• DocumentCode
    454647
  • Title

    On Variable Rate Frame Independent Predictive Speech Coding: Re-Engineering ILBC

  • Author

    Garrido, Christopher M. ; Murthi, Manohar N. ; Andersen, Soren Vang

  • Author_Institution
    Dept. of Electr. & Comput. Eng., Miami Univ., FL
  • Volume
    1
  • fYear
    2006
  • fDate
    14-19 May 2006
  • Abstract
    The Internet low bit-rate coder (iLBC) is now widely used for voice over Internet protocol (VoIP) applications. Unlike speech coders such as those based on code excited linear prediction (CELP), the iLBC achieves superior robustness to packet loss by avoiding inter-frame coding dependencies. While robustness to packet loss is essential, a VoIP codec should also possess the flexibility to change its source coding rate in order to counter network congestion and facilitate joint source channel coding for wireless channels. Previously, we presented a new variation of the iLBC encoding procedure which yielded a more efficient, rate-flexible result. In an effort to improve performance at lower source rates, we present various improvements to the original framework. Specifically, we reallocate bits from the adaptive codebook procedure; reduce the length of the start state vector; utilize an adaptive pulse gain quantization scheme; and extend the use of entropy coding. Overall, the various combined improvements result in the modified iLBC (with entropy coding) achieving a rate reduction of 2.0 to 2.9 kbps when compared to the original fixed-rate iLBC without any loss in quality. In comparisons with adaptive multi-rate (AMR), the modified iLBC coder remarkably exhibits equivalent perceptual evaluation of speech quality (PESQ) scores as the AMR coder at 10.2 and 12.2 kbps, and out-performs AMR for all packet loss rates. This is a significant result as the modified iLBC performs equivalent to AMR without exploiting inter-frame redundancies
  • Keywords
    Internet telephony; adaptive codes