speech parameter encoding method capable of transmitting a spectrum parameter at a reduced number of bits in a speech parameter encoding method of encoding an input speech signal into a sequence of encoded signals vector quantization is carried out through a plurality of vector quantizers which are connected in cascade to one another through subtractors and which cooperate with code books respectively such a cascade connection of the vector quantizers may be supplied with the input speech signal either at every frame or at every subframe shorter than the frame to produce a set of code vector candidates from each of the vector quantizers each set of the code vector candidates is sent to a cumulative distortion calculator to select an optimum combination of the code vector candidates and to produce as the encoded signals an index representative of the optimum combination alternatively a sequence of coefficients which represent a spectrum parameter is calculated at every frame and is divided into a plurality of coefficient groups each of which is subjected to vector quantization to produce a plurality of code vector candidates for each coefficient group cumulative distortions are calculated for combinations of the code vector candidates to select an optimum combination

variable frame rate fixed bit rate vocoding method a method of operating a vocoder so that a variable frame rate results while maintaining a constant bit rate is disclosed a base frame rate remains constant however spectral change is measured to determine an appropriate subframe rate within the base frame only one of a plurality of separate vector quantization processes is selected in response to spectral change for performance of a vector quantization operation on lpc spectrum coefficients each of the plurality of vector quantization operations utilizes its own codebook that contains a different quantity of reference patterns from that contained in the other codebooks each vector quantization operation produces a reference pattern descriptor code that contains a different number of bits from that produced by the other vector quantization processes vector quantization operations producing smaller less spectrally accurate outputs are selected when more subframes are included with a base frame and vector quantization operations producing larger more spectrally accurate outputs are selected when fewer subframes are included within a base frame

frequency selective harmonic coding the present invention relates to a method of encoding speech comprised of processing the speech by harmonic coding to provide a fundamental frequency signal and a set of optimal harmonic amplitudes processing the harmonic amplitudes and the fundamental frequency signal to select a reduced number of bands and to provide for the reduced number of bands a voiced and unvoiced decision signal an optimal subset of magnitudes and a signal indicating the positions of the reduced number of bands whereby the speech signal may be encoded and transmitted as the pitch signal and the signals provided for the reduced number of bands with a bandwidth that is a fraction of the bandwidth of the speech

a direct approach for the frequency adaptive feedforward cancellation of harmonic disturbances this paper is concerned with the robust rejection of harmonic disturbances with unknown frequency and amplitude affecting uncertain linear system the developed control scheme combines the properties of adaptive feedforward cancellation afc techniques with the phase and frequency detection capabilities provided by a nonlinear frequency estimation algorithm under mild assumptions on the nominal model of the system to be controlled the proposed scheme is proven to achieve the complete rejection of harmonic disturbances at the input or at the output of the system a detailed stability analysis based on averaging provides useful informations on the effect of the tuning parameters over the convergence of the estimator the effectiveness of the proposed approach is demonstrated by a two time scale averaging analysis

scheduled control for robust attenuation of non stationary sinusoidal disturbances with measurable frequencies attenuation of sinusoidal disturbances with uncertain yet online measurable frequencies is considered the disturbances are modeled as the outputs of an undisturbed parameter dependent exogenous system with a skew symmetric system matrix obtained in response to nonzero initial conditions the problem is formulated for a parameter dependent plant as the synthesis of a parameter dependent controller in a way to ensure internal stability as well as a desired level of steady state disturbance attenuation in the face of all admissible parameter variations the solvability of this problem is first related to the existence of bounded solutions to a matrix differential regulator equation subject to an asymptotic norm constraint reformulating this as a parameter dependent state feedback like synthesis based on which suitable solutions to the differential regulator equation can be obtained online tractable solvability conditions are then provided in the form of parameter dependent matrix inequalities controllers that solve the generalized asymptotic regulation problem are also parameterized in terms of the suitable solutions of the differential regulator equation and some free parameter dependent matrices that are to be designed off line to ensure stability a procedure is then developed to design the free parameters in a way to achieve desirable transient behavior the use of the developed synthesis procedure is illustrated on a simplified version of the course control problem in ship steering

