METHOD AND SYSTEM FOR MODELLING A SOUND CHANNEL AND SPEECH SYNTHESIZER USING THE SAME

Abstract: The invention is associated with speech synthesis and with the producing of speech by electronic methods. The object of the invention is to create a new method e.g. for the modelling of the human speech mechanism's acoustic characteristics, i.e., of speech producing. The acoustic transfer function modelling the sound channel is approximated by subdividing it by mathematical methods into partial transfer functions of simpler spectral structure. Each partial transfer function is separately approximated by realizable rational transfer functions. The last mentioned rational transfer functions are realized, each separately, by means of equivalent electrical filters, which have been interconnected in parallel and/or series in the manner implied by the acoustic transfer function which is to be modelled. The models produced by the method of the invention may also be utilized in speech identification, in the estimation of the parameters of a speech signal and in so-called Vocoder apparatus. The invention is also applicable in electronic music synthesizers.

Description: METHOD AND SYSTEM FOR MODELLING A SOUND CHANNEL AND SPEECH SYNTHESIZER USING THE SAME The present invention concerns a model of the acoustic sound channel associated with the human phonation system and/or music instruments and which has been realized by means of an electrical filter system. Furthermore, the invention concerns new types of applica¬ tions of models according to the invention, and a speech synthesizer applying models according to the invention. The invention also concerns a filter circuit for the modelling 1 of an acoustic sound channel. In its most typical form, this invenτ on is associated with speech synthesis and with the artificial producing • of speech by electronic methods. One object' of the invention is to create a new model for modelling e.g. the acoustic characteristics of the human speech mechanism, or the producing of speech, Modelj ρro- duced by thυ method may also be used in speech identifica¬ tion, in estimating the parameters of a genuine speech signal and in so-called Vocoder apparatus, in which speech messages are transferred with the aid of speech signal analysis and synthesis %/ith a minor amotint of information e.g. over a low capacity channel, at the same time endeav¬ ouring to maintain the highest possible level of speech quality and intelligibility. Since the model of the invention is intended to be suit- able for the modelling of events taking place in an acoustic tube in general, the invenτion is also applica¬ ble to electronic music synthesizers. The mctliods of prior art serving tho artificial producing of speech are divisible inτo two main groups. By the meth¬ ods of the first group onl such speech messages can be produced which have at some earlier time been analysed, encoded and recorded from corresponding genuine speech productions. Best known among these procedures are PCM (Pulse Code Modulation), DPCM (Differential Pulse Code Modulation) , DM (Delta Modulation and ADPCM (Adaptive Dif¬ ferential Pulse Code Modulation) . A feature common to these methods of prior art is that they are closely asso¬ ciated with signal theory and %/ith the general signal processing methods worked out on its basis and therefore imply no detailed kno% ledge of the character or mode of generation of the speech signal. The second group consists of those methods of prior art in which no genuine speech signal has been recorded, nei¬ ther as such or in coded form, instead of which the sueech is generated by the aid of apparatus modelling the func- tions of the human speech mechanism. First, from genuine speech are analysed its recurrent and comparatively in¬ variant elements, phonetic units or phonemes and variants thereof, or phoneme variants, in varying phonetic environ¬ ments. Zn the speech synthesizing step, the electronic counterpart of the human speech system, which is referred to as - a terminal analog, is so controlled that phonemes and combinations of phonemes equivalent to genuine speech can be formed. To date, these are the only methods by which it has been possible to produce synthetic speech from unrestricted text. In the territory between the said two groups of methods of prior art is located Linear Predictive Coding, LPC, /1 / J.D. Markel , A.II. Gray Jr. : Linear Prediction of Speech, New York, Springer-Verlag 1 97-3 . Differing from other coding methods, this procedure necessitates utili¬ zation of a model of speech producing. The starting as¬ sumption in linear prediction is that the speech signal is produced by a linear system, to its input being sup¬ plied a regular succession of pulses for sonant and a random succession of pulses for surd speech sounds, It s usual to employ as transfer function to be identified, an all-pole model (cf. cascade model). With the aid of speech signal analysis, estimates are calculable for the coefficients (a.) in the denominator polynomial of the transfer function. The higher the degree of this poly¬ nomial (which is also the degree of the prediction) , the higher is the precision with which the speech signal can be characterized by the aid of the coefficients a.. The said filter coefficients a. are however nonperspic- uous from the phonetic point of view. To realize a digi- tal filter using these coefficients is also problematic, " for instance in view of the filter hardware structures and of stability considerations. It is partly owing to these reasons that one has begun in linear predicting to use a lattice filter having a corresponding transfer function but provided with a different inner structure and using coefficients of different type. In a lattice filter of prior art, bidirectionally acting and structurally identical elements are connected in cas- cade. Vith certain preconditions, this filter type can be made to correspond to the transfer line model of a sound channel composed of homogeneous tubes with equal dimen¬ sion. The filter coefficients bi. will then corresp " ond to the coefficients of reflection (|b.! < 1). The coefficients b. are determinable from the speech signal by means of the so-called Parccr (Partial Correlation) method. Even though the coefficients of reflection b. are more closely asso¬ ciated with speech