METHOD OF AND DEVICE FOR SPEECH SIGNAL CODING AND DECODING BY SUBBAND ANALYSIS AND VECTOR QUANTIZATION WITH DYNAMIC BIT ALLOCATION

Abstract: According to the coding method, the digital speech signal is split into subbands; each subband is coded independently of the oth­ ers. The first two subbands undergo an adaptive linear prediction operation yielding a residual, normalized with respect to a quantized r.m.s. value; the residual samples are quantized and coded by a varia­ ble bit number. As to the third subband the samples normalized with respect to a quantized gain term are directly quantized and coded by a variable bit number. While splitting the subband signals into sample block, for the first two subbands the coefficient vectors of subband linear pre­ diction filters relevant to each block are also determined by vector quantisation and linear prediction inverse filtering operations; quan­ tized r.m.s. values are also determined to calculate the numbers of bits permitting subband signals coding. The coded signal consists of subband signal quantization level indices, of r.m.s. value quantiza­ tion level indices, and of linear prediction filter coefficient vector indices. A coding device is also described (Fig. 1).

Description: The present invention concerns adaptive-predictive speech-­ signal coding systems, and more particularly a method of and a device for speech signal coding and decoding by subband analysis and vector quantization with dynamic bit allocation. While implementing speech signal coding systems for medium bit-rate transmission (e.g. 16 kbit/s), the main problem to resolve is that of the quality of representation of the signal by exploiting the whole knowledge of production and perception speech-signal models, max­ imizing reproduction quality versus the wished bit rate. From that point of view the systems which analyse total-band speech signal do not optimally exploit its characteristics, since the spectrum is non-uniform, i.e. its characteristics are considerably different in function of the frequency region considered. Then, by previously splitting the speech signal spectrum into independently-analyzed subbands, a piece-wise approximation is obtain­ ed; i.e. a more accurate shaping of the full spectrum, rendering the coding method more flexible and efficient. A speech-signal coding system by a band splitting is describ­ ed in the paper by R.S. Cheung, S.Y. Kwon: "The design of a 16 kbit/s split-band adaptive predictive coder for noisy channels", Proceedings of International Conference on Acoustics, Speech and Signal Proces­ sing, Atlanta, 1981, pp. 631-635. According to this method the speech signal is split into two subbands. Each subband is quantized by using a method of adaptive li­ near prediction and dynamic bit allocation providing also the calcula­ tion of additional parameters such as spectral parameters which are scalarly quantized. The sampling frequency used is 6.5 kHz. The splitting into two subbands is insufficient to approxima­ te the signal spectrum as well as possible. More particularly, subband quantizers determination results hardly accurate and flexible, and a dynamic bit allocation over two subbands renders it difficult to follow with sufficient accuracy energy variations inside the spec­ trum. Besides, the scalar quantization of spectral parameters en­ tails the transmission of a considerable number of bits devoted to ad­ ditional information, with consequent reduction of those devoted to subband quantization; it furthermore reduces acoustic background noise immunity, since spectral parameters are calculated on the input-signal spectrum which can be noise-affected; which noise alters also quanti­ zed parameters. In addition, the use of predictive algorithms for hi­ gher speech-band frequencies is hardly efficient owing to the scanty correlation existing between said subband samples. Finally, the used sampling frequency entails the additional insertion of apparatus to allow interfacing with standard transmission devices, e.g. on telepho­ ne network operating at 8 kHz frequency. These problems are resolved by the present invention of a speech-signal coding method and of a variant providing voice-band splitting into 4 subbands; this splitting is the best compromise bet­ ween speech-spectrum approximation, optimal dynamic bit allocation ex­ ploitation, and computing complexity. The first two subbands (frequencies up to 2 kHz) are quanti­ zed by an adaptive linear predictive (ALP) method whose spectral para­ meters are vectorially quantized; in this way the number of coding bits devoted to said parameters is considerably reduced, thus allowing an increase in those devoted to subband quantization; in addition, noise-immunity is increased, since spectral parameters are selected out of a parameter-vectors codebook determined, once for all, in the conditions of nearly-total noise absence. The third subband (frequencies ranging between 2 and 3 kHz) is quantized with a non-predictive method, thus avoiding computations which would result useless, owing to the scanty correlation existing between said subband samples. In a preferred embodiment of the method, to be applied e.g. in transmission systems (such as telephone transmission) in which bit-­ rate is medium-low, and the number of bits to be transmitted is to be reduced as much as possible, the fourth subband (frequencies beyond 3 kHz) is not to be taken into account during the coding process: since sound energy contribution at these frequencies is extremely low, its exclusion entails just a slight quality degradation of the reproduced speech-signal, to the great advantage of a reduction of calculations to be executed and an increase in the number of coding bits available for the other subbands, bit rate remaining equal. In an alternative embodiment of the method, to be applied whenever desired, to reproduce the speech signal in each frequency component, the fourth subband undergoes a coding similar to the third subband. Besides, the sampling frequency used is 8 kHz, which value permits an easy interfacing of the coding device with standard devices of telephone-transmission on voice band channels. It is a particular object of the present invention, the meth­ od for speech signal coding and decoding as in claims 1 and 2, and a variant as in claim 3. The invention also extends to the method of generating sub­ band quantizers as in claim 4. There are also provided by the present invention an apparatus for coding/decoding the speech signal as in claims 5 to 14 and a va­ riant as in claim 15. The preferred embodiment of the coding and decoding method provided by the invention will be now described. The coding method can be subdivided into three main steps as follows: A) subband splitting of the speech signal converted into digital; B) calculation of parameters to be used in the subsequent quantiza­ tion step; C) independent quantization of each subband with dynamic bit alloca­ tion: the coding bit number is different for each subband and pe­ riodically updated. Each of said main steps will be hereinafter described, with reference to an application to a low bit-rate transmission system: A) The speech signal converted into digital is filtered by a bank of FIR (Finite Impulse Response) digital filters of QMF (Quadrature Mirror Filter) type according to known techniques described e.g. in the paper by C.R. Galand, H.J. Nussbanner "New Quadrature Mir­ ror Filter Structures", IEEE Transactions on Acoustics, Speech and Signal Processing, Vol. ASSP-32, No. 3, pp. 522-531, June 1984. The bank of filters has a tree structure composed of pairs of ca­ scaded filters, wherein the filters of the pairs have specular transfer functions with respect to transition frequencies. By said filtering the digital speech signal is split into four equal-width subbands, hereinafter denoted by X₁, X₂, X₃, X₄ with an increasing frequency order. Considering in a non limiting example a speech signal band ranging between 0 and 4 kHz, the following subdivision will be obtained for the bands: X₁ = 0÷ 1 kHz; X₂ = 1÷2 kHz; X₃ = 2÷3 kHz; X₄ = 3÷4 kHz. B) The digital samples of subband X₁ are split into blocks X₁(j), where j is the index of a sample in the block (1≦j≦J). The blocks of digital samples X(j) are then filtered in known man­ ner using linear prediction inverse filtering, i.e. PCL inverse filtering, the transfer function H(z), in the Z transform, being in a non-limiting example: where z⁻¹ represents a delay of one sampling interval; a₁(i) is a vector of linear prediction coefficients (O≦i≦L); L is the filter order and also the size of vector a₁(i), a₁( O ) being equal to 1. Coefficient vector a₁(i) must be determined for each block of digital sample X(j) with an updating frequency equal to J subband samples. Said vector is selected from a codebook of vectors of quantized linear prediction coefficients a 1h (i), where h is the vector index in the codebook (1≦h≦H). The selected vector allows the optimal inverse filter to be built up for each block of samples X₁(j); the selected vector index will be hereinafter denoted by h 1ott (J). The technique used to generate the codebook of vectors of quantized linear prediction coefficients a 1h (i) is the known vector quantization technique involving measurement and mini­ mization of the spectral distance d LR (likelihood ratio measure) between normalized gain linear prediction filters described for instance in the paper by B.H. Juang, D.Y. Wong, A.H. Gray, "Distortion Performance of Vector Quantiza­ tion for LPC Voice Coding", IEEE Transac tions on ASSP, vol. 30, n. 2, pp. 294-303, April 1982. The same technique is also used to choose the coefficient vector a 1h (i) in the codebook. This coefficient vector a 1h (i), which allows the building of the optimal LPC inverse filter is that which allows the minimiza­ tion of the spectral distance d LR (h) derived from the rela­ tionship: where C 1X (i), C 1a (i, h) C* 1a (i) are the autocorrelation coeffi­ cient vectors respectively of blocks of digital samples X₁(j), of coefficients a 1h (i) of generic LPC filter of the codebook, and of filter coefficients calculated by using current samples X₁(j). Minimization of the distance d LR (h) is equivalent to finding the minimum value of the numerator of the fraction in (2), since the denominator depends solely on the samples X₁(j). Vectors C 1X (i) are computed starting from the input samples X₁(j) of