technical communique performance of an adaptive algorithm for sinusoidal disturbance rejection in high noise the paper proposes a discrete time algorithm for the rejection of sinusoidal disturbances of unknown frequency the algorithm is adapted from an existing continuous time algorithm and the overall nonlinear system is analyzed using a linear approximation the paper shows that the noise rejection properties may be predicted using an in phase quadrature decomposition of the noise similar to the one encountered in the theory of phase locked loops estimates are obtained for the standard deviations of the plant output and of the adaptive parameters simulations validate the predictions of the analysis despite the nonlinear nature of the adaptive system and the high level of noise applied

a stability instability boundary for disturbance free slow adaptation with unmodeled dynamics the instability of adaptive schemes for small values of parameter adjustment gain has the form of a slow drift of adjustable parameters with an averaging analysis we derive not only a sufficient condition for stability of this drift but also a sufficient condition for it to be unstable a sharp boundary between stability and instability is signal dependent and much less demanding than the usual strict positive realness property the new positivity condition has a simple signal energy interpretation

adaptive feedforward disturbance rejection in nonlinear systems this paper investigates the problem of adaptive feedforward compensation for a class of nonlinear systems namely that of input to state and locally exponentially convergent systems it is shown how under suitable assumptions the proposed scheme succeeds in achieving disturbance rejection of a harmonic disturbance at the input of a convergent nonlinear system with a semi global domain of convergence the effectiveness of the proposed solution is demonstrated by combining results from averaging analysis with techniques for semi global stabilization an illustrative example shows the effectiveness of the scheme

quantization of harmonic amplitudes representing speech in a speech coding and decoding system in which a timewise segment of an acoustic speech signal is represented by a frame of a data signal characterized by a fundamental frequency and spectral harmonics a current frame is reconstructed using a set of prediction signals based on the number of spectral harmonics for the current frame and a preceding frame and reconstructed signal parameters characterizing the preceding frame the number of spectral harmonics for the current and preceding frames are reconstructed from at least a pair of digitally encoded signals that are generated using error protection codes for all of their bits

speech parameter coding method and apparatus a speech parameter coding method and apparatus which can quantize a spectrum parameter of a speech signal with a smaller number of bits smaller than ever a dividing section divides a predetermined order number of spectrum parameters obtained by division of a speech signal into frames for each order number of spectrum parameters smaller than the divisional order number a vector quantizing section searches codebooks for the divided spectrum parameters for each order number and outputs a plurality of candidates of codevector in order of magnitude from the minimum one an accumulated distortion calculating section calculates accumulated distortions for the entire order number for combinations of codevectors a minimum judging section selectively outputs a combination of codevectors which minimizes the accumulated distortion thereby to quantize the spectrum parameter

constant data rate speech encoder for limited bandwidth path a speech signal has its characteristics extracted and encoded 16 transmitted over a limited data rate path 18 and is decoded 20 and synthesized 22 at the receiving end the characteristics include line spectral frequencies lsf pitch and jitter the lsf are extracted by autoregression and split vector quantized svq in a single frame and in parallel in blocks of two three and four frames the svq codes have equal length and are evaluated for distortion in conjunction with a threshold the threshold is varied in such a manner as tend to select for transmission those codewords which maintain a constant data rate into a transmit buffer a single bit jitter bit and encoded pitch value are product coded with the selected lsf codeword and all are transmitted over the data path 18 to the receiver the receiver decodes the characteristics and controls a pitch generated 1226 in response to the pitch value and a random pitch jitter in response to the jitter bit two sets of line spectrum filters receive random noise and the pitch signal respectively the filtered signals are modulated by multipliers 1222 1230 controlled by the lsf codes and the filtered signals are summed and applied to a final lsf controlled filter

fourier transform vector quantization for speech coding design algorithms and simulation results are presented for vector quantizers for fourier transformed data transforming the data prior to quantization has two potential advantages first each sample in the transform domain depends on many samples in the original domain thus even scalar quantization in the transform domain is a form of vector quantization or block source coding in the original waveform domain and the basic coding theorems of information theory show that such block codes can provide better performance than scalar codes even for memoryless sources second vector quantization of fourier transformed speech waveforms provides distinctly better subjective quality than ordinary vector quantization of the waveform using codes of comparable complexity while the system is of course more complicated due to the need to take fourier transforms its envisioned application is as a coder for the output of fft chips currently available or under development the proposed implementation of a fourier transform vector quantizer ftvq uses a product code structure providing different codes for different coefficient vectors corresponding to different frequency bands this is a form of subband coding and yields a simple means of optimizing bit allocations among the subcodes two coding structures with corresponding distortion measures are considered those that quantize vectors of pairs of real and imaginary coefficients and those that quantize separate vectors of magnitude and phase coefficients both structures yield good performance for the given complexity in comparison to waveform vector quantizers for speech coding a magnitude phase ftvq yields better subjective quality than a real imaginary ftvq when the rate allocation is properly chosen