production, i.e., with its articulatory aspect, generation of these coefficients by regular syn- thesis principles has also turned out to be difficult. It is thus understood that speech synthesis apparatus of the terminal analog type, known in prior art, implies that speech production is modelled starting ' - out rom an acoust¬ ic-phonetic basis. For the acoustic phonation system, con¬ sisting of larynx, pharynx and oral and nasal cavities, an electronic counterpart has to be found of which the trans— fer function conforms to the transfer function of the acoustic system in all and any enunciating situations. Such a time—variant filter is referred to as a terminal analog because its overall transfer function from input to output, or between the terminals, aims at analogy with the corresponding acoustic transfer function of the human phonation system. The central component of the terminal analog is called the sound channel model. As known, this is in use e.g. in vowel sounds and partly also when syn¬ thesizing other sounds, depending on the type of model that is being used. Since the human phonation system is extremely complex of its acoustical properties , a number of simplifications and approximations must be made when .formulating -models for practical applications. A problem of principle which figures centrally in such model formulation is that the sound channel is a subdivided system '.vit an acoustic transfer function composed of transcendental functions . Creation of a corresponding terminal analog arrangement using lumped electrical components requires that the acoustic transfer function can be approximated with the aid of rational, eromorphic functions. Another centrally important point is the controllability of the model, that is the number and type of control par¬ ameters required in the model to the purpose of creating speech, and the degree in which the group of control par¬ ameters meets the requirements of optimal, "orthogonal" and phonetically clear-cut selection. in the following is described in detail n reference being made to Figs A- in the attached drivings "c e state of art associated with the present .nvention, and its theoretical foundation. — Fig. A presents u series (cascade) model as known in prior art Fig. 3 presents a parallel model as known in prior art. Fig. C presents a combined model as known in prior art. Figs' t) . E and F present, with a view ro illustrating the problems constituting the starting point of the present 10 invention, the graphic result of computer simulation. As ηown in pi-ior art, in constructing sound channel mod¬ els , the acoustic sound channel is simplified by assuming it to be a straight homogenous tube, and for this the 15 transfer line equations are calculated (cf. jzf G. Fant: Acoustic Theory of Speech Production, the Hague, Mouton 1970, Chapters 1.2 and 1.3j and /3/ J.L. Flanagan: Speech Analysis Synthesis and Perception, Berlin, Springer- Verlag 1972, p. 21*4—228). The assumption is made that the 20 tube has low losses and is closed at one end; the glottis-, or the opening between the vocal cords, closed; and the other end opening into a free field, The acoustic load at the mouth aperture may be simply modelled either by a short circuit or v a finite iπroedance Zr. The acoustic 25 transfer function that is being approximated will then have the form: (1) 30 where γ (s) = α + j ~ = propagation coefficient α = attenuation factor 3 = ω /c = phase factor ω = angular frequency * ^\ c a velocity of sound 35 Z = acoustic load impedance Zo - characteristic impedance of the channel 2. = length of the channel. Assuming that the losses of the channel are minor and Ό tha t the channel tuπiiiiiα Lus in short ci rcui t ( '/._ = 0 ) , or that the channel is loss-free and Z is re si stive , r * equation ( l ) be co mes : whore Λ, a niul k arc real. The logarithmic amplitude graph of the absolute value of the transfer function II. ( ω ) is shown in the attached Fig. 7« The homogeneous 0 sound channel chosen as starting point for the approxi¬ mation is most nearly equivalent to the situation en¬ countered w ex pronouncing a neutral vowel ( 3 ) « The profile of the sound channel and its transfer function are altered for other sounds. 5 The method commonly known in prior art for approximation by rational functions of the idealized acoustic transfer function IT,( ω) is to construct an electronic filter out of second order low-pass or band-pass ' filter elements 0 with resonance. Most commonly used arc the cascade cir¬ cuit of low-pass filters, depicted in Fig. A, and the parallel circuit of band-pass filters, shown as a block diagram in i- ' ig. B. 5 If in an acoustic channel when the channel profile changes its adjacent resonances approach each other, this causes the signal components in their ambience to be am¬ plified, similarly as occurs in series-connected elec¬ tronic resonance circuits. As a result, the cascade model 0 (Fig. Λ) of prior art is more advantageous titan the par¬ allel model (lig. il) . la order that the amplitude propor¬ tions of the resonances (or or ants ) might arrange them¬ selves as desired, it is necessary in the parallel model to adjust each amplitude separately (coefficients A1 -A 35 in Fig. 3). ∑n the cascade model , the amplitude relations automatical adjust themselves to be approxima ely cor¬ rect, and separate adjustments are not absolutely needed. It is true, though, that in this model too, considerable errors arc incurred in the fomiaπts ' amplitude propor¬ tions in certain circumstances, as will be shown later on, i a view to synthesis of consonant sounds on the other hand the parallel model is more favourable than the cas¬ cade model. By reason of its separate amplitude adjust¬ ments its transfer function can always be made to conform fairly well to the acoustic transfer function. Synthesis of consonant sounds is not successful with the cascade model without additional circuits connected in parallel and/or series with the channel. A further problem ' with the cascade model is that the optimum signal/noise ratio is hard to achieve. The signal must be altematingly derivat- ed and integrated, and this involves increased noise and disturbances at the upper frequencies. Owing to this fun¬ damental property, the model is also non-optimal with a view to digital realisations. The computing accuracy re¬ quired by