each block, after weighting according to the known Hamming curve over a length of F samples and with superposition between consecutive windows such that the F consecutive samples are cen­ tred around the J samples of each block. Vector C 1X (i) is given by the relationship: Vectors C 1a (i,h) on the other hand are extracted from a correspon­ ding codebook in one-to-one correspondence with that of vec­ tors a 1h (i). Vectors C 1a (i,h) are derived from the following relation­ ship: For each value h, the numerator of the fraction present in relationship (2) is calculated using relationships (3) and (4); the index h 1ott supplying the minimum value of d LR (h) is used to choose vector a 1h (i) from the relevant code­ book. As a result of inverse LPC filtering a residual signal sample R₁ is obtained for each block of subband X₁. The sequence of R₁ samples is divided into blocks R₁(k), with k sample index in the block (1≦k≦K); it is to be noted that in gene­ ral it results K‡J. In the non limiting example described here the following values are used: J=40, K=20, corresponding to time periods of 20 and 10 ms respectively if the above described subband splitting is used. Each block R₁(k) undergoes an estimate of its r.m.s. value σ₁(κ) according to the following relationship: where β₁ is a constant experimentally determined so as to maximize the correlation between the actual r.m.s. value and the estimated one. The operations of inverse LPC filtering and estimate of resi­ dual signal R₁ are made necessary by the different upda­ ting period of coefficients a 1h (i) and r.m.s. value σ₁(κ) respectively updated every J and K digital samples of sub­ band Y. In fact it is known that the numerator of relationship (2) supplying spectral distance d LR (h) is also the measurement of the energy associated with the residual signal obtained by filtering the samples of subband X₁ by using coeffi­ cients a 1h (i). Then, if it were J=K, the minimum value of numerator of d LR (h) would correspond also to the energy of the considered block of samples X₁, and hence the r.m.s. value calculation would be imme­ diate, as dependant on a simple relation with the energy va­ lue. The deriving complexity increase is justified by improved perfor­ mances in subband quantization operations (which will be described hereinafter) obtained by using a 10 ms updating period for the r.m.s. values σ₁(κ), instead of 20 ms (used for the vectors of coefficients a 1h (i)). On the other end , if a 10 ms period were used also for the updating of coefficients a 1h (i), coded speech signal transmission rate also ought to be increased, owing to the presence of a higher data number to be transmit­ ted. The digital samples of subband X₂ undergo the same operations de­ scribed up to now for X₁, thus obtaining optimal vector indices h 2ott (J) and r.m.s. values σ₂(κ). As to the digital samples of subband X₃, only the esti­ mate of r.m.s. value σ₃(κ) is effected directly on them, without carrying out prediction operations. In this case samples X₃ are split into blocks X₃(k) and r.m.s. va­ lue σ₃(κ) is given by the following relationship derived from (5): The samples of subband X₄ are not considered since their ener­ gy contribution is extremely low and their exclusion does not entail relevant errors in the decoded reconstructed si­ gnal. The terns of r.m.s. values σ y (κ), with y=1, 2, 4 subband index, under a quantization operation. A scalar quantization separately performed on each one of three r.m.s. components is preferred instead of a vector quantization on the r.m.s. tern which would be justified by the high degree of correlation between them. In fact since quantized-value indices are to be transmitted, quan­ tization and hence line transmission of the three components one at a time offers higher reliability against possible channel errors: in case of scalar quantization in fact, a channel er­ ror altering a component does not affect the value of the others, while in case of vector quantization the whole tern is altered. Besides scalar quantization increases transmission rate, but it is less complex than vector quantization and requires fewer opera­ tions. Quantized r.m.s. values y (κ) are obtained from scalar quantiza­ tion. Indices s y (K) of the levels selected in quantizers are sent onto the line. The determination of the number of subband coding bits re­ quired for the operations of point C) is carried out at this point and is made to depend on the relative energy of three subbands Xy: it is carried out at block frequency (every K samples), so as to minimize the total mean dis­ tortion D(K) introduced with quantization and given by the following relation: where σ y ²(K) is the square of quantized r.m.s. value, i.e. the energy associated with the signal of subband X y ; d(y,b γ ) re­ presents the means square error relating to an optimal quanti­ zer with variance equal to 1; b γ represents the generic number of bits assigned to the quantization of the y-th subband, wherefrom number 2 b y of levels of the quantizer to be used in the subsequent step C) is obtained, wherein, as it will be seen, the signals of subbands X y are quantized with a variable bit number. Hence relation (7) is calculated for all the possible combinations of terns of numbers of bits b y . It is to be noted that the total number NT, given by the following relation, of subband encoding bits is constant and is a function of the desired transmission ra­ te on line: Hence tern combination number b y is limited. The tern which supplies minimum distorsion D(K) is selected as op­ timal and supplies the optimum number of encoding bits B y (k) for each subband X y to be used every K subband samples. C) The sample blocks of subbands X₁, X₂ submitted to quantization with a known adaptive predictive coding method whose main steps will be described hereinafter. From the samples of subbands X y(m) (now it is y=1, 2 and m is the sample index) there are extracted the corresponding sam­ ples of residual signal E y (m) in a way to be described hereinafter. Residual E y (m) is divided by the previously-calculated correspon­ ding quantized r.m.s. value (value y (K) changes at a frequency of K samples of X y (m)), thus obtaining a normalized residual EN y (m) which is quantized by using a quantizer out of a set of quantizers having a different number of levels. A set of quantizers is used for each subband. The previously-determined number of bits B y (κ) selects, every K subband samples, the set quantizer which has then 2 B y levels. Index e y (m) of the level selected in the quantizer is sent onto the line. Quantized normalized residuals N y (m) are obtained as a quantiza­ tion effect. Residuals E y (m) are generated as follows: quantized normalized residuals N y (m) are multiplied by the corresponding quantized r.m.s. values already used, y (k), thus obtaining quantized re­ siduals y (m) whereupon the so-called "backward" prediction operation is performed to generate a predicted signal P y (m-1) starting from the reconstructed signal y (m) according to the relation: y (m) = y (m) + P y (m-1) (8) Then a linear filtering operation P y (m) is carried out of the following type: where m is the sample index; a yhott (i) is the i-th component of the linear prediction coefficient vector selected out of the cor­ responding codebook as already seen at point B). Then residuals E y (m) are obtained by the following difference: E y (m) = X y (m) - P y (m-1) (10) However, the samples of subband X₃ are submitted to non-­ predictive quantizazion; samples X₃(m), m il still the sample index, are directly divided by the corresponding-­ quantized r.m.s. value ₃(κ) thus obtaining normalized va­ lues EN₃(m), which are quantized by using also in this ca­ se a quantizer out of a set of quantizers having a diffe­ rent number of levels and thus obtaining quantized normali­ zed values N₃(m). The previously-determined number of coding bits B₃(K) is used to select the set quantizer. Index e₃(m) of the level detected in the quantizer is sent onto the line. Hence, on line there are transmitted: - indices h yott (J) (with y=1, 2) of linear prediction coef­ ficient vectors a yh (i) selected in the pertaining code­ books of subbands X y , with an updating period of J subband samples; - indices s y (κ) (with y=1, 2, 3) of quantized r.m.s. values y (κ) selected in the corresponding quantizers, with updating period equal to K subband samples; - indices e y (m) (with y=1, 2) of quantized normalized residuals N y (m), and indices e₃(m) of quantized normalized values N₃(m) identified in the quantizers selected in the relevant set at a subband sample frequency. The technique of generation of quantizer sets used for sub­ bands X₁, X₂, X₃ is now going to be described. Subband sample training sequences are generated starting from a pre-selected speech signal training sequence. As to subbands X₁, X₂, the codebooks of linear-prediction coefficient vectors a yh (i) are first generated, as mentioned at point B) of the method, the optimal vectors for the training sequence are then determined. For each number of coding bits b y and for each one of sub­ bands X₁, X₂ the following operations are executed: 1) 2 b quantizer levels are initialized by optimal values for a Laplacian-distribution speech signal with unit variance and null mean value known in the literature; 2) the adaptive predictive algorithm also used during coding and de­ scribed at point C) of the method is applied to the corresponding training sequence, to determine normalized residual values EN y (m), by using optimal linear-predictive coefficient vectors a yhott (i) of the training sequence; 3) 2 b subsets (one for each quantizer level) of normalized residuals EN y (m), obtained at point 2), are generated, by including in each subset the values having minimum distance from the same quantizer level; the minimum distance d WF is calculated by looking for the minimum of values: d WF = y (K)² · [EN y (m) - N y (m)]² (11) where N y (m) is the generic quantizer level. Thus 2 b distance va­ lues will be computed for each EN y (m), and then compared with each other; 4) the centroid of each subset is computed and taken as a new quantizer level. The method described is repeated till the new levels are no longer basically different from the preceding ones. Generally two ite­ rations are sufficient. For generating the set of quantizers used for subband X₃, from the