brief robust adaptive compensation of biased sinusoidal disturbances with unknown frequency asymptotically stable observable linear systems of order n which are not required to be minimum phase and are affected by an additive noisy biased sinusoidal disturbance with unknown bias magnitude phase and frequency are considered the problem of designing an output feedback compensator which regulates the output to zero for any initial condition and for any biased sinusoidal disturbance with no noise is addressed under the assumption that the system parameters are known this problem is solved by a 2n 6 order compensator which generates asymptotically convergent estimates of the biased sinusoidal disturbance and of its parameters including frequency the robustness of the closed loop system with respect to sufficiently small additive unmodelled noise is characterized in terms of input to state stability

vector quantization in a harmonic speech coding arrangement a harmonic speech coding arrangement where vector quantization is used to improve speech quality parameters are determined at the analyzer 120 of an illustrative coding arrangement to model the magnitude and phase spectra of the input speech a first codebook of vectors is searched for a vector that closely approximates the difference between the true and estimated magnitude spectra a second codebook of vectors is searched for a vector that closely approximates the difference between the true and the estimated phase spectra indices and scaling factors for the vectors are communicated to the synthesizer 160 such that scaled vectors can be added into the magnitude and phase spectra for use at the synthesizer in generating speech as a sum of sinusoids

digital speech coder having improved vector excitation source an improved excitation vector generation and search technique fig 1 is described for a code excited linear prediction celp speech coder 100 using a codebook of excitation code vectors a set of m basis vectors vm n are used along with the excitation signal codewords i to generate the codebook of excitation vectors ui n according to a vector sum technique 120 of converting the selector codewords into a plurality of interim data signals multiplying the set of m basis vectors by the interim data signals and summing the resultant vectors to produce the set of 2m codebook vectors the entire codebook of 2m possible excitation vectors is efficiently searched by using the vector sum generation technique with the m basis vectors without ever having to generate and evaluate each of the 2m code vectors themselves furthermore only m basis vectors need to be stored in memory 114 as opposed to all 2m code vectors

adaptive rejection of non linear disturbances in extended non linear output feedback systems this paper deals with asymptotic rejection of disturbances generated from non linear exosystems for uncertain nonlinear systems in an extended output feedback form which allows the vector field coupled with the system input to have different nonlinear functions of the system output as its elements a new internal model design is proposed to deal with nonlinear functions of the system output that are coupled with the input and the unknown disturbance adaptive control techniques are then used to deal with the uncertainty in the system the proposed adaptive disturbance rejection algorithm with the new internal model design ensures the asymptotic rejection of the unknown nonlinear disturbance and the boundedness of all the variables

methods for speech transmission the quantized parameter bits are grouped into several categories according to their sensitivity to bit errors more effective error correction codes are used to encode the most sensitive parameter bits while less effective error correction codes are used to encode the less sensitive parameter bits this method improves the efficiency of the error correction and improves the performance if the total bit rate is limited the perceived quality of coded speech is improved a smoothed spectral envelope is created in the frequency domain the ratio between the actual spectral envelope and the smoothed spectral envelope is used to enhance the spectral envelope this reduces distortion which is contained in the spectral envelope

solution of second order difference equations via the τ method abstract classically the solution of many functional equations is sought in the form of an infinite series of some kind in practice to get numerical results the series must be truncated in recent times alternative finite series approximations have been sought as solutions of a functional equation slightly perturbed from the given one this concept was first introduced by lanczos who named it the method in previous studies the author has applied the method to get polynomial and rational approximations for differential equations mostly of hypergeometric type in the present paper we apply the method concept to get like approximations for solutions of difference equations in particular we study the second order difference equation satisfied by the bessel function of the first kind except for some recent work on the gamma function by the author wimp and ting the literature is void on use of the method idea for solutions of difference equations

