is based on the trigonometric identity: ( ) ( ) 2.10

The product on the left hand side represents the multiplication mixing of a sine wave with another sine wave. The right hand side shows that the resulting signal is the difference of two sinusoidal terms, one at the sum of the two original frequencies, and one at the difference, which can be considered to be separate signals. Using this trigonometric identity, the result of multiplying two sine wave signals ( ( ), and ) ( ( ) ) can be calculated: [ ( ) ] [ ( ) ] 2.11

The result is the sum of two sinusoidal signals, one at the sum f1 + f2 and one at the difference f1 - f2 of the original frequencies. Hence when two coherent ultrasound waves will meet in a non-linear material like air they will undergo heterodyning to produce an audio wave.

2.4 Modulation scheme

The nonlinear interaction mixes ultrasonic tones in air to produce sum and difference frequencies. A DSB-AM modulation scheme with an appropriately large baseband DC offset, to produce the demodulating tone superimposed on the modulated audio spectra, is one way to generate the signal that encodes the desired baseband audio spectra. This technique suffers from extremely heavy distortion as not only the demodulating tone interferes, but also all other frequencies present interfere with one another. The modulated spectrum is convolved
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with itself, doubling its bandwidth by the length property of the convolution. The baseband distortion in the bandwidth of the original audio spectra is inversely proportional to the magnitude of the DC offset (demodulation tone) superimposed on the signal. A larger tone results in less distortion.

Further distortion is introduced by the second order differentiation property of the demodulation process. The result is a multiplication of the desired signal by the function - in frequency. This distortion may be equalized out with the use of preemphasis filtering. By the time convolution property of the fourier transform, multiplication in the time domain is a convolution in the frequency domain. Convolution between a baseband signal and a unity gain pure carrier frequency shifts the baseband spectra in frequency and halves its magnitude, though no energy is lost. One half-scale copy of the replica resides on each half of the frequency axis. This is consistent with Parseval's theorem. The modulation depth m is a convenient experimental parameter when assessing the total harmonic distortion in the demodulated signal. It is inversely proportional to the magnitude of the DC offset. THD increases proportionally with m1. These distorting effects may be better mitigated by using another modulation scheme that takes advantage of the differential squaring device nature of the nonlinear acoustic effect. Modulation of the second integral of the square root of the desired baseband audio signal, without adding a DC offset, results in convolution in frequency of the modulated square-root spectra, half the bandwidth of the original signal, with itself due to the nonlinear channel effects. This convolution in frequency is a multiplication in time of the signal by itself, or a squaring. This again doubles the bandwidth of the spectra, reproducing the second time integral of the input audio spectra. The double integration corrects for the - filtering characteristic associated with the nonlinear acoustic effect. This recovers the scaled original spectra at baseband. The harmonic distortion process has to do with the high frequency replicas associated with each squaring demodulation, for either modulation scheme. These iteratively demodulate and self-modulate, adding a spectrally smeared out and time exponentiated copy of the original signal to baseband and twice the original center frequency each time, with one iteration corresponding to one traversal of the space between the emitter and target. Only sound with parallel collinear phase velocity vectors interfere to produce this nonlinear effect. Even-numbered iterations will produce their modulation products, baseband and high
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frequency, as reflected emissions from the target. Odd-numbered iterations will produce their modulation products as reflected emissions off the emitter. This effect still holds when the emitter and the reflector are not parallel, though due to diffraction effects the baseband products of each iteration will originate from a different location each time, with the originating location corresponding to the path of the reflected high frequency self-modulation products. These harmonic copies are largely attenuated by the natural losses at those higher frequencies when propagating through air.

2.4.1 Pulse Width Modulation

Practical implementation and computer simulation has shown that the quality of the sound or its fidelity can be improved using modulation technique like Frequency Modulation or Pulse width modulation. In pulse width modulation the width of the clock pulse is varied acording to the amplitude of the signal thus the time duration and power of each pulse depends upon the amplitude of the signal.the lerger tha amplitude greater is the width of the pulse. The modulation is carrien out by a comparatorwhich is given modul;ating signal and choping signal as input

Figure 2.3 : Generation of PWM signal

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Pulse-width modulation uses a rectangular pulse wave whose pulse width is modulated resulting in the variation of the average value of the waveform. If we consider a pulse waveform ( )with a low value
min,

a high value

max

and a duty cycle D (see figure

1), the average value of the waveform is given by: ( ) 2.12

As

( )is a pulse wave, its value is ymax for 0 < t < D.T and ymin for D.T < t < T. The

above expression then becomes:

( (

) 2.13

This latter expression can be fairly simplified in many cases where

as

. From this, it is obvious that the average value of the signal ( ) is directly dependent on the duty cycle D. The simplest way to generate a PWM signal is the interceptive method, which requires only a sawtooth or a triangle waveform (easily generated using a simple oscillator) and a comparator. When the value of the reference signal (the red sine wave in figure 2) is more than the modulation waveform (blue), the PWM signal (magenta) is in the high state, otherwise it is in the low state.

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Figure 2.4 : Wave diagram of generation of PWM

Above figure depicts a simple method to generate the PWM pulse train corresponding to a given signal is the interceptive PWM: the signal (here the red sine wave) is compared with a sawtooth waveform. When the latter is less than the former, the PWM signal is in high state (1). Otherwise it is in the low state (0).

2.5 Piezoelectric transducer

The piezoelectric effect is understood as the linear electromechanical interaction between the mechanical and the electrical state in crystalline materials with no inversion symmetry. The piezoelectric effect is a reversible process in that materials exhibiting the direct piezoelectric effect i.e. the internal generation of electrical charge resulting from an applied mechanical force also exhibit the reverse piezoelectric effect i.e. the internal generation of a mechanical strain resulting from an applied electrical field. For example, lead zirconate titanate crystals will generate measurable piezoelectricity when their static structure is deformed by about 0.1% of the original dimension. Conversely, those same crystals will change about 0.1% of their static dimension when an external electric field is applied to the material.
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Piezoelectricity is found in useful applications such as the production and detection of sound, generation of high voltages, electronic frequency generation, microbalances, and ultrafine focusing of optical assemblies. It is also the basis of a number of scientific instrumental techniques with atomic resolution, the scanning probe microscopes and

everyday uses such as acting as the ignition source for cigarette lighters and push-start propane stoves. A piezoelectric speaker contains a piezoelectric crystal coupled to a mechanical diaphragm. An audio signal is applied to the crystal, which responds by flexing in proportion to the voltage applied across the crystal's surfaces, thus converting electrical energy into mechanical. The conversion of electrical pulses to mechanical vibrations and the conversion of returned mechanical vibrations back into electrical energy is the basis for ultrasonic testing. The active element is the heart of the transducer as it converts the electrical energy to acoustic energy, and vice versa. The active element is basically a piece of polarized material (i.e. some parts of the molecule are positively charged, while other parts of the molecule are negatively charged) with electrodes attached to two of its opposite faces. When an electric field is applied across the material, the polarized molecules will align themselves with the electric field, resulting in induced dipoles within the molecular or crystal structure of the material. This alignment of molecules will cause the material to change dimensions. The active element of most acoustic transducers used today is a piezoelectric ceramic, which can be cut in various ways to produce different wave modes. A large piezoelectric ceramic element can be seen in the image of a sectioned low frequency transducer. Preceding the advent of piezoelectric ceramics in the early 1950's, piezoelectric crystals made from quartz crystals and magnetostrictive materials were primarily used. When piezoelectric ceramics were introduced, they soon became the dominant material for transducers due to their good piezoelectric properties and their ease of manufacture into a variety of shapes and sizes. They also operate at low voltage and are usable up to about 300. The first piezoceramic in general use was barium titanate, and that was followed during the 1960's by lead zirconate titanate compositions, which are now the most commonly employed ceramic for making transducers. New materials such as piezopolymers and composites are also being used in some applications.

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Figure 2.5 the Construction of a Piezoelectric Transducer

