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The concepts of digital signal processing are playing an increasingly important role in the area of multirate signal processing, i s . signal processing algorithms that involve more than one sampling rate. In this paper we present a tutorial overview of multirate digital signal processing as applied to systems for decimation and interpolation. We fmt discuss a theoretical model for such systems (based on the sampling theorem) and then show how various structures can be derived to provide efficient implementations of these systems. Design techniques for the linear-time-invariant components of these systems (the digital fdter) are discussed, and finally the ideas behind multistage implementations for increased efficiency are presented.

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A rationale is advanced for digitally coding speech signals in terms of sub‐bands of the total spectrum. The approach provides a means for controlling and reducing quantizing noise in the coding. Each sub‐band is quantized with an accuracy (bit allocation) based upon perceptual criteria. As a result, the quality of the coded signal is improved over that obtained from a single full‐band coding of the total spectrum. In one implementation, the individual sub‐bands are low‐pass translated before coding. In another, “integer‐band” sampling is employed to alias the signal in an advantageous way before coding. Other possibilities extend to complex demodulation of the sub‐bands, and to representing the sub‐band signals in terms of envelopes and phase‐derivatives. In all techniques, adaptive quantization is used for the coding, and a parsimonious allocation of bits is made across the bands. Computer simulations are made to demonstrate the signal qualities obtained for codings at 16 and 9.6 kb/s.