speech parameter encoder a speech parameter encoder capable of encoding spectrum parameters at a bit rate of 1 kb s or less with comparatively small amount of operations and memory capacity a spectrum parameter calculation unit 130 derives a spectrum parameter representing the spectrum envelope of a discrete input speech signal through division thereof into frames each having a predetermined time length a weighted coefficient calculation unit 150 derives a weighted coefficient corresponding to an auditory masking threshold value through derivation thereof from the speech signal a spectrum parameter quantization unit 160 receives the weighted coefficient and the spectrum parameter and centeses the spectrum parameter through search of a codebook such as to minimize the weighting distortion based on the weighted coefficient

disturbance cancellation for linear systems by adaptive internal models abstract the adaptive disturbance cancellation problem is considered for single input single output linear systems when the linear exosystem which generates the disturbances to be rejected and or the reference signals to be tracked is unknown that is both its order 2 r 1 and its parameters 1 r are unknown the proposed regulator makes use of two observers and contains an adaptive internal model which tunes m internal parameters on the basis of the output regulation error it is shown that if m r that is the adaptive internal model can reproduce the required input the regulation error tends exponentially to zero if m r that is the adaptive internal model can only approximate the required input within an approximation error the regulation error tends exponentially into a closed ball whose radius is proportional to provided that is sufficiently small so that singularities are avoided

unified analysis of iterative learning and repetitive controllers in trial domain while iterative learning control ilc and repetitive control rc have much ground in common they fundamentally differ in the initial conditions at each repetition this difference has lead to distinct analysis techniques hereby clouding the interrelations between both control strategies to facilitate the transfer of results this paper presents a unified approach to ilc and rc both control problems are formulated in the trial domain using so called system lifting for a given system the corresponding ilc and rc trial domain models differ and a thorough system theoretic analysis and comparison of these models is performed to illustrate the value of a unified formulation of ilc and rc the analysis of the most commonly used ilc and rc structures is harmonized this analysis reveals central differences and similarities between various stability monotonic convergence and steady state performance conditions

methods for speech quantization and error correction the redundancy contained within the spectral amplitudes is reduced and as a result the quantization of the spectral amplitudes is improved the prediction of the spectral amplitudes of the current segment from the spectral amplitudes of the previous is adjusted to account for any change in the fundamental frequency between the two segments the spectral amplitudes prediction residuals are divided into a fixed number of blocks each containing approximately the same number of elements a prediction residual block average prba vector is formed each element of the prba is equal to the average of the prediction residuals within one of the blocks the prba vector is vector quantized or it is transformed with a discrete cosine transform dct and scalar quantized the perceived effect of bit errors is reduced by smoothing the voiced unvoiced decisions an estimate of the error rate is made by locally averaging the number of correctable bit errors within each segment if the estimate of the error rate is greater than a threshold then high energy spectral amplitudes are declared voiced

brief paper asymptotic rejection of asymmetric periodic disturbances in output feedback nonlinear systems this paper deals with asymptotic rejection of periodic disturbances which may have asymmetric basic wave patterns this class of disturbances covers asymmetric wave forms in the half period such as alternating sawtooth wave form some disturbances which are generated from nonlinear oscillation such as van de pol oscillators as well as disturbances with symmetric half period wave forms such as sinusoidal disturbances and triangular disturbances etc the systems considered in this paper can be transformed to the nonlinear output feedback form the amplitude and phase of the disturbances are unknown the novel concept of integral phase shift is introduced together with the newly introduced half period integration operator to investigate the invariant properties of asymmetric periodic disturbances they are used for the estimation of unknown disturbances in the systems together with observer design techniques to deal with nonlinearity the proposed control design with the disturbance estimation asymptotically rejects the unknown disturbance and ensures the overall stability of the system

robust optimal design and convergence properties analysis of iterative learning control approaches in this paper we address four major issues in the field of iterative learning control ilc theory and design the first issue is concerned with ilc design in the presence of system interval uncertainties targeting at time optimal fastest convergence and robustness properties concurrently we formulate the ilc design into a min max optimization problem and provide a systematic solution for linear type ilc consisting of the first order and higher order ilc schemes inherently relating to the first issue the second issue is concerned with the performance evaluation of various ilc schemes convergence speed is one of the most important factors in ilc a learning performance index q factor is introduced which provides a rigorous and quantified evaluation criterion for comparing the convergence speed of various ilc schemes we further explore a key issue how does the system dynamics affect the learning performance by associating the time weighted norm with the supreme norm we disclose the dynamics impact in ilc which can be assessed by global uniform bound and monotonicity in iteration domain finally we address a rather controversial issue in ilc can the higher order ilc outperform the lower order ilc in terms of convergence speed and robustness by applying the min max design which is robust and optimal and conducting rigorous analysis we reach the conclusion that the q factor of ilc sequences of lower order ilc is lower than that of higher order ilc in terms of the time weighted norm