A transducer which is constructed out of piezoelectric material will have a natural frequency of resonance and it is appropriate that the transducer should be excited with alternating electric field which matches the natural resonant frequency of oscillation of the material. Transducers which are used for ultrasound imaging have to be tuned for different frequencies. For a transducer material in which ultrasound waves travel at the speed c, with a resonant frequency f, the thickness of the material is related by the formula f=c/2d. Hence, it is possible to tune various transducers constructed of the same material to different frequencies by adjusting the thickness of the material. The ultrasound transducer can be excited by a continuous wave, a pulsed wave, or a single voltage pulse depending on the requirements. The rear face of the piezoelectric crystal material is usually supported by a backing material which is tungsten loaded araldite, so that the vibrations in the piezoelectric material are rapidly damped after the initial excitation. The acoustic parameters of an ultrasound transducer include its nominal frequency, the peak frequency which is the highest frequency response measured from the frequency spectrum, the bandwidth of the transducer which is the difference between the highest and the lowest 6 dB level in the frequency spectrum, the pulse width response time of the transducer, which is the time duration of the time domain envelope which is 20 dB above the rising and decaying cycles of a transducer response. Ultrasound transducers are fairly rugged and the piezoelectric material does not loose its properties unless exposed to high
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temperatures approaching the Curie temperature for the material are reached or there are strong alternating or direct electrical fields opposing the direction of poling for the material. Mechanical stresses imposed on the piezoelectric materials should not exceed the specified limits and although the specified limits vary for different types of materials, mechanical stress in excess of 2.5 MPa may be considered as likely to cause permanent damage.

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3. DESIGN AND CONSTRUCTION

The Block diagram of the system in shown in figure 3.1. The first requirement is a pulse width modulator. Since designing and implementing it using discrete components would prove too difficult, we opted for an integrated circuit TL494.

Audio Signals

Pulse Width Modulator

A Half Bridge Driver

Array of Piezoelectric Transducers

Driver Figure 3.1 Block Diagram of Directional Speaker System

The first requirement is a pulse width modulator. Since designing and implementing it using discrete components would prove too difficult, we opted for an integrated circuit TL494. TL494 is a 16 pin IC consists of 5V reference voltage circuit, two error amplifiers, flip flop, an output control circuit, a PWM comparator, a dead time comparator and an oscillator. It can be operated in the switching frequency of 1 KHz to 300 KHz. It can provide extremely accurate PWM signals according to input audio signal. It gives two complimentary output. We have used one but other can be used if more transducer are required to increase range. The second requirement is to amplify the PWM signals so that they are able to drive the large number of piezoelectric transducers. Piezoelectric transducer require high voltage for their operation. Also due to large number and due to their behavior as capacitors large current is also required.

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We solved this problem by using H-bridge circuit. A H-Bridge uses MOSFETs for switching the power supply, this leads to far superior efficiency than BJTs. This is due to the fact that a MOSFET switch is faster and the channel resistance in MOSFET is very low so that current can flow without any resistance within the transistor.

Figure 3.2 : H-Bridge

The MOSFET gate cannot be driven using the PWM signal generated by the modulator IC. To solve this problem we used a Half Bridge driver IC IR2110.

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3.2 Final Circuit

Based on recommendations from datasheets of ICs and as per our need we designed the following circuit. For connecting piezoelectric speakers following circuit is used. They are all connected in parallel to ensure coherent propagation of ultrasound which is essential for the speaker to work

Figure 3.3 : Final Circuit Diagram

. For connecting piezoelectric speakers following circuit is used. They are all connected in parallel to ensure coherent propagation of ultrasound which is essential for the speaker to work

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Figure 3.4 : Piezoelectric transducer circuit

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3.3 Components used

Integrated circuits LM7812 IR2110 TL494 IRF540

Manufacturer National Semiconductor International Rectifiers Fairchild Semiconductors International Semiconductors

Quantity 1 1 1 1

Table 3.1 : List of ICs used

Component Resistor Resistor Variable Resistor Capacitor Capacitor Capacitor Capacitor Diode Piezoelectric Tranducers 3.5mm Audio Jack PCB

Value 470 1500 20 k 100 nF 100 F 1000 f .0003 F IN4007 N/A N/A

Quantity 3 1 2 3 1 1 1 5 50 1 2

Table 3.2 : List of components used

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3.4 PCB Layout

We used Diptrace Software for creating the PCB for circuit.

Figure 3.5 : PCB layout

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3.5 Circuit description

The main aim of the system it to create ultrasound which is modulated by audio signals for this we first took a loudspeaker level audio input and fed it into the IC TL494. The audio is first filtered using a 100 nF ceramic capacitor to remove any DC from previous stage. PWM is created by comparing the input signal with a saw-tooth wave. The IC TL494 contains a built in oscillator whose frequency can be controlled by varying the value of variable resistor at terminal no 6. The frequency of oscillation decides the frequency of the pulses and hence the frequency of sound produced. The input is applied at the terminal 3 and 4. The PWM comparator compares the input with sawtooth pulse from the oscillator.