this model is higher than in the parallel-con¬ nected model, . In Fig, C. has been depicted a fairly recent problem solu¬ tion of prior art, the so-called Klatt model, which tries to combine the good points of the parallel and series-con¬ nected models j / J. Allen, It. Carlson, 3. Graaistrδiπ, 5, Hunn cutt, D, Klatt, D, Pisoni: Conversion of Unre¬ stricted English Text to Speech, Massachusetts Institute of Technology 1 79 * This combination model of prior art requires the same group of control parameters as the par¬ allel model. The cascade branch F1-F is mainly used for synthesis of sonant sounds and the parallel branch Fl '-FV for that of fricatives and transients. The English speech synthesized with this combination model represents perhaps the highest quality standard achieved to date with regular synthesis of prior art. Λn obstacle hampering the practi- cal applications of the combination model is the complex¬ ity of its structural embodiment.The combination model requires twice the group of formant circuits compared with equivalent cascade and parallel models. E,-ren. though the s cii-cuits in different brunches of the combination associ¬ ated with the same formants are controllable by the same variables (frequency, Q, value), the complex structure im¬ pedes the digital as well as analog realisations. Approximation of the acoustic transfer function with the parallel model is simple in principle. The resonance fre¬ quencies F1...Y and Q values 1..-.Q of the band-pass filters are adjusted to conform to the values of the acoustic transfer function, the filter outputs are summed with such phasing that no zeroes are produced in the transfer function, and the final step is to adjust the amplitude ratios to their correct values by means of the coefficients A1,.,A- , f-, The use of the parallel model is a rather straight orward approximation procedure and no particularly strong mathematical, background is associated with it. In contrast, the method by which the -cascade model is ι'τeatcd is more distinctly based on mathematical analysis (see /" / * , P» 214— ), v/hen the load of a low-loss acoustic tube is represented by a short circuit, equation (l ) ob¬ tains the form (3) R ^ ~ cosh γ (s) Applying here the series expansion derived for functions of complex variables converts the expression to 2 W l _ - n cosh γ (s) 1 n=l (s-s )(s-s K ) n n where sn = first zero- of the function cosh (s)' s* = the complex conjugate of tlia above ω = the resonance frequency corresponding to the zero. According to equation (4), the acoustic transfer function of the ou of equal bandwidth resonances at uniform intervals on the frequency scale (see Fig. 7) > can be written as a product of rational expressions. Each rational expression repre¬ sents the transfer function of a second order low-pass filter with resonance. The desired transfer function may thus in principle be produced by connecting in cascade an infinite group of low-pass filters of the type mentioned. In realisations of practice, as known in the art, three to four lowest resonances are taken into account, and the in- fluences of higher for ants on the lower frequencies are then approximated by means of a derivating correction fac¬ tor (correction of higher poles, * see /Z/ p, 50-5 ' ) > T e correction factor calculated from the series expansion is graphically shown in Fig. D (curve a). The overall trans- fer function of the cascade model with its correction fac¬ tor is shown as curve b in the same Fig. D, The curve c in Fig. D illustrates the error of the model, compared with the acoustic transfer function. The error of approxi¬ mation is- exceedingly small in the range of the formants included, in the model. In actual, truth when speech is being formed, the profile of the sound channel and its transfer function are varied in large extent. It is important from the viewpoint of speech synthesis that the terminal analog that is used is able to model acoustic phenomena in any phases and varia¬ tions of speech. In addition to the difficulties already described, the cascade-connected model of prior art has presented problems in the modelling of the sound channel's transfer- functions. In cases of an inhomogeneous channel, which constitute the greater part of situations occurring in real speech, the cascade model causes errors in the amplitude proportions of the formants. With - view to Vocoder applications, attempts have been made to eli i.- nate this problem by a patented design based on afterward correction of the spectrum /j/ C Fant: Vocoder System, U.S. Patent No. 3,3^6,695, Oct. 10, 1967. Particularly controvorsial requirements are imposed by the tone bal- BAD ORIGIN anciπg of f ont and back vowels . The problem touched upon in the foregoing is illustrated in Figs E and F by the aid of computer simulations . In the 5 simulations, the acoustic sound channel has been modelled with two low-loss homogeneous tubes with different cross section and length. (cf. / / » P« °9-72). The cascade model has been adapted to the acoustic transfer function of this inhomogeneous channel so that the formant frequencies and 10 Q . values are the same as in the acoustic transfer func¬ tion. The transfer function of the cascade model is shown as curves a in the figure and the error incurred, as curves b. Fig. E represents in the first place a back vowel /o/ and Fig, F, a front vowel /' / . 15 Figs E and F reveal that the cascade model causes a quite considerable error in front as well as back vowels. The errors are moreover different in type, and this makes their- compensation more difficult. 20 In the foregoing, the most generally known methods for the modelling of speech production have been reviewed. These considerations may be summarized by observing that the following problems are encountered in the models of 25 prior art, the solving of which, in part at least, is one of the objects of the present invention. Cascade models (Fig. A) : - not applicable as such in the synthesis cf fricatives , 30 nor of several other consonant sounds - giving rise to problems of dynamics - causing errors in the amplitude relations even of vowel sounds ; a particular problem being that of find¬ ing a tone balance between front and back vowels. 