system for embedded coding of speech signals the set of possible excitation signals is subdivided into a plurality of subsets the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network whilst the others supply a contribution which when added to that of the first subset causes a rate increase by successive steps at the receiving side a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate whilst for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used

efficient codebook structure for code excited linear prediction coding a speech communication system using a code excited linear prediction speech decoder the decoder using a first codebook containing a first digital value sequence selected from the set of binary values 0 1 the decoder also using a second codebook containing a second digital value sequence having values selected from the set of binary values 1 0 the first digital value sequence and the second digital value sequence are combined to become a third digital value sequence having a set of ternary values from the set of 1 0 1

repetitive control system a new type servo system for periodic exogenous signals a control scheme called repetitive control is proposed in which the controlled variables follow periodic reference commands a high accuracy asymptotic tracking property is achieved by implementing a model that generates the periodic signals of period l into the closed loop system sufficient conditions for the stability of repetitive control systems and modified repetitive control systems are derived by applying the small gain theorem and the stability theorem for time lag systems synthesis algorithms are presented by both the state space approach and the factorization approach in the former approach the technique of the kalman filter and perfect regulation is utilized while coprime factorization over the matrix ring of proper stable rational functions and the solution of the hankel norm approximation are used in the latter one

technical communique asymptotic rejection of unknown sinusoidal disturbances in nonlinear systems this paper deals with global disturbance rejection of nonlinear systems the disturbance is assumed to be sinusoidal with completely unknown phases amplitude and frequencies but the number of distinct frequencies or the order of the corresponding unknown linear exosystem is known different from the common structural assumptions of nonlinear systems needed in literature for disturbance rejection of nonlinear systems the proposed method only requires the information of control design with a known lyapunov function when the system is disturbance free and a mild assumption needed for internal model design the proposed disturbance rejection algorithm extends complete global rejection of unknown sinusoidal disturbances for nonlinear dynamic systems beyond the common nonlinear models such as the strict feedback forms and the output feedback forms

a general theory of linear time invariant adaptive feedforward systems with harmonic regressors establishes necessary and sufficient conditions for an adaptive system with a harmonic regressor i e a regressor comprised exclusively of sinusoidal signals to admit an exact linear time invariant lti representation these conditions are important because a large number of adaptive systems used in practice have sinusoidal regressors and the stability and performance of such systems having lti representations can be completely analyzed by well known methods the theory is extended to applications where the lti conditions do not hold in which case the harmonic adaptive system can be written as the parallel connection of a purely lti subsystem and a linear time varying ltv subsystem an explicit upper bound is established on the induced two norm of the ltv block which allows systematic treatment using emerging robust control methods applicable to lti systems with norm bounded ltv perturbations

integrals of jacobi functions in this paper we study f 1 x a 1 x bp a x dx and its partial derivatives with respect to a and b where p a x is the jacobi function our expressions generalize the work of blue gautschi and gatteschi the results are useful to derive integration formulas for integrands with algebraic and logarithmic singularities

nonstationary spectral modeling of voiced speech the main purpose of this paper is to present a novel model for voiced speech the classical model which is being used in many applications assumes local stationarity and consequently imposes a simple and well known line structure to the short time spectrum of voiced speech the model derived in this paper allows for local non stationarities not only in terms of pitch perturbations but in terms of vocal tract variations as well the resulting structure of the short time spectrum becomes more complex but can still be interpreted in terms of generalized lines the proposed model supports new forms of spectral prediction which can be put to advantage in speech coding applications experimental results are presented supporting the validity of both the model itself and the prediction relationships finally a new class of speech coders denoted harmonic coders based on the presented model is proposed and a specific implementation is presented

