This paper is a tutorial which describes "main stream" sonar digital signal processing functions along with the associated implementation considerations. The attempt is to promote further cross-fertilization of ideas among digital signal processing applications in sonar, radar, speech, communications, seismology, and other related fields.

Digital signal processing for sonar

A unified approach to the analysis of periodically timevarying digital filters is introduced. It is shown that this method may be used to describe both time and frequency domain responses. By considering multirate filters as a special class of periodically time-varying filters we have characterized the frequency response of a general multirate filter. A time domain synthesis procedure for periodically varying difference equations is also presented.

A unified analysis of multirate and periodically time-varying digital filters

http://ieeexplore.ieee.org/iel6/53/26079/01162225.pdf

The acoustics, speech, and signal processing society - A historical perspective

In this paper a general theory of multistage decimators and interpolators for sampling rate reduction and sampling rate increase is presented. A set of curves and the necessary relations for optimally designing multistage decimators is also given. It is shown that the processes of decimation and interpolation are duals and therefore the same set of design curves applies to both problems. Further, it is shown that highly efficient implementations of narrow-band finite impulse response (FIR) filters can be obtained by cascading the processes of decimation and interpolation. Examples show that the efficiencies obtained are comparable to those of recursive elliptic filter designs.

Optimum FIR digital filter implementations for decimation, interpolation, and narrow-band filtering

The application of modern signal processing techniques to the production and processing of musical sound gives the composer and musician a level of freedom and precision of control never before obtainable. This paper surveys the use of analyis of natural sounds for synthesis, the use of speech and vocoder techniques, methods of artificial reverberation, the use of discrete summation formulae for highly efficient synthesis, the concept of the all-digital recording studio, and discusses the role of special-purpose hardware in digital music synthesis, illustrated with two unique digital music synthesizers.

Signal processing aspects of computer music: A survey

The design of bandwidth-limiting filters for the purpose of sample-rate reduction is considered. Realization of linear-phase finite-duration impulse-response (FIR) filters for this application by direct convolution is shown to be more efficient than the recursive realization [1]. The degree to which the Nyquist rate (relative to the desired signal bandwidth) must be exceeded at the filter output in order to avoid aliasing is suggested as a measure of filter effectiveness. Direct convolution is faster than the fast convolution for FIR equiripple [2] filters designed to operate within 10 percent of the Nyquist rate with 60- to 70-dB stopband attenuation at a 2:1 sample-rate reduction. This advantage improves with the log of the sample-rate reduction ratio. Several comparisons made with recursive realizations of elliptic filters give the advantage to direct convolutional realization of FIR filters for sampling within about 20 percent of the Nyquist rate at 60- to 70-dB attenuation. Elliptic filters become more efficient at higher complexities (of about eight poles and eight zeros). Two design techniques that exploit the reduced output sample rate in the design of FIR filters by direct convolution are suggested. The effects of quantization of FIR filter coefficients on the frequency response are considered and several examples illustrated.

A. Crooke

Papers

Digital filters for sample-rate reduction