Fast method of channel equalisation for speech signals and its implementation on a DSP

Electronics Letters

Volumn 35, Issue 16, 1999, Pages 1309-1311

  Theobald, B ^a,b   Cox, S ^a   Cawley, G ^a   Milner, B ^a  

^a UNIVERSITY OF EAST ANGLIA   (United Kingdom)

^b MARTLESHAM HEATH   (United Kingdom)

Author keywords

[No Author keywords available]

Indexed keywords

ADAPTIVE FILTERING; ALGORITHMS; CIRCUIT THEORY; COMMUNICATION CHANNELS (INFORMATION THEORY); DIGITAL SIGNAL PROCESSING; EQUALIZERS; POLES AND ZEROS; TRANSFER FUNCTIONS;

BLIND EQUALIZATION; CHANNEL EQUALIZATION; NONLINEAR QUANTIZATION; SPEECH SIGNALS; STATIONARY POLES;

SPEECH PROCESSING;

EID: 0032637088     PISSN: 00135194     EISSN: None     Source Type: Journal    
DOI: 10.1049/el:19990912     Document Type: Article

References (5)

1. HAYKIN, s.: 'Adaptive filter theory' (Prentice-Hall, 1991)
2. STOCKHAM, T.G., and CANNON, T.M.: 'Blind deconvolution through digital signal processing', Proc. IEEE Trans. Acoust. Speech Signal Process., 1975, ASSP-63, (4)
3. SPENCER, P.S., and RAYNER, P.J.W.: 'Separation of stationary and time-varying systems and its application to the restoration of gramophone recordings'. Proc. IEEE Int. Symp. Circuits Syst., 1989, pp. 292-295
4. ATAL, B.S., and HANAUER, L.S.: 'Speech analysis and synthesis by linear prediction of the speech wave', J. Acoust. Soc. Am., 1971, 50, pp. 637-655
5. RABINER, L.R., and SCHAFER, R.W.: 'Digital processing of speech signals' (Prentice Hall, 1978)