training sequence a corresponding sequence of normalized va­ lues EN₃(m) is obtained; for each quantizer, i.e. for each value b₃, 2 b ³ subsets are identified as at the preceding point 3); the subset centroids are computed and, without carrying out further iterations, are taken as definite values of quantizer thresholds. The decoding method provides the following operations. By using received indices s y (K) the corresponding quantized r.m.s. values y (K) are identified with operations equal to those com­ puted during coding. Hence the terns of optimal coding bit numbers B y (K) are iden­ tified with a method equal to that described for coding. By using received indices e y (m) the corresponding quantized normalized residuals N y (m) are identified in the quantizers chosen through B y (K) values in the sets of quantizers already used during co­ ding. Values N y (m) of subbands X₁, X₂ are multiplied by the corresponding quantized r.m.s. values y (K) thus obtaining quantized residuals y (m) as during coding, and on the latter the same "backward" linear predictive adaptive operation described at point C) of the coding method [computation of equations (8), (9)] is executed to generate reconstructed signals y (m) of subbands X₁ and X₂. The values N y (m) of subband X₃ are multiplied by the corre­ sponding quantized r.m.s. values y (K), thus directly obtaining the samples of reconstructed signal ₃(m) for subband X₃. The samples of the three subbands of the reconstructed signal y (m) are then filtered by a QMF filter bank, dual of that used in step A) of the coding method, to generate the reconstructed digital speech signal which, converted into analog, gives rise to the recon­ structed speech signal during decoding. The method variant takes also into account the samples of the fourth subband X₄, which are submitted during coding to operations si­ milar to those envisaged for the third subband X₃. R.m.s. values σ₄(K) are obtained from the samples of subband X₄ in accordance with relation (6) wherein it is enough to replace subscript 3 by subscript 4. By scalar quantization operations, quantized r.m.s. values ₄(K) are obtained with corresponding quantization level indices s₄(K). Total mean distorsion D(k) to be minimized defined in (7) is now to be extended to the four subbands, thus obtaining sets of four numbers of optimal coding bits B y (K) with y = 1, 2, 3, 4. From subband samples, there are then obtained, utilizing a set of quantizers of the fourth subband, quantized normalized values N₄(m) and indices of quantization level e₄(m). With respect to the preferred embodiment, by that variant ad­ ditional coding information is generated consisting of level indices s₄(K) of quantized r.m.s. values, and of level indices e₄(m) of normal­ ized quantized values. The set of quantizers of the fourth subband X₄ is generated in accordance with what described for the third subband. Samples of reconstructed signal of the fourth subband ₄(m) are obtained during decoding, which samples, filtered by a QMF filter bank, contribute to generate the reconstructed speech signal. An example of device object of the invention is now descri­ bed, which implements the preferred embodiment of the method and is given by way of a non limiting example, with reference to the annexed drawing in which: - Fig. 1 shows a general block diagram of the device; - Fig. 2 shows a circuit diagram of block PAR of Fig. 1; - Fig. 3 shows a circuit diagram of block QNT of Fig. 1. In Fig. 1 QMF denotes a block consisting of a bank of FIR di­ gital filters of QMF type to filter the input digital signal IT at the coding side, in accordance with what described at point A) of the method description. The bank of QMF filters has a two-level cascaded structure; the first level consists of filters using 32 coefficients to split the input signal band into two subbands, while the second level consists of filters using 16 coefficients for the further splitting into the four subbands X₁,...,X₄. The coefficients used are e.g. those quoted in the paper by J.D. Johnston "A filter family designed for use in Quadrature Mirror Filter banks", Proceedings IEEE International Conference on ASSP, April 1980, pp. 291-294. The choice made on the number of filter coefficients permits the best compromise between computational complexity and out-of-band signal attenuation. Input signal IT is obtained from the analog speech signal by band limitation to 3,4 kHz and sampling at frequency f c =8 kHz with known type circuits not represented in the Figure. Subband signals outgoing on the respective buses of the pa­ rallel type X₁, X₂, X₃, X₄ have a bit-rate of 2kbit/s determined by a suitable subsampling frequency. Subband signal X₄ is not considered by the coder. QMFD denotes a block comprising a second filter bank having a structure exactly dual to that of block QMF. QMFD block issues digital signal UR reconstructed during de­ coding by recombining the reconstructed digital signals X₁,X₂,X₃ of the three subbands present on buses XR1, XR2, XR3 arriving from block QNT; the signal of subband X₄ is simulated by a sequence of logic 0's. Signal UR is then converted into analog by an ordinary digital-to-­ analog converter, non shown in the Figure, to obtain the reconstructed speech signal. Implementation of blocks QMF and QMFD is no problem to skil­ led in the art starting from what is known in the art and from the da­ ta supplied by the present description. PAR denotes a block to compute the parameters necessary to coding-decoding. More particularly PAR implements the functions described at point B) of the coding method and computes; every 20 ms indices h yott (J) and corresponding coefficient vectors a yhott (i) of linear predictive inverse filters for subbands X₁, X₂; every 10ms, indices of levels s y (K) and the corresponding quantized r.m.s. values y (K), as well as the numbers of coding bits B y (K) for subbands X₁, X₂, X₃. PAR computes also coefficient vectors a yhott (i), quantized r.m.s. values y (K), and the number of coding bits B y (K) necessary to decoding phase. PAR receives from QMF subband signals on buses X₁, X₂, X₃, it re-emits duly delayed on buses XD1, XD2, XD3 towards block QNT. PAR receives from block DMX indices s y (K), h yott (J) for decoding phase on buses SR and HR, and emits on buses ST and HT towards block MPX indi­ ces s y (K), h yott (J) for coding phase; it emits on buses BLC, S, A, to­ wards block QNT the corresponding values B y (K), y (K), a yhott (i) for coding and decoding phases. Block PAR structure will be examined in details in connection with Fig. 2. QNT denotes a block to carry out direct quantization of sub­ band signals described at point C) of the coding method, and inverse quantization to obtain subband signals reconstructed during decoding. QNT receives subband signals on buses XD1, XD2, XD3 from block PAR and emits on bus ET towards block MPX the corresponding in­ dices e y (m) during coding; it receives indices e y (m) from block DMX on bus ER and emits on buses XR1, XR2, XR3 towards block QMFD subband si­ gnals, X₁, X₂, X₃, reconstructed during decoding. In addition QNT receives from block PAR values a yott (i), (K), B y (K) on corresponding buses A, S, BLC for both phases. The structure of block QNT will be examined in detail in con­ nection with Fig. 3. MPX denotes an ordinary multiplexer generating the coded di­ gital signal emitted on output bus UT. MPX sequentially makes up the signals present on input buses ST, HT, ET and generates coded signal frames on bus UT which frames can be sent to a line coder, of any known type, inserting suitable frame synchronism bits. Said frames can undergo a modulating operation in order to be transmitted on translated band lines. DMX denotes an ordinary demultiplexer carrying out a function dual to that of multiplexer MPX; DMX splits coded-signal frames arri­ ving, through the input bus on the decoding side IR, e.g. from a line decoder dual to the decoder provided at MPX output. MPX and DMX are activated through signal T/R during coding and decoding respectively. BT denotes a block acting as a time base, supplying the whole device with control and synchronization signals. BT can be controlled by the signal at sampling frequency f c and, in addition to usual synchromism signals which will be no longer mentioned for the sake of brevity, emits the signals on buses CKJ, CKL, CKK, CKV, CKR, towards block PAR, and on bus CKM towards block QNT; it emits also signal T/R controlling the sequence of coding and decoding steps. The implemention of block BT is no problem to the skilled in the art, and can be deduced from the operation of the device de­ scribed hereinafter. In Fig. 2 CHA1 denotes a block, whose structure is surrounded by the dotted line, determining linear prediction coefficient vectors a 1hott (i) and corresponding h 1ott (H) for subband X₁ during coding and decoding, and also residual signal R₁(m). CHA1 comprises circuit blocks BF1, LPCF1, CXH, MX1, VOCA, D1, IND1. BF1 denotes a random-access memory organized into two equal and interchangeable units to temporarily store and group into blocks the samples of subband X₁ it sequentially receives on bus X₁. Each BF1 unit has F=60 sample capacity. Every block of J sub­ band samples, a unit of BF1 temporarily stores the last 10 samples of the preceding block, the 40 samples of the present block and the first 10 samples of the subsequent block; which is necessary for the subse­ quent weighting operation of blocks of samples according to the over­ lapping technique between subsequent blocks, already mentioned at point B) of the method description. At the same time the other unit of BF1, containing the sam­ ples of the preceding period, is sequentially read through a high speed addressing supplied by block IND1 on bus 28, and supplies all the F samples on bus F1 to block CXH, and then on bus 26 and after­ wards on XD1 supplies central J samples of the preceding block. At the subsequent block of J samples the two units are mu­ tually exchanged. Then sample writing in BF1 memory units takes place synchro­ nous with subband bit rate, and is controlled by suitable synchronism signal present on bus CKJ arriving from time base BT of Fig. 1. Reading in BF1 takes place, on the contrary, at higher bit rate to allow downstream circuits to carry out corresponding real-time operations. block IND1 comprises a counter, synchronized by fast synchro­ nism signal arriving, on wire CKL, from time base BT of Fig. 1, sup­ plying BF1 reading addresses on bus 28. IND1 is activated during co­ ding phases by signal on wire T/R coming from BT. CXH denotes a block computing indices h 1ott (J) of coefficient vectors a 1hott (i). CXH sequentially receives, through data bus F1, F samples stored in BF1. CXH computes autocorrelation coefficient vectors C 1X (i) of sample blocks X₁ in accordance with relation (3); using then auto­ correlation coefficient vectors C 1a (i,h) of coefficients a 1h (i) belon­ ging to a codebook contained in one of its internal memories , CXH calculates for each of them the numerator of relation (2), compares with each other the results obtained to find the minimum supplying the desired index h 1ott (J); said index is sent onto bus HT1. The implementation of block CXH is no problem to the skilled in the art, and can be derived from what stated at point B) of the method description, with reference to what known from the art and de­ scribed e.g. in the above-cited paper by B.H. Juang and al. Bus HT1 is connected to an input of multiplexer MX1 to whose second input bus HR1 is connected, whereupon indices h 1ott (J) are present for the decoding step arriving from block DMX of Fig. 1. MX1 connects one of the two input buses to output bus 21 on the basis of the logic level of the signal on wire T/R applied to con­ trol input. Indices h lott (J) of coding and decoding phases will then be present on bus 21 and applied as addresses to ROM memory VOCA. VOCA contains the codebook of linear prediction coefficient vectors a lh (i). In the example described here the codebook for subband X₁ contains H=128 vectors of L=6 component each: hence indices h₁(J) will have 7-bit size. Coefficient vectors a 1hott (i) will be present on VOCA data output on bus 27 both for the coding and decoding phase. D1 denotes a known structure block performing as a delay line for coefficient vectors present on bus 27, supplied on output bus A1. The delay is introduced only during coding to take into ac­ count the time spent by block CSS1 to carry out its operations, so as to render data emission on buses A1, ST1, S1, XD1 synchronous. The si­ gnal on wire T/R controls a delay introduced by D1. LPCF1 denotes a known structure block performing as LPC in­ verse filter according to relation (1) using the coefficient vectors of coding-step present on bus 27. LPCF1 sequentially receives the samples of subband X₁ on bus 26, and for each of them it determines a sample of residual signal R₁(m), which it sends to the output on parallel bus R1. LPCF1 is acti­ vated during coding phase by the signal on wire T/R. CSS1 denotes a block, whose structure is surrounded by a da­ shed line, determining quantized r.m.s. values ₁(K) during coding and decoding and indices s₁(K) for the coding alone. CSS1 consists of cirucit blocks VEF1, CFR1, MX3, CT1, MX2, MQ1, BF2. VEF1 is a block calculating r.m.s. value σ₁(K) of sets of K consecutive samples of residual signal R₁(m) received on bus R1, in accordance with relation (5). VEF1 can be e.g. implemented by an adder-accumulater which, at a frequency of K samples of R₁(m), determined by synchronism signal CKK it receives from time base BT of Fig. 1, supplies the sum of the moduli of the samples present on bus R1 to a multiplier which multi­ plies it by constant β₁: the multiplier output on bus 22 carries r.m.s values σ₁(K). MQ1 denotes a ROM memory containing the levels of the quanti­ zer used to quantize r.m.s. values, i.e. all possible quantized r.m.s. values ₁(K). In the example described here, 16 levels are used; hence indices s₁(K) have 4-bit size. The memory is read by using as addresses said indices arri­ ving from the output of multiplexer MX2, which is controlled by the signal on wire T/R: for coding phase, indices s₁(K) are present on bus 23 connected to an input of MX2, while during decoding they are pre­ sent on bus SR1 which arrives from block DMX of Fig. 1 and is connec­ ted to another input od MX2. CT1 denotes a modulo-16 counter which at each synchronism si­ gnal pulses CKK starts a counting cycle to generate indices s₁(K) which sequentially address the whole memory MQ1 supplying the quanti­ zer levels on data bus 24. CT1 is also synchronized by fast synchronism signal CKV arri­ ving from time-base BT of Fig.1, and which has such a frequency that it determines a complete counting of CT1 for each CKK pulse. Bus 23 is also connected to block CFR1, which is to determine for each block of K samples of residual signal R₁(m) quantized r.m.s. value ₁(K) of coding phase emitted on bus 25. CFR1 compares values σ₁(K), present on bus 22 and connected to one of its inputs, with all the values ₁(K) present on bus 24, kee­ ping stored, in an internal register, such value ,(K) that gives rise to the minimum difference with respect to the value on bus 22 and the corresponding index s₁(K): said values are then supplied on buses 25 and 29 at the end of each comparison cycle. The sending on buses 25, 29 of stored values is controlled by pulses of synchronism signal CKK. Block CFR1 implementation is no problem to the skilled in the art. Buses 24 and 25 are connected to two inputs of a multiplexer MX3 which is controlled by signal T/R: at the output of MX3 on bus 20 there are present quantized r.m.s. values for the coding phase (arriving from bus 25) and for decoding phase (arriving from bus 24). BF2 denotes an ordinary buffer register which keeps tempora­ rily stored two consecutive pairs of values ₁(K), s₁(K) present on buses 20, 29; the pairs are those computed on two consecutive blocks of K subband samples corresponding to the block of J samples whereupon CHA1 calculates vector a 1hott (i) present on bus A1. BF2 emits pairs ₁(K), s₁(K) on buses S1, ST1 respectively on the basis of the pulses of synchronism signal CKR arriving from time base BT of Fig. 1. BF2 acts as block D1, i.e. it aligns in the time values ₁(K), s₁(K) with subband signal samples present on bus XD1. It is to be noted that the updating frequency of data present on buses S1, ST1 is twice as high as that relevant to the data present on bus A1. CHA2 denotes a block whose internal structure is similar to that of block CHA1, and processes the samples of subband X₂ present on bus X₂ and supplies: on bus A2 linear prediction coefficient vectors a 2hott (i) for coding and decoding phases; indices h 2ott (J) on bus HT2, samples of residual signal R₂(m) on bus R2, and subband signal delayed on bus XD2 for coding phase. CHA2 receives synchronism signals on bu­ ses CKJ, CKL, the signal on were T/R, indices h 2ott (J) for decoding phase on bus HR2. For subband X₂, there is used in CHA2 a codebook of linear prediction coefficient vectors a 2h (i) containing H=32 vectors of L=4 components each, with subsequent size of indices h₂(J) of 5 bits. BF3 denotes a block containing a RAM memory of BF1 type and an addressing unit for reading of the type IND1; BF3, controlled by synchronism signals on buses CKJ, CKL and the signal on wire T/R, tem­ porarily stores the samples of subband X₃ present on bus X₃, and re-­ emits them duly delayed on buses R3 and XD3, time-aligned with those present on buses R1 and XD1 respectively. CSS2, CSS3 denote two blocks whose internal structure is si­ milar to that of block CSS1. CSS2 receives the samples of residual R₂(m) during coding on bus R2 and indices s₂(K) during decoding on bus SR2, and determines quantized r.m.s. values ₂(K) emitted on bus S2 for coding and deco­ ding phases, in addition to indices s₂(K) for coding phase emitted on bus ST2. CSS3 receives during coding delayed subband signal samples present on bus R3 and indices s₃(K) during decoding on bus SR3 and de­ termines quantized r.m.s. values ₃(K) for both phases emitted on bus S3, in addition to indices s₃(K) during coding emitted on bus ST3. CSS2, CSS3 receive synchronism signals on buses CKK, CKV, CKR and the signal on wire T/R. The level number of σ₂(K), σ₃(K) quantizers is equal to that of σ₁(K), while level values are different. MX9 denotes a multiplexer which, based on the logic level of signal T/R applied to control input, connects on output bus 210 to­ wards block RM1 the tern of indices s y (K) during coding present on bu­ ses ST1, ST2, ST3 or, during decoding, present on buses SR1, SR2,SR3. RM1, RM2, RM3, RM4 denote 4 memories of ROM type forming a simple example of circuit embodiment which determines the number of bits B y (K) during coding of signals of subbands X y . This simple implementation is attained on the basis of the following considerations: in the method description it is mentioned that the total coding bit number NT of the three subbands is taken as a constant; besides being y (K) quantized values, they can only assume a finite number of combinations of terns of values (in the example de­ scribed here, as there are 2⁴ values of y (K) for each subband, there will be obtained 2¹²=4.096 terns of values); an only configuration of number of bits, giving the minimum value of D(K) according to relation (7), corresponds to each y (K) tern. If for example a 16 kbit/s on-line transmission rate is desi­ red, NT=7 is to be selected, as will be seen also hereinafter. Besides it has been experimentally determined that each b y can assume only 5 values, from 1 to 5 bits, thus obtaining a total number of 15 configuations of b y terns, according to the following table: Then for each tern of values y (K) there have been previously computed the corresponding 15 values D(K) and their minimum has been determined: the number of the corresponding configuration of bit num­ bers b y (one of the numbers of the left column of the table) has been written in memory RM1 in such a position that the tern of indices s y (K) present on bus 210 is the reading address. Hence, for both steps, RM1 will supply on bus 211, for each index tern present on bus 210, the corresponding configuration number which in turn is the reading address for memories RM2, RM3, RM4. In RM2 there are stored, for each coding bit configuration, three addresses for the memories of subband quantizer sets, which will be mentioned with relation to Fig. 3. Each address relates to the first level of quantizer chosen for the corresponding subband. RM2 emits on bus B1 the address for the quantizer of the first subband, on bus B2 that for the second, on bus B3 that for the third. In RM3 there are stored, for each coding bit