IMPROVEMENTS IN OR RELATING TO LOUDSPEAKERS

Abstract: A loudspeaker has a digital input signal port connected to a digital interpolator to increase the effective sampling rate whose output is fed to a signal delay and magnitude detector after which the short-term dynamic range of the input signal is possibly reduced before application to a unary encoder which encodes the digital input signal into a plurality of unary signals which are then differentially delayed and pulse shaped before application to a plurality of substantially identical acoustic transducers via transducer drivers whose average power drive levels may be controlled by signals derived from the magnitude detector and from a human operator to alter the sound volume produced.

Description: ITvffROVFMENTS IN OR RFI .ATING TO I .OI mSPF.AKI-.RS Technical Field This ωvention relates to loudspeakers for providmg sound from electrical signal, which may be analogue or digital in nature Background & Prior Art Conventional analogue loudspeakers are well known and highly developed They rely for their operation on the motion of a diaphragm (usually a single diaphragm) which is driven by some type of electromechanical motor, moving coil being the most common, though electrostatic, piezoelectric and lomsation devices all been tried and used The analogue loudspeaker as a whole attempts to reproduce the desired sound by moving all or part of the diaphragm closely m synchronism with a smoothly varying analogue electrical signal which is usuall interpreted as representing the instantaneous sound pressure that a listener to the loudspeaker device should hear The inherent limitations of such analogue loudspeakers are related to. the mute (ι∞_-_____) stiffness of the diaphragms used (bending causes the diaphragm to not move as a rigid body, causing sound distortion, sometimes called "diaphragm break-up") which is a seπous problem at the higher audio ireqencies in any device where the diaphragm is not driven w ith a uniform force per unit area over its entire extent, which is the case in most loudspeaker types except electrostatic devices Exotic high-stiffness materials have been used to minimise this problem. the finite (non-zero) mass of the diaphragms used (mass causes the diaphragm to have significant inertia so thai the loudspeaker dynamics are no longer dominated by the mass and resistance of the air which is to be moved to generate the sound) limit the rate of change ol diaphragm velocity that can be achieved with practical motors and agam cause problems at the higher audio frequencies In general the previously mentioned stiffness limitations and the mass limitations interact - a diaphragm of a given mateπal may be made suffer but only usually at the expense of added mass. the linearity and efficiency of and power available from electromechanical motors with adequate bandwidth 20Hz to 20 Hz) A ptrvsically bigger motor will generallv allow greater power output but because ol limitations on materials and heat removal capacitv the power output tends to πse as the square of the linear dimensions whereas the mass of its moving parts will increase roughly as the cube of its linear dimensions resulting in its bandwidth tailing because ol the smaller high-frequency limit on performance There are also heat removal problems associated with small but high powered motors which limit the materials usable. the acoustic power deliverable at the lowest operating frequency of a loudspeaker of a given frontal area is directlv proportional to the maximum amplitude the diaphragm can achieve whilst operating within its approximately lmear region (sometimes known as the "throw") Moving coil motors impose limits on (he throw as the working section ot the coil must at all tunes move in a linear magnetic field if its output is to be linear, and there are practical limits on the length ol uniform field achievable with practical magnets, and practical limits on the overhang allowable of moving coils beyond magnet gaps The diaphragm suspension, whose function is to keep the diaphragm centred axially dunng motion and longitudinally in the no-dπve condition, SUBSTITUTE SHEET (RULE 26) also imposes limits on the throw achievable with acceptable Imeanty In general the suspension will produce a restraining force proportional to the deflection of the diaphragm from its rest position and at low frequencies where such forces generally dominate this will produce significant non-lineanties. hich may be overcome by position-feedback systems, but at the expense of increased power dissipation and heatmg of the motor, and which method- cause very audible distortion once the limit of their control region is approached, giv ing rise to very poor overload response Electrostatic loudspeakers impose other limits on throw, at least in part because of the requirement of close spacing of driver electrodes to the diaphragm in order lo produce reasonable output power from relatively low dπve voltages As the electrode spacing produces an absolute limit on diaphragm throw there is no way around this problem other than lo use higher dπve voltages or larger diaphragms in order to achieve higher acoustic power Piezoelectric motors have inherently very small throw because the piezoelectnc effect is a small effect (typically, direct deflections of < 10 'm volt are available though these are generally increased by some sort of mechanical lever, e g a "bender" configuration) , a compromise solution to some of the abovemenuoned limitations is to array two. three, or even four narrow¬ band loudspeakers each optimised for operation over then- respective freqency range, whose overall limits cover the whole audible frequency range. Input signals are split into suitable frequency bands by a "crossover" unit (some combination of lowpass. bandpass, and highpass filters) either pre- or post- power amplification This technique merely reduces the limitations descnbed. whilst introducing problems of its own including non- uniform frequency response in the cross-over regions, unbalance between the individual loudspeaker bands, phase distortions due to the filters, power loss and reduction in damping for post-amplification filters, and increase tn cost and complexity for pre-amplification filters. the nett effect of all the abovementioned (plus other) causes of nonlineant . result in a distortion figure for a good quality hi-fidehty loudspeaker rarely better than about 1% Even very expensive and state of the an devices cannot achieve distortion levels much better than 0.5% over the audible range as a whole: because the mass of the air moved by a diaphragm is generally very much smaller than the mass of the diaphragm itself (less so in an electrostauc loudspeaker) and because the viscous resistive reaction forces of the air on the diaphragm are generally much smaller than the spring force produced on the diaphragm by its suspension (again, less so tn the electrostatic loudspeaker), a great pan ol the electncal dnvtng power applied to the loudspeaker motor is consumed in moving the loudpseaker components and only a small pan in moving air or creating sound power Thus, such loudspeakers are highly inefficient Horn loading of a suitable loudspeaker can increase its efficiency but unless the horn has a small flare constant non-lineanties can become significant, and the physical size ol a bom tor operation at the lower audio frequencies can be unpractically large A typical hi-fidelity loudspeaker mav have an efficiencv (acoustic power output divided bv electncal power input) of order of 1%. with horn loudspeakers perhaps going as high as 30% to 40% within their frequency range. the enclosure tn which the analogue loudspeaker umt(s) is are housed also has a very significant effect on the output sound quality and the nature of current technology generally results in enclosures which are either heavy . or large, or both, tf hi-fidelity reproduction is to be achieved at significant power levels For domestic use. bulky enclosures are a distinct disadvantage Bulky enclosures are also problematical in portable sound reproduction systems such as those used for concerts by travelling performers. because of the linear drive requirements of most analogue loudspeakers, linear dπving amplifiers of appreciable power are required, and standard high qualit Class A designs are themselves quite -.efficient (say 30%) whilst even Class B designs where very low distortion is harder to achieve are rarely better than 60% Pulse width modulated amplifiers (PWM) have been used and their efficiencies can be much higher, in the 90°/o-95% region, but arc difficult lo produce lo the most exacting hi-lϊdelitv standards ol low noise and low distortion Taken together ith the low efficiency of the loudspeakers themselves, the combination of a linear analogue amplifier and a linear analogue loudspeaker will often have an electncal power input to acoustic power output efficiency as low as 0.3% and in hi-fidelity. rarely higher than I % This is a problem for portable batten- operated equipment as the size and weight of the battenes can be considerable and most of their energy is wasted as heat In domestic audio sv tems the wasted heat has lo be removed lo keep the equipment within its operatmg temperature range and generally results in larger heatsinking areas than would otherwise be necessary With the current prevalence of high quality digital audio matenal available, frequentlv in 16-bit binary format with an inherent distortion level of close to 0 002%. it is clear that analogue hi-fidelity loudspeaker systems operatmg close to the 1% distortion level (500 times worse) are now the limiting factor in audio quahtv when listemng to reproduced sound (including radio, television, compact discs (CD), and digital tape) Recent trends in electronic equipment have also been to mmtmise power consul mpoon. not onh to reduce power wastage, but also to reduce equipment operatmg temperatures thus allow mg miniaturisation and high reliability , as well as portability, and allowing operation from small battenes Agam. the linear analogue power amplifier/loudspeaker combination operating at the ϋ 3% to 1% efficiency level is out of step w th these trends Lastly, even though digital audio source matenal is now commonplace and becoming increasingly so with the advent of digital radio and television, all conventional hi- fidelity systems for the reproduction of digital source material need to contain a digital to analogue convener (DAC) at some point in the system, to produce analogue signals for application to the analogue loudspeaker. The DACs themselves produce further noise and distortion that adds to that already present in the system, and also add extra cost anv attempts have been made to develop a digital loudspeaker design that o ercomes some or all of the limitations of analogue loudspeakers mentioned above I hese fall into sev eral categories Pseudo-digital loudspeakers wherein a dev ice with a digital input terminal contains digital signal processing (DSP) equipment that is used amongst other purposes to compensate the inpul signals for the kno n distortions produced bv the linear amplifier and analogue loudspeakers also contained uiihin the dev ice and which create the output sound Such a sy tem is able to produce marginally unproved distortion performance overall, but no improvement in efficiencv . and still has a distortion figure significantlv worse than 0 1% It inherently suffers from all the limitations previously described, but compensates slightly lor some of them using the DSP system Moving Coii Digital Loudspeakers herein a tapped "voice-cod" <ι e divided into multiple sections) is used on an otherwise conventional analogue moving co loudspeaker The idea is that if the taps are connected to differenl numbers of turns, where the turns-ratio between the taps is ol the lorm 1 2 4 8 16 32 64 128 256 etc then direct binary digital input to the taps '. tll produce a net ampere-turn effect in the voice coil equivalent to that of the lmear signal bemg represented by the bmary digital signal The device is flawed in practice because a) It relies for its operation on a conventional (lmear) moving cod loudspeaker and thus brings with it all the problems of that device (and alleviates none ol them) b) As conventional loudspeaker voice coils have onlv a small number of turns (typically 30 to 100). dividing this accurately into as lew as 8 binarv -related turn taps requires a fractional turn tap for one or more of the least significant bits with all the attendant accuracy problems, given that the tap wires have to be brought out from the narrow air gap in the magnet wherein lies the voice cod Extending the dev ice to 10 or 12 bits is impractical Thus the dev ice is alreadv operatmg at around the 0 5% to 1% error (distortion) level c) Because die tapped voice coil is dπven by binarv digital signals, whenever an all-the-ones code changes to the next upwards code a large transient occurs because of inevitable skew in the timing of the signals, inductive spikes caused by the leakage inductance of the separate voice coil sections, and due to imprecision tn the matching and symmetry of the separate tapped sections Thus force-output hneaπty is generally much worse than current-input lincantv fed to the lapped sections by the digital driver, resulting in additional output distortion ov er and abov e that descπbed above for the analogue loudspeaker alone Those relying on piezoelectric and electrostatic drivers, where the area of the diaphragm is divided into separate regions whose areas are in the ratios 1 2 4 8 16 32 64 128 256 512 etc (see e g Elude d'un haut- parleur piezoelectnque assurant la conversion numenque-analof>ιque in ACL STIC A l ' υ/ 60 (1986) by Bnssaud M and D Noterman. also see Japanese Pats JP 83658S6 ιh304l4ι. JP 25236. JP 3106300. JP 58121897. JP 89244151] Because the total driving force applied to an electrostatic speaker s diaphragm is proportional to the diaphragm area, feeding bmary digital signals directly to such binary-related areas via separate conductors will in principle produce an appropnate force on the diaphragm for any particular digital input signal In practice there are senous problems with getting the areas preciselv in the right ratio when taking into account edge effects, plus transient problems at code change points as descnbed above tor the tapped voice coil digital loudspeaker Lastly, the device relies on the inherent Imeanty of the electrostatic loudspeaker (which is at best approximate in a real dev ice) and carries w ith it all of the problems of size, cost and complexitv that the conventional line— ' electrostatic loudspeaker has Similar considerations applv to the piezoelectric digital loudspeaker (here force is again proportional to area ol electrodes on the piezo matenal) but additional problems anse if one piece of piezo matenal is to be used for the hole device as such a large piece (as would be neeαed for high output) would be expensive (difficult to make) and edge effects again cause problems A venation on this theme is to be found in I S Pat 4515997 where the bmar -related areas are fabricated from an appropπate number of equal area transducers (i e 1 2. 4. 8. 16. 