• DocumentCode
    1103524
  • Title

    Bit allocation in time and frequency domains for predictive coding of speech

  • Author

    Honda, Masaaki ; Itakura, Fumitada

  • Author_Institution
    Nippon Telegraph and Telephone Corporation, Musashino, Tokyo, Japan
  • Volume
    32
  • Issue
    3
  • fYear
    1984
  • fDate
    6/1/1984 12:00:00 AM
  • Firstpage
    465
  • Lastpage
    473
  • Abstract
    Adaptive predictive coding with dynamic bit allocation is presented for speech encoding at low to medium bit rates (6.4 kbits/s to 16 kbits/s). In this system, a split-band predictive coding scheme and a bit allocation scheme are employed in order to remove the redundancies due to a periodic concentration of the prediction residual energy, as well as the nonuniform nature of the speech spectrum. Quantization bits are dynamically allocated, both over the subbands (in the frequency domain) and over the subintervals (in the time domain), in accordance with the distribution of the residual energies in the time-frequency domain. Optimum bit allocation is derived based on the mean square error criterion on the speech waveform. The SNR gain is presented as the sum of the spectral SNR gain Gf, equivalent to the prediction gain, and the temporal SNR gain Gt. Although Gtis much smaller than Gf, temporal bit allocation greatly improves the actual SNR performance of the APC system to more than the value expected from its SNR gain in the bit rate range of less than 2 bits/sample. A study on the segmental SNR performance for various coder designs shows that the coder design using three subbands, four subintervals, and a fourth-order predictor in each subband is most appropriate for speech encoding in the bit rate range of 6.4 kbits/s to 16 kbits/s. This system is evaluated in terms of the segmental SNR and subjective speech quality. The results show that the system results in a substantial improvement compared with the conventional full-band APC system in regard to SNR performance and predictor loop stability. It is also shown that this system can provide speech quality subjectively equivalent to 7 bit log-PCM at 16 kbits/s, and to 6 bit log-PCM at 9.6 kbits/s.
  • Keywords
    Bit rate; Encoding; Frequency domain analysis; Mean square error methods; Performance gain; Predictive coding; Quantization; Redundancy; Speech coding; Time frequency analysis;
  • fLanguage
    English
  • Journal_Title
    Acoustics, Speech and Signal Processing, IEEE Transactions on
  • Publisher
    ieee
  • ISSN
    0096-3518
  • Type

    jour

  • DOI
    10.1109/TASSP.1984.1164350
  • Filename
    1164350