configuration, three numbers of levels of the three selected subband quantizers. RM3 emits on bus L1 the number of levels of the quantizer of the first subband, on bus L2 that of the second subband, on bus L3 that of the third. In RM4 there are stored, for each coding bit configuration, suitable bit configurations forming command signals for circuits of block QNT of Fig. 3, emitted on buses C1, C2, C3. As shown, said memories do not contain numbers of bits B y (K), but a coding of them which is function of how it will be used by cir­ cuit QNT of Fig. 3. Hence this coding is to be intended as non limi­ ting but typical of the example of embodiment described here. Buses ST1, ST2, ST3 form bus ST of Fig 1; buses HT1, HT2 form bus HT; buses SR1, SR2, SR3 bus SR; buses HR1, HR2 bus HR; buses B1, B2, B3, L1, L2, L3, C1, C2, C3 form bus BLC; buses S1, S2, S3 bus S; buses A1, A2 form bus A. In Fig. 3 PRB1 denotes a block, whose structure is emphasized inside the dashed line it is delimited by, carrying out "backward" adaptive linear prediction function, described at point C) of the meth­ od, on the samples of subband X₁ in both phases. PRB1 comprises circuits SM1, DV1, MX4, ML1, SM2, LPCF2. SM1 denotes a block executing the difference, in accordance with relation (10), between the samples of subband X₁, present on bus XD1 arriving from block CHA1 of Fig. 2, and the samples of the predic­ ted signal P₁(m-1) present on bus P1, supplying on bus 31 residual samples E₁(m); the latter samples are divided in divider DV1 by quan­ tized r.m.s. values ₁(K) present on bus S1 arriving from block CSS1 of Fig. 2, obtaining the samples of normalized residual EN₁(m) sup­ plied on bus N11. MX4 denotes a multiplexer which, controlled by signal T/R, supplies on bus 33 the samples of quantized normalized residual N₁(m), present for the coding phase on bus N12 or for the decoding one on bus N13. ML1 denotes a multiplier which executes the product of sam­ ples present on bus 33 and quantized r.m.s. values ₁(K) present on bus S1, obtaining samples of quantized residual ₁(m) supplied on bus 32. SM2 denotes an adder computing, in accordance with relation (8), the sum of samples present on buses 32 and P1 to obtain the sam­ ples of reconstructed signal ₁(m) supplied on bus XR1 for both pha­ ses. LPCF2 denotes a block carrying out linear-predictive filte­ ring operation, in accordance with relation (9), of the samples of re­ constructed signal ₁(m) present on bus XR1, using the vectors of li­ near prediction coefficients a 1hott (i) present on bus A1 and arriving from block CHA1 of Fig. 2, and obtaining the samples of predicted si­ gnal P₁(m-1) supplied on bus P1 for both phases. LPCF2 can be implemented by two linear prediction filters of known structure, one for coding the other for decoding, alternately activated by logic levels of signal T/R: this to keep from one frame to the other the contents of the filter registers forming delay lines for coding and decoding reconstructed signal which, otherwise would be altered by the presence of each phase of the signal samples of the ot­ her phase. Otherwise LPC2 can be implemented by a unique filter where only the delay line registers are doubled and alternately used. QUA1 denotes a block, whose structure is emphasized within the delimiting dashed line, determining the samples of the quantized normalized residual N₁(m) for both phases, and indices e₁(m) for co­ ding phase relevant to subband X₁. QUA1 comprises circuit blocks CFR2, CT2, MX6, SM3, SM4, MX5, MQ2. MQ2 denotes a ROM memory containing quantizers used for quan­ tizing the samples of normalized residual EN₁(m) of subband X₁. In the non-limiting example described here MQ2 consists of 5 quantizers, one for each number of bits b₁ by which values EN₁(m) can be quantized. For decoding phase, MQ2 emits on data bus N13, directly ap­ plied to an input of multiplexer MX4, the quantizer level addressed by the value present on address bus 40. For coding phase, MQ2 emits in succession on data bus N13 all the levels of the selected quantizer, on the basis of sequential ad­ dressing received on bus 40. Quantizer levels are supplied to a comparison input of compa­ rator CFR2 to whose second comparison input there is connected bus N11 carrying values to be quantized. CFR2 compares all the levels present on bus N13 with the va­ lue present on bus N11, keeping stored in an internal register the le­ vel which has originated minimum difference with respect to the value present on bus N11, and the corresponding index e₁(m) supplied on bus 35: said values are then supplied on buses N12 and ET1 at the end of the comparison cycle, at the occurrence of a synchronism signal pulse CKM, and represent respectively the quantized normalized residual N₁(m) of coding phase supplied at an input of multiplexer MX4, and index e₁(m) to be sent over the line. Each comparison cycle of CFR2 must be carried out during the time the values to be quantized remain on bus N11; CFR2 is activated by signal T/R. CT2 denotes a usual programmable counter emitting on bus 35 all indices e₁(m) of the quantizer to be addressed in memory MQ2. CT2 receives on wire CKM the signal of start of count and on bus L1, ar­ riving from memory RM3 of Fig. 2, the number of levels of the selected subband quantizer, which is the programmed value for the counter. MX6 denotes a common demultiplexer receiving on bus B1, co­ ming from memory RM1 of Fig. 2, the address of the first level of se­ lected subband quantizer, and emits it for coding phase on bus 36 and for decoding phase on bus 37, dependant on the logic level of signal T/R applied to control input. SM3 denotes a conventional adder supplying on bus 38 the sum of the contents of buses 35 and 36 to generate the addresses for memory MQ2 for coding phase: the value present on bus 36 address in MQ2 the selected quantizer, while the various values present on bus 35 address all its levels. SM4 denotes another adder supplying on bus 39 the sum of the contents of buses 37 and ER1, whereupon indices e₁(m) of decoding pha­ se at present, to generate the addresses for MQ2 in decoding: the va­ lue present on bus 37 addresses the selected quantizer, while the va­ lue present on bus ER1 addresses the quantizer level. MX5 denotes a multiplexer supplying on bus 40, as addresses for MQ2, the contents of bus 38 or of bus 39 on the basis of the logic level of signal T/R. PRB2 denotes a block analogous to PRB1, carrying out the function of adaptive linear prediction for subband X₂ in both phases. During coding, PRB2 receives the subband samples on bus XD2 and quantized normalized residual EN₂(m) on bus N22 and emits on bus N21 the samples of normalized residual EN₂(m). ring decoding instead, PRB2 receives samples N₂(m) on bus N23 and emits on bus XR2 subband samples. Further, PRB2 receives on buses S2 and A2 quantized r.m.s. values ₂(K) and coefficient vectors a 2hott (i) for both phases; fur­ thermore, it receives the signal on wire T/R. DV3 denotes a usual divider carrying out the division between the samples of subband X₃ present on bus XD3 and quantized r.m.s. values ₃(K) present on bus S3, thus obtaining normalized values EN₃(m) during coding on bus N31. ML3 denotes a usual multiplier executing a product between quantized normalized values N₃(m) during decoding present on bus N33 and those present on bus S3, obtaining on bus XR3 the samples of sub­ band X₃ during decoding. QUA2, QUA3 denote two blocks analogous to QUA1, determining samples of quantized normalized values N₂(m), N₃(m) for both phases and indices e₂(m), e₃(m) during coding respectively for subbands X₂ and X₃. QUA2 receives the normalized residual samples in coding on bus N21 from block PRB2, and emits thereto the samples of quantized normalized residual signal during coding on bus N22 and during deco­ ding on bus N23; it emits indices e₂(m) during coding on bus ET2 and receives those during decoding on bus ER2; in addition it receives on bus B2 the addresses of the first levels and on bus L2 the numbers of levels of the quantizers to be addressed in the internal memory of the set of quantizers of subband X₂. QUA3 receives normalized value samples in coding on bus N31 and emits on bus N33 quantized normalized values in decoding, while quantized normalized values in decoding, present at output N32, are not used since, as already seen, for subband X₃ no prediction is car­ ried out. Besides QUA3 emits indices e₃(m) in coding on bus ET3 and re­ ceives those in decoding on bus ER3; it also receives on bus B3 the addresses of the first levels and on bus L3 the number of levels of quantizers to be addressed in the internal memory of the set of quan­ tizers of subband X₃. MX7 denotes a multiplexer setting on bus ET the bits relating to indices e y (m) in coding extracting them from buses ET1, ET2, ET3, on the basis of control signals present on buses C1, C2, C3 arriving from memory RM4 of Fig. 2, in accordance with the table of coding bit configurations above. According to said table, the control signals of bus C1 deter­ mine the connection on bus ET of the number of wires of bus ET1 corre­ sponding to the number of coding bits allotted to subband X₁; the si­ gnals of buses C2, C3 carry out the equivalent task on the wires of buses ET2, ET3 relating to subbands X₂ and X₃ respectively. DX8 denotes a demultiplexer carrying out a task of opposite kind to that carried out by multiplexer MX7, i.e. it extracts the bits of the terns of indices e y (m) in decoding present on bus ER coming from block DMX of Fig. 1, and distributes them on buses ER1, ER2, ER3 on the basis of control signals present on buses C1, C2, C3 in accor­ dance with the table of coding bit configurations. According to said table the control signals of bus C1 de­ termine the connection on bus ER1 of the number of wires of bus ER corresponding to the coding bit number allotted to subband X₁; signals of buses C2, C3 carry out the equivalent task on the wires of buses ER2, ER3 relating to subbands X₂, X₃ respectively. In the example described here, buses ET1, ET2, ET3, ER1, ER2, ER3 consist of 5 wires to contain maximum number of coding bits to be allotted to each subband (5 bits), while buses ET, ER consist of 7 wi­ res to contain the total number of coding bits of each tern of indices e y (m) which is constant (7 bits). The outputs of MX7 and DX8 are activated respectively for co­ ding and decoding phases by the logic