Figure 3.6: Internal Circuit of TL494 IC

The output from the comparator then goes through an and gate for output control. The output from the and gates g drives the transistors which switch the terminal C1/C2 at pin 8/11 respectively with terminal E1/E2 at terminal 9/10.

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The output from the terminal E1 is applied to the input of the Half bridge driver IC IR2110 at the terminals HIN and LIN at pin 10 and 12 respectively. The output can be taken at the terminal HO and LO which is taken as the gate input for the MOSFETs

Figure 3.7 : Internal circuit of IR2110 IC

The gates of the MOSFET control the path of the circuit. When gate voltage is high the channel is open and the current flows. When the gate voltage of transistor Q1 is high the channel is open and the current flows into the piezoelectric transducers and when the LO is high the MoSFET Q1 is off and Q2 is on and the charge can return to complete the pulse cycle.

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3.6 Software Used

Various kinds of softwares were used in running the schematics and designing the PCB.

3.6.1 DipTrace

DipTrace 2.07 proved to be a very handy & easy- to-use tool for the PCB layout process. Many of its features were utilized leading to an accurate & efficient design. It has Design Error Check & Electrical Rule Check tools which proved to be helpful in the design. It is loaded with a huge component list that is categorized in various libraries for giving simplicity. Placement of components is also very easy and they can be rotated in 360 to customize the design.

3.6.2 Multisim NI Multisim was an excellent tool for designing and running the schematics. It has a huge component library and a very easy and user friendly environment which give an ease of application with perfection in quality

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4. APPLICATIONS

Various applications of directional sound can be

4.1 Exhibitions and Museums

In situations like Exhibitions and Museums the information about an article may be needed to repeated again and again by the presenter. Traditional speaker cannot be used because they can produce interference with the speaker of neighboring articles. Using

directional sound it would of course be better to record the info and then automatically communicate it in front of exhibits. One can sound distribute the products with the needed info without disturbing the surroundings.

4.2 Planes, trains, buses

We can watch movies, listen to the radio or surf the internet when traveling in today's transports. In these transport vehicles the sound distribution is done by the headphones for one use only, and it's not very ecological or economical approach. Our directional sound solution enables us to remove this headphones barrier and creates the individual listening zones. . 4.3 Use as repellent

We all know that the sound can be very uncomfortable and that it's permanent distribution can be bothering. With regard to technological feasibility, we can broadcast uncomfortable directional sound signal to particular locations to draw back animals or persons. Even today we can use the directional sound as an invisible repellent against pigeons
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at the monuments, or as repellent against homeless people from the locations they are associating and harassing.

4.4 Addition for the security systems and alarms

Imagine the security system for the building, which will register the person in front of the doors of the secured object who's interested in breaking into the object. The security system will send sound beam to the doors: "Leave the secured space and drop what you are doing. If you won't, the security system will call the police immediately". This can be used also to trick the trespasser, when sending the human voice: "We can see you and we are calling the police now" to upset the burglar and to increase the probability that he will leave the object. We can use directional sound with access terminals like failure detectors etc.

4.5 Cars

Directional sound can be used functionally in cars where we can provide every passenger with individual sound. Today we can see ideal usage in sound distribution for the driver seat, thereby only the driver will be distributed with sound information about the route from GPS. We can also achieve discrete calls through hands-free or the driver could be alerted with system messages about the petrol status and important information. Presently Mercedes - Benz buses are fitted with audio spotlighting speakers so that individual travelers can enjoy the music of there on interest.

4.6 Waiting rooms, platforms etc...

It is difficult to find a quite place in an area like a waiting room of a railway station or a railway platform because of the megaphone. Usage of the Omni-directional speakers is making the train station strongly contaminated environment with sound. Directional sound
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allows us to solve this problem at halls or platforms. One can create restricted sound locations, where arrivals and departures times are communicated with sound and the rest of the dispatching area will be free of the sound pollution.