32. etc) each such group beam dπven in parallel by the same pulse signal This is claimed to help ith the problems of mass production of the transducer elements (as they are all identical) but as they are switched in groups of binary-related sizes they still suffer from the matching and transient problems at code change points as previouslv described for the tapped oice-coil case Pulse-width-modul ύon (PWM) has also been used in the context of "digital loudspeakers" Here an analogue or digital input signal is converted into a two-level (bmary in some sense) digital waveform whose instantaneous mark-space ratio is proportional to the instantaneous value of the input signal, ith 50% mark- space ratio corresponding to zero input signal The frequency of the PWM waveform may or may not be constant but needs to be much higher than the highest input frequency, and for audio applications this implies it must in practice be greater than about 40KHz So long as that cntenon is satisfied the actual frequency is not cπUcal With a digital input signal, it is possible lo produce a PWM wa eform entirclv digitallv However, when it comes to producing sound output, the PWM signal is applied to conventional lmear transducers (e g moving coil loudspeakers) The result is that the inertia of the transducer causes it to respond to the average value of the PWM waveform (which instantaneously is the same as the mark-space ratio) which in turn is equal to the mstantaneous value of the input signal Sound is then produced corresponding to the input audio signal As the device relies on the Imeanty of the transducer, this system has all of the disadvantages of analogue loudspeakers phis some extra ones related to the PWM conv ersion process, and so is really a digital amplifier technology . not a digital loudspeaker technology It does have the virtue of higher efficiencv than lmear amplification It is mentioned here for completeness General problems encountered when using binary digital coding at the output transducers All previous attempts at buildmg a digital loudspeaker system have assumed that bmary digital code was the digital signal medium, not only at the mput of die device (which is a reasonable assumption m practice) but also right through to the output transducers This causes senous technical problems m practice In an n-bit system, (be transducer used for the least significant bit (LSB) of the output operates at a power level 2"* 2 tunes less man the most significant bit (MSB) (discounting an assumed sign bit included in the n-bits) In an 8-bιl system (the least thai is uselul tor reasonable sound reproduction) there is thus a 64 limes power ratio between MSB and LSB transducers Because of the necessarily mechanical nature of sound producmg devices (sound is a mechanical movement of air) this wide dynamic range imposes senous design constraints on the types of devices used for LSB and MSB transducers, and thus makes matching ol the devices very difficult - the problems are much worse when one considers a more realistic 10 or 12 bit system where the ratios in power levels between the MSB and LSB transducers are then of the order of 250 to 1000 tunes, and for a 16-bit system the ratio nses to greater than 16000 In a binar -weighted transducer (orli sducer -array) system, there are senous transient problems caused al points where the code changes from a value with many consecutive low order zeroes or ones to the next level (up or down) where there are mam consecutive low order ones or zeroes For example, consider a 9-bit bmary code where the signal level changes from (decimal) 255, 0 = (bmary) 01 1 1 1 1 1 1 1, to (decimal) 256, 0 = (binary) 100000000, At this transition, the signal itself has changed by one least significant bit i e a very small change The bmary code representation has changed from a zero plus all-ones code to a one-and-all-zeroes code The effect of this on a system where Ihe code bits each dnve bmary weighted transducers (and also bmary weighted transducer-arrays as desenbed in ' _ Pat 4515997 which does not address this problem) is that tn the first state all transducers except the most significant w ill be on. and in the second state all w ll be off except for the most-significant Thus two half full-power acoustic transitions occur at this code point change wiuch will inevitably produce considerable sound energ . even though the code change represents only a least significant bit change in signal amphtude which normally ould be expected to be ncarlv inaudible Other such ones-to- zeroes and zeroes-lo-ones transitions occur throughout the signal amphtude range, and become more of a problem as the total number of bits increases, as the power of the transient increases relative to the system's least significant bit power level Thus, increasing the resolution of the system by adding more bits makes the problem worse, not better In addition to the switching transient problem outlined above, there is also a level error associated with such zeroes-to- ones and ones-to-zeroes code changes This is because in a real system the transducers cannot easily be matched precisely enough that the most significant bit transducer is precisely one least significant bit greater in effective power or amphtude than the sum of all the lesser-bit transducers actmg in concert The same is also true to a smaller extent for the next most significant bit transducer and its lesser-bit transducers acting in concert Such unavoidable errors can m practice easily dominate the accuracy of the sy stem and quite independently of the transient effects desenbed above can lead to large distortion components In a binary weighted transducer or transducer-arrav s stem (as desenbed in U S Pat 4515997 which does not address this problem either), only extreme mechanical precision can eliminate this problem even in principle, which will inevitably lead to high manufacturing costs even if the precision required is achievable In practice, the transducers will necessarily be spatially separated and the matchmg problems at such transition points then become intractable In a 16-bιl system, compatible with current digital audio standards, it is highly unlikely that the necessary precision could be achieved at any cost Another problem not adequately addressed by existing digital loudspeaker designs is that of transducer dynamics and appropnate dnve waveforms for producmg the desired acoustic sound output waveform All previous designs appear to make the assumption that the application of a square dπve pulse (of voltage or current as appropπate) to the output transducers will produce a square acoustic output pulse This is almost never the case in practice and leads to senous distortion in the generated acoustic waveform I- or example, in the common case where the transducer moving mass is the dominant factor, and the principal forces lo be overcome are tnertial. then the application of a square dπve pulse to such a transducer will produce approximatetv constant acceleration of the diaphragm w hich in turn w ill produce, to first approximation, a triangular or ramped acoustic output pulse, which will continue after (he end of the input dπve pulse at approvimateh constant amphtude as the diaphragm continues to "coast" due to its inertia For the other common case where the diaphragm restoring spring forces are the dominant tactor. then the appbcation ol a square drive pulse to such a transducer w ll produce a very rapid initial acceleration of the diaphragm causing it to move quickly to the point where the spring restoring force equals the dnving torce after which it will overshoot (depending on the damping of the system), and then settle around that point ol equilibrium, after hich the end of the dnve pulse will produce a similar velocity profile in reverse This motion ill produce, to first approximation, a pair of narrow impulsive spike acoustic output pulses of opposite sign, separated by a time interval approximately equal to the input dnving pulse length Only m Ihe case where the dominant forces on the moving mass of the transducer are resistive (e g due to friction or viscosity of the an * bemg moved by the diaphragm) will its motion be of approximately constant velocity when dπven by a square dπve pulse, and only m this case will the output acoustic pulse be approximately ol square pulse waveform Wh_ this means m practice is that electrostatic transducers with excepuonally light diaphragms ( hich constitute the onlv movmg mass in this type of transducer) are the onlv devices where a square dnve pulse might be expected to produce an approximatelv square acoustic output pulse Definition of Terms used in the Disckwure. Abstract & Claim. A digit is a single symbol representing a unique integer number A decimal digit can take on any of the ten values 0. 1. 2. 3. . 8. 9 Decimal integer positional notation uses the normal convention thai digits within a decimal number represent factors tunes powers often, with the nght hand digit representing a factor tunes 10° = 1 , the 2nd digit from the nght a factor tunes 10' = 10. the 3rd digit from the nght a factor tunes 10 * = 100. etc The value of the number represented by the decimal posTboαal code is the sum of the factors tunes each of their respective powers of ten So, e g 357,„ = 3 x 10 2 + 5 x 10' + 7 x 10° = 300 + 50 + 7 = 357 A binary digit can take on either of the two values 0. 1 Binarv integer positional notation is similar to decimal integer positional notation except powers of-κγj are used instead of powers of ten Thus the 4th binary digit from the nght in a bmary positional integer represents a factor of 1 or 0 limes 2 J So. e g 1 1010, = 1 x 2 + l x 2 J + O x 2 : + 1 x 2' + 0 x 2° = 16 10 -r 10 + 0 + 2 10 + 0 = 26, 0 A unary digit can also take on any of the two values 0. 1 or alternativ elv can be defined to take on onlv the single value 1, and its absence is then used to represent 0 somewhat as m Roman Numeral notation Unary integer positional notation is similar to binary or decimal positional notation except that integer powers of one are used instead of powers of 2 or 10 As all positive integer powers of 1 are equal to 1. it is clear thai with a unary representation, all digits have equal weight, and that weight is unity and that the position of a unarv digit in a unary positional notation number is irrelevant only its value of 1 or 0. or alternatively, its presence or absence, having anv significance Thus the 4th unary digit from the nght in a unarv positional integer represents a factor of 1 or 0 tunes 1 ' = 1 and the first unarv digit at the nght represents a factor ol 1 or 0 tunes 1° = 1 So e g 1 1010, = I x I 4 + 1 x T + 0 x 1 : + 1 x 1 ' + 0 x 1° = 1 10 + 1 10 + 0 + 1 , 0 + 0 = 3 10 . which is just the number of 1 -digits in the number Thus digit position becomes irrele ant in unarv numbers It is for this reason thai the 0 is not needed since its use as place-keeper in positional notations is irrelevant in the unarv case Thus we av j ust as precisely wnle the number 1 1010, as 1 1 1 with both representations having the decimal value 3 , 0 Unarv numbers are simplv a formal name for the marks people frequent Iv use tor lallving when, e g counting Hems It is important to realize that in unarv representation it is the number of one-digits that matters not the position of the one-digits Note also that an unsigned \ digit unarv code can represent \+l different values, because zero is represented by all the unarv digits bemg 0 or absent and does not need an additional unarv digit A signed Λ digit unary code where one particular digit is reserved to represent the sign often number (e g 0 for positive 1 for negative) can represent 2 \- 1 values (I e 0 and ± 1 to ±.V- 1 ) It follows, that where a digital signal is to be represented in unary code, that a unipolar signal that can take on N distinct levels can be represented by N-l unary digits Ibis is to be compared with bmary notation where N- l bmar digits are able to represent as many as 2 s ' distmcl levels I nan, representation therefore requires many more digits than bmary representation to represent a given range of v alues. just as bmary representation requires many more digits than decimal representation to represent a giv en range of v alues Terminology rhere is no special name for decunal digits It is conventional to abbrev late the phrase hman digit lo bit. Similarly, it is conventional to abbreviate unan digit to unit However, as the word unit is easily confused in this unfamiliar role ith its more conventional meaning, we use the phrase unary digit D_.lo.ur. of the Invention According to one aspect of the invention a loudspeaker comprises a plurality of substantially identical transducers each capable of convertmg an electncal signal mto a sound wave, whereω the transducers are capable of being dπven each independ-nth. of all the others by unary encoded signals representative of the sound to be produced by the loudspeaker According to another aspect of the in ention a loudspeaker comprises encoder means tor convertmg an put signal mto unary digital signals and a plurality of transducers each operative to convert a corresponding one of the unary digital signals mto a sound pulse so that the cumulative effect of the transducers is to produce an output sound representative of the put signal According to another aspect of the invention a loudspeaker compnses encoder means for convertmg an mput signal mto unary digital signals and pulse shaper means for convertmg the standard unary digital signals mto a vanety of square and non-square profile pulse signals appropπate to the type of transducers used and a plurality of transducers each operative to convert a corresponding one of the pulse-shaped unary signals mto a sound pulse so that the cumulative effect of the transducers is to produce an output sound representative of the input signal The transducers are preferably identical and in a preferred embodiment each transducer is bipolar, being capable of producmg positive and negative pressure changes dependent on the polarity significance ol the unary signal applied In one preferred embodirnent the transducers are arranged in a two-dimensional array 1 he shape ol each transducer may be such that they tcsscllate in two dimensions, e g bemg tnangular. square, rectangular or hexagonal Gaps between the transducers ma\ optionally be prov ided m this case Alternativ ely the shape of each transducer may be such that the transducers do not tcssellate. e.g. bemg circular or oval, gaps bemg provided between adjacent transducers The presence of these gaps in a first array of transducers can be exploited by providmg a further array of transducers behind the first aπa . each transducer m the second array bemg located behind a corresponding gap in the first array . making the arrangement three — n cnsional This process may be repealed to provide a composite transducer aπay w ith any number of lavers BAD ORIGINAL As the transducer aπay is distributed in space, either m two or three dimensions, a listener will be distanced from the transducers by varying amounts, dependent on the position of any particular transducer in (he aπay . with the effect that acoustic pulses emitted simultaneously by the transducers w ll amve at the position of the listener at different tunes This effect can be corrected by mtroducmg a delay means for differentially delaymg the mput signals to the transducers m dependence upon their distance from the listener such that the acoustic pulses from all the transducers resulting from a single mput signal change to the loudspeaker amve at the position of the listener simultaneousl . Further, the delay means may be adjustable to vary the delay applied dependent upon a chosen, and possibly var ing, position of the listene According to another aspect of the invention a loudspeaker comprises a plurality of pairs, tnplels or quadruplets of transducers arranged m a two or three dimensional aπay . each transducer of a pair, tnplet or quad capable of converting an electncal signal into a sound wave, wherem the pairs, triplets or quads of transducers are capable of bemg driven each independently of all the odier pans, triplets or quads by unary encoded signals representative of the sound to be produced by the loudspeaker, with each of Ihe transducers comprising a pair, tnplet or quad of transducers bemg positioned m the array such that the position of the centre of gravity of each pair, triplet or quad taken as a whole lies as close as possible to the vertical or horizontal centre lines of the aπay . or both, and m the case of a three dimensional aπay as close as possible also lo the front-to-back centre lme of the aπay . m such a way as to localise the perceived sound from the loudspeaker to as small an area near the centre of the aπay as possible The output sound produced by the aπay of transducers is the additive effect of the individual sounds produced by the individual transducers No individual transducer reproduces the desired sound. In Ihe case where the level of dnve to each transducer is fixed quieter sounds w ill be reproduced as a result of activation of a smaller number of transducers than louder sounds. The effect of encoding the input signal into a unary format is that an .1/ out of ,V encoding is produced where λ ' is the number of distinct levels representable by the mput signal and is also the maximum number of transducers required, and A. is the instantaneous mput signal level and is also the number of transducers activated by that mput signal level. Unlike a bmary coding (or any other higher order position-weighted coding system such as ternary, decimal etc ) a unary coding has the property that each indi idual unary digit represents the same identical value, i.e. one (arbitrar ) unit and it is solely the number of unary digits present or active that corresponds to the value of the number represented by any particular unary digital code word Thus unary coding has the unique property amongst all digital codes, that all members of a particular subset of indiv idual unary digits are alwa s off. inactive or absent when the value represented by a code word is below some specific number and one or more of them are always on. active or present when the value represented by a code word is equal to or above that same specific number I he relevant feature of this property to digital loudspeakers is thai using a unary encoding scheme at the output transducers completely eliminates the senous problems encountered when us g a bmary . ternary or higher-order digital representation scheme at the output transducers, which problems include the large transients produced when such higher-order codes traverse a value wbere a higher order transducer turns on or off and all lower order transducers turn off or on simultaneously This is because in a unary encoding scheme it is not required that the effect of one transducer bemg turned on should partially cancel the effect of other transducers bemg simultaneously turned off. In fact, m any given (unipolar) transition whatsoever of mput digital word value, unary transducers will either be turned on. or turned off. or neither, but never BAD OR.C; both at the same time Thus their operation m the application to digital loudspeakers is tree of such partial-cancellation transients Other distinct advantages of the unar encoding scheme at the output transducers is (hat all transducers are required to be substantially identical and small vanations between them will have little effect on the overall loudspeaker output signal level. Since it is the effect of the lotal number that are activated at any one tune, that represents the mput code word, a statistical averagmg over their individual sensitivities occurs, ensuring that the precision of the aπay as a whole is better than the precision of the mdiv ldual transducers If the deviations ol mdiv idual transducer sensitivities from a nominal value are randomly gaussianly distributed as would be expected, then a loudspeaker with Λ transducers would have a precision of .V that of the mdiv idual transducers. E.g. if Λ= 10.000 and the transducers are matched to withm 5%. then the Imeanty of the loudspeaker as a whole, ignoring other