levels of signal T/R. The operation of the device provided by the invention will be now described. Blocks QMF and QMFD (Fig. 1) work synchronously with the sam­ pling and subsampling frequencies and product continous sample se­ quences of digital signals at their outputs. Also the writing operations of subband samples in memory BF1 (Fig. 2), in its equivalent of block CHA2, and in memory BF3 are syn­ chronous with the continuous flow of samples at their inputs. However, reading operations in said memories take place block by block; the length of said blocks (J samples) determines the frequency at which the two units of memories are mutually exchanged as well as the dimen­ sions of the intervals into which the device operation is split. Each interval consists of a coding and decoding phase which must take place while the units of said memories are filled with sub­ sequent sample blocks. The alternance of the two phases is controlled by the logic level of the signal T/R whose period corresponds to a block of J samples. During coding, sample blocks are read three times in the units of said memories. The first reading involves blocks of subbands X₁, X₂ which are supplied to circuit CXH and its equivalent in block CHA2 to compu­ te indices h 1ott (J), h 2ott (J) supplied on buses HT1, HT2 and identify vectors a 1hott (i), a 2hott (i) supplied on buses A1, A2. The second reading allows to calculate residual signals R y (m) supplied on buses R1, R2, and to supply the samples of subband X₃ on bus R3. Hence the internal circuits of blocks CSS1, CSS2, CSS3 deter­ mine the two pairs of values y (K) s y (K) supplied then on buses S1, ST1, S2, ST2, S3, ST3. Indices s y (K) identify on memories RM1, RM2, RM3, RM4 data supplied then through bus BLC to block QNT (Fig. 1) for addressing subband quantizers. It is worth noting that data on buses BLC, S, ST are updated at a frequency which is twice as high as that on buses A1, A2, being J=2K in the example of implementation described here. The third reading allows to supply said subband samples through buses XD1, XD2, XD3 to block QNT (Fig. 3). The first samples of the three blocks appear on said bus at the same time as corresponding coefficient vectors on buses A1, A2, and the first tern of pairs on buses S1, ST1, S2, ST2, S3, ST3, in addition to the first set of data on bus BLC. When on buses XD1, XD2, XD3 the (K+1)-th sample of the corre­ sponding blocks appears, the second tern of pairs of values appears on buses S, ST and the second set of data appear on bus BLC. For each tern of input subband samples, QNT circuits compute the corresponding tern of indices e y (m) supplied on buses ET1, ET2, ET3, using the data supplied through buses BLC, S, A by block PAR (Fig. 2); hence block MX7 (Fig. 3) supplies on bus ET the coding bits of the tern of samples. The two terns of values present on bus ST, the two indices present on bus HT and the coding bits of the three blocks of J subband samples present on bus ET are supplied to block MPX (Fig. 1) which or­ ganizes them into a frame, and sends them on output bus UT. In the subsequent decoding phase, DMX (Fig. 1) supplies to the outputs the data relevant to blocks of J samples of the three sub­ bands it receives from bus IR; on buses HR1, HR2 (Fig. 2) the indices allowing blocks CHA1, CHA2 to supply on buses A1, A2 the pair of coef­ ficient vectors during decoding a 1hott (i), a 2hott (i); then, on buses SR1, SR2, SR3, it supplies the first of the two terns of indices du­ ring decoding s y (K) allowing blocks CSS1, CSS2, CSS3 to supply on buses S1, S2, S3 a first tern of r.m.s. values y (K), and memories RM1, RM2, RM3, RM4 to supply on bus BLC a first set of data for block QNT (Fig. 3). Then DMX (Fig. 1) supplies on bus ER the terns of indices e y (m) relating to the first K samples of the three blocks, which are distributed on buses ER1, ER2, ER3 through block DX8 (Fig. 3). QNT extracts the quantized values it presents then on buses N13, N23, N33, and hence the samples of reconstructed subband signal, it send through buses XR1, XR2, XR3 to block QMFD (Fig. 1) which ex­ tracts therefrom samples of reconstructed digital speech signal sup­ plied on bus UR. After sending the K-th sample of the three blocks, DMX sends onto bus SR the second tern of indices s y (K) replacing the first; a second tern of r.m.s. values y (K) appears on buses S1,S2, S3 (Fig. 2,3) and a second set of data for block QNT (Fig. 3) appears on bus BLC. Then DMX (Fig. 1) supplies on bus ER the terns of indices e y (m) relating to the second K samples of the three blocks, whereupon QNT (Fig. 3) extracts other samples of reconstructed subband signal it sends to block QMFD (Fig. 1). At the end of these operations a new coding phase begins. Modifications and variations are possible without going out of the scope of the invention. More particularly for the embodiment of the variant of the above-described method, the following circuit mosifications are possi­ ble: - block PAR (Fig. 2) will comprise an additional circuit portion, for processing the samples of subband X₄ it receives from filter bank QMF of Fig. 1, similar to that provided for the samples of subband X₃ (blocks BF3, CSS3); besides, memories RM1, RM2, RM3, RM4 will have a greater capacity necessary to contain four sets of data relevant to the four subbands; buses ST, SR, BLC will also comprise data words relevant to the fourth subband; - block QNT (Fig. 3) will comprise an additional circuit portion, always for processing the samples of subband X4 it receives de­ layed by the additional buffer provided in block PAR (Fig. 2), si­ milar to that provided for the samples of subband X₃ (block DV3, ML3, QUA3); multiplexer MX7 and demultiplexer DX8 will also com­ prise a portion relevant to indices e₄(n) of the fourth subband also present on buses ET, ER; - multiplexer MPX and demultiplexer DMX (Fig. 1) will have now to process the information relevant to all the four subbands, while filter bank QMFD will also receive the samples of the reconstruc­ ted signal of the fourth subband from block QNT.

Claims: 1. A method for speech signal coding and decoding, charcterized in that for coding it provides the following steps: - said speech signal converted into digital is filtered by pairs of cascaded digital filters having specular transfer functions with respect to their transition frequencies and is split into subbands; - the samples of a first subband, at the lowest frequencies, are subdivided into a first sequence of blocks of J samples and submitted to a linear prediction inverse filtering opera­ tion with vector quantization techniques, and for each block there is selected, from a codebook of quantized filter coef­ ficient vectors a1h(i), vector a1hott(i) having index h1ott(J) and forming the optimal filter thus obtaining sam­ ples of residual signal R₁(m), m being the progressive sample index; the samples of residual signal R₁(m) are then split into a second sequence of blocks of K samples (with K submul­ tiple of J), and for each block a quantized r.m.s. value ₁(K) is computed by choosing from a quantizer a level of in­ dex s₁(K); from said first subband samples, samples of norma­ lized resudual EN₁(m) are extracted, each of them being quan­ tized by choosing a level of index e₁(m) in a quantizer of a set of quantizers of the first subband having a different number of levels; said normalized residual being obtained by dividing by said quantized r.m.s. values ₁(K) a residual signal obtained by subtracting from said signal of the first subband a predicted signal P₁(m-1) obtained by sub­ mitting said quantized normalized residual, previously multiplied again by said quantized r.m.s. value ₁(K), to a linear prediction filtering operation by using said vector of filter coefficients a1hott(i) forming the op­ timal filter; - the samples of a second subband adjacent to the first undergo the same type of operations carried out for said first sub­ band, thus obtaining vectors of quantized coefficients a2hott(i) having index h2ott(J), quantized r.m.s. values ₂(K) having index s₂(K), samples of quantized normalized re­ sidual having index e₂(m); - the samples of a third subband adjacent to the second subband are split into said second sequence of blocks and for each block a quantized r.m.s. value ₃(K) is computed by choosing in a quantizer a level of index s₃(K); each sample of the third subband is then divided by said quantized r.m.s. value ₃(K) obtaining a normalized value EN₃(m) which is quantized by choosing a level of index e₃(m) in a quantizer out of a set of quantizers of the third subband having diffe­ rent number of levels; - each tern of quantized r.m.s. values ₁(K), ₂(K), ₃(K) is used to compute the minimum of a mean total distortion func­ tion, said distorsion being introduced with the quantization of subband samples, said function depending on the number of bits used to quantize subband signals, said minimum supplying the terns of number of levels choosing the quantizers out of said sets of subband quantizers; said level indices e₁(m), e₂(m), e₃(m), said indices h1ott(J), h2ott(J) of linear-prediction optimal filter coefficient vec­ tors, and said indices s₁(K), s₂(K), s₃(K) of quantized r.m.s. values forming that speech signal coding; also cha­ racterized in that for decoding the following steps are provided: - level indices s₁(K), S₂(K), s₃(K) obtained during coding identify in the corresponding quantizers the terns of quanti­ zers r.m.s. value ₁(K), ₂(K), ₃(K), which, in turn, are used, as for the coding phase, to compute the terns of num­ bers of levels choosing the quantizers in said sets of sub­ band quantizers; - level indices e₁(m), e₂(m), e₃(m) obtained during coding de­ tect, in the chosen subband quantizers, quantized normalized values N₁(m), N₂(m), N₃(m); - quantized normalized values N₁(m), N₂(m) are multiplied by said quantized r.m.s. values ₁(K), ₂(K) and the result is submitted to a linear prediction filtering operation by using quantized filter coefficient vectors a1hott(i), a2hott(i), chosen in the relevant codebooks by said indices h1ott(J), h2ott(J) obtained during coding, and obtaining reconstructed signal samples of the first and second subbands; - quantized normalized values N₃(m) are multiplied by quanti­ zed r.m.s. values ₃(K) obtaining samples of the reconstructed signal of the third subband; - said reconstructged signals of the three subbands undergo an inverse filtering operation with respect to that used during coding to obtain the reconstructed speech signal. 