4.7 Theater and Stage hint system

Giving hints in a places like theaters and staged can be difficult task. Actors and presenters have to wear and carry wireless communication devices and use earphones. This system can be complex and often prone to malfunction. Directive sound can be used in such situations easily the direction of the propagation can be easily changed on changing of person in interest Students would appreciate for sure hints at school using this technology :)

4.8 Production lines and manufacturing processes

With all the automatization of the production the directional sound brings into this sector great revolution in communication on production lines or manufacturing processes. Today it is normal that most of the production is handled by automatization, but there is always human labor as well and because of this we are still trying to achieve the best possible communication between machine and human. Until now there is only the visual input, f.e. the next vehicle is approaching etc... If the human wants to work effectively, he can't, because he attends to visual communication and at this time he can work only with restrictions. If we would use the sound information instead, the worker can steadily dedicate himself for work.

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4.9 Retail advertising In stores and supermarket directional sound can be used to advertise and inform the costumer about the products.

4.10Audio/Video conferencing Project the audio from a conference in four different languages, forma single central device without the need for headphones.

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5. FURTHER IMPROVEMENTS AND SCOPES

5.1 Improvement in Fidelity

Even a best designed and constructed Ultrasonic Directive Speaker system has one major drawback that is lack of fidelity. Fidelity is the range of frequencies which a system can produce. The frequency response of a Ultrasonic Directive Speaker at low frequency is quite low that is it cannot produce low frequency sound efficiently low frequency sound or Bass gets severely attenuated in this kind of system. Due to this defect this system cannot be used to produce music, in which the low frequency components are essential. This makes the system unsuitable in entertainment places like clubs where this kind of system can have great use otherwise.

5.2 Reduction in attenuation

High frequency waves suffer a great deal of attenuation in air which increases with increase in the frequency of the ultrasound used. This leads to significant reduction in the range of the system. This leads to decrease in the efficiency of the system.

5.3 Reflections

Even if the produced sound is non-scattering there is still a possibility that the sound waves could be reflected from the surrounding objects. This could lead to decrease in directivity of the sound beam. On the other hand reflection can be helpful for propagation of sound to a listener who is not in a line of sight or there is a object obstructing the path of the beam.

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5.4 Safe use of high levels of ultrasound

For the nonlinear effect to occur relatively high intensity ultrasonics are required. The SPL involved was typically greater than 100dB of ultrasound at a nominal distance of 1m from the face of the ultrasonic transducer. Exposure to more intense ultrasound over 140dBnear the audible range (2040 kHz) can lead to a syndrome involving manifestations of nausea, headache, tinnitus, pain, dizziness and fatigue, but this is around 100 times the 100dB level cited above, and is generally not a concern. Dr Joseph Pompei of Audio Spotlight has published data showing that their product generates ultrasonic sound pressure levels around 130 dB (at 60 kHz) measured at 3 meters. The UK's independent Advisory Group on Non-ionising Radiation (AGNIR) produced a 180 page report on the health effects of human exposure to ultrasound and infrasound in 2010. The UK Health Protection Agency (HPA) published their report, which recommended an exposure limit for the general public to airborne ultrasound sound pressure levels (SPL) of 100 dB (at 25 kHz and above). OSHA specifies a safe ceiling value of ultrasound as 145dB SPL exposure at the frequency range used by commercial systems in air, as long as there is no possibility of contact with the transducer surface or coupling medium (i.e. submerged).This is several times the highest levels used by commercial Audio Spotlight systems, so there is a significant margin for safety. In a review of international acceptable exposure limits Howard et al. (2005) noted the general agreement amongst standards organizations, but expressed concern with the decision by United States of Americas Occupational Safety and Health Administration (OSHA) to increase the exposure limit by an additional 30 dB under some conditions (equivalent to a factor of 1000 in intensity). For frequencies of ultrasound from 25 to 50 kHz, a guideline of 110dB has been recommended by Canada, Japan, the USSR, and the International Radiation Protection Agency, and 115dB by Sweden in the late 1970s to early 1980s, but these were primarily based on subjective effects. The more recent OSHA guidelines above are based on ACGIH (American Conference of Governmental Industrial Hygienists) research from 1987. Lawton (2001) reviewed international guidelines for airborne ultrasound in a report published by the United Kingdoms Health and Safety Executive, this included a discussion of the guidelines
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issued by the American Conference of Governmental Industrial Hygienists (ACGIH), 1988. Lawton states This reviewer believes that the ACGIH has pushed its acceptable exposure limits to the very edge of potentially injurious exposure. It should be noted that the ACGIH document also mentioned the possible need for hearing protection.

5.5 Use of Digital Signal Processing

Use of digital signal processing can improve both the quality and efficiency of the system. Equalizer circuits and preprocessing of the sound signal before propagation can make the system much better.