effects, is approximately 0 05% at full scale, illustrating the potential of this technology to produce extremely high precision with easy to manufacture low precision components. Thus the unary encoding at the output transducers eliminates the problems associated with. e.g. bmary weighted output transducers, where the transducer connected to the most significant bit has to be extremely precisely matched to all the lower-weighted transducers. By comparison with the example just given, a 13 bit bmary weighted digital loudspeaker system is able to reproduce just over 8000 levels, and its most significant transducer must be matched to the sum of all the lower order transducers to better than 1 part in 40% to produce output Imeanty better than 1 bit requiring a precision of better than 0 02% manufacturing which would be exceedingly difficult to produce in practice. Because m a unary encoded digital loudspeaker system all transducers have the same. unit, output power level or 'weight', from the point of vie of resultant sound output pressure, it does not matter which particular A/ transducers are turned on out of the total set of .V transducers, to produce an output pressure level of A/ V of the maximum available. Thus a degree of flexibility is avadable choosing subsets of transducers from the hole aπay which may be used to enhance performance a vanety of ways. Preferably . the transducers associated with similar mput signal levels are physically adjacent the aπay so that a reasonably localised sound source results, particularly at low amplitudes of reproduced sound According to one aspect of the invention, pairs ol identical transducers may be counecled directly parallel, one such pair bemg connected lo each of the unary encoded outputs of the driver circuits, and the transducers makmg up each such pair arc mounted one on either side of the loudspeaker ertical centre line and equidistant from it on a horizontal I me. so that their resultant effect is more closely similar to the effect that w ould hav e been achiev ed by locating the transducer pair on the v ertical centre line In this w ay . the undesirable horizontal spatial effect of a large aπay of transducers may be reduced According to another aspect of the invention, pairs of identical transducers may be connected directly parallel, one such pan- bemg connected to each of the unary encoded outputs ol the driver circuits, and the transducers making up each such pair are mounted one above and one below the loudspeaker horizontal centre Imc and equidistant from it on a vertical line, so that iheir resultant effect is more closely similar to the effect that ould have been achieved by locating the transducer pair on the horizontal centre Line In this w ay . the undesirable vertical spatial effect of a large aπay of transducers may be reduced According to a further aspect of the invention, four identical transducers are connected m parallel to each of the unary encoded outputs of the driver circuits, and the transducers making up each such quadruplet are mounted at the corners of a rectangle whose centre is approximately comcident with the centre of the loudspeaker aπay . so that their resultant effect is more closely similar to the eflect thai would have b. n achieved by locating all four transducers very close to the centre of the aπay In this w ay . the undesirable horizontal and vertical spatial effects of a large aπay of transducers may be reduced The method may be extended to other numbers ol grouped and paralleled transducers (e g tnplets located at the vertices of equilateral mangles whose centres are approximately comcident with the centre of the loudspeaker aπay. qumtuplets. sextets, etc ) This mechanism helps to localise the perceived source of sound for a listener because of the psychoacoustic effects m the ear and brain which cause a group of 2 or more similar pulses heard closely spaced time to be perceived as a single pulse at the average tune of arnval of the separate pulses fhe effect ol spacing pairs, tnplets. quadruplets or greater numbers ol transducers syinmctncaHy about the centre ol die aπay . all sunultaneouslv activated by one of the unarv dri er signals, is to give the appearance to a listener that the source of the sound is very close to the centre of the aπay of transducers, independently of the actual location of the transducers m the paralleled grouping, so long as then- joint centre of gravity (or centre of spatial symmetry) lies close lo the centre of the aπay. This technique allows the construction of a large digital loudspeaker comprised of arrays of transducers where the spatial extent of the aπay is comparable lo the distance of the listener from the loudspeaker, whilst still producmg the illusion of bemg a spatially small sound source w ith a definite location, close to the centre of the aπay In order to reduce the acoustic emission ol the loudspeaker m the frequency region above the limit of human hearing (uhτaso_ c emission), e g at frequencies greater lhan about 20KHz. an acoustic low pass filter may be added between the output transducer aπay and the listening space This may be implemented by the positioning of an appropπate quantity of matenal ith high sound absorption the region above 20 I lz and low sound absorption below that frequenc . between the acoustic ouput transducers and the listemng space This is desirable since domestic animals are frequently sensitive to such high frequency acoustic emissions and may be alarmed or distressed by them A second approach to reduce such ultrasonic emission from the loudspeaker is to increase the digital sampling rate as much as possible Standard digital audio matenal such as that avadable from compact discs and other common sources has a sampling rate the 401_iz to 50KHz region When reproducing a 20 I 1/ audio mput signal with such a sampling rate, only two or three samples will occur within each cycle of the mput signal 11 the same sampling rate is earned through all the way to the acoustic output transducers, significant acoustic energy ill be emitted below 100 KHz and smaller amounts at higher frequencies If the sampling rate is raised to e g 100 1 then the lowest strong supersonic emission wdl occur at a significantlv higher frequency and its amplitude will be proprotionately less than that of the components prcviou.sk present A method ol raising the effective sampling rat is to digitally interpolate the mput signal to the loudspeaker Such a process is alreadv done in digital signal processing sy ste s including lor example the digital to analogue converters used m the better quality Compact Disc players, pnncipallv to ease the electrical filtering requirements after the conversion to analogue electncal signals Here a similar interpolation process is used lo ease the acoustic filtering requirements after digital to analogue sound conversion The encoder means may have a plurality of parallel outputs coπesponding to the number of unary signals and the number of transducers An alternative arrangement is for the unary signals to be compressed m tune and for the encoder to have fewer outputs, in the limit a s gle output, means then bemg provided to reconstitute the unary signals as a parallel stream for application to the transducers Preferably , the loudspeaker assembly according to the in ention includes transducer dnvers connected between the encoder means and the transducers, the transducer dnvers conv ertmg the unary output signals from the encoder means lo appropπate cuπent and voltage levels to dnve the transducers. According to the dynamics of the transducers to be dπven from the unary output signals from the encoder means, possibly via transducer dnvers. there may additionally be pulse shaping means controlling the shape of the driving waveform to the transducers, and m particular the pulse shapmg means ma\ prov ide dnving pulses deviating from the nominal square shape of a standard digital pulse. Where the transducers are such that over the range of speeds of operation appropriate to use as elements of a digital loudspeaker their dynamics are dominated by resistive or viscous drag forces, then a square drive pulse will provide approximately constant velocity operation while the pulse is on and thus approximately constant pulse output pressure will result. Where the transducer dynamics are such that spring-like restoring forces (compliance) dominate, generally the case with transducers operatmg below their resonant frequencies and with low damping, then the pulse shapmg means may provide lmear ramp shaped dnving pulses which then also result in constant velocity operation for the duration of the mput pulse and approximately constant pulse output pressure Where the transducer dynamics are such that mertial forces dominate, generally the case with transducers operatmg above their resonant frequencies and with low damping, then the pulse shapmg means may provide bipolar impulse shaped dnving pulses comprising a short pulse comcident ith the leadmg edge of the mput pulse and a second short pulse of reverse polanty comcident with the trading edge of the mput pulse, which then also results m essentially constant velocit operation for the duration of the mput pulse (because the initial impulse from the pulse shaper gives the transducer moving parts some initial momentum after which they effectivel "coast" for the duration of the input pulse at constant velocity until the reverse impulse from the pulse shaper removes the momentum at the end of the mput pulse) and approximately constant pulse output pressure Where the dynamics of the transducers are composites of these three cases (he pulse shapmg means may provide a driving pulse waveform such as to produce essentially constant pulse output pressure for the duration of each mput pulse The pulse shapmg means may m all cases cited above be combined directly with the transducer dn ing means mto a composite structure Alternativ civ . the pulse shapmg means may be interposed between the encoder means and the transducer dπver means Another alternative is to interpose the pulse shapmg means between the transducer dπver means and the transducers b order to preserve the generally high power efficiency of digital pulse dπv e electronics when producmg square dπve pulses, the pulse shapmg means may be implemented using pulse width modulation techniques (PWM) herein the effective shape of the dπve pulse to the transducer is the mean value of a rapidly changing square waveform with many cycles occurring ithin the duration of one unary mput pulse and whose mark-space ratio is vaπed continuously as required m order to produce the desired effective pulse shape suited to the transducer dynamics ( I BAD ORIGINAL The eocodmg means may have grouping means for dividing n mput bmary digits or bits (if the mput were bmary) mto k groups of n k bits and may then have a pluralit ol encoders coπespondmg in number to k each with n k input bits, and each with a much reduced number of logic gates, the transducer dnvers then having some additional gatin The encodmg means for producing λ unary signals from n mput bmary (e.g.) bits, where Λ =2"- 1 or if one of the n mput bits is used as a sign bit then Λ=2" " '- 1. may be built m a modular fashion so that a number of identical encodmg sub- modules may be connected to a data bus carry g the complete mput bmary (e.g.) data words rcprescntmg the electncal input signal to be reproduced as sound. The encoding sub-modules, which are each designed to encode P unary digits where P<\ and where usually there would be O such modules such that P \ Q = V, may be pre-programmed before activation as encoders by sending to them control signals via a control bus and progranuning data via the data bus or the control bus such that each of the Q sub-modules after programming is responsive to a different group of 5 mput signal levels and encodes just that group of 3 input signal levels into P unary output signals. The nett effect is to encode all .V possible mput signal levels into P \ Q = N unary output signals but without the complexity of a brute-force n-bit binary (e.g.) to unary encoder, instead using Q identical modules which are easier to design and mass produce, and also easier to expand to different numbers of input signal bits n. The programming system may be made extremely simple by arranging for each of the 0 encoder sub-modules lo contain a flip-flop, and aπanging for the control-bus connection between the modules to interconnect the O flip-flops so as to form a senal mpul shift register. At programming tune a single pulse is introduced to the input of the shift-register so formed and which shift register is physically distributed amongst all 0 encoder sub-modules, and which is then clocked via a common clock signal present on the control bus. through the shift register one flip-flop at a time. As only one pulse is introduced into the mput of the flip-flop dunng programming, only one module can contain the pulse after each clock pulse has moved it to the next stage, and therefore if the flip-flop m each module is used to activate that module for programming if and only if it contains the shifted pulse, each module may be programmed uniquel in turn by introducing programming information onto the common data bus for example, and issuing a programming pulse onto the control bus common to all modules whereupon only the module containing the shifted pulse m its flip-flop will respond to that programming instruction. Thus by shifting the pulse through the O modules one module at a tune by means of the clock signal and issumg programming information after each such shift operation, the entire chain of modules may be programmed each ith information specific lo that module, even though the modules are logically identical and have no unique address as such. This module programming technique is widely applicable to any programmable modular structure connected to a common bus and is not restncted m its use to the digital loudspeaker design presented here The encoding means for convertmg digital mput m one form, e g bmary . to unary digital output, may be simplified where the mput form represents a signed quantity . by taking the sign information outside of the encodmg scheme and using it together ith the encoder outputs to control the transducer dnvers or pulse shapmg means directly to control the sign of the output signals. In the case of a bmary to unary encoder with n mput bits, where one of the mput bits is a sign-bit if the other n- l bits are fed lo an unsigned n- 1 bit bmary to unary encoder and the 2" " '-l unary digital output signals are fed to the transducer dnvers together with the mput bmar sign-bit a considerable savmg m circuitry results with no loss of information. BAD ORlG/NΛf ^ The loudspeaker assembly may have mierpolatmg means for mteipolatmg the digital mput signals to mcrease the effective sampling rate and thereby reduce spuπous high frequency outputs from the transducers The effective amplitude of the acoustic output pulses emitted by the unary output transducers may be adjusted whilst still maintaining high efficiency by gating them on and oft ' w ith a high frequency signal superimposed on top of ihe dπve signals from the unary encoder outputs and any pulse shapmg circuitry . in a logical AND manner, where the mark space ratio of the high frequency signal is contmuously v aπable from 0 lo 1 This is similar to pulse idth modulation but is an additional modulation lo that alread produced by the loudspeaker circuitry An alternative or possibly additional way of altering the effective amplitude of emitted acoustic output pulse from the transducers is lo pulse width modulate the power supply to the transducer dm er circuitry w hich agam may be done w ith high efficiency Both of these techniques allow a volume control function to be incorporated mto the loudspeaker whilst maintaining the highest possible signal to noise ratio as the effective attenuation of the volume control occurs nght at the output end of the loudspeaker system and thus attenuates any internally generated noise equally w th the signal. The method described m the preceding paragraph may be used to reduce the number of transducers required for a unary digital loudspeaker without reducing the effective resolution of the sound output This is preferably achieved by the incorporation in the loudspeaker assembly of power control means such as desenbed in Ihe previous paragraph which dynamically vary the output power of each transducer m dependence upon the amplitude of the mput signal The power control means ma mclude a digital delay device capable of storing, at full mput signal resolution of - bits (e.g. if the input signal is encoded m bmary). at least one half cycle of the mput signal at its lowest frequency, storage means for storing the maximum amphtude attained by the mput signal in the tune duration for which the mput signal is stored m the delay device, means for selecting the most significant consecutive mput signal bits (p <= n) containing a 1 and not a 0 in the most significant bit position of that group of p bits and discounting the sign bit for transfer to the unary encoder, and means for selecting the output power level of the transducers in dependence upon die maximum amphtude attained m the storage means, the selected power level prevailing whilst the stored mput signal is read out of the digital delay device In this manner, a digital loudspeaker capable of encoding <= p bits mto unary encoded signals drrvmg 2 P output transducers is able to produce a dynamic range of n bits (/- <= n) whilst avoiding the extra complexity of prov ldmg the additional circuitrv and transducers required b an • > bit unary encoder and output