Multirate signal processing by P. M. Grant Multirate digital signal processing, in which the sample rate is changed to match the bandwidth of the signal at different points within a system, has grown in importance in recent years. It is a key technique in the efficient design of transmultiplexers and digital filterbanks with applications in spectrum analysis, digital audio systems, speech and image coding, among others. This paper introduces the underlying concepts of decimation and interpolation and describes the techniques for minimising degradations caused by the aliases and images which they introduce. Examples of the application of these techniques are presented. 1. Introduction Mulirate systems"? difer from singlerate processing systems nthat the sample rate isakered at diferent places Within the system. The motivation for this isto achieve the ‘most appropriate value for the sampling rte, ‘commensurate withthe signal bandwidth, to minimise the {otal number of processor operations. As the bandwidth of signal varies following filtering, changes can be made in the sample rate from pointtepoint within the processor provided thatthe Nyquist criterion is always satisfied. The aim is to operat, at each point in the processor, at the lowest sampling rate possible, without introducing aliasing effects, Variable samplerate processing is widely used in the ‘discrete Fourier ransform (DFT).*'As the bandwidth of each ofthe N parallel channels in a DFT filterbank is only V/Mth of the total input signal bandwidth, the output ‘sample rate in each channel can be reduced by the factor N a process somewhat erroneously called ‘decimation’ — Without incurring any los of information. Over the last two decades there have beea major advances in the design of variable samplerate systems. These systems find application in communications, speech processing, spectrum analysis, ralar systems, and antenna systems. This paper inially reviews the basic concepts of variable samplerate signal processing; the later sections ‘are devoted to a discussion of the major variable sample- rate digital signal processing (DSP) applications 2. Decimation, interpolation, imaging and ing Fig. shows he symbols fr the baste decimation (sample. rate reducing) and interpolation (sample-ate increasing) ‘components. Using the normal notation the Mold decimatoris characterised by the input-output relation: soln) =x(0) o ‘which implies that the output at sample is equal t the ingat at sample Mx. As a consequence, only the input samples with sample numbers equal toa multiple of Mare retained a the output and the remainder are disc i Fig. 1 Variable sample-ate processing operations: (@) M4old decimator (6) -fold interpolator Fig. 2 Time domain signals: (a) before decimation; (6) after decimation by two 4 ELECTRONICS & COMMUNICATION ENGINEERING JOURNAL FEBRUARY 1996‘Sample-ate reduction by a factor of M is demonstrated in Fig. 2for the ease of M=2, where every second sample is discarded, (This figure shows the input and output signals ‘on the same absolute time scale as would be seen on an oscilloscope.) Fig. 3a showsa representation ofthe spectrum, X(o), of ‘the original sampled input signal Fig. 2a), which is also conveniently drawn here withthe same triangular shape. ‘The solidsine component represents the spectrum ofthe ‘unsampled signal. The dashed components are some of the aliasing terms due to sampling. Decimation by 2, Fig. 2 causes the output sample rate to be halved and hence the separation between the unsampled_ signal spectrum and thealis component centred onthe sampling frequency is aiso halved, Fig, 3. The reduced output ‘sample rate Of fogya/2 thus moves the aliased frequency domain components to half the frequency of their predecimation position.” In Fig. 3¢ this separation has ‘been quartered and the alias components centred on two and three times the sampling frequency also overlap the base spectral information, which is represented by the sold linesin Fig. 3. Inthe decimation by M operation M~ 1 ‘uniformly shifted components wil thus overlap the input signal spectrum. However if() is bandlimited to [ol 2/2 isnot small enough to be Aiscarded and we cannot decimate x(n) without casing significant Be te aliasing. It does seem unfortunate ee — that the smal faction of ener in Ht ia the highrecuency region should A Drevent us from obtaining any | mre a However the signal can be | conveniently split into two | frequency bands* by using an Le analysis bank, Fig. 166, The ay = Subband signal) now has much “ if Jess ener than) and so canbe encoded with many fewer bits than 4y(n).Asan example, letx(9) bea 10 Signal spit into Kz bandwidth signal