35 Parallel model (Fig. B) : - a large group of control parameters required - the values of the amplitude parameters are difficult to generate by regular synthesis - the model fails to realize the cascade principle of the sound channel, Combination models (Klatt) (Fig, C) - regarding the parallel and cascade branches, the prob¬ lems are in principle the same as in the equivalent par¬ allel and cascade models, but said branches complement each other so that many of the problems are avoidable thanks to the parallel arrangement of two branches of different type - structural complexity and difficult control of parame¬ ters. LPC synthesis! - the filter parameters are difficult to generate by reg¬ ular synthesis - problems associated with the speech production model employed by LPC synthesis, which impair the quality of the synthetic sound (cf, e.g. D.Y. Wong: On Understand¬ ing the Quality Problems of LPC Speech, ICA SSP 80, Denver, Proc, p. 725-72S). The sound channel models produced by the method of the in- vention are also applicable in speech ara lysis and speech identification, where the estimation of the speech sig¬ nals' features and parameters plays a central role. Such parameters are, for instance, the formant frequen- cies, the formants 1 values, amplitude proportions, sonant/surd quality, and the fundamental frequency of sonant sounds. Usually the Fourier transformation is ap¬ plied to this purpose, or the estimation theory, which is lαiown from the field of control technology in the first place * Linear prediction is one of the estimation methods. The basic idea of estimation theories is that there ex- 1 2 ists an a-priori model of the system which is to be esti¬ mated. The principle of estimation is that when to the model is input a similar signal as to the system whic is to be identified, the output from the model can be made 5 to con orm to the output signal of the system to be iden¬ tified, the better the greater the accuracy with which the model parameters correspond to the system under anal¬ ysis. Therefore it is clear that the results of estima¬ tion obtainable with the aid of the model increase in re- 10 liability with increasing conformity of the model used in estimation to the system that is being identified. The object of the present invention is to provide a new- kind of method for the modelling of speech production. It 15 is possible by applying the method of the invention to create a plurality of terminal analogs which are struc¬ turally different from each other.The internal organisa¬ tion of the models obtainable by the method of the inven¬ tion may vary from pure cascade connection to pure paral- 20 lei connection, also including intermediate forms of these, or so-called mixed type models. In all configura¬ tions, however, the method of the invention furnishes an unambiguous instruction as to how the transfer function of the individual transfer function should be for achieve— 25 ment of the best approximation in view of equation (2). The general aim of the present invention is to attain the objects set forth above and to avoid the dra backs that have been discussed. To this end, the model of the inven- 30 tion is mainly characterized in that the transfer function of said electrical filter system is substantially consist¬ ent with an acoustic transfer function function modelling said sound channel which has been approximated by dividing said transfer function by mathematical means into partial 35 transfer functions with simpler spectral structure, which have been approximated, each one separately, by realizable rational transfer functions , and that to each said ratio¬ nal transfer function separately corresponds an electronic mutually connected in parallel and/or series to the pur¬ pose of obtaining a model of the acoustic sound channel. A further object of the invention is the use of channel models according to the invention in speech analysis and identi ication, the use of channel models according to the invention as estimation models in estimating the par¬ ameters of a speech signal, and the use of the transfer function representing a single, ideal acoustic resonance, obtainable by repeated use of the formula (6) to be pre¬ sented later on, in speech signal analysis, parametration and speech identification. A further object of the invention is a speech synthesizer comprising input means, a microcomputer, a pulse genera¬ tor and • noise generator, a sound channel model and means by which the electrical signals are converted into acoustic signals, and ' in this synthesizer said input means being used to supply to the microcomputer the text to be synthe- sized, and the coded text transmitted by said input means going in the form of series or parallel mode signals throug the microcomputer's intake circuits to its temporary memory and the arithmetic-logical unit of said microcomputer oper¬ ating in a manner prescribed by the program stored in a permanent memory, and in said speech synthesizer the micro¬ computer reading the input text from the intake circuits and storing it in the temporary memory, and in said speech synthesizer, after completed storing of the symbol string to be synthesized, a control synthesis program being start- ed, which analyses the stored text and with the aid of tables and sets of rules forms the control signals for the terminal analog, which consists of the pulse and noise gen¬ erator and the sound channel model. The above-defined speech synthesizer constituting an object of the invention is mainly characterized in that a parallel-series model ac¬ cording to the invention serves as sound channel model in the speech syntliesizer. The invention differs from equivalent methods and models of prior art substantially in that the acoustic transfer function having the form (2) is not approximated as one whole entity, but it is instead first divided by exact procedures into partial transfer functions having a sim¬ pler spectral structure. The actual approximation is only performed after this step. x-oceeding in this way, the method minimizes the approximation error, whereby the transfer functions of the models obtained are no longer in need of any correction factors , not even in inhomogene¬ ous cases. The most appropriate range of application of the method of the invention of which the inventor is aware is found ' in the implementation of mixed type models. In the description of the mixed type models of the invention, which are of a cex'tain kind