Suppression of transients in variable recursive digital filters with a novel and efficient cancellation method

A new method for suppressing transients in recursive infinite impulse response (IIR) digital filters is proposed. The technique is based on modifying the state (delay) variables of the filter when coefficients are changed so that the filter enters a new state smoothly without transient attacks, as originally proposed by Zetterberg and Zhang (1988). In this correspondence, we modify the Zetterberg-Zhang algorithm to render it feasible for efficient implementation. We define a mean square error (MSE) measure for transients and determine the optimal transient suppressor to cancel the transients down to a desired level at the minimum complexity of implementation. The application of the method to all-pole and direct-form II (DF II) IIR filter sections is studied in detail. Time-varying recursive filtering with transient elimination is illustrated for tunable fractional delay filters and variable-bandwidth lowpass filters.

[1]  Yinong Ding,et al.  Filter Morphing of Parametric Equalizers and Shelving Filters for Audio Signal Processing , 1995 .

[2]  Jean-Marc Jot,et al.  Digital Signal Processing Issues in the Context of Binaural and Transaural Stereophony , 1995 .

[3]  W. Verhelst,et al.  A modified-superposition speech synthesizer and its applications , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[4]  John G. Proakis,et al.  Digital Signal Processing: Principles, Algorithms, and Applications , 1992 .

[5]  Julius O. Smith,et al.  J.O. Smith III Comments on Sullivan Karplus-Strong Article , 1991 .

[6]  Sanjit K. Mitra,et al.  Efficient audio coding using perfect reconstruction noncausal IIR filter banks , 1996, IEEE Trans. Speech Audio Process..

[7]  Vesa Välimäki,et al.  Energy-based effective length of the impulse response of a recursive filter , 1998, Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181).

[8]  John Princen The design of nonuniform modulated filterbanks , 1995, IEEE Trans. Signal Process..

[9]  Mark J. T. Smith,et al.  Nonuniform filter banks: a reconstruction and design theory , 1993, IEEE Trans. Signal Process..

[10]  Efstathios D. Kyriakis-Bitzaros,et al.  Theory and Real-Time Implementation of Time Varying Digital Audio Filters , 1990 .

[11]  Leonard T. Bruton,et al.  The design of N-band nonuniform-band maximally decimated filter banks , 1993, Proceedings of 27th Asilomar Conference on Signals, Systems and Computers.

[12]  K. W. Cattermole Theory and Application of the Z-Transform Method , 1965 .

[13]  L. H. Zetterberg,et al.  Elimination of transients in adaptive filters with application to speech coding , 1988 .

[14]  V. Valimaki,et al.  Delayless signal smoothing using a median and predictive filter hybrid , 1996, Proceedings of Third International Conference on Signal Processing (ICSP'96).

[15]  Vesa Välimäki,et al.  Elimination of Transients in Time-Varying Allpass Fractional Delay Filteres with Applications to Digital Waveguide Modeling , 1995, ICMC.

[16]  P. P. Vaidyanathan,et al.  Cosine-modulated FIR filter banks satisfying perfect reconstruction , 1992, IEEE Trans. Signal Process..

[17]  John Mourjopoulos,et al.  A dedicated processor for time-varying digital audio filters , 1993 .

[18]  Udo Zoelzer,et al.  Strategies for Switching Digital Audio Filters , 1993 .

[19]  Vesa Välimäki,et al.  Physical Modeling of Plucked String Instruments with Application to Real-Time Sound Synthesis , 1996 .

[20]  Fabrizio Argenti,et al.  Design of pseudo-QMF banks with rational sampling factors using several prototype filters , 1998, IEEE Trans. Signal Process..

[21]  Vesa Vlimki,et al.  Discrete-Time Modeling of Acoustic Tubes Using Fractional Delay Filters , 1998 .

[22]  Matti Karjalainen,et al.  Implementation of fractional delay waveguide models using allpass filters , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[23]  P. P. Vaidyanathan,et al.  Role of Anticausal Inverses in Multirate Filter-Banks-Part 1: System Theoretic Fundamentals , 1993 .

[24]  Vesa V Alim Aki Discrete-Time Modeling of Acoustic Tubes Using Fractional Delay Filters , 1995 .

[25]  Jelena Kovacevic,et al.  Perfect reconstruction filter banks with rational sampling factors , 1993, IEEE Trans. Signal Process..

[26]  Nirmal K. Bose,et al.  Fast evaluation of an integral occurring in digital filtering applications , 1995, IEEE Trans. Signal Process..

[27]  Truong Q. Nguyen,et al.  On perfect-reconstruction allpass-based cosine-modulated IIR filter banks , 1994, Proceedings of IEEE International Symposium on Circuits and Systems - ISCAS '94.

[28]  Soo-Chang Pei,et al.  Design of 1-D and 2-D IIR eigenfilters , 1994, IEEE Trans. Signal Process..

[29]  Thierry Blu,et al.  An iterated rational filter bank for audio coding , 1996, Proceedings of Third International Symposium on Time-Frequency and Time-Scale Analysis (TFTS-96).

[30]  Rudolf Rabenstein Minimization of transient signals in recursive time-varying digital filters , 1988 .