sv stem In order to allow analogue signal sources as well as digital signal sources to be reproduced through the digital loudspeaker that is the sub j ect ol this invention, an analogue to digital conv rter niav additionallv be incorporated mto the loudspeaker assembly to facilitate this function Because all the transducers have unit weight (i e they are identical and not relaied in a 1 2 4 8 16 elc relationship), the relative error of each one is the same, and even a 50% eπor m the sensitivity ol one ol the transducers would produce onlv a half least-signdicant-bit error rcfeπed to the bmary mput signal BAD ORIGINAL Λ Because unary code has no sequential all-ones to all-zeroes adjacent code changes (unlike bmary for example, as is illustrated by when it changes from 255 l0 = 01 1 1 1 1 1 1 1 : to 256 10 = 100000000.) there are no transients m a unary based digital loudspeaker associated with such digital pattern changes Because the transducers are all identical and each has unit weight (i e has only to produce the output required for Ihe smallest mcrcmental change in output (i e 1 bit)) they are low power dev ices, cheap and easy to make and edge effects are irrelevant as they are identical for each transducer, which is all (hat is required Because the transducers all handle exactly the same power level there are no engmeeπng problems associated with matchi g devices operatmg at vastly different power levels Each transducer turns on and off at most once dunng each cycle of a sinusoidal audio output waveform independent of the digital sampling rale, so the output transducers are not required to have bandwidth significantly greater than die bandwidth of the audio signals to be reproduced The performance required of the transducers is independent of the resolution (number of bits) required of the digital loudspeaker - one simply needs to add more identical units for higher resolution The unary digital loudspeaker sy tem that is the subject of this invention works because the pressure generated by M transducers bemg turned on is M times the pressure generated by one transducer bemg turned on As M transducers are turned on wben die mput digital code represents the number M men the output sound pressure is a faithful reproduction of the mput signal Because bmary (or higher order encoding) is not used at die output transducers, a unary encoded loudspeaker is not subject to any of the distortion generating problems previously desenbed here resulting from bmary or higher encoding at the output transducers Because sound is a longitudinal wave and can require both increases and decreases m pressure, it is practice preferable to provide both positive and negativ e pressure changes These can be produced with separate positive and negative pressure __nsduccrs. or with the same transducers dπven in a bipolar manner To reproduce silence, all transducers are turned off To produce a positive pressure ώe front face of the transducers is made to move outwards relative to the off- state To produce a negative pressure the front lace of the transducers is made to move in ards relative lo the off state If separate unary digit signals from the output ol the binarv to unary decoder represent positive and negative pressures, it is possible lo either apply these signals to separate positive-pressure and negative-pressure producmg transducers, or to dπve individual transducers a push-pull or bipolar manner with a pair of unary signals dnving each one The latter scheme reduces the number of mdrvidual transducers required for a digital loudspeaker of a given resolution by a factor of two Altcrnatrvelv . the sign bit of a bmary mput signal may be omitted from the bmary to unary encoder and separately used to control the polanty of pressure pulses from the acoustic transducers dπven by the (positive) unary outputs of die encoder This scheme also reduces the number ol transducers required for a digital loudspeaker of a given resolution bv a factor of two BAD ORIGINAL A practical digital loudspeaker of this type will require a large number of transducers L g to handle an 8-bit bmary input requires the representation of 256 sound pressure levels As le el 0 requires no pressure, no transducer is required for this level Therefore. 255 transducers (maximum) are required this example If the transducers are dn en m a bipolar manner, either, each by a pair of unarv digital signals representing a positiv e and negativ e unit step of pressure, or. by a sign control bu and unipolar unary digital signals, then 128 transducers w ill suffice In general, tor a system handling n-bit bmary mput either 2"- 1 or 2" " ' transducers are required, depending on w hethcr adv antage is taken of the bipolar dπvmg scheme desenbed abo e Whilst it is possible to use discrete transducers for this purpose, it is possibly advantageous to use mtegrated multiple transducers to reduce cost and manulacturmg complexity For example, if electrostatic transducers were to be used, it would be possible to produce a large number of electrodes of equal area, each with a separate connection to separate unarv digital signals, on one phvsical transducing device, thus producmg a transducer array If piczo-electnc transducers were to be used, then one piece of piczo-electnc matenal could be divided up mto a large number of equal area regions each with its own electrodes for separate connection to distinct unary digital signals, again resulting in a transducer array Sunilariy. with an electromagnetic transducer, a set of separately connected wires each producmg identical ampere-turn effects ithin the magnetic field of the dev ice. and individually connected to distinct unary digital signals, would agam result m a transducer aπay AU of these aπay structures could additionally be operated m a bipolar or push-pull fashion so that each transducer element ol the aπav had separate connections to two distinct unarv digital signals or a unary digital signal and a sign control bit for producmg posiliv e and negative output pressures All such arrav structures hav e the great adv antage of requiring multiple identical elements w hich assists with matching and simpler manufacture Brief Description of Drawing. Fig 1 is a block diagram illustrating the relationship between the vanous basic components of a digital loudspeaker according to the present invention Fig 2 illustrates simple logic lor a unipolar 1 -bit binarv to unarv conv erter Fig 3 illustrates simple logic lor a unipolar 2-bit bmary to unan converter Fig 4 illustrates simple logic lor a unipolar " >-bιt buian to unarv conv erter Fig 5 illustrates simple logic tor a Vbit offset binarv to unarv converter Fig 6 illustrates a method of push-pull (bipolar) dn ing ol transducers by pairs ot unarv signals ol opposite polantv significance Fig 7 illustrates simple logic for a 3 -bit 2s-complenιenl bmary to unary converter L 'BAD ORIGINAL jm Fig. 8 illustrates the essential puts and outputs of an /--bit umpolar binary to unar encoder with typical complexity of- (w- 1 )2" simple logic gates. Fig. 9 illustrates a method of assembling an »-bιt unipolar bmary lo unary encoder from two rTi/2)-bιt bmary to unary encoders and some additional simple logic Fig 10 illustrates details of one of the simple additional logic blocks shown in Fig 9 Fig 1 1 iDustratcs a scalable and extendable bus-based binary to unary encoder constructed out of a plurality of identical logic modules connected to the bus and programmed by a bus controller Fig. 12 illustrates m greater detail the possible structure of one of die bussed modules of Fig 1 1. which encodes a specific range of the input signals applied to i mto unary. Fig. 13 illustrates details of the simple flip-flop logic incorporated mto each of the modules of Fig 1 1. m order to allow each to be uniquely programmed by the bus controller illustrated m Ftg. 1 1 Fig. 14 illustrates an example unary signal aveform and associated suitable dπv e waveforms for acoustic transducers with various dynamical properties m order to produce acoustic pulses of approximately square shape Fig 15 illustrates simplified logic for a digital pulse width modulation (PWM) system for producmg lmear ramp PWM waveforms from a unary signal and a sign (polaπty) signal. Fig. 16 illustrates die conventional manner of interconnecting die counters and the m gnitude comparator thai are components of the system shown m Fig. 15 Fig 17 illustrates tvpical PWM waveforms produced by the circuit of Fig 1 w ith (he interconnection pattern shown Fig 18 illustrates an improved method of interconnecting the counters and magnitude comparator of Fig 15 Fig 1 illustrates the improved PWM waveforms produced by the interconnection pattern of Fig 18 when applied to the circuit of Fig 15 Fig 20 illustrates a simple logic circuit for producmg a dual bipolar impulse dnve for a transducer with lnertia- dominated dynamics Fig. 21 illustrates typical waveforms for the circuit shown Fig. 20 Fig. 22 illustrates the time-delay problems caused by extended aπays of multiple acoustic transducers Fig 23 illustrates how much more compact an aπay of acoustic transducers may be made by aπay in g them two dimensions rather than one Fig 24 illustrates in section how multiple two-dimensional aπavs ol transducers may be slacked three dimensions to produce a more compact sound source when gaps are placed each aπay to allow through the passage of sound Fig. 25 illustrates plan view a similar aπangemcnt to Fig 24 Fig. 26 illustrates a compact t o-dimensional aπay of transducers each posiuoned so thai it is adjacent to tiansducers associated with adjacent signal levels Fig. 27 illustrates the different path lengths between a listener and the vanous transducers m an aπay of transducers. Fig. 28 illustrates a method of apply mg differential signal delays to each transducer ui an aπay in order to compensate for drfferential listcner-to-transducer path lengths Fig. 29 illustrates how a delay system as illustrated m Fig. 28 may be made v aπable and programmable either dynamically or statically Fig. 30 ϋlustrates m block diagram form a system to select the most significant non-zero bits of an mput signal and apply them to a number of unary transducers so as to maintain the desired input-output relationship whilst maintaining the highest possible resolution. Fig 31 illustrates m block diagram form a loudspeaker incorporating most of the inventions and designs described herein Best Mode for Carrying Out the Invention Fig 1 illustrates the fundamental inno ation in this in ention Λ digital mput signal ol some kind representing a sound pressure avetorm is received by the apparatus al the mput buffer 1 on n mput signal paths This digital signal may be m any digital code form (e g bmary code, decimal code, m senal or parallel formal) For the purposes of explanation only we assume a bmary code input signal ol n bits, where one of the bits is a sign bit indicating the polanty of the mput signal though the mvention is not ui any way limned to this mput format The mput buffer 1 presents a uniform impedance to the input sιgnal(s) and performs am necessary level conversion and/or senal to parallel conversion before presenting the n parallel bmary bits to the encoder 2. The encoder 2 encodes the - bit bmary mput code mto .V unar signals, one of which is a unary sign or polanty signal indicating whether die remaining Λ ' -l unary signals represent a positive or negative quantity, and where .V = 2" " '. and where the prmctpal function of die encodmg is that hen die (positive) magnitude of the --bit mput signal is equal lo A/ then A/ of the Λ- 1 non-sign unar signals " ill be turned on (i.e. logic one) and the remainder turned off (i.e logic 0). The V output signals mcluding the sign signal from the encoder 2 are applied to the transducer dnvers 3 which con ert the \ unary signals from the encoder 2 mto V- 1 scls ol signals of appropriate current, voltage, pulse shape and polanty to separately dπve \- l acoustic output transducers shown collectively in Fig. I at 4 to which the transducer dnvers 3 are connected The Y- 1 sets of signals may be each a smgle bipolar signal with all Λ- 1 transducers sharmg a common return (nol shown), or alternatively, may each be a pan of push-pull signals The set of Λ- 1 acoustic output transducers 4 each of which is substantially identical to the others convert the electncal dπve signals lo uniform acoustic pulses each of amphtude a and polanty as mdicated by the mput sign bit As the action of the encoder 2 is to activ ate A / of the .Y- l non-sign unary signals when the digital mput signal has magnitude A/, each of which results m an acoustic pulse of amphtude a bemg emitted, the total emitted amplitude from the aπay of transducers is A/ x a = A Thus d e total acoustic output pressure from the system illustrated in Fig. I has the same polanty as the input signal and amplitude A/ x a when the digital input signal has magnitude A/, and thus faithfully reproduces as sound die digital electncal input signal albeit with some quantisation noise dependent on die number of transducers .V- 1. The mput buffer 1 is straightfor ard and will not be desenbed further The unary digital encoder 2 may be implemented in any of the standard ways familiar to those versed m the art of digital electronics, including suitably connected gates, programmable logic devices and read-only memory look up tables. The definition of the decoding function to be implemented will be illustrated for the case of an --bit signed binary mput. The encoder 2 will then have n bmary mputs b 0 . b,. b,. . _„.,. and .Y outputs u-. u,. u : . ... u v .,. where Y = 2" " ' The output u- w ill be die unary output sign signal indicating whether die unary output number whose magnitude is encoded m the remaining Λ'- l outputs is to be interpreted as positive or negative. It is defined as: u- = b_,. where bmary mput b.., is the sign-bit of the mput signal. The remaining n-l bmary mputs b-. b,. b : . ... b_. : represent an unsigned bmary number whose magnitude 1 ' may range between 0 and 2"* '- 1 = Y- l The remaining Y- i unary outputs u,. u-. .. u v .,. are defined as u, = 0 ιf l ' i. otherwise ., = 1. for O < ; - ' .V Thus if 1 ' = 0 (which is < 1 ) then all of the unary outputs are zero Otherwise, there are I unary oulputs w ith v alue I when _ιe mput bmary magnitude = I ' . where 0 < I ' <Λ Thus the sign bit is tnviaUv passed from mput bit b.., straight through to output unary sign bit u. The remai i g circuitry essentially implements an »• 1 bit unipolar bmary to unary encoder F ig 2 illustrates a tnv tal 1 -bit unipolar bmary lo unary encoder, while Fig 3 illustrates a 2-bιl version and Fig 4 a 3-bit version ol unipolar bmary to unary encoders Fig 5 illustrates a -bit oil set bmary to unary encoder 5 In this case there is no mput sign-bit as such Instead, when the bmary code is interpreted as a bipolar signal (offset bmary code), one has to define the code representing zero output sound pressure rat a 3-bιt offset bmary system this is usually laken as code 01 1. or 100. The table below shows how this would be encoded mto unary signals, some representing positive pressure outputs outputs, where we have assumed code 100. represents zero The decimal v alue ol the analogue mput value represented is also shown m the colurin labelled i/p decimal bmary negative outputs positi e outputs In this truth table, i p is given m decimal and represents a bipolar mput signal level, and bits 0 to 2 arc the same dung in bmary The unarv digit outputs op 1 lo op4 will be used to dnve negative pressure transducers, hilst unary digit outputs op5 to o 7 w ill be used to dπve positive pressure transducers As can be seen from the truth table, none of the negative pressure outputs op 1 to op4 are on (value 1 ) when any of the positive pressure outputs op 5 to op 7 are on (value 1) Thus it. for example, op I was paired with op5. op2 with op6. and op3 wιth op7 each pair dπvmg the opposite sides of a bipolar pressure transducer, then it will be seen that d e transducers ill propertv produce positive or negative pressure steps according to the code m die table, without mutual interference between the positive and negative outputs I e any one transducer will never be dπven by a positive and a negative signal of value=l simultaneously Thus die number of transducers required may be approximately halved with respect to the case where each of the unarv ouputs is used to dπve a separate transducer In Fig 6 max be seen how pans of outputs from die encoder 5 can be appbed to acoustic transducers 6 to provide bipolar dπve and thus bipolar pressure wave outputs As described above op l is paired w ith op5 to dnve one transducer and op 2 & op6 and op & op7 are sumlarly paired op4 can either be used for unipolar dn e ol an additional transducer 7 or simptv not used al all as no matching positive output signal is av adable to pair it with This w ill always be the case when deπv me unarv outputs from binary mputs because binarv codes necessanlv represent an even number of levels (all powers of two) one of which will be used to represent zero leaving an odd number which cannot be shared evenly between positive and negative levels So m general w ith u-bit offset binarv mput 2"- l unarv digital outputs will be derived bv the encoder and one of these will be un-pairable leaving 2"-2 unarv signals These w l tiien be aπanged mto 2""'- 1 pairs and applied to the same number of audio transducers possibh v la some transducer dnving circuitrv An alternative to offset-bman LS twos-complement binarv (commonlv used in modem digital computers for the representation of signed integers because of die ease of domg anthmeϋc witii this code) The truth table for a twos- complement