of parallel-series models , the name PAIICΛS model is used, this being derived from the word combination Parallel ÷ Cascade. The PAHCAS models of the invention are realizable by means of structurally simple filters. In spite of their simplicity, the models of the invention afford a better correspondence and accuracy than heretofore in the rπodel- ling of the acoustic phenomena in the human phonation system. In the invention, one and the same structure is able to model effectively all phenomena associated with human speech, without any remarkable complement of exter¬ nal additional filters or equivalent ancillary structures, The group of control parameters which the P RC S models require is comparatively compact and orthogonal. All par¬ ameters are acoustically-phonetically relevant and easy to generate by regular synthesis principles. As taught by the invention, the PARCA5 models combine the advantages of the series and parallel models, while the drawbacks are eliminated in many aspects. The model of the invention gives detailed instructions as to the required type e.g. of the individual for ant cir¬ cuits F1. , , used in the model of Fig. 1 regarding their filter characteristics to ensure that the overall transfer function of the model approximates as closely as possible the acoustic transfer function of equation (2). The pro¬ cedure of the invention is expressly based on subdividing equation (2) into simpler partial transfer functions which hav fewer resonances , compared with the original transfer function, within the frequency band under consideration. The subdivision into partial transfer functions can be done fully exactly in the case of a homogeneous sound channel. The next step in the procedure consists of ap¬ proximation of the partial transfer functions e.g. by the aid of second order filters. In the following, the invention is described in detail with reference being made to certain embodiment examples of the invention, presented in the figures of the attached drawing, to the details of which the invention is in no way narrowly confined. Fig, 1 presents, in the form of a block diagram, a paral¬ lel-series (PARCAS) model according to the invention. Fig. 2 presents an embodiment of a single formant circuit according to the invention by a combination of transfer functions of low, high and., band-pass filters. Fig. 3 presents, in the form of a block diagram, a speech synthesizer applying a model according to the invention. Fig. shows, in the form of a block diagram, the more de¬ tailed embodiment- of the speech synthesizer of Fig. 3 and the communication between its different units . Fig. shows the more detailed embodiment of a terminal analog based on a PARCAS model according to the invention. Fig. 6 shows an alternative embodiment of the model of the invention. In Figs 7, S, 9» 10, 11 , 12 and 1j are reproduced various amplitude graphs , plotted over time , obtained by computer simulation, serving to illustrate the advantages over prior art gainable by the model of the invention. In Fig. 1 is depicted a typical PARCAS model created as taught by the invention. It is immediately apparent from Fig. 1 that the PARCAS model realizes the cascade princi¬ ple of the sound channel, that is, adjacent formants (the blocks F1...F4) are still in cascade with each other (FI and F2 , F2 and F'3 , 3 and F , and so on). Simultaneously the model of Fig. 1 also implements the property of par¬ allel models that the lower and higher frequency compo¬ nents of the signal can be handled independent of each other with the aid of adjusting the parameters A_ , A w , k„ , lc . This renders possible the parallel formant cir- cuits FI ,F3 and F2,F in the filter elements A and 3. As a result of this structural feature , the PARCAS model of Fig, 1 is suitable to be used in the synthesis not only of sonant sounds, but very well also in that eig. of fricatives - both sonant and surd - as well as transient- type effects. For instance, the fifth formant circuit po¬ tentially required for the s sound may be connected either in parallel with block A in Fig. 1 or in cascade with the whole filter system. The 250 Hz formant circuit required by nasals may also be adjoined to the basic structure in a. number of ways. Thanks to the parallel structures of ■ blocks A and 3 in Fig. 1 , it is possible with the PARCAS model to achieve signal dynamics on a level w-ith the par¬ allel model, and a good signal/noise ratio. For the same reason, the model is also advantageous from ine vievrpoint of purely digital realisations. In the following the analytical foundation of the model of the invention shall be considered in detail- In the transfer function of equation (2) the amplitude co¬ efficient A may be omitted in the subsequent consideration, whereby the transfer function appears in the form (5) H A (ω) S cos x + j a sin x where a is a real coefficient (a < 1 ) depending on the losses of the channel and/or its acoustical load, and x = kω . The expression in equation (5) can be exactly written as tho product of tvo partial functions, as fol¬ lows: (<S) cos x + j a sin x (b cos x_ + j c sin x_) (b cos x. + j c sin whore x_ (x-π/2)/2 (x+π/2)/2 b c (/ " T+a - ■ / l-a)/ " T The partial 'transfer functions of equation (6 ° ) may also be written in the form (7) b* b cos x ± + j cs n x + cos x + + j a' sin x„ ^ where a* = (l-/l-a 2 )/a b' = 1/b - c/a - (/ " I+F- = a)/( T * a) Equations (β) and (7) show that the original transfer function (2) can bo divided into two partial transfer functions, which are in principle of the same type as the original function. However, only every second re¬ sonance of the original function occurs in each partial transfer function. In the analysis just presented, the original acoustic transfer function was divided into two parts. By apply- IS ing the same procedure, again, to the parts , both parts can be further subdivided into partial transfer functions with fewer resonances. In Fig. 7 is graphically presented the original acoustic transfer function H. ( _ ) in the case of 3. = 100 Hz (con¬ stant bandwidths). The function K., „( '* ) reoresents one of 1 the two partial transfer functions obtained by the first division, and H { '- ) represents the transfer function ob- tained by further subdivision of the latter. The partial transfer function H _( '- ) has the sam$ shape as H..