internal model based design of repetitive and iterative learning controllers for linear multivariable systems repetitive and iterative learning control are two modern control strategies for tracking systems in which the signals are periodic in nature this paper discusses repetitive and iterative learning control from an internal model principle point of view this allows the formulation of existence conditions for multivariable implementations of repetitive and learning control it is shown that repetitive control can be realized by an implementation of a robust servomechanism controller that uses the appropriate internal model for periodic distrubances the design of such controllers is discussed next it is shown that iterative learning control can be implemented in the format of a disturbance observer compensator it is shown that the resulting control structure is dual to the repetitive controller and that both constitute an implementation of the internal model principle consequently the analysis and design of repetitive and iterative learning control can be generalized to the powerful analysis and design

processing of acoustic waveforms a sinusoidal model for acoustic waveforms is applied to develop a new analysis synthesis technique which characterizes a waveform by the amplitudes frequencies and phases of component sine waves these parameters are estimated from a short time fourier transform rapid changes in the highly resolved spectral components are tracked using the concept of birth and death of the underlying sine waves the component values are interpolated from one frame to the next to yield a representation that is applied to a sine wave generator the resulting synthetic waveform preserves the general waveform shape and is perceptually indistinguishable from the original furthermore in the presence of noise the perceptual characteristics of the waveform as well as the noise are maintained the method and devices disclosed herein are particularly useful in speech coding time scale modification frequency scale modification and pitch modification

brief paper robust high order repetitive control optimal performance trade offs high order repetitive control has previously been introduced to either improve the robustness for period time uncertainty or reduce the sensitivity for non periodic inputs of standard repetitive control schemes this paper presents a systematic semidefinite programming based approach to compute high order repetitive controllers that yield an optimal trade off between these two performance criteria the methodology is numerically illustrated through trade off curves for various controller orders and levels of period time uncertainty moreover existing high order repetitive control approaches are shown to correspond to specific points on these curves

low data rate speech encoder with mixed excitation a speech signal has its characteristics extracted and encoded 16 transmitted over a limited data rate path 18 and is decoded 20 and synthesized 22 at the receiving end the characteristics include line spectral frequencies lsf pitch and jitter the lsf are extracted by autoregression and splitvector quantized svq in a single frame and in parallel in blocks of two three and four frames the svq codes have equal length and are evaluated for distortion in conjunction with a threshold the threshold is varied in such a manner as tend to select for transmission those codewords which maintain a constant data rate into a transmit buffer a single bit jitter bit and encoded pitch value are product coded with the selected lsf codeword and all are transmitted over the data path 18 to the receiver the receiver decodes the characteristics and controls a pitch generated 1226 in response to the pitch value and a random pitch jitter in response to the jitter bit two sets of line spectrum filters receive random noise and the pitch signal respectively the filtered signals are modulated by multipliers 1222 1230 controlled by the lsf codes and the filtered signals are summed and applied to a final lsf controlled filter

spectral magnitude representation for multi band excitation speech coders a method for encoding a speech signal into digital bits including the steps of dividing the speech signal into speech frames representing time intervals of the speech signal determining voicing information for frequency bands of the speech frames and determining spectral magnitudes representative of the magnitudes of the spectrum at determined frequencies across the frequency bands the method further includes quantizing and encoding the spectral magnitudes and the voicing information the steps of determining quantizing and encoding the spectral magnitudes is done is such a manner that the spectral magnitudes independent of voicing information are available for later synthesizing

small amplitude chaos and ergodicity in adaptive control abstract the theory for continuous iterated maps on the real line is used to study the dynamics of a continuous family of one parameter adaptive systems we establish sufficient and necessary conditions for uniform asymptotic stability and ultimate boundedness exact bifurcation boundaries and sufficient conditions for the existence of chaotic attractors and ergodicity of the estimated parameter it is argued that the chaotic dynamics are structurally stable that they persist in higher dimensional adaptive control systems and that they do not necessarily lead to poor performance when the controller is modified to deal with the infinite drift and small divisor problems a novel modification involving a sign change in the adaptive law is proposed

predictive optimal iterative learning control a new optimization based iterative learning control algorithm is proposed and its properties derived an important characteristic of this algorithm is that it uses present and future predicted errors to compute the current control in a similar manner to model based predictive control using a receding horizon in particular it enables the algorithm designer to achieve good control over convergence rate the actual implementation has a multimodel structure but uses standard linear quadratic regulator methods for a causal formulation in the iterative learning sense of what is originally a non causal algorithm the results are illustrated by simulations