to unary bipolar encoder is shown below BAD ORIGINAL oΛ Fig 7 shows a sunple implementation of _us truth table, and once agam illustrates that the conversion can be achieved with just 2 levels of sunple gates The gates marked nt in Fig 7 are simple non-mv ertmg gates used to more equally balance the propagation delay between mput and output of all the paths through the encoder In a practical digital loudspeaker, one needs to encode possibly 8-bιl binarv or more The complexity of Ihe encoder then increases considerably at least tn die sense that a large number of outputs have to be produced, and the number of gates increases accordmgl Also the level of gat g w ill mcrease if the individual gates themselves are to be kepi as simple as possible It should be noted that the absolute total gate delay through the encoder is usually of no consequence if it is less than 1 sec total Of some importance is the relative delay of the different input-to-output paths, since m an ideal encoder, all outputs would change simultaneously Such a condition can be well approximated bv keepmg the level of gating between mput and each output the same For example, it will be seen that m Fig 7. every path between mput and output traverses precisely gates Four gates, marked m (non-inverting), ha e been used to provide this path-length matching and have no other logic function It should be noted dial all of die logic circuits given above for encoders can be optimised bv die standard methods The circuits presented are for illustration onlv and do not attempt to minuiuze the number ol gates used As die extension to a bipolar encoder is tπvial. as illustrated above, we shall consider unipolar encoders only, below The number ol sunple gates m a straightforward unipolar binarv to unarv encoder increases roughlv exponentially ith the number ol bits ol umpolar binarv mput to be encoded (number ol simple gales ~ <n- 1 ) 2" lor n-bit unipolar binar ). so it is worthwhile looking at wa s to reduce the complexity ol an encodmg system 1 he requirements ol the umpolar unarv encodmg scheme are that A/ unarv outputs should be on when die digital mput number represented has magnitude \l Referring to hg 8 consider an n-bit unipolar binarv mput decoder 8. where n is even The n mput signals can be thought ol as 2 sets of n/2 bits The 2"-l outputs together with a virtual 0 output can be thought of as 2" : sets of 2 *: outputs (note that 2" : x 2 n = 2") The straightfor ard n-bit encoder will have order of (n-l ) 2" gates The scheme shown m Fig 9 uses two n 2 bit encoders 9 between which die n bmary bits 11 are equall y divided, and a tittle additional simple gatmg m die logic blocks 10 The logic blocks 10 m this example, of which only 4 out of a total of 2*' are shown. each dnve 2 π7 -l unary outputs and have the same number of standard inputs, each ol which turns on one output. In addition, each of the 2"~ logic blocks 10 is provided with an ALL mput which when turned ON. turns on all the outputs of the logic block: in addition there is an ENABLE input, which must be turned ON if any of die standard mputs is to be able to turn on its corresponding output, f he 2" : - 1 unary outputs from the upper encoder 9 are used m addition to the 2" : - l outputs from each ol the 2" : logic blocks 10 in the relative positions shown in He. 9 I aken together this results m 2*-- i + (2" : -l ).(2' l2 ) = 2 n -i outputs m lotal as required for an n-bit unipolar binary to unary encoder. Each of the n/2 bit encoders 9 has order of (n/2 -1).2' Λ gates, so two of them will have — («/2 -l ).2 r,:* 1 gates This can be very much less than the number of gates in an n-bit encoder For example, if n= 10 (a reasonable value for good sound quality ), then (n- l ) 2" = 9216 is the approximate number of gales for the standard 10-bit unary encoder, whereas (nil -l).2 r*:' ' = 4.2" = 256 is die approximate number ofgahs for a sta__rad 5-bιt unary encoder, a pair then requiring onl 512 gates which is ver much less than 9216. Thus the cost of the encoder fabricated in this manner can be much reduced as it is much simpler and uses multiples of me same device ( this example, two of the n/2 bit encoders). This decomposition scheme is not limited to the n to (n 2 times 2) scheme described here for illustration One can split up the mput bits mto groups in many other ways and still realise savings in the number of gates and the overall complexity For example, if n w as a multiple of 3. then one could split die n input bits mto 3 groups of n 3. (e.g. ύ n = 12. then instead ol a smgle encoder with order ol ( 12- 1 ) 2 >: = 45056 gates, one could use three 4-bit encoders) and m general, when n is a multiple of k. men the mput bits can be split into k groups of nlk. Fig. 10 shows an m mput logic block 10. containing extra gatmg to prov ide the ALL and die ENABLE functions. Approximately 2m simple gates are required for an m mput logic block. The gating required is v cry sunple. and because multiple such units are required (for our example above where a 10-bιt bmary mput ι> decomposed mto two groups of 5-bιts. one would need 32 (= 2') logic blocks each with 32 outputs) then unit cost is much reduced A different scheme for implementing an n-bit binary to unar encoder is illustrated schematically Fi 1 1. where die n-bit bmary mput signal 12 is applied via the bus controller 13 to a common data and control bus 14. lo which are connected a sencs of r identical encoder sub odules m,. m ; . in,. . ni, shown at 15. each ith p encoded unary ouputs grouped as Ci,. ( G,. ... G r shown at 16. The nature of d e bus structure allows a vanable number of identical modules IS to be added to the bus 14 to provide a v anable bit-w idth bmary to unary encodmg scheme, in operation, the bus controller 13 imtiahses all die modules IS via the control and data bus 14 at power -on or reset time, giving each of them a unique address After that initialisation, n-bit bmary data fed mto the bus controller al 12 is passed m parallel to all the sub_-θdιι s IS. and depending on each module's address (initialised by the bus controller as described) die modules IS each decode a unique range of die of the n-bit bmary mpul signal values mlo p unary outputs, for a total of p.r unary outputs. In one implementation, p is an integer power of 2. e.g. p = 2". and the submodules 1 S are implemented as shown in Fig. 12 where die n-bit bmary mput data section of bus 14 is seen to split mto two groups of bits. 21 and 22. with group 21 compnstng the q low order data bits and group 22 compnsmg the n-q-l high order data bits (excludin die data sign bit) of die bus 14. The q low order data bits 21 are fed into a <,-btt unipolar bmary to unary encoder 18 where / PAD they are converted ιnto/j output unary signals at 23 which m turn are connected to the um mputs of logic block 20 Logic block 20 acts as a switch between the p unarv signals 23 at its mput um and die p unarv signals 16 al its output uout. and is controlled bv two mput hoes AU and Enable If the All mput is on (logic 1 ) then all p unarv outputs 16 arc turned on. independent ol all other mputs to block 20 If die Enable mput is on. then each ol the p unary outputs 16 take on die state of the corresponding one of the /? unarv mputs 23. providmg a straight-through gatmg function I uiallv if neither ABDOT Enable is on. then all p unarv outputs 16 are turned off The n-q-l high order data bits (excludmg the data sign bit) 22 of die bus 14 are connected to a latch 17 and a magnitude comparator 19 rhe latch is controlled bv signals from control block 25 which m turn connects to die data and control bus 14 via die bus signals 24. which serve lo allow the bus controller 13 lo load a unique value into each latch 17 at system initialisation time v la a mechanism desenbed below (and dlustraled in I ig 13 desenbed later) In operation, after initialisation the »-<-- / bit v alue stored in latch 17 which is fed to mput B of n-q-l bit magnitude comparator 19. is contmuously compared to die value on the upper n-q-l data lines (excludmg ihe sign bit) of the data and control bus 14 which is fed to mput A of magnitude comparator 19 The A>B output of comparator 19 is connected to die All mput of logic block 20 while the A =B output of comparator 19 is connected to die Enable mput of block 20. The net! effect of this circuitry is that whenever die binarv mput data has a value less than the value held m die latch 17 (taking mto account its bit significance) then none of the p unarv outputs of a submodulc IS will be on Whenever the bmary mput data has a v alue greater than the value held m the latch 17 (takmg mto account its bit significance) then all of die p unan outputs of a block 1 S w ill be on Finalh w hen Ihe bmary mput data has a value equal to die value held m die latch 17 (taking mto account its bit signdicance) men the remaining q low order bits will be encoded mto p unarv outputs by encoder 18 If r such submodules IS are connected parallel to the bus 14 each programmed with a ddferent value m their respective latches 17 then the whole assembly wdl completely encode die n-bit binarv mput value onto tite p r unarv output tines as required This configuration has die values of simplicity, modulaπty. and easv extensibdity to greater numbers of identical modules IS and greater numbers of mput bits n Fig 13 illustrates a general method of allowing die interconnection of multiple identical modules on a parallel bus structure and providmg means of mdependendv controlling each of them even though Ihev contain no hard ired unique identification codes In P ig 11 27 represents a data and/or control bus lor connection lo multiple modules parallel similar to e g Ihe data and control bus 14 Fig 12 One ol the lines m die bus 27 is shown separatelv as 26 and 32. and is broken at each module connection location on die bus as shown at 35 The end of die bus in the direction of die bus controller (e 13 in Fig 12) is shown as 33 and the control I e 26 from that direction is connected lo die D mput of a standard D-tvpc flip-flop 30 whose O output is connected so as to dnve ihe lme 32 ol die bus awav from the bus controller and towards (he next module on the bus The flip-flop 30's clock and reset control mputs are connected to suitable bus control lines as shown as shown at 29 and 28 Whenever lme 31 m a module is at logic I that module will respond to programming information on me bus lines 27 (because of other circuitrv in die module nol shown, but controlled bv ime 31 ), bul will otherwise ignore it In operation, to control separaieh and uniquetv all of the modules on __ bus 27 each with control circuitrv as shown m Fig 13. die bus controller (nol shown m Fig 13) first issues a reset signal on die lme 28 which clears die flip-flops 30 m every module, after which it places a logical high on die Din lme 26 which connects only to die first module on die bus The bus controller ώen issues a smgle clock pulse on die clock lme 29 which clocks every flip-flop 30 As all flip-flops 30 were previoush reset, and all but die first on die bus receive D uφut signals from die O outputs of previous flip-flops 30 on die bus lines 32. onlv (he first tup-flop 30 on the bus w ill clock m a logic I . all the rest clockmg in a logic 0 At mis pomt the bus controller places a logic 0 on die Din control line 26 and issues anv required control signals destmed u quelv lor the first bus module Onlv module 1 will respond to die control inioπnauon as it dl be the onh module whose lme 31 is at logic hι_h as prev IOUSIV desenbed Thereafter die bus controller maintains a logic 0 on the Dm control lme 26. and issues successive clock pulses on lme 29 which have the effect ol shifting out die logic high on lme 31 of oue module mto the flip-flop 30 of the next module, hilst shifting in logic zeroes everywhere else, die hole structure operatmg sumlarlv to a senal shift register, and between successive clock pulses die bus controller issues programming and control lnfoπnalion for the one module currentlv activ ated bv its control lme 31 which is cuπently holdmg the single pulse in the shift register structure so formed If desired, die bus controller may additionally be connected lo the tar end of the control bus (aw av from die bus controller) via a spare lme. connected to tine 32 m the last module only on the bus. and this way. by awaitmg the arnval of the shifted-lhrough pulse die shift register, die bus controller can determine dial all modules on die bus have been piugianuned and may also count die number of such modules present, which is useful m a vanable modular structure where flexible programmabihty is desired As die transducers 4 m F lg 1 are what produce die external sound (acoustic) power from the electncal signals ithin the digital loudspeaker, die transducer dnvers 3 have to raise the digital signal levels to a power level adequate to produce die desired output sound power, taking mto account the efficiency of the output transducers Ihe level required will depend on the type of transducers used, c g piezo-electnc. electrostatic, moving-coil magnetic, magnetorestnclrve In digital logic lerms. die transducer dnvers 3 are simply pulse amplifiers In practical terms, thev may also be required to produce some pulse shapmg. to compensate for the transfer function of Ihe transducers 4. so as to maintain an approxtmatelv square sound-pulse shape It should however be noted dial die total output sound power of a digital loudspeaker with V output transducers is P = \ xp. where p is die output sound power of a smgle transducer Thus, if e g -one watt of sound power is required (roughly equivalent to that from a conventional movmg cod hi-fi loudspeaker dnven ith 100W of electncal power), then an 1 1 -bit digital loudspeaker with - 1024 transducers would require /? = - lmVλ output trom each mdiv idual transducer Such power levels are easilv av ailable dircctlv Irom logic gales Allowing lor low efficiency electro-acoustic U-ansducers (sav as low as - 1%) still oulv requires ~ l OmW ol electncal dnve power per transducer which is easily accommodated by a standard buffer logic gate operatmg at for example 5V and 20mA Thus practical digital loudspeakers mav hav e their transducer aπav elements directlv dnven from suitablv selected standard logic components Fig 14 shows live electncal waveforms 36, 7, 38, 39, 40 as a function of tune along Ihe horizontal axis 36 represents a typical bipolar unarv electncal signal av alable trom the binarv to unary encoder after combination with the sign information, hich corresponds to die desired pressure output from one transducer ol die digital loudspeaker that is die subject of this invention The waveform section shown encompasses a penod ol zero pressure demand between lime O and time 1. a constant positive pressure demand from time i to tune B. a further penod of zero pressure demand from B to C followed bv a penod of constant negative pressure demand from C to D. and thereafter zero pressure demand To a first approximatioiL the transducer diaphragm must move w ith constant v elocity lo produce constant pressure, and zero velocity to produce zero pressure, and therefore waveform 37 is illustrative of d e required velocitv profile ith tune tor a transducer to produce die pressure profile shown in 36 We make the simplifying assumption here, that is gencraih true tor common transducers, that the mput dnve voltage or current aveform to die transducer corresponds to the force produced on die transducer diaphragm For a transducer where the dominant reaction tor , on the diaphragm is resistive or viscous due to die resistance ol the air bemg moved m the production ol sound then wavclorm 37 (which is substantially die same as 36 apart from perhaps scale) represents a suitable force-time profile to achieve die desired pressure waveshape m 36. and m turn is also a suitable electncal dπve waveshape, so essentially no pulse shapmg is required m this case For a transducer where the dommant reaction force on Ihe diaphragm is a restoring force proportional lo the deflection ol Ihe diaphragm, as might be produced by a diaphragm suspension then waveform 38 represents a suitable force-time profile to achie e die desired pressure w veshape m 36. where it wdl be seen to consist of constant ramp rate sections between .-1 and B. and C and D. of opposite slope, and constant levels of zero slope elsewhere, since such constant ramp rates correspond to lmear increases of force, and thus displacement with tune, resulting m approximately constant pressure output for titese periods For a transducer where die dommant reaction force on the diaphragm is tnertial due to die mass of the movmg parts of the transducer and en tr a med air. then waveform 39 represents a suitable force-time profile to achieve die desired pressure waveshape m 36. where at tune .