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REFERENCES

1. Yoneyama M. and Fujimoto J.The audio spotlight: An application of nonlinear interaction 2. Pompei, F. Joseph (1999). "The use of airborne ultrasonics for generating audible sound beams". 3. Westervelt, P. J. (1963). "Parametric acoustic array" 4. Bellin, J. L. S.; Beyer, R. T. (1962). "Experimental investigation of an end-fire array". 5. Mary Beth, Bennett; Blackstock, David T. (1974). "Parametric array in air" 6. Muir, T. G.; Willette, J. G. (1972). "Parametric acoustic transmitting arrays" 7. Berktay, H. O. (1965). "Possible exploitation of nonlinear acoustics in underwater transmitting applications". 8. Kite, Thomas D.; Post, John T.; Hamilton, Mark F. (1998). "Parametric array in air: Distortion reduction by preprocessing". 9. Bass, H. E.; Sutherland, L. C.; Zuckerwar, A. J.; Blackstock, D. T.; Hester, D. M. (1995). "Atmospheric absorption of sound: Further developments" 10. Jacqueline Naze, Tjtta, Sigve (1980). "Nonlinear interaction of two collinear, spherically spreading sound beams". 11. Pompei, F Joseph (Sept 1999). "The Use of Airborne Ultrasonics for Generating Audible Sound Beams" 12. Howard et al. (2005). "A Review of Current Ultrasound Exposure Limits" 13. Lawton (2001). Damage to human hearing by airborne sound of very high frequency or ultrasonic frequency 14. Graf, Rudolf F. (1999). Modern dictionary of electronics, 7th Ed. 15. Manbachi, A. and Cobbold R.S.C. (November 2011). "Development and Application of Piezoelectric Materials for Ultrasound Generation and Detection"

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BIBLIOGRAPHY

Wikipedia the free encyclopedia en.wikipedia.org Articles on Sound from Ultrasound, Ultrasound Heterodyning, Parametric array, Piezoelectricity, Directional Sound, Pulse-width modulation, H-Bridge Elektor Electronics edition of March 2011

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APPENDIX

Sound - is a mechanical wave that is an oscillation of pressure transmitted through a solid, liquid, or gas, composed of frequencies within the range of hearing and of a level sufficiently strong to be heard, or the sensation stimulated in organs of hearing by such vibrations.

Frequency - is the number of occurrences of a repeating event per unit time. It is also referred to as temporal frequency. The period is the duration of one cycle in a repeating event, so the period is the reciprocal of the frequency. For example, if a newborn baby's heart beats at a frequency of 120 times a minute, its period (the interval between beats) is half a second.

Wavelength of a sinusoidal wave is the spatial period of the wavethe distance over which the wave's shape repeats. It is usually determined by considering the distance between consecutive corresponding points of the same phase, such as crests, troughs, or zero crossings, and is a characteristic of both traveling waves and standing waves, as well as other spatial wave patterns.

Amplitude is the magnitude of change in the oscillating variable with each oscillation within an oscillating system. For example, sound waves in air are oscillations in atmospheric pressure and their amplitudes are proportional to the change in pressure during one oscillation. If a variable undergoes regular oscillations, and a graph of the system is drawn with the oscillating variable as the vertical axis and time as the horizontal axis, the amplitude is visually represented by the vertical distance between the extreme of the curve and the equilibrium value.

Sound pressure or acoustic pressure is the local pressure deviation from the ambient (average, or equilibrium) atmospheric pressure caused by a sound wave. Sound pressure in air can be measured using a microphone, and in water using a hydrophone. The SI unit for sound pressure p is the Pascal (symbol: Pa). Sound pressure diagram: 1. silence, 2. audible sound, 3. atmospheric pressure, 4. instantaneous sound pressure Sound pressure level (SPL) or sound level is a logarithmic measure of the effective sound pressure of a sound relative to a reference value. It is measured in decibels (dB) above a standard reference level. The commonly used "zero" reference sound pressure in air is 20 Pa RMS, which is usually considered the threshold of human hearing (at 1 kHz).
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Direction is the information contained in the relative position of one point with respect to another point without the distance information. The speed of sound is the distance travelled during a unit of time by a sound wave propagating through an elastic medium. In dry air at 20 C (68 F), the speed of sound is 340.00 meters per second (1,115 ft/s). This is 1,236 kilometers per hour (768 mph), or about one kilometer in three seconds or approximately one mile in five seconds.