-(ω ), with the formant peaks located at the second and fourth formants. The partial transfer functions H.. ( ω ) , H^(ω ) and I (ω ), respectively, are obtained by shifting the H»(ω ) graph along the frequency axis. The original acoustic transfer function cah be divided according to similar principles also into three, four, etc., instead of two, mutually similar partial transfer functions. However, subdivision into two parts is the most practical choice, considering channel models com¬ posed of four formants. wien the equation (β) is once applied to equation (2), the result is a PARCAS structure as shown in Fig. 1. On repeated application of equation (6) on the partial transfer functions R- _ and H ; , the outcome is a model • J — ~ - with pure cascade connection, where the transfer function of every formant circuit is — or should be — of the form PI . It is thus also possible by the modelling method of the invention to create a model with pure cascade connec¬ tion. Differing from prior art, the formants of this new model are closer to tbe band-pass than to the low-pass type. If one succeeds in approximating the transfer f nc- tions of the H_ type with sufficient accuracy, no spec¬ tral-correction extra filters are required in the model. The dynamics of the filter entity have at the same time improved considerably, compared e.g. with the casc model of prior art (Fig. A) . . Generally sponkiπ,'7, the r.Lπoiplo jtu.t dnjic ±boc ] may bo applied to subdivide the acoustic transfer function H. of a homogeneous sound channel according to equation (5) into n partial transfer functions , in which every n-th formant of the original transfer function is present, and by the cascade connection of which exactly the original transfer function H. is reproduced. The following table shows the kinds of partial transfer functions obtained in the spe¬ cial cases n = 2 and n ~ » and in the general case. Table 1 also reveals which formants belong to which partial transfer function. TΛBLt) I n H, H.«, Δ_ ?m , F~, F-,...J "I -- { 4' F δ"" } H 14 Δ { F χ , F 4 , F ? ,...} A * • " 25 - { F 2' F 5* F 8'-" } H-g ._ { ^» , , F Q ,...; General form : H n(2n) - - - n>' ^ * 2„n>' * 3wn' - The equation ( 5 ) is also divisible into two transfer functions , the original function being obtained as their sum. cos x_ + j sin x_ ( S ) 1 cos x + j a sin x b-c b cos x + j c sin cos , + j sin x . b cos x_ + j c sin 20 where x , x. , b ar-d c are as in equation (6). The transfer functions obtained differ from those pre¬ sented in equation (6) only by the phase factors in the numerator. By applying equation (8) first to equation (2) and thereafter to the partial functions which have been obtained, a parallel model is produced, in which the transfer functions of the individual formant circuits have the form H_ . Equation (8) may equally be applied in the division of partial transfer functions H. _ and H_. into parallel elements H- and H 5 . Hereby a more precise picture can be obtained of how the lower and upper form¬ ants shoτild be approximated and how the phase relations should be arranged for the combined transfer function constituting the objective to be produced. It is obvious that it is difficult to find-an accurate, and at the same time simple^ polynomial approximation for a function of the H„ type. he amplitude graph of an acoustic resonance is symmetrical on a ' linear frequency s le , whic is not true for most of the simple transfer functions of second order filters. This accuracy require¬ ment is essential in the pure cascade model, whereas the • oure parallel model is not critical in this respeci Sound channel models obtained by the method of the inven¬ tion may be applied e.g. in speech synthesizers, for in¬ stance in the way shown in Tig. 3. Over the input device 10, the text Cl to be synthesized (coded text), converted into electrical form, is supplied to the microcomputer 11. The part of the- input device 10 may be played either by an alphanumeric keyboard or by a more extensive data pro¬ cessing system. The coded text Cl transmitted by the input device 10 goes in the form of series or parallel mode signals through the input circuits of the microcomputer 11 to its temporary memory (RAM) . From the microcomputer 11 the control signals C2 are obtained, which control both the pulse generator 13 and the noise generator 1 , the latter being connected by interfaces C to the PΛRCΛ3 model 15 of the invention. The output signal Ok from the PARCAS model is an electrical speech signal, which is converted by the loudspeaker 16 to an acoustic signal 05 - The microprocessor 11 consists of a plurality of integrat¬ ed circuits of the kind shown in Fig, k f or of one inte¬ grated circuit comprising the said units. Communication between the units is over data, address and control buses. The arithmetic-logical unit (C.P.U,) of the microcomputer 11 operates in the manner prescribed by the program stored in the permanent memory (ROM). The processor reads from the inputs the text that has been entered and stores it in the temporary memory (RAM). On completed storing of the text to be synthesized, the regular system program starts to run. It analyses the stored text and sets up tables • and, using the set of rules, controls for the terminal analog, which consists of the pulse and noise generator 13*1^ and of the sound channel model 15 o the invention. The more detailed structure of the terminal analog based on the PARCAS model is shown in Fig. 5 • ^ ~ - the case of sonant sounds, the pulse generator 13 operates as main signal source, its frequency of oscillation and ampli- tude Λ0 being separately controllable. In the case of fricative sounds, the noise generator 1 k serves as • source. In the case of sonant fricatives, both signal sources 13 and 1 k are in operation simultaneously. The impulses from the sources are fed into three parallel- connected ilters T 1- -1. F.1. and F„1 _ • ) over amplitude con- trols. The amplitudes of the higher and lower frequen¬ cies in the spectra of both sonant and fricative sounds are separately controllable by the controls V ,VH and FL,?H respectively. The signals obtained from the fil- ters F- - . F.. - and F- m are added up. Either before this summing operation or in its connection, the signal from the filter . ^ is attenuated by the actor lc. and that from filter F.. _ by the factor 1 «• The summed signal ' from filters F ' -..... .. _ is carried to the filters F.. and F 1 .. In parallel, with the filters mentioned has been con¬ nected a nasal resonator N (resonance frequency 250 Hz), of which the output is summed with the signals from fil- ters F 1 and Fm , . while at the same time the signal com¬ ponent that has passed through the filter F.. , ( is attenu¬ ated by the factor lc. - . T< Q other parameters of the ter¬ minal analog include the values of the formants (Q11 , 1 , Q13, Q1 , . The output signal can be made to cor- respond to the desired sounds by suitably controlling the parameters of the terminal analog. The terminal analog of Fig. 5 represents _ one of the reali¬ sations of the PARCAS principle of the invention. The same basic design may be modified e.g. by altering the position of the formant circuits F. 1,-., and . Fig, 6 presents one such variant. It could be established both by computer simulation runs. and in practical laboratory tests that it is possible by the PARCAS model of the invention to atta n in the approx¬ imation of the transfer function a higher accuracy than by any other designs. This is mainly due to the internal structures of the filter elements A and B (Fig. 6). If it is desired, for instance, to construct a pure cascade model of transfer functions of H_ type (Fig. 7), such a transfer function should be approximable accurately within the whole frequency band under consideration. But this is found to bo difficult in practice. In Fig. 2 is illustrated the approximation of E. by means of a low-pass filter LP, a lev/—pass and band-pass filter combination LP/BP and a low-pass and high-pass filter com¬ bination P/HΓ. The said filters can ho realized e.g. by the parameter filter principle shown in Fig. 2 . In the em¬ bodiment example of Fig. S, the low-pass approximation introduces the largest and the LP/H? combination the smallest error. The error of approximation is high a~ the top end of the ' frequency b.-tπcl in all instances. In PARCAS models where the ' transfer functions to be ap- proximated arc of the form FL _ (Fig. 9), it is possible to make the error of approximation very small over a wido band. In Fig. 9 » H. « has een approximated with the par¬ allel connection of LI'/U -ami IIP/llP filters, and it is ob¬ served that,the error IS.,- is exceedingly small on the central frcqtoncy band. Fig.' 10 shows the approximation of II . by low-pass and high-pass filters alono. The error E^. is small on the avoi-agc ore too. Fig. 11 displays the overall transfer function of tho PARCAS model consistent with tho principles of the inven- tion obtained as tho combined result of approximations as in Figs 10, and tho error IS compared with the acoustic transfer function. The coefficients of the model (sec Fig. l) are in this case k.. s -0.2, = O. and A- - ~ A-.» Tho ' said values of the coef icients k. re— present tho case of a neutral vowel. In the inhomogeneous case the said coe fϊcients have to be adjusted consistent with the formants' Q values as follows: If the band widths arc constant, e.g. Bi = 100 11:2, the coefficients may be defined directly from the resonance f equcncics : (10) ^ * F3 • k, = F 2 /F .4 : By adjusting the coefficients k. as indicated by equation (lθ), highcx* accuracy is achieved wi h tho PARCAS model in iho case υf all vowul sounds. I11 Figs 12 and 13, this principle has been followed in simulating the vowels /o/ and /i/, and it is seen that the ei-ror of approximation remains, in these inhomυgenous channel cases, in tho most central frequency rangu significantly smaller than with 2 k tho cascade model (cf. Fi s IS and p) . The example presented above shows that the PARCAS design according to the present invention eliminates marϊy of the 5 cascade model's problems. At the same time the model of the invention is substantially simpler than the -'cascade model of prior art, for instance because it requires no corrective filter, and moreover it is more accurate in cases of inhomogeneous sound channel profiles. ' . 10 As was observed earlier in the introductory part of the disclosure, the invention may also be applied in .connec¬ tion with speech identification. The models created by the method of this invention have been found to be simple 15 and accurate models of the acoustic sound channel. It is therefore obvious that the use of these models is advan¬ tageous also in estimation of the parameters of a speech ' signal. Therefore the use of models produced by the method above described in speech identification, in the process 20 of estimating its parameters, is also within the protec¬ tive scope of this invention. Furthermore, by using the formula (6) repeatedly (without limit) , the trans er function representing one single 25 (ideal) acoustic resonance can be produced, This transfer function too, and its polynomial approximation, has its uses in the estimation of a speech signal's parameters, in the first place of its formant frequencies. The form¬ ant -frequencies are effectively identifiable by applying 30 the said ideal resonance to the spectrum of a speech signal. Therefore the use of said ideal formant in speech signal analysis is also within the protective scope of th .3 invention. 35 In the following are stated the claims, different details of the invention being allowed to vary within the scope of the inventive idea defined by these claims. BAD ORIGINAL