-I a short duration positive dπv mg lorce terminating al tune A ' is produced to give a positive impulse ol momentum lo the transducer's movmg mass, after which die mass coasts at approximately constant positive velocity until tune B. where it is given a short negative impulse until time B' to bring it quickly to rest agam. whereafter at tune C a further short negative impulse is given until time C to grve die movmg mass a negative impulse of momentum, followed by a further coasting penod at substantial constant negative v elocity until tune D at which a short positive impulse until tune D' is applied once more bringing die movmg mass to rest For a transducer with mixed dynamics where die dommant forces are some combination of die diree identified above, a composite dπve w aveform. an example of w hich is shown at 40. comprising some suitable lmear combination of 37, 38 and 39 may be applied to produce square-form acoustic pulse pressure output as required Wavelorm 37 mav be produced with high electncal elliciencv in a standard pulse amplifier W av eform 38 mav also be produced with high electncal eifieienev by means of pulse- idth-modulation (PWM ) ol a suitably high frequency pulse waveform in die following manner Fig 15 ύhistrates a novel digital pulse width modulation ramp generator, compπstng a high frequencv clock generator 41 which feeds mto the clock mput of a A-bil binary counter 42 w ith parallel binarv output OR at 52 which in turn feeds mto one of d e two parallel binarv mputs (_" in this case) of a A-bit binarv magnitude comparator 43 There is also a digital div ider 44 connected to the clock 41. the output d ol 44 shown as 51 bemg connected to one of two mputs of an AND- ate 45 The unary signal I shown al 47 which is lo be shaped as a ramp as shown tor example at 38 m Fig 14. is connected to die other mput of Ihe AND gale 45 w iti die result that w henever U„ is at logic one clock pulses from ./ of divider 44 issue from die output of die \ND gate 45 from where tiiev are connected to the clock mput of A-btl bmary up/down counter 46. and otherwise the output of the AND gate is at logic low Up/down counter 46 has its up/down control mput connected to die sign-bit (or unary sign signal) of die digital loudspeaker circuitrv and determines whether counter 46 wdl count up when a clock pulse amves at its mput. or count down Ihe reset mput of counter 46 is aπanged to set the counter to half full count when activated (e g d A were equal to 10. so that 46 had a maximum count of 1023 l0 = 1 1 1 1 1 1 1 1 1 1. (bmary ) then reset might be arranged to set e counter to 51 1 ,„ = 01 1 1 1 1 1 1 1 1 : ) and is connected to an external signal Res shown at 49 which might for example be issued by die bus controller 13 al system initialisation tune, or possiblv al other limes when the desired output signal from die unarv output I ) „ as zero The A-bit parallel binarv output UI ol up/down counter 46 is connected to die parallel bmarv mput I ol comparaior 43. such at die comparator conlmuouslv determmes the magnitude of the output Ol of 46 relative to the magnitude of the output OR of 42. output -1 B of comparator 43 en bemg at logic high whenever _*/ > OR Details of data synchroαizatioo are not shown for simplicity The effect of this logic circuitry is dial after reset tune (i e after a R s pulse has been sent to 49 from outside die circuit block) and while I τ r remains at logic low. die counter 46 stavs static al hall-full count while counter 42 cvcles throughout its A-bit count range with penod P = 2*'/ where/is die frequency of digital clock 1. and thus output PWλl shown at 50 derrv ed from output A B of 43. spends precisely half its tune at logic low and half al logic high This output 50 therefore has penod P and mark-space ratio of IT. Starting from this stale, when U„ goes to logic high, dien depending on die state of die Sign mput. counter 46 eι__- counts up or down from its initial half full count al a constant rate determined by die divided clock signal 51 so dial its mstantaneous output value 1 ' available m parallel bmary at 01 from counter 46 is linearlv varying witii tune at a rate of fD counts per sec. where D is die division ratio of clock divider 44 II the clock rate / of counter 42 is large compared wnh/D 0 e if D»l ) then I may be assumed substantially constant over a counter 42 penod P. m which case die PWM signal 50 will be high tor die fraction !7(2'-l) of the penod P where 0-^ I ' < (2*-1 ). which is exactly die condition required for the signal 50 to be a lmear pulse widti modulated representation of value 1 ' It can be shown that even when die condition/ »/D does nol hold, that the circuit still produces lmear pulse idth modulated signals at output 50 As die value 1 ' increases or decreases linearly with tune when IJ_ is at logic one (dependmg on whether Sign is at logic high or low), die effective value of die PH A/ output 50 (which is just die time average of output SO dunng a penod as long as or longer than die peiod P) is a linear ramp while U_ is on and a static value hen U n is off. which is precisely die condition required to generate die type of waveform dlustrated at 38 Fig 14. for dnving spnng-umited transducers to produce clean square digital acoustic pulse outputs In practice additional circuit refinements are useful, one of which is to configure counter 46 as a dead-end counter so tiiat when it reaches either maximum or minimum count it will nol roll-ov cr. but instead remam al its terminal couni v alue until die count direction ( up or dow n ) rev crscs and the next clock pulse amves This adds significant stability lo the PWM generator it is not essential that the clock mput elk on counter 46 be denved from the same clock 41 as used tor counter 42 as shown, although this agam assists with stabdity Additional stability can be achieved m ihe digital loudspeaker application of this circuit bv s y nchronizing clock 41 with die digital loudspeaker input-data sampling clock and separatelv bv dm mg (he Res input 49 high, whenever ihe mput data word v alue on ihe control bus represents zero Also tor die digital loudspeaker application, it is essential that the full-count penod ol counter 46. hich is T = 2*D// " ιs greater than or equal to the half-penod of ihe lowest trequencv audio signal that it is desired to be faithfully reproduced bv die speaker, tvpieallv 25ms for a 20Hz lower cut-off trequencv In die application of this PWM generator tor a digital loudspeaker it should be noted dial circuit components 41,42 and 44 mav be shared amongst a large number of mdividual PWM generators assigned each lo a distinct unary output U„. so die saving m parts is considerable It should be noted that this digital metiiod of creating pulse idth modulated (PWM) waveforms has applications outside of digital loudspeakers wherever PWM is useful A common requirement of PWM systems is a low pass filter system lo reduce die high frequency switchmg noise m die final output dπve waveform Such low pass filters are more complex and expensive 10 construct, the .loser die PWM clock rate is to the highest modulation frequency required to be reproduced m die low pass filtered output A method, which uses no extra components, of maximizing this frequency ratio for a PWM generator of die k d illustrated Fig 15 is now desenbed. F ig 16 sho in some more detail the conventional method ol interconnecting the two counters 42 and 46 to die magnitude comparator 43. where it wdl be seen that the least significant bit outputs ol the counters 46 and 42 shown as </<> q,, q^ arc connected lo the least significant bit mputs of the comparator 43 shown a . lg. A,. A ,. ... and B 0 B t _ > ... and die rest of die bits are connected in die same sequence nght through to die most significant bits q connected to. ),., and _ k , This mediod of connection results m a PWM output w aveform al 50 tiiat has a pulse period of 2"/ " where/is die clock frequency of counter 42 and A is die number of bits in that counter As an example, for die sunple case where -3. and where the value (assumed static) represented by die output of counter 46 is 101. = 5 l0 tiι_n Fig. 17 shows at 52 die expected waveform at output 50 ith mark space ratio of 5 3. and at 51 die clock mput signal to counter 42 and marks die counter 42' count state above each clock pulse as 0. 1. 2. 7, 0. etc.. Waveform 52 results because while counter 42 is in the first five states 0 \o 4 its output is less tiian that of counter 46 assumed static at value 5 in this example, and so die A B output of die comparator is at logic high for these stales, after which it goes low for the rest of the counter 42's cycle. In die improved version of die circuit shown in Fig 1 . it w ill be seen that die bit order of die connections between the bit outputs of counter 42 and die bit mputs of comparator 43 have been rev erscd so tiiat die most significant output bit of die counter q t ., is now connected to die least significant bit mput B 0 of comparator 43 and tins bit-order reversal is earned through for the other bit connections between these devices nght ihrough to q 0 connected to B, , The effect of this bit reversal is to alter die count sequence seen on die B. (0 t<k) mputs of comparator 43 when viewed m die conventional bit order, w ith B 0 bemg die least significant bit of this comparator mput The actual count sequence seen for die previously given example (with A- = 3. and where counter 46 has v alue 5 on its outputs) is shown at S3 m Fig. 19 The resulting P\\ A/ output from 50 in die modified circuit ι- shown at 54 m ig 19. where it will be seen dial whilst it still has die same required average mark-space ratio as die previous aπangement (shown at 52 in Fig. 17) of 5:3. it now consists of diree cycles dunng one penod of counter 42 rather ώan just one This is precisely die effect required to reduce die effort of low pass filtering It wdl be apparent to those versed in the art that this novel technique ol raising die effectiv e pulse rate of the PWM output w aveform is generalh applicable to all applications of pulse width modulation and is not restneted to use m die digital loudspeaker invention presented here Other rcordenngs of die bit connecnons between 42 and 43 than the one shown here are useful m this respect but it can be shown tiiat die bit-reversal ordering gives die maximum number ol outpul transitions over die full range of PWM output states In particular, die bit-reversed orderi g produces an output al 50 that transitions on each clock pulse lo counter 42 when die counter 46 is at half-full count, which is die maximum possible output frequency from such a circuit widi 50% or 1 1 mark-space ratio A digital mediod of producmg a waveform of die type shown at 39 in Fig. 14. for die dnving of mass-limited (inertia dominated) acoustic transducers is illustrated m Fig 20. where a unary mput signal . at 58 and the Sign signal dnve die clock mputs ( A- of a pair of flip-flops 57 and 60 via a pair of exclusive-OR gates 59 and 62 and an inverter 64. m such a way as to clock in a logic- 1 mto the D mput of one of die flip-flops on die leadmg edge of die L ' _ signal and into BAD ORIGIN;., j die other flip-flip on the trailing edge, which flip-flop responds to which edge bemg dependent on die Sign signal In die configuration shown when Sign is at logic low then llip-flop 57 is clocked bv the lead g (nsmg) edge of I „ and flip-flop 60 by die trailing (falling) edge The leadmg and falling edges of ' „ are aπanged to be synchronous with die nsmg edges of die clock signal Clo k at 55 (whose wavelorm is shown scheniaticallv at 65 m Fig 21 ) which is apphed dirough an mverter 56 to die reset inputs R of die flip-flops 57 and 60 1 he neti etiect ol diis configuration is that hen ' _ goes high ( see F ig 21 wavelorm 66) one ol die flip Hop outputs goes high and is then reset lo 0 hall a clock evele (of Clock) later as shown at 67 and when i „ next goes low the other flip-flop output goes high for half a cycle ol Clock and dien goes low loo as shown at 68 If the two flip-flop outputs l and _ _ are used to dnve a transducer 63 m push- pull as illustrated in Fig 20 either directlv or via transducer driver circuitrv Ihe transducer is dnven bv then- duTerence signal which is shown at 69 in F ig 21 This wav eform is preciselv ol the lonn required as shown al 39 in F ig 14 to dnve an inertia dommated acoustic transducer m order to produce clean acoustic pulses Because die output of die digital loudspeaker is synthesised from a large number of pulses, ratiter than smooth analogue waveforms, there will be f equency components in die output outside die normal range of hearing, generally reckoned to be ~20Hz to ~20KI Iz As tiiese components are by definition inaudible to humans, it is possible to simply ignore diem However, loud sounds in die range 20KHz to 60KHz can cause a certain amount of alarm and distress in domestic animals, and il mav be necessary to reduce such emissions as much as possible One approach is to place an acoustic low-pass filter over die output transducer aπav to absorb such frequencies directly at their poml ol generation A matenal with a heavy sound absorption above ~20 Hz but which is practically acousticallv transparent below ~20KHz will provide die required filtering A second approach is to minimise die high frequency emissions from the transducers diemsefves This can be done by ensuring that even at die highest frequency of operation, die resolution of the digital loudspeaker (m terms of bits, or unary digits) is kept as high as possible The ny quist dieorem tells us mat to adequately reproduce a 20KHz sine wave from digital samples we need to sample at a frequency of at least 40 Hz In practice reproduction of a sme wave from . ' few samples (ι e j ust 2 per evele when sampling at the nvquist rate) can onh be achieved with a perfect low pass filter The filtering requirements can be much reduced if instead we sample at a rate much higher dian the nyquist rate If the digital mput signal is avadable at a suitably high sampling rate then no more needs to be done other than to maintain thai sampling rate throughout die digital loudspeaker If howev er one was lo dnve a practical digital loudspeaker lron_ sav digital audio signals derived trom a Compact Disc which is sampled at ~44F_ Iz then one would need to interpolate the digital samples to create a higher sample rate Such a process is alread y done to some extent in better qualm compact disc plavers to ease the filtering requirements when convertmg die digital signals to electrical analogue signals lor further amplification Here we are suggesting that a similar process be performed on low- sampling rale digital mput signals to ensure that the sound output signals trom the digital transducers of the digital loudspeaker should ha e less spunous high trequencv content This design of digital loudspeaker using unary code at die output transducers, ensures thai mdividual transducers w l turn on and ofl ' only once per cycle of sine wave output, mdependent of the resolution of the digital output cnablmg tins digital mterpolation process to be earned out to any degree without mcreasmg the specifications of die output transducers in terms ol thev frequency response This independence is not the case d bmary . ternary . or oiher number- ( greater than or equal to 2) -based digital coding is used In this design of digital loudspeaker, die output sound is produced by an array of transducers actmg together and m concert No individual transducer akne reproduces die desired output sound signal. It is therefore important dial a human listener be placed such a position that an mtegrated effect is heard from all of die transducers equally . as far as is possible if the transducers have finite size then they w ill iiecessaπh be separated m space from one another Arrangements of transducers that minimize the path lengtii differences between mdividual transducers and die listener therefore are desirable For example, if diere were 1024 (= 32 x 32) transducers 70 as shown m Fig. 22. and if die transducers 70 were circular and say 30mm m diameter, men one possible arrangement would be to lay die transducers all out along a lme. Even if the spacing between diem was zero, die lme of transducers would be 30 x 1024mm =--31 metre which in itself is already impractical. Therefore, (refer to Fig. 22) tor die path length d from die middle transducers to the listener 71 at L to be no less than. say. 10% smaller than die path length h from die outer pair of transducers to die listener, it is necessary for die listener distance to the aπay d to be at least -32m For much closer than 10%pa_ι length m»ir_ιιnp rhis listener distance increases proportionately. Clearly this is unpractical for most (domestic) listening purposes. In order to minimize die least acceptable listening distance, it is tiierefore important to minimize die spatial extent of die transducer aπay This can be done by laying out die transducers 70 m as tight a two dimensional aπangement as possible, and regular circular, hexagonal and square array shapes are close to optimum from this perspective. For die example above, d a square aπay of 1024 30mm diameter transducers 70 was laid out as per Fig 23. dien die array becomes 960mm wide by 960mm high (i.e. 32 times shorter than the linear aπa sketched m Fig 22. and die corresponding listening distance falls from 32m to ~lm which is much more practical A second w ay to minimize die extent of die aπay of a given number of transducers, is to make die transducers -temscrves smaller across tiieir aperture. For example, for e aπay shown in Fig 23. dthe transducers 70 were 3mm across their aperture mstead of 30mm. then die aπay of 2 x 32 transducers could be made as small as 96mm square, and die minimum listening distance according to die same cntenon as desenbed above, would fall lo 10cm So. compact two-dimensional aπays ol small transducers are best for listening from practical distances If the transducers Ihemselves are dim. front to back, men die aπay size can be reduced even further by producmg a multi-layered three dimensional aπangement of transducers where a front two-dimensional aπay of transducers is placed over one or more further two-dimensional arrays of transducers behind it wuh die sound from e rear aπays passmg dirough gaps between die front aπays. or through holes in die transducers ihemselves If die transducers are necessanlv circular (for example, because of their method ol construction), then any regular aπay of circular devices necessaril has gaps m it as circles of one sue do nol tesseUale This multi-layered two dimensional