optimal periodic disturbance reduction for active noise cancelation abstract the design of an optimal internal model based imb controller by extending standard discrete time optimal control theory for imb controllers is described the optimal observer and state feedback gains of the imb controller are given via the solution of discrete time algebraic riccati equations the design method is applied to an acoustic system that is subjected to disturbances from a server fan periodic disturbances from the server fan appear as harmonics of the fundamental frequency of the fan parametric models for the plant and non periodic part of the disturbance are identified from experimental data an internal model is designed in discrete time and the internal model principle is used to design a feedback controller that rejects periodic disturbances in the acoustic system the controller is implemented in real time and successfully attenuates the first four harmonics of the fan noise

output regulation for unknown stable linear systems the regulation problem is addressed for single input single output linear unknown stable systems p s of unknown order with exosystems generating biased multi sinusoidal references and or disturbances containing at most q known frequencies omega i 1 leq i leq q sufficient conditions are presented which require the knowledge of rm sign p 0 and either rm sign re p j omega i or rm sign im p j omega i 1 leq i leq q the constructive proofs lead to the design of regulators with minimal order 2q 1 provided that the gains are sufficiently small for stable plants p s such that rm re p j omega 0 for any omega in bbr a universal modular regulator is designed without gain limitations

solution of the difference equation for gauss hypergeometric function by use of the tau method abstract in two previous papers the idea of the tau method to get approximate solutions of differential equations was extended to get like results for difference equations the models treated were the difference equation for the reciprocal of the gamma function and the difference equation for the entire part of the bessel function of the first kind in this paper we apply the tau method as described in the title of this paper some numerics are presented which indicate that the process is convergent in this connection an a posteriori error analysis for the bessel function case is presented how this analysis could be extended to the model of the present paper is discussed but further analysis is deferred to future research

asymptotic rejection of unmatched general periodic disturbances with nonlinear lipschitz internal models this article deals with asymptotic rejection of general periodic disturbances in a class of nonlinear systems by exploiting observer design with lipschitz output nonlinearities for the internal model design it is shown that a class of general periodic distances can be modelled as outputs of linear systems with nonlinear output functions based on this observation a nonlinear observer design with output lipschitz nonlinearities is investigated and integrated for the internal model design to estimate the phase and amplitude of the desired feedforward input for asymptotic disturbance rejection in a class of nonlinear systems with input to state stable zero dynamics a number of problems on disturbance rejection can be formulated in the disturbance rejection form shown in this article to which the proposed nonlinear lipschitz internal model can be applied two examples are included to demonstrate the proposed design strategies

asymptotic rejection of unmatched general periodic disturbances in a class of non minimum phase non linear systems this article deals with asymptotic rejection of unmatched general periodic disturbances in nonminimum phase non linear output feedback systems the steady state responses are defined for unstable systems subject to general periodic disturbances and the generalised gain and phase are defined for stable systems subject to general periodic disturbances based on the new definitions new results are obtained for the equivalent input disturbance and disturbance estimation an lp convergent estimate of the equivalent input disturbance is incorporated in the control design to ensure the asymptotic rejection of unmatched general periodic disturbances while maintaining the stability of the non linear system

brief paper an adaptive learning regulator for uncertain minimum phase systems with undermodeled unknown exosystems the design of an adaptive learning regulator is addressed for uncertain minimum phase linear systems with known bounds known upper bound on system order known relative degree known high frequency gain sign and for unknown exosystems with unknown order uncertain frequencies on the basis of a known bound on system uncertainties and a known bound on the modeled exosystem frequencies a new adaptive output error feedback control algorithm is proposed which guarantees exponential convergence of both the output and the control input errors into residual bounds which decrease as the exosystem modeling error decreases exponential convergence of both errors to zero is obtained when the regulator exactly models all exosystem excited frequencies while asymptotic convergence of both errors to zero is achieved when the actual exosystem is overmodeled by the regulator the new algorithm generalizes existing learning controllers since in the case of periodic references and or disturbances the knowledge of the period is not required

brief damping of harmonic disturbances in sampled data systems parameterization of all optimal controllers optimal damping of harmonic disturbances of known frequencies is studied for sampled data systems a sampled data output feedback controller is designed to minimize the intersample variations of the controlled variable the set of all stabilizing optimal controllers is obtained in terms of the youla parameterization and a set of interpolation conditions at the disturbance frequencies which ensure that the stationary cost is minimized