normally Sora chan Fig. 17 _Proctical subband speech coder. Typically 92-5 bivsample input "io 5 subbands of width 05-1 ki? for transmission fovera channel at 16-32 KBs ELECTRONICS & COMMUNICATION ENGINEERING JOURNAL FEBRUARY 1996 uter Gress BS dere ia EkconiEageig fete Herat Une sd' Ri de rom (Gero Engh He wre in reecorunios fa ney Congr bere movin te Unsere oth oy te penises ae ‘rent Poe flere Pcs anise sites penance tnarcarap sb etn eet achieve an allpass response by using a quadrature mirror filter where FZ(o) comprises the image (about 2/2 — hence the name ‘quadrature’ of the Hi(o) response. The allased distortions due to interband leakage and images are then exactly compensated when the channel signals are combined following the synthesis fltors.™ Further generalisations follow immediately: che signal can be split into. subbands, Fig 17, with each subband ‘signal decimated by M and independently quantised, The coding in each subband is typically more sophisticated ‘han just quantisation. Fig. 17 provides typical numbers of bit/sample for 16-32 kbit/s tansmission-ate speech coding systems. Subband coding. techniques are also being applied to obiain higher Sdelity wideband (>4 kHz) telephony transmission while sil using only 64 Kits of| PCM orevea lower transmission rates, Thisis achieved by coding over wider analogue bandwidths such as 8 kHz and spliting this into 0-4 Kitz and 4-8 KHz subbands for subsequent quantisation. Thistechniqueis used in the ITU 6.722 high-quality audio coder, which employs two sublands to code a wideband 8 kHz signal at 61/96/48 kbit/s. Thisisalow-complesty coder. implemented one ISP microprocessor, which can operate ata bit error rate of 10° for teleconferencing and loudspeaker telephone ‘applications “Two dimensional subband coding is also used for data compression of video and other image transmission systems, This is important for the digital storage of broadcast quay signals, for example high-definition TV 5 Summary This paper has provided an introduction tothe concepts of Aecimation and interpolation in variable samplerate DSP systems, where the sample rateis adjusted commensurate with the signal bandwidth The spectral images and aliases that are created by these operations have been deseribed and further techniques have been introduced to minimise these degradations. It has been shown that considerable simplification in computational complexity canbe obtained with variable sampleate techniques when implementing gh-speed signal samplers and iter banks The application ofthese techniques tothe analysis of speech and efficient encoding of ther signal types has also heen described. References 1 CROCHIERE, RE. and RABINER, LR: ‘Mulirate digital ‘Signal eocescing’ (Prentice Hal Engelwood Ci, NJ 1988) 2 VAIDYANATHAN, PP: Malate systems and filterbank (Prentice Hall, Englewood CEs, NJ, 1) 8 BRIGHAM, B.0:The as Fourier tansform! (Prentice Hal, 1974) 4 GRANT, PMC Digital signal processing Part: Dig ters and the DET, Bletron. & Comaan. Big. J, 198, 5, () poas-2 5 JACKSON, LB (awe 1986) 6 HAMMING, RM Digia ters renter Hal, 1958) 7 SANDLER, MB: Diglahtoanalogue conversion wing pase width molto Elston. & Commun. Bg J, 188,5, (5, pp39-348 8. SHELSWELL, P: The COFDM modklaton syste the heart of cigtal auto broadcasting letrm. & Commun. En. J 1986, 7, (9, pp 127-196 9 BLAIR. GM. review ofthe discrete Fourier transform. Part 1 Manialaing the powers, Electron. Commun. Em 41885, 7, (8), 99. 269-126, 10 VARY, P, and HEUTE, Us" shoretime specuum analyser ith polyphase network and DFT, Signal Processing January 1880, 2, 0). pp5-65 1. KAY, SM "Modern spectral estimation’ (Prentice-Hall, 1968) 12 MARPLE, SL ‘igh spectral analyse! (Prentice Hal, Englewood lil, NJ, 1987) 18 WAITE,3.W7'A malate bank of digital bandpass ers for scousic applications, Healt Pacha J, Ape 1985, 4, 2), mrss M4 COHEN, Li ‘Timedtequency analysis! Prentice Hal Baglewood Cli NI, 1995) 15 BENTLEY, PM, and MeDONNELL, JT.E: ‘Wavelet transforms an introduction’, Electron, © Commun, Eng J 1984, 6,0) ,pp.175188 16 MASSOPUST, PR: ‘Fractal functions, facta surfaces and wavelets (Academic Press, 18) 17 Special las, /EEE ASSP Mag, October 1985, 2, (0) 18 CROCHTERE, RE! Subband coding, Gell ix ‘Tok September 1961, 60, 7p 1685-1654 19 JAYANT, NS, and NOLL, 2 Digital cong of waveforms ‘Prentice, Englewood Cis, 154) ‘Digital fiers and signal processing! IEE: 1985 Received 22nd June 1985, Peter Grant is Professor of Electronic Signal Processing inthe Departoent of Electrical Engineering, University of Edinburgh ‘The King’s Bullings, Edinburgh, EH9 SJL, UR. He isan TEE Fellow 2 ELECTRONICS & COMMUNICATION ENGINEFRING IOURNAL FEBRUARY 1986