Acoustics is the interdisciplinary science that deals with the study of all mechanical waves in gases, liquids, and solids including vibration, sound, ultrasound and infrasound. A scientist who works in the field of acoustics is an acoustician while someone working in the field of acoustics technology may be called an acoustical or audio engineer. The application of acoustics can be seen in almost all aspects of modern society with the most obvious being the audio and noise control industries.

MOSFET - The metaloxidesemiconductor field-effect transistor (MOS-FET, or MOS FET) is a transistor used for amplifying or switching electronic signals. Although the MOSFET is a four-terminal device with source (S), gate (G), drain (D), and body (B) terminals,[1] the body (or substrate) of the MOSFET often is connected to the source terminal, making it a three-terminal device like other field-effect transistors. When two terminals are connected to each other (short-circuited) only three terminals appear in electrical diagrams.

Capacitor is a passive two-terminal electrical component used to store energy in an electric field. The forms of practical capacitors vary widely, but all contain at least two electrical conductors separated by a dielectric (insulator); for example, one common construction consists of metal foils separated by a thin layer of insulating film. Capacitors are widely used as parts of electrical circuits in many common electrical devices. When there is a potential difference (voltage) across the conductors, a static electric field develops across the dielectric, causing positive charge to collect on one plate and negative charge on the other plate. Energy is stored in the electrostatic field. An ideal capacitor is characterized by a single constant value, capacitance, measured in farads. This is the ratio of the electric charge on each conductor to the potential difference between them.

Resistor is a passive two-terminal electrical component that implements electrical resistance as a circuit element. The current through a resistor is in direct proportion to

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the voltage across the resistor's terminals. Thus, the ratio of the voltage applied across a resistor's terminals to the intensity of current through the circuit is called resistance. Color coding of carbon film resistor-

band A is first significant figure of component value (left side) band B is the second significant figure band C is the decimal multiplier band D if present, indicates tolerance of value in percent (no band means 20%) Color Black Brown Red Orange Yellow Green Blue Violet Gray White Significant figures 0 1 2 3 4 5 6 7 8 9 Multiplier 100 101 102 103 104 105 106 107 108 109 Tolerance Temp. Coefficient (ppm/K) 1% 2% (5%) 0.5% 0.25% 0.1% 0.05% (10%) 250 100 50 15 25 20 10 5

Diode is a type of two-terminal electronic component with nonlinear resistance and conductance (i.e., a nonlinear currentvoltage characteristic), distinguishing it from components such as two-terminal linear resistors which obey Ohm's law. A
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semiconductor diode, the most common type today, is a crystalline piece of semiconductor material connected to two electrical terminals. A vacuum tube diode (now rarely used except in some high-power technologies) is a vacuum tube with two electrodes: a plate and a cathode. The most common function of a diode is to allow an electric current to pass in one direction (called the diode's forward direction), while blocking current in the opposite direction (the reverse direction). Oscillator is an electronic circuit that produces a repetitive electronic signal, often a sine wave or a square wave. Square wave is a kind of non-sinusoidal waveform, most typically encountered in electronics and signal processing. An ideal square wave alternates regularly and instantaneously between two levels. Its stochastic counterpart is a two-state trajectory. Saw tooth wave (or saw wave) is a kind of non-sinusoidal waveform. It is named a saw tooth based on its resemblance to the teeth on the blade of a saw. The convention is that a saw tooth wave ramps upward and then sharply drops. However, there are also saw tooth waves in which the wave ramps downward and then sharply rises. The latter type saw tooth wave is called a "reverse saw tooth wave" or "inverse saw tooth wave". Printed circuit board, or PCB, is used to mechanically support and electrically connect electronic components using conductive pathways, tracks or signal traces etched from copper sheets laminated onto a non-conductive substrate. It is also referred to as printed wiring board (PWB) or etched wiring board. Comparator is a device that compares two voltages or currents and switches its output to indicate which is larger. Amplitude modulation (AM) is a technique used in electronic communication, most commonly for transmitting information via a radio carrier wave. AM works by varying the strength of the transmitted signal in relation to the information being sent. Single-sideband modulation or (SSB-SC) is a refinement of amplitude modulation that more efficiently uses electrical power and bandwidth.

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