asymptotic rejection of finite frequency modes of general periodic disturbances in output feedback nonlinear systems this paper deals with asymptotic rejection of individual frequency modes in nonlinear systems under general periodic disturbances an iterative observer design is proposed for disturbance estimation and it is shown that the proposed observer design successfully extracts individual frequency modes from general periodic disturbances the proposed iterative observer design is then applied to asymptotic rejection of nonlinear systems under general periodic disturbances

method of and device for coding speech signals with analysis by synthesis techniques the set of possible excitation signals is subdivided into a plurality of subsets the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network while the others supply a contribution which when added to that of the first subset causes a rate increase by successive steps at the receiving side a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used

brief paper design of noise and period time robust high order repetitive control with application to optical storage repetitive control is useful if periodic disturbances or setpoints act on a control system perfect asymptotic disturbance rejection is achieved if the period time is exactly known the improved disturbance rejection at the periodic frequency and its harmonics is achieved at the expense of a degraded system sensitivity at intermediate frequencies a convex optimization problem is defined for the design of high order repetitive controllers where a trade off can be made between robustness for changes in the period time and for reduction of the error spectrum in between the harmonic frequencies the high order repetitive control algorithms are successfully applied in experiments with the tracking control of a cd player system

harmonic generation in adaptive feedforward cancellation schemes the paper investigates the generation of harmonics in adaptive feedforward cancellation schemes specifically it is shown that an algorithm designed to reject a certain number of frequency components may in fact be capable of reducing higher order harmonics as well this effect originates in the time variation of the adaptive parameters a consequence is that the adaptive system may perform better than the algorithm with the fixed ideal parameters surprisingly the response to higher order harmonics can be calculated precisely using a laplace transform analysis the origin of the numerical procedure is in the equivalence between the adaptive feedforward cancellation scheme and a control scheme based on the internal model principle the implication of the results on design and implementation issues is discussed

adaptive algorithms for the rejection of sinusoidal disturbances with unknown frequency abstract two algorithms are presented for the rejection of sinusoidal disturbances with unknown frequency the first is an indirect algorithm where the frequency of the disturbance is estimated and the estimate is used in another adaptive algorithm that adjusts the magnitude and phase of the input needed to cancel the effect of the disturbance a direct algorithm that uses the concept of a phase locked loop is also presented in which frequency estimation and disturbance cancellation are performed simultaneously approximate analyses are presented for both schemes and the results are found useful for the selection of the design parameters simulations are given which demonstrate the validity of the analytical results and the ability of the algorithms to reject sinusoidal disturbances with unknown frequency the indirect algorithm is found to have a larger capture region for the parameter estimates whereas the direct algorithm has superior convergence properties locally about the optimum parameter estimates

identification of periodic signals with uncertain frequency this paper presents a new algorithm to identify periodic signals with uncertain frequency the approach is based on the behavior of an internal model in an error feedback system as such the signal is fed to a fictitious plant with a feedback controller the feedback controller is based on a traditional controller in parallel with an internal model which identifies and cancels the periodic disturbances under ideal circumstances the phase plot of the states of the internal model form an ellipse the speed of rotation about this ellipse can be used to calculate the difference between the nominal frequency of the model and the true frequency of the periodic signal an integral controller or a least squares estimator can be used to drive this error to zero simulations demonstrate the validity of this approach with time varying frequency and the algorithm is then applied to some data collected from a spot welder that has been corrupted by a sinusoidal signal whose frequency is between 1 hz and 5 khz

asymptotic rejection of general periodic disturbances in output feedback nonlinear systems this note deals with asymptotic rejection of general periodic disturbances in nonlinear systems a novel approach based on integrations over the half of disturbance period is proposed to estimate disturbances which may not be harmonic or even smooth the proposed algorithm uses the period and the knowledge of the basic disturbance patterns while the amplitude and phase of the disturbances are unknown and they are estimated based on the estimated disturbance the disturbance rejection algorithm is then proposed for linear systems and nonlinear systems in the output feedback format which asymptotically rejects the periodical disturbances the proposed algorithm extends asymptotic rejection of unknown disturbances from sinusoidal disturbances to a more general class of periodical disturbances which includes zero mean square waves symmetric triangular waves etc