aπangement then becomes attractive, and allows a very compact aπay to be constructed, even when using a large number of transducers Staggering successive two dimensional aπays allows the centres ol the rear transducers to align with the centres of the gaps or holes in die front transducer aπay s F ig 24 showmg a side v iew and Fig 25 showmg a front view of such a diree-dnnensional array of transducers illustrates these principles In order to synchronize the sound pulses received by die listener from die different layers in die durd dimension, it may be desirable to add differential digital delay to die signals m each layer to compensate for theu different distances from die listener (see below) Because unary digital code has no particular positional significance, we are free lo connect die unary digital outputs from the transducer dnvers. to transducers in die arr_y(s) in any spatial manner suitable. As quieter sounds wdl be reproduced wiώ smaller lotal numbers of transducers active tiian louder sounds, it is optimal to keep transducers related to adjacent input signal levels, physically adjacent m die output transducer aπay In this manner, the overall size of the sound source is kept as compact as possible at all sound output levels. Furthermore, if die geometnc centre of die group of transducers used to reproduce any particular sound leveL is kept as close to die geometnc centre of die whole aπay as possible, men die apparent sound source position will appear to move the least ith changes m reproduced sound level. Thus, good patteras of mtercoonection of transducers to transducer dnvers. include tight spirals centred on die geometnc centre of the aπay (with die obvious extension lo three dimensions d a multi-layered aπay is used) Fig. 26 illustrates this principle for die special case of a hexagonal two dimensional aπay of circular transducers Its extension to square aπays and otiier regular two-dimensional and duee-dimensional aπay structures is straightforward In order for die listener ' s bearing system (ear & bram) to be able to properly integrate the aπay of pulses from die digital loudspeaker so as to reconstruct die desired sound, it is important that sound pulses from die different transducers m me output aπay amve w ιtfa die correct tune relationship (i.e. at die same relative tunes as die parts of the original input signal toey represent) As die transducer aπay is distributed m space m two or three dimensions, a listener not placed a very long way from die loudspeaker ill hear die different sound pulses at tunes atiecicd bv their spatial positions m the aπay This is illustrated in Fig 27 where a listener is on the left at L and a t or three dimensional aπay (drawn schematically m cross section) is at the nght with just a few (1 ,. I -, T,. T,. T„) ol many transducers shown It will be seen that all die transducers T, ( ι= 1. m general are at unique distances /) - d from L. w here D is die distance to the nearest transducer to 1.. shown for example at I . in Fig 2 * " 1 he time of amv al i ot pulses al 1. emitted simultaneously from all transducers 1 at time t will be t, = r ->- (D * jk | where r. is the v elocitv ot sound), which in general ls ddferent for every transducer It is possible to completely correct for this undesirable effect for any one given listener position L. and approximately correct for a wide range of listener positions, by adding differential digital lime delay to die signals to each transducer Fig. 28 illustrates bow dns LS achieved - Y unary signals from die encoder are fed ui al die left each to a separate digital delay lme 73 producmg delays t,. u. . I . chosen such ihat - and d uλ is die maxunum value of t/,, and t, is the appropπate v alue of delay for transducer 1 s al distance />•_, from L. As die signals to each transducer are one-bu or unary digital signals, delay devices can be very sunple (e g. a 1 bit wide shift register, or suitably addressed RAM storage elements). The aπangeroenl of the delays is uch dial die transducers closest to the listening position are delayed the most and tiiose furthest from it e least or not at all so as to give die sound from the distant transducers a head-start' In practice, die delay applied would be quantised to die nearest mtegral multiple of a suitable delay time (e.g. 5μs. for maxunum 10% eπor on a 20KJ L* output signal). Thus a digital delay system with a 200KHz clock and vanable length real or synlhesLscd shift registers each transducer dnver path would suffice This could be mtegrated w ith die transducer dnve electronics mto an ASIC (Application Specific Integrated Circuit), and die actual pattern of delays required for a particular transducer and __tener-geometry programmed mto e driver chip. This program could even be varied to suit different listening positions dunng use of die loudspeaker. This is illustrated m Fig 29 where die .Y unary coded mputs amve at die left at 77 and are fed mto Y programmable vanable digital delay generators 76 with delays I,. U .. to I ^ each digital delay bemg controlled by a signal from die digital delay program store 78 which receives program delay information from mput 75 The outputs of die delay generators 76 eventually dnve transducers 74 as shown schematically The program delay information 75 can either be set up once before each use of the loudspeaker, or could alternatively be vaned dynamically throughout the use of die speaker, and one application of dns would be to track die listener's position relative to die loudspeaker and optimise die delay pattern I, to t for his current position The digital nature of die output transducers allows a method of volume level control dial ensures thai maxunum signal resolution and maximum signal -to-noise ratio is attained at all levels of listening, wiih particular advantages al low listening levels. With a conventional (analogue) hi-fi amplifier, die system volume control is usually placed somewhere before die mam power amplifier m me mpul-to-output path of die system The effect of ihis is tiiat die power amplifier itself is always operating at die same power-level regime (i e. always capable of full power output) and more importantly, always produces the same absolute level n of self-generated spunous output noise When listening at hieh levels, where die peak output power p is approached then the perceived si nal-noise-ratio ( snr) of the amplifier is p n However, when listening at much reduced (and much more common domestic) listening levels, e g ? / where / might typically be 100 (e g a listening level of 1 W electncal for a lOOW electncal amplifier) then the perceived amplifier snr is (pIDIn. l e die snr is reduced bv the factor / This can bring the amplifier sclf-noisc lev els mto prominence m a hi-fi system Ihe DI.S allows at least two methods of volume reduction the amplifier ilself . nght at die power generation pomt so mat noise and signal are reduced together, thus ma tammg ihe inherent snr ot die DLS/ampldier combination Mediod 1 LS lo supply the output pulse amplifiers from a vanable po er supply level, so that smaller pulse amplitudes are produced when a lower volume setting LS used. To __φlement such a scheme, die power supply output voltage would be made to depend in some ay . on the volume level setting chosen. In this case, output power is proportional to die square of supply voltage, giving a wide power output range hilst keepmg die supply level within operating limits of the pulse amplifiers / 1-δAD Method 2 applies pulse- width control to the output transducer dnvers. so that whereas normally the transducers would either be on or off for an entire digital clock cycle, by contrast with pulse w idth control, die transducers dial were previously on for die whole of each digital clock cycle would all be gated off for a same proportion ol each such cvcle If die proportion ol a cvcle gated off was t% dien output power would be reduced to ( 100-r)%. so with this method control is proportional to v' However, apart from limitations caused by die finite nse and fall tunes of die transducer dπver output pulse amphfiers. this mediod allows very wide range power level control, and mav be implemented entireh digitally Finally botii Methods 1 and 2 could be used together d required to optimize their respective benefits The mediod desenbed above for volume control and low-level Listening noise reduction, can also be used to reduce die total number of transducers needed m a DLS witiiout reducing die effective resolution of die sound output The mediod works by dvnamicallv applying die low-level listening technique as a function of the actual level of die mput signal Thus, when mput signal amphtudes are small, die output power provided bv each output transducer is reduced proportionately, and when die mput signal approaches its maxunum permitted v alue die output transducers are aπanged to provide then maxmium power For example, consider a sy stem with 16-bιl signed bmarv digital mput and only 1 23 (=2 '°- l ) unary output transducers, corresponding to 1 1 -bit signed bmarv signals I hen. it whenever the mput signal magnitude was small enough diat it could be expressed wtih lO or less bits, we could connect die lowest 10 (excludmg the sign bit) mput bits to a 10-bit umpolar bmary to unary encoder and dnve all die output transducers from that, but each with output power reduced from full load to 1/64 (= I 12*) of full power then we could reproduce Ihe low level signal with exactly die same output resolution as ύ we had 65535 (=2 "- 1 ) transducers for intermediate levels ol mput it is necessary to connect die 10-ιnput bits of the encoder m me example to bits I to 10. men 2 lo 1 1 , etc up to 5 to 15 for die highest level of mput signals In this way. mput signals greater titan I /64 of maximum mput level are always quantised to 10-bit precision nght dirough to die output, and smaller signals to exactfv the same precision as tiiev would be widi a full 16-bit DLS The dynamic range of a 16-bιt system and die precision of an 1 1 -bit system are attained at much greater simplicity than with a full 16-bit system The fact mat a 16-bit CD digital system sounds adequately accurate even when reproducing music at levels ven much lower than lull amplitude mdicatcs that the lull 16-bit precision is not needed for adequate sound quality It is necessary however lor dynamic range The scheme just desenbed provides both of tiiese features by effectivelv usmg a floating-point representation of die digital signal Fig 30 illustrates a possible mediod ot implementing this scheme Here an n-bit bmary mput signal 88 is to be reproduced widi an -bit digital loudspeaker where m<n I he n-bit mput signal 88 is fed mto a digital buffer memorv M al 7 9 capable ol sionng at least one hall-cycle ol ihe mput signal at its lowest trequencv (e g 50ms tor a 201 Iz low frequencv limit) It is simultaneously fed to a comparator C at 85 and maxunum value storage latch X at 86. such dial dunng die half cvcle. successively greater mput values become stored m die maxunum value latch 86 A zero crossing detector Z at 81 provides a signal at die end of each mput half cycle At dns pomt die value m die maximum value latch 86 mdicates how large is die biggest signal stored m the buffer 79. containing die digital mput samples yet to be reproduced (as tiiey have been buffered m 79 and thus delayed) Dunng die next half cycle, these stored samples wdl ,BAD ORIGINAL be read out of 79 and reproduced by die loudspeaker whilst new values for die next half cycle are bemg stored At die end of die half cycle, then, die value m 86 LS latched mto die number-of-bits register B at 84. which outputs an mteger in die range 0 to n-m (where m<n as before) This number is used to select the power level lo be used bv die output transducer dnvers 83 for die half cycle (which may be controlled by a pulse idth modulation technique, or a supply voltage vanauon lechnique. or botii in combmation. as desenbed prcviouslv ) It is also used lo scicci which of die unsigned mput bits 0 to n-2 will be passed lrom die buffer 79 lo the bmarv to unary encoder I) al 82 I " his selection is done by an m-bit wide, n-m way selector block S at 80. which takes its digital signal mput from buffer 79 (n bits wide) and sends m of those bits onto die unary encoder 82 Which bits it selects are determined by die signal from register 84 It always selects an adjacent group of bits, m bits wide, with die lowest output bit selected bemg in the range bit 0 to bit n-m. and die highest output bit selected bemg m die range m- 1 to n-2. bit n- 1 bemg the sign bit m this example Typical savings provided by tins technique are for a 16-bιt audio digital signal and a 10-bit digital loudspeaker provided with die floating pomt bit representation system just discussed, die digital loudspeaker would require only 1023 transducers and dnvers to reproduce sounds to 10 bit (umpolar) precision over a 16-btt dynamic range, as compared to a 16-bιt DLS which would require 32767 transducers - I e a sav g of over 31.000 transducers Thus, dns technique makes practicable die construction of unarv -based digital loudspeakers with ery high dynamic range and adequate precision, at relatively low cost A specdic embodiment of die invention incorporating many of die features desenbed above wdl now be desenbed by way of example with reference to Fig 31 m which a digital mput signal consisting of n bmary bits is applied to mput buffer 1. with an optional analogue mput at 92 connected to analogue to digital converter 90. producmg p bmary output bits where/? - n. die outputs of die buffer 1 and converter 90 bemg connected to die mputs of digital -elector/mixer 89. which is controlled by mput from a user at 96 Depending on die user mput some combmation of mput signals is sent m n-bit bmary form from mixer 89 to a digital interpolator 97 where die effective sampling rate of die mput signal is optionally increased before the n-bit data is passed to die signal delay store and magnitude detector 95 where die mput signal is delaved for a tune up to half die penod of die lowest audio trequencv to be reproduced, and where die peak magnitude ol the delaved signal is determined whose v alue is used to produce control signals 91 and 93 The delaved n-bit bmarv signal data are then passed to an m-bit (n-m ) wav selector (where m n- controlled bv signal 91. which emits m output bmarv (non-sign) bits connected to die mput of binary -to-unarv encoder 2 The sign bit is separately connected from delav store 95 directlv to encoder 2 as shown at 98 Λ he effect ol store 95 and selector 80 are to select up to m bits plus sign ot the n-bit mput signal m such a wav that the m most significant adjacent non-sign bits with a leadmg I are contained m die m-bit subset Encoder 2 converts die m bmary bits and sign bit mto \ unary signals where \ = 2" "1 and where one ot die \ unarv signals is a sign or polanty signal The \ unarv signals are then ted lo a vanable length digital delav generator 76 controlled bv delav programmer 78 hose mode of operation may be modified by die user at 96. where die differential delay of die vaπous \ unarv signals may be adjusted lo compensate for transducer 4 positions relative to listening position The possibly delayed V unarv signals ti.cn pass dirough pulse shaper 88 which may modify the square profile mput pulses (not die sign signal) to different pulse shapes appropnate to die type of transducers 4 used after which die Λ unary signals pass to die Y- 1 transducer drivers 3 which dnve die Y- 1 transducers __ ° ° IGIN L 4. the latter prov iding acoustic pulses which m combmation constitute die output sound reproduced The transducer dnvers 4 arc controlled by a signal from die power level controller 94 which turn is controlled by an mput from die magnitude detector in 95 and also by user mput which may clude volume lev el selection Ihe effect of this control signal on dnvers 3 LS to modify the mean output power dehvered to each of the transducers 4 when activated bv a pulse signal in such a way that die mean power level trom die aπay ol transducers 4 mav be \ ancd. in ihe presence ot a fixed level mput signal In «i -rtrial Applicability The digital loudspeaker thai LS die subject ot this invention has application in all instances where analogue loudspeakers are currentiy used mcluding reproduction of music and speech and odier sounds domestic and commercial equipment, including radios, televisions, record compact disc and tape players, music centres, hi-fidelity sound systems, public address systems, sound reinforcement systems, home tiieatres. cinemas, theatres, background music systems, bands, portable sound reproduction equipment, in-car entertainment systems, and mmiatunsed form m headphones. The advantages of dns digital loudspeaker design over existmg loudspeaker designs in these appbcations mclude higher quality and lower distortion reproduction, flatter form factor than most cabmet-enclosed analogue loudspeakers: greater stabdity due to digital rather than analogue electronics, elimination of requirement for separate lmear power amplifier from die sound reproduction system, tighter weight: greater portability easier manufacture to consistent high quality standard: mass production techniques may be apphed to the transducer aπay assemblies, higher efficiency and therefore lower power consumption and consequent longer operation tune from battery power sources: scalable design allows balancing of required precision aga st cost and complexity m a uniform manner as lower distortion may be achieved simply by die addition of more components of identical precision, produces essentially zero output noise when mput signal is zero (i.e. very high signal to noise ratio).

