Digital Speech: Coding for Low Bit Rate Communication by A. M. Kondoz

By A. M. Kondoz

Construction at the good fortune of the 1st version Digital Speech deals huge new, up to date and revised fabric established upon the newest study. This Second Edition keeps to supply the basic technical heritage required for low bit expense speech coding and the most well liked advancements in electronic speech coding concepts which are acceptable to evolving communique structures.

  • Features new chapters on Pitch Estimation and Voice-Unvoiced type of Speech, Harmonic Speech Coding and Multimode Speech Coding
  • Presents a comprehensively revised bankruptcy entitled Analysis by means of Synthesis LPC Coding together with particular examples of renowned speech coders resembling CELP (Code-Excited Linear Predictive) Coding
  • Contains an up-to-date bankruptcy on effective LPC Quantization tools together with MSVQ and anti-aliasing filtering
  • Discusses Voice job Detection (VAD) methods
  • Offers increased assurance of speech enhancement strategies akin to echo cancellation and noise suppression

Written by means of a widely known, hugely revered educational, this authoritative quantity may be worthwhile to practicing engineers, community designers, desktop scientists and complicated scholars in communications, electric and digital engineering.

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Extra resources for Digital Speech: Coding for Low Bit Rate Communication Systems, 2nd Edition

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A. Jasiuk (1990) ‘Vector sum excited linear prediction (VSELP) speech coding at 8 kb/s’, in Proc. of Int. Conf. , Speech and Signal Processing, pp. 461–4. April 1990. Albuquerque, NM, USA [43] T. Honkanen, J. Vainio, K. Jarvinen, and P. Haavisto (1997) ‘Enhanced full rate speech coder for IS-136 digital cellular system’, in Proc. of Int. Conf. , Speech and Signal Processing, pp. 731–4. May 1997. Munich [44] T. E. Tremain (1982) ‘The government standard linear predictive coding algorithm: LPC-10’, in Speech Technology, 1:40–9.

K. Paliwal (Eds), pp. 49–78. Amsterdam: Elsevier Science [51] W. Wong, R. Mack, B. Cheetham, and X. Sun (1996) ‘Low rate speech coding for telecommunications’, in BT Technol. , 14(1):28–43. 1 Introduction In digital communication systems, signal processing tools require the input source to be digitized before being processed through various stages of the network. The digitization process consists of two main stages: sampling the signal and converting the sampled amplitudes into binary (digital) codewords.

49–78. Amsterdam: Elsevier Science [51] W. Wong, R. Mack, B. Cheetham, and X. Sun (1996) ‘Low rate speech coding for telecommunications’, in BT Technol. , 14(1):28–43. 1 Introduction In digital communication systems, signal processing tools require the input source to be digitized before being processed through various stages of the network. The digitization process consists of two main stages: sampling the signal and converting the sampled amplitudes into binary (digital) codewords. The difference between the original analogue amplitudes and the digitized ones depend on the number of bits used in the conversion.

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