Claims: A (received by the International Bureau on 25 June 1982 (25.06.82)) Original claims 1. to 14 . ( de leted) 15.Model of the acoustic sound channel associated with the human phonation system and/o music instruments and which has been realized by means of an electrical filter system, characterized in that the transfer function of said electrical filter system is substantially consistent with a acoustic transfer function mαdelling said sound channel which has been approximated by ing the acoustic transfer function of a homogeneous sound channel* conforming to the eq (5) stated below into two or more (n) partial transfer functions H.., in which only every n-th formant of t original transfer function is still present (Table 1)» the model of the sound channel is consistent with that model which is obtainable by app mating said partial transfer functions H.. with realizable "rational transfer functions, to each of which separately corresponds an electronic filter in the electrical filter system. said filters have been connected mutually both in parallel and in series in the way impli the acσustic sound channel, and said connection of the filters has been so arranged that formant circuits (FI and F25 F2 F3 > F3 and F4| ..«) which are mutually adjacent on the frequency scale are in cascade with each other. 16 . Model of an acoustic sound channel according to claim 1 , characterized in that in the summing of the output amplitudes of the formant circuits connected in parallel their wei factors are constant. 17 . Parallel-series model αf order n = 2 according tα claim 1 or 2, characterized in that th transfer functions H. ^ and H, . αf the electrical filter system have been approximated b low-pass filter ( P), a low-pass and band-pass filter combination (LP/SF) and a low-pas high-pass filter combination ( F/HP) (Figs 2, 10 and 11). 18 . Parallel-series model according to- claim 3, characterized in that the k coefficients (F 1 ) have been selected in accordance with equation (9,10) as follows: k. = 0.3/2.3 and b - B , / ~ ~ J> the model's different branches has also been provided the difference of the signals sα that zeros, αr anti-resonances, are produced in the transfer function. 20 . Parallel-series model according to claim 3, characterized in that the amplitudes αf the sienals entering - the filter element- (H i.].) are controlled indeo • endent of each other ( A X, and 21. The use of a sound channel model according to claim 1 , 2, 3, 4, 5 or ό in speech identification. 22 . The use αf a sound channel model according to claim 1, 2, 3, 4, 5 or 6 as estimation mode in estimating the parameters αf a speech signal. 23 . The use of a sound channel model according to claim 1, 2, 3» 4, 5 αr 6 as sound channel model (15) of a speech synthesizer. 24". Speech synthesizer comprising input means (10), a microcomputer (i l), a pulse generato (13 and noise generator (14), a sound channel model (15) and means (16) by which the elec¬ trical signals are converted to acoustic signals, and wherein by mediation of said input device (10) is given to the microcomputer (11) the text tα be synthesized ( C and the coded text (C. ) entered by said input device (10) goes in the form αf series αr parallel mode sig¬ nals through said microcomputer's (10) input circuits to its temporary memory (RAH) and th arithmetic-logic unit (CPU) of said microcomputer (i i) operates in the manner prescribed b the program stored in a permanent memory (ROM), and wherein the microcomputer reads the entered from the input circuits and stores it in a temporary memory (RAM) and wherein aft completed storing of the symbol string to be synthesized a control synthesis program is st ed, which analyses the stored text and sets up tables and, by using sets of rules, the contr signals (C,) for a terminal analog (13 14,15), consisting of a pulse and noise generator ( 13,14) and of a sound channel model, characterized in that said sound channel model consis cf an electrical filter system αf which the transfer function is substantially consistent wit an acoustic transfer function modelling said sound channel which has been approximated by dividing the acoustic transfer function αf a homogeneous sound channel, conforming tα the equation (5) stated belαw into two or more (π) αartial transfer functions H... in which αnlv every n-th formant αf the i] - 7 original transfer function is still present (Table 1 ), the. odel of the sound channel is consistent with that model which is obtainable by approx ' __ said filters have been connected mutually both in parallel and in series in the way implie the acoustic sound channel, and said connection of the filters has been so arranged that formant circuits (FI and F2J F2 F3; F3 and F4? ...) which are mutually adjacent on the frequency scale are in cascade with each other. 25 . Speech synthesizer according to claim 10, characterized in that as signal source in th case of sonant sounds has been arranged to act mainly the pulse generator (14), its frequ of oscillation (F0) and pulse amplitude (AjΣO being separately controlled, and that as sαur for fricative sounds has been arranged to act mainly the noise generator (14), and that in case of sonant fricatives both signal sources (13,14) have been arranged to operate simultaneously. 26 . Speech synthesizer according to claim 11, characterized in that the impulses obtained from said signal sources (13,14) are supplied to three parallel-connected filters (F. . » F. and F 3 ) aver amplitude controls (VL,VH,FL,FE)i that the signals from said filters ( F^, and F. are summed (S3, that either before or after said summing the signal from one αf filters (F. ) is attenuated by a given factor (k, , ), that the signal from a second of said filters ( F. g ) is attenuated by another factor «(k,. ), that the summed signal from said filte (F 11 r-.mr . m.) is conducted to second filters (F, , and F 4 .), and that in parallel with the abov mentioned filters (F. . ...F. has been connected a nasal resonator (N)» its output being summed with the signals from the latter filters (F. ,, and F. J, while at the same time the l— l τ signal component that has passed through one αf the two last-mentioned filters (F .) is attenuated by a given factor (k. ^). 27 . Speech synthesizer according to claim 12, characterized in that as other parameters o said terminal analog are used the formants' Q values ( GL ^G^ , Q 3' Q i ,C W' aπd * hat al - parameters of the terminal analog are so controlled that the output signal αf the termina analog is made consistent, with sufficient accuracy, with the sounds to be synthesized in instance. 3 . ^ Filter circuit for the modelling of a sound channel, consisting of resonance filters ( F^F^F^F connected together, characterized in that the filter circuit consists αr π filter units ( A,S) comprising connected in parallel, corresponding to every n-th resonance the original transfer function to be modelled, one filter so that filters cor- responding to resonances which are mutually adjacent an the frequency scale (F 1 ,F..;F.- / ,F ^ .; l 1 — — m/ are physically incorporated in different filter units, and that said filter units (A,Ξ) have

Inventors: LAINE UNTO
Applicants: EUROKA OY, LAINE UNTO
Application year: 1981
Main classification: G10L1
IPC codes: G10H1, G10L13, G10L15, H03H11