Claims: CLAIMS A loudspeaker compnsmg a plurahty of substantially identical transducers each capable of convertmg an electncal signal mto a sound wave, wherein die transducers are capable of bemg dnven each independently of all die others by unary encoded signals representativ e of die sound lo be produced by die loudspeaker. A loudspeaker compnsmg encoder means for converting an mput signal mto a plurality of unary digital signals and a plurality' of substantially identical transducers each operative to convert a coπespondmg one of the unary digital signals mto a sound pulse so that the cumulative effect of die transducers is to produce an output sound representative of the mput signal. A loudspeaker compnsmg encoder means for converting an mput signal mto a plurahty of unary digital signals and pulse shaper means for converting die unary digital signals mto a vanety of square and non-square profile pulse signals appropπate to die type of transducers used die pulse-shape bemg chosen such tiiat die output acoustic pulse from the transducer when dnven with said pulse-shape is approximately square m profile, and a plurahty of substantially identical tiansducers each operative to convert a coπesponding one of die pulse- shaped unary signals mto a sound pulse so tiiat die cumulative effect of the transducers is to produce an output sound representativ e of die mput signal. A loudspeaker as claimed m Claim 1 or Claim 2 or Claim 3. wherein each transducer is bipolar and tiius able to produce both positive and negative acoustic pressure pulses dependent upon the polanty significance of die unary signal apphed. A loudspeaker as claimed m any preceding claim, wherein die transducers are aπanged m a two dimensional arra . or a three dnnensional array with gaps m between transducers to allow sound energy from all transducers to pass to the loudspeaker listemng area. \ loudspeaker as claimed in any preceding claim, wherein delay means are provided between die loudspeaker mput signal and each acoustic output transducer in order that the tune ol arm al of acoustic pulses from each transducer may be independently adjusted eitiier statically or dynamically order to optimise some one or more parameters of die loudspeaker behaviour, including Ihe simultaneity ot amval al the position of a Listener of coπelated signals trom the transducers \ loudspeaker as claimed any preceding claim, wherein die prev iously desenbed transducers are each replaced bv a pair tnplet quadruplet or other multiple groupi g all members of die group bemg connected to die same unar digital dnve signal, and positioned phy ically such dial die geometnc centre of each multiple grouping of transducers is as close to a common pomt as possible, m such a way as to minimise the apparent spatial distnbution of die sound source as perceived by a listener to die loudspeaker, symmetric placmg aπangements of transducers each multiple group g bemg preferable. I A loudspeaker as claimed in any preceding claim, herein die transducers or multiple groupmgs of transducers are so connected to unary digital outputs such thai transducers associated with adjacent mput signal levels are physically adjacent or m close proximity m Ihe two or three dimensional aπay layout so as to minimise anv apparent motion of die sound source as sound level changes A loudspeaker as claimed in any precedmg claim, wherein an acoustic low pass filter is placed between Ihe output transducers and the listenmg area so as to mmunize unwanted acoustic ouput due to the pulsed nature of the outputs of individual transducers in die frequency range abo e die normal limits of human hearin Λ loudspeaker as claimed in any preceding claim, wherein the uiput signal is digitally ov ersampled in order to mcrease the unary digital signal sampling rate of signals applied to die output tiansducers and thus reduce unwanted acoustic ouput due to die pulsed nature of die outputs of individual transducers die frequency range above Ihe normal limits of human hearing A loudspeaker as claimed in any preceding claim, wherein an analogue lo digital converter is incorporated to allow analogue mput signals lo be applied lo and reproduced dirough the loudspeaker. A loudspeaker as claimed m any precedmg claim, wherein transducer dnvers are connected between the encoder means and the acoustic output transducers which convert the unary digital output signals from the encoder means to appropπate cuπent and voltage levels to dnve die required electncal power mto die transducers. A loudspeaker as claimed in die precedmg Claim, wherein pulse shaping means are incorporated between Ihe unary digital signal ouputs of die encoder means and die acoustic output transducers, either before the transducer dnvers or mtegrated with the transducer dnvers. such that the electncal dnve pulse shape lo each transducer compensates for its electroacoustic transfer function in such a way as lo optimise the pulse shape of its acoustic output pulse, such pulse shapmg means mcludmg but not bemg restneted to producmg lmear ramps, square pulses, bipolar impulse pairs and lmear combinations thereof tor the compensation of restonng- forcc dommated. resistance dommaled and mass do inated transducers Λ loudspeaker as claimed in Ihe precedmg Claim, whercm the pulse shaping means are implemented using pulse w idth modulation techniques in order to improv e the electncal drive efficiency to ihe transducers An encoder means for convertmg a digital mput signal comprised ot n mput digits mto a plurality of unary signals having grouping means for dividing n input digits mto A groups ol n k mput digits each group connected to a separate encodmg means capable ol encoding n mput digits mto unary signals, ith die output signals from the A separate encoding means combined using digital logic to produce the requisite total number of unary output digital signals, m tius manner reducing die total number of sunple logic gates required to implement die encoder means An encoder means for converting a digital mput signal compnsed ol ■■ mput digits mto a plurality of unary signals, budt m a modular manner from a number k of identical sub-encoders each capable ol encodmg n k mpm digits all connected to a common mput signal and control bus and programmable so as each to respond lo and encode lo unarv a ccrtam unique range of mput digital signals so as to encode die entire n mput digits of the mput signal when all actmg m concert A structure capable of allowing a number of identical modules all connected to a common bus w herein certain line or lines of die bus are daisy chamed dirough die modules and here a bus controller means is situated at one end of die bus structure, and where flip-flops incorporated lor the purpose each module are so interconnected by die common and daisy chamed bus structures as to effectively form a senal mput shift register wrtii the bus controller situated at die mput end of die shift register so formed, and wherein a smgle pulse from die bus controller may be shifted dirough die modules m sequence and one at a tune by means of a common clock signal impressed on die bus by die bus controller and readable by all modules, and wherein die modules are so constructed such that when die pulse m die shift register is within a module that module will respond to programming signals available to it on die bus from the bus controller logic and not otiicrweise. and m this way die bus controller may provide unique programming information to each and every module on die bus notwidistandmg die fact that die modules are logically identical and contain no unique hard- wired or pre¬ programmed unique address information, dins allowing completely standard -.programmed modules to be added to die bus and yet be uniquely programmable and thereb distinguishable m function, and wherein an additional bus control signal may be connected only to die last module connected to die end of die bus that is furthest from die bus controller m such a way as to signal to die bus controller dial die smgle pulse inserted mto the previously mentioned shift register has reached die last module on die bus m dns way allowing the bus controller to determine the number of identical modules connected to die bus simply by counting die number of shift register clock pulses issued \ loudspeaker as claimed in anv ol the precedmg Claims incorporating anv ol die encodmg means claimed m any of the precedmg Claims A loudspeaker as claimed in any ot die precedmg Claims incorporating interpolating means tor mteipolatmg the digital put signals or analogue mput signals alter conversion to digital signals, o as to mcrease die effective signal sampling rate and thereby reduce unwanted high trequencv acoustic emission from Ihe output transducers A loudspeaker as claimed m any of die preceding Claims incorporating mean-amplitude control ot the acoustic ouput pulses from die transducers by gating them on and off with a high frequency digital waveform m addition to any unarv and pulse-shapmg modulation already present and m this way providmg an effective volume control for the whole loudspeaker widiout reducing the effective number of transducers m use and thus preserving die resolution of die loudspeaker and thus nol raising its internally generated signal to quantisation noLse level, and without lo ering ihe electncal dπve efficiency ot the transducers A loudspeaker as claimed die precedmg Claim, herem die mean-amplitude control mechanism desenbed therein is automatically controlled so as to raise die mean-amplitude w hen the loudspeaker is reproducmg louder sounds and lower die mean-amplitude when reproducmg less loud sounds in such a way as lo always maxunize die number of transducers ihal may be dnven subject to die total output amplitude corresponding as clsoely as possible to die required ouput amphtude. and thus to mamtam high resolution over a wide dynamic range ithout necessanl providing as many separate output transducers as ihere are distmcl levels in the digital input signal A digital pulse idth modulation generator compnsmg an up/down digital counter, a digital magnitude comparator, and a second digital counter, wherem the magnitude comparaior compares me relative sizes of the counts in die two counters connected to die magnitude comparator m the conventional sense with least significant bit outputs connected to least significant bit mputs of the comparator and tiiereafter m bit significance sequence, and whose grcater-than or less-than outputs prov ide pulse width modulation signals, when die second counter is clocked contmuously by a constant rale clock signal and when the up/down mput of die up/down counter is controlled by the sign bit of die digital mput signal and the clock mput of die up/down counter is gated on and off by the presence or absence of die unary mput signal, wherem die pulse widtii modulated output signal or signals convert a steady pulse on die unary mput mto a pulse width modulated ramp at die output die slope of die ramp bemg determined by the polanty of the sign bit mput. An unproved digital pulse widtii modulator based on that described m die preceding claim and identical to it in all respects save tiiat the outputs of die second counter are connected to one of die mput ports of the magnitude comparator m bn -reversed order that is m the reverse of the conventional ordering of bits and m particular with die least significant bit output of die counter connected to the most significant bit uipout of die comparator and vice ersa, such that die pulse idth modulated output trom the comparator now makes many more transitions per total count c cle of the second counter whilst still maintaining die required average mark space ratio, thus significantly eas g die usually required low pass filtering of die pulse idth mouulated output \ means ot generating a suitable pulse wavelorm lor application to an clectroacoustic transducer whose dvnamics over die frequency rage of interest are dominated by die inertia ol the moving mass ol die transducer, when it LS required to produce an approximately rectangular acoustic pulse from the transducer from a unary inpul signal and a sign signal, wherem a pan of flip-flops is connected each w ith one Q output to one terminal of the output transducer, possibly v la transducer dπ er means, and whercm die unary mput signal is apphed to each of die clock mputs of the flip-flops v ta one mput ot two separate two-input exclusive-OR gate, and wherem die sign mput signal is applied m antiphase to die odier mputs of the excusrve-OR gates, and wherem die D mputs of die flip-flops are tied to logic 1. and wherem die Reset mputs of die flip-flops are activated by SUBSTITUTE SHEET (RULE 26) die trading edge of die unary mput signal synchronization clock, such that the effective dπve lo the transducer is an unpulse one half synchronization clock penod wide at die start ot the unary mpul signal whose polanty is controlled by die sign mput signal and a second impulse of equal w idth but opposite polanty at the end of the unary mput ignal pulse Λ loudspeaker substantially as described herein w ith reference lo Figures 1 -3 I of the accompaiiv in drawi g jBAD ORIGINAL $)

Inventors: HOOLEY ANTHONY, HOOLEY, ANTHONY
Applicants: HOOLEY ANTHONY, HOOLEY, ANTHONY
Application year: 1996
Main classification: H04R5
IPC codes: H03M1, H04R1, H04R3, H04R27