2. A method, as in claim 1, characterized in that subband splitting gives rise to a fourth subband, at the highest speech-signal fre­ quencies taken as a null signal during coding and decoding. 3. A method as in claim 1, characterized in that said subband split­ ting gives rise to a fourth subband at the highest speech-signal frequenices, for which the following additional operations are carried out during coding: - the samples of said fourth subband undergo the same kind of operations carried out for said third subband, thus obtaining quantized r.m.s.values ₄(K) of index s₄(K), and samples of a quantized normalized value of index e₄(m); - said minimum of the mean total distortion function is also computed on the basis of said quantized r.m.s. values ₄(K), thus obtaining four sets of numbers of levels to select the quantizers out of the sets of quantizers of the four sub­ bands; said indices e₄(m) of quantized normalized value, and said in­ dices s₄(K) of quantized r.m.s. value cooperating in building up that speech-signal coding; further characterized in that during decoding: - said indices s₄(K) obtained during coding identify, in the corresponding quantizer, quantized r.m.s. values ₄(K) coope­ rating in determining, as in the coding phase, the four sets of numbers of levels choosing the quantizers in said sets of quantizers of the four subbands; - said inidces e₄(m) obtained during coding identify, in the chosen subband quantizer, quantized normalized values of the fourth subband which are multiplied by said quantized r.m.s. values ₄(K) to obtain samples of reconstructed signal of fourth subband, they too submitted to said inverse filtering operation, with respect to coding operation, to obtain said reconstructed speech-signal. 4. A method according to any one of the preceding claims, characteri­ zed in that to generate said sets of subband quantizers the follo­ wing steps are provided: a) from a speech-signal training sequence there are extracted sequences of subband samples and corresponding coefficient vectors a1hott(i), a2hott(i) forming optimal linear predic­ tion filters; b) subband quantizer levels are initialzed by optimal values for a Laplacian distribution speech-signal with unit variance and null means value; c) for each quantizer of the sets relating to the first and se­ cond subbands the following operations are carried out: c1) from the sequence of samples of the corresponding subband there are extracted said normalized residual signal sam­ ples EN₁(m) or EN₂(m), by using said vectors a1hott(i) or a2hott(i) obtained in step a); c2) samples EN₁(m) or EN₂(m) extracted in step c1) are split into a number of subsets equal to the number of levels of the quantizer to be generated, grouping into each subset the samples having a minimum distance from the same quan­ tizer level; c3) the centroids of each subset are computed and taken as new quantizer levels; c4) steps c2), c3) are repeated for a determined number of times; d) for each quantizer of the sets relating to the third and fourth subbands the following operations are carried out: d1) from the sequence of samples of the corresponding subband there are extrcted said samples of normalized values split into a number of subsets equal to the number of le­ vels of the quantizer, grouping in each subset the sam­ ples having a minimum distance from the same quantizer level; d2) the centroids of each subset are computed and taken as quantizer levels. 5. A speech-signal coding and decoding device characterized in that it basically comprises: - a first bank of digital filters (QMF1) organized in cascaded pairs, the filters of the pairs having specular transfer functions with respect to transition frequencies, said bank splitting into subbands the digital speech signal to be coded; - a first circuit (CHA1) which during coding receives the sam­ ples of a first subband, at the lowest frequencies, from said first bank (QMF) and reemits them delayed to the output, and comprises means (BF1, IND1) to subdivide samples into blocks of J samples, which form a frame time, linear prediction in­ verse filtering and vector quantization means (LPCF1, CXH, VOCA) to obtain, from each block, residual samples of first subband R₁(m), m being the progressive sample index, a first vector index h1ott(J) and a corresponding first vector of quantized coefficients a1hott(i) of the optimal filter during coding; said first circuit (CHA1) receiving, for each block of J samples, a first index h1ott(J) of the optimal filter during decoding and determining a corre­ sponding first vector of quantized coefficients a1hott(i) during decoding; - a second circuit (CHA2) whose structure is similar to that of said first circuit (CHA1), and which, during coding receives from said first bank (QMF) the samples of a second subband ad­ jacent to the first, and which for each block of J subband samples issues samples of second subband residual R₂(m), a second vec­ tor index h2ott(J) and a second quantized coefficients vector a2hott(i) of the optimal filter during coding; the second circuit (CHA2) receiving a second vector index h2ott(i) of the optimal filter and determining a corresponding se­ cond vector of quantized coefficients a2hott(i) during decoding; - first memory means (BF3) to temporarily store and subdivide into blocks the samples of a third subband adjacent to the second; - a third cirucit (CSS1) which during coding calculates an r.m.s. value of blocks of K samples of said first subband re­ sidual R₁(m), quantizes them obtaining for each block of K samples a first quantized r.m.s. value (K) and a corresponding first index of quantization level s₁(K) during coding, and receives a first index of quantization level s₁(K) and determins a first quantized r.m.s. value ₁(K) du­ ring decoding; - a fourth circuit (CSS2) whose structure is similar to that of the third circuit (CSS1), and which obtains from said second subband residual R₂(m) a second quantized r.m.s. value ₂(K) and a corresponding second index of quantization level s₂(K) during coding, and receives a second index of quantization level s₂(K) and determines a second quantized r.m.s. value ₂(K) during decoding; - a fifth circuit (CSS3), similar in structure to said third circuit (CSS1), and which obtains from each block of K samples received from said first memory means (BF3) a third quantized r.m.s. value ₃(K) and a third index of quantization level s₃(K) du­ ring coding, and receives a third index of quantization level s₃(K) and determines a third quantized r.m.s value ₃(K) du­ ring decoding; - first decoding means (MX9, RM1, RM2, RM3,RM4) which receive said first s₁(K), second s₂(K), third s₃(K) index of quanti­ zation level during coding and decoding and extract quantizer selecting signals (B1, B2, B3, L1, L2, L3); - a sixth circuit (PRB1) carrying out an adaptive linear pre­ diction function on samples of the first subband it receives from said first circuit (CHA1), by using at every block of samples said first quantized coefficient vector a1hott(i) du­ ring coding and decoding, and samples of a first quantized normalized residual N₁(m) during coding and decoding, and issuing during coding samples of a first normalized resi­ dual EN₁(m) with respect to said first quantized r.m.s. value ₁(K) it receives from said third circuit (CSS1), and during decoding first-subband reconstructed-signal samples (XR1); - a seventh circuit (PRB2), whose structure is similar to that of said sixth circuit (PRB1), receiving the samples of second subband and the second vector of quantized coefficients a2hott(i) from said second circuit (CHA2), said second quan­ tized r.m.s. value ₂(K) from said fourth circuit (CSS2), and samples of second quantized normalized residual signal N₂(m) during coding and decoding, and issues during coding the samples of a second normalized residual EN₂(m) and during decoding samples of second-subband reconstructed signal (XR2); - an eigth circuit (QUA1) which, by using quantizers selected out of a set of first subband quantizers on the basis of said signals of selection of a quantizer it receives from said first decoding means, quantizes the samples of said first normalized residual EN₁(m) it receives from said sixth circuit (PRB1), issuing samples of said first quantized nor­ malized residual N₁(m) and indices of first-subband quanti­ zation levels e₁(m) during coding; and it issues samples of said first quantized normalized residual N₁(m) during deco­ ding on the basis of indices of first - subband quantiza­ tion levels e₁(m) during decoding it receives at the in­ put; - a ninth circuit (QUA2), whose structure is similar to that of said eigth circuit (QUA1), receiving signals for selecting a quanti­ zer out of a set of second subband quantizers from said first decoding means, samples of said second normalized residual EN₂(m) from said seventh circuit (PRB2), indices of second. subband quantization levels e₂(m) during decoding, and ex­ tracts samples of said second quantized normalized residual N₂(m) during coding and decoding supplied to said seventh circuit (PRB2), and second-subband quantization-level indices e₂(m) during coding; - a tenth circuit (QUA3), whose structure is similar to that of said eigth circuit (QUA1), receiving selection signals of a quanti­ zer of a third-subband quantizer set from said first decoding means, samples of a normalized value EN₃(m) obtained by a di­ vider (DV3) of the samples of the third subband (XD3), recei­ ved from said first memory means (BF3), by said third quanti­ zed r.m.s. value ₃(K) during coding, indices of third-sub­ band quantization levels e₃(m) during decoding, and extracts third-subband quantization level indices e₃(m) during coding, and samples of a quantized normalized value N₃(m) which are multiplied in a first multiplier (ML3) by said third quanti­ zed r.m.s. value ₃(K) during decoding obtaining samples of third-subband reconstructed signal (XR3); - a first multiplexer (MPX) combining in subsequent frames said first h1ott(J) and second h2ott(J) vector index during coding it receives from said first and second circuits (CHA1, CHA2), said first s₁(K), second s₂(K) and third s₃(K) quantization level indices during coding it receives from said third, fourth and fifth circuits (CSS1, CSS2, CCS3), said quantiza­ tion level indices of first e₁(m), second e₂(m) and third e₃(m) subband during coding it receives from said eighth, ninth and tenth circuit (QUA1, QUA2, QUA3), relating to sub­ sequent blocks of J subband samples, and sends each frame to the output (UT) as a coded speech signal; - a first demultiplexer (DMX) which receives and splits coded speech signal frames, and generates said first h1ott(J) and second h2ott(J) vector indices during decoding for said first and second circuits (CHA1, CHA2), said first s₁(K), second s₂(K) and third s₃(K) quantization level index during deco­ ding for said third, fourth and fifth circuits (CSS1, CSS2, CSS3), said quantization level indices of first e₁(m), second e₂(m) and third e₃(m) subband during decoding for said eigth, ninth, and tenth circuits (QUA1, QUA2, QUA3); - a second bank of digital filters (QMFD), structurally dual to said first bank of digital filters (QMF), which receives from said sixth (PRB1), seventh (PRB2) circuits and first multi­ plier (ML3) said samples of reconstructed signal of first, second and third subband (XR1, XR2, XR3) and generates a di­ gital speech signal reconstructed during decoding. 6. A device as in claim 5, characterized in that it also comprises a time base (BT) generating synchronism signals for the circuits of the device and a first signal (T/R) whose logic level splits each frame time into a coding and decoding phase. 7. A device as in claim 6, characterized in that said first circuit (CHA1) comprises: - a first memory (BF1) temporarily storing subsequent blocks of J samples of first subband (X₁); - a first counter (IND1) generating reading addresses of said first memory to sequentially read three times each block of samples during coding; - a circuit (CXH) computing an autocorrelation function of each block of samples of first subband it receives through a first reading addressing of said first memory, said circuit determining said first vector index h1ott(J) during coding for each block of samples; - a second multiplexer (MX 1) emitting at the output the first index h1ott(J) in coding or in decoding, both applied to the inputs, on the basis of the logic level of said first signal (T/R); - a second memory (VOCA) storing a codebook of quantized coef­ ficients a1h(i), which, addressed by the output of said se­ cond mutliplexer (MX1), emits said first quantized coeffi­ cient vector a1hott(i) of the optimal filter in coding or in decoding; - a linear-prediction inverse digital filter (LPCF1) for said first- subband samples it receives, during a second reading addressing of said first memory (BF1), said filter using the first vectors of quantized coefficients of the optimal filter in coding received from said second memory (VOCA), and generating said first- subband residual sam­ ples R₁(m); - a first delay element (D1) for said first vectors of quanti­ zed coefficient of the optimal filter in coding which re-­ emits at the output (A1) synchronised with a third reading addressing of said first memory (BF2). 8. A device as in claim 7, characterized in that said third circuit (CSS1) basically consists of: - a computing circuit (VEF1) of the r.m.s. value σ₁(K) relevant to blocks of K samples of said first-subband residual R₁(m) it receives from said first circuit (CHA1) during coding pha­ ses; - a third memory (MQ1) containing the levels of a quantizer of said r.m.s. value σ₁(K); - a second counter (CT1) which when computing said r.m.s. value σ₁(K) generates reading addresses of said third memory (MQ1); - a third multiplexer (MX2) which, controlled by said first si­ gnal (T/R), emits as addresses towards said third memory (MQ1) the output of said second counter (CT1) in coding or said first quantization level s₁(K) in decoding; - a first comparator (CFR1) which during coding compares each r.m.s. value σ₁(K) computed by the relevant circuit (VEF1) and all the levels read in said third memory (MQ1), and sup­ plying at the output (25) the level originating the minimum difference, the address of said level emitted by said second counter (CT1) being said first quantization level index s₁(K) during coding; - a fourth multiplexer (MX3) which, controlled by said first signal (T/R), supplies at the output (20) as a first quanti­ zer r.m.s. value ₁(K) for the coding phase the output level (25) of said first comparator (CFR1), and for the decoding phases the output (24) of said third memory (MQ1); - a fourth memory (BF2) which temporarily stores said first quantization level index s₁(K) and said first quantized r.m.s. value ₁(K), during coding, and re-emits them synchro­ nous with said third reading addressing of said first memory (BF1). 9. A device as in claim 8, characterized in that said sixth circuit (PRB1) basically consists of: - a first adder (SM1) computing the difference between the first-­ subband samples (XD1), it receives from said first memory (BF1) through said third reading addressing, and the samples of a predicted signal; - a first divider (DV1) of the output of said first adder by said first quantized r.m.s. value ₁(K) supplied by said fourth memory (BF2), which issues said samples of first nor­ malized residual EN₁(m) during coding; - a fifth multiplexer (MX4) which, controlled by said first si­ gnal (T/R), supplies at the output said samples of first nor­ malized quantized residual N₁(m) during coding or decoding it receives at the inputs; - a second multiplier (ML1) of the output of said fifth multi­ plexer (MX4) by said first quantized r.m.s. value ₁(K) du­ ring coding or decoding; - a second adder (SM2) of the output of said second multiplier and said predicted signal (P1), the output of said second adder being during decoding said first-subband reconstructed signal (XR1); - a linear-prediction digital filter (LPCF2) which filters the samples emitted by said second adder (SM2) during coding and decoding using the first vectors of quantized coefficients of the optimal filter it receives during coding from said first delay element (D1), and during decoding from said second me­ mory (VOCA), said filter obtaining the predicted signal (P1). A device as in claim 9, characterized in that said eighth circuit (QUA1) basically comprises: - a fifth memory (MQ2) which contains said set of first-subband quantizers, and which at each reading during coding supplies in sequence all the levels of a quantizer of the set, and du­ ring decoding a quantizer level; - a third, programmable counter (CT2) which, for each sample of said first normalized residual EN₁(m) during coding progres­ sively counts up to a level number (L1) of a quantizer of said set; - a third adder (SM3) of the output of said third counter (CT2) and a first level address (B1) of a quantizer of said set, the output (38) of said third adder being the reading address of said fifth memory (MQ2) during coding; - a fourth adder (SM4) of a first level address (B1) of a quan­ tizer of said set and a first-subband quantization level in­ dex e₁(m) during decoding, the output (39) of said fourth ad­ der being the reading address during decoding of said fifth memory (MQ2) which emits said first quantized normalized re­ sidual N₁(m) during decoding; the outputs of said fourth or third adder being supplied at the address input of said fifth memory through a sixth multiplexer (MX5) controlled by said first signal (T/R); said first level addresses (B1) and level number (L1) forming said quantiser selecting signals; - a second comparator (CFR2) which during coding compares each sample of said first normalized residual EN₁(m) it receives from said first divider (DV1) with all the levels of a quan­ tizer read in said fifth memory (MQ2), supplying at the output N12 the level originating the minimum difference, said level being said first quantized normalized residual N₁(m) during coding, the index of said level emitted by said third counter (CT2) being said first-subband quantiza­ tion level index e₁(m) during coding. 11. A device as in any one of claims 5 to 10, characterized in that, each of said sets of quantizers of first, second and third subband consists of quantizers having a different number of levels corre­ sponding to a different number of bits of the corresponding quan­ tization level indices, and in that the sum of the number of bits of each tern of said quantization level indices of first, second and third subband is constant. 12. A device as in claim 11, characterized in that each quantization level index has a number of bits higher than zero. 13. A device as in claim 12, characterized in that said first decoding means comprise: - a seventh multiplexer (MX9) which, controlled by said first signal (T/R), sends to the output (210) said first s₁(K), se­ cond s₂(K), third s₃(K) quantization level index during co­ ding or decoding applied to its inputs; - sixth memory (RM1) containing, for each tern of quantiza­ tion level indices received from said seventh multiplexer as reading address, a bit number configuration to be allotted to said quantization level indices of first, second and third subband; - a seventh memory (RM2) containing, for each configuration of bit numbers it receives from said sixth memory as a reading address, said addresses of a first quantization level (B1, B2, B3) of a quantizer of first, second and third subband; - an eighth memory (RM3) containing, for each configuration of bit numbers it receives from said sixth memory as a reading address, said level numbers (L1, L2, L3) of quantizers of first, second and third subband. 14. A device as in any of claims 5 to 13, characterized in that said first bank of digital filters (QMF) produces a fourth subband, at the highest speech-signal frequencies, which signal is transferred as a null signal to the inputs of said second bank of digital filters (QMFD). 15. A device as in any of claims 5 to 13, characterized in that said first bank of digital filters (QMF) produces a fourth subband at the highest speech-signal frequencies, and in that it comprises additional means for processing the samples of said fourth subband in a way similar to that of said third subband, so that to the inputs of said second bank of digital filters (QMFD) there are transferred reconstructed signal samples of the fourth subband.

Inventors: COPPERI, MAURIZIO, SERENO, DANIELE
Applicants: CSELT CENTRO STUDI E LABORATORI TELECOMUNICAZIONI S.P.A.
Application year: 1986
Main classification: H04B1
IPC codes: H03M7, G10L19, H04B1, G10L11, H03M3