• DocumentCode
    281575
  • Title

    Modern developments in speech coding

  • Author

    Xydeas, C.

  • Author_Institution
    Dept. of Electr. Eng., Manchester Univ., UK
  • fYear
    1989
  • fDate
    32566
  • Firstpage
    42370
  • Lastpage
    42377
  • Abstract
    Modern speech coding algorithms are based on models which attempt to successfully describe short-term speech signals. These models can be defined in the `time´ or in the `frequency´ domain and analysis by synthesis (AbS) estimation procedures are often used to determine the short term model parameters. The parameters can then be vector quantised if the coder is designed to operate at low bit rates (less than 8 kbits/sec). Frequency domain algorithms decompose the speech signal into sinusoidal components, with time varying amplitudes and frequencies, and hence the speech is modelled as a time varying line spectrum. Medium to low bit rate frequency domain coding systems include: adaptive transform coding, subband coding, harmonic coding, sinusoidal transform, multiple excitation, codebook excitation, backward excitation recovery
  • Keywords
    analogue-digital conversion; encoding; frequency-domain analysis; speech analysis and processing; time-domain analysis; adaptive transform coding; analysis by synthesis; backward excitation recovery; codebook excitation; estimation procedures; frequency domain; harmonic coding; multiple excitation; short-term speech signals; sinusoidal transform; speech coding; subband coding; time domain;
  • fLanguage
    English
  • Publisher
    iet
  • Conference_Titel
    Spectral Estimation Techniques for Speech Processing, IEE Colloquium on
  • Conference_Location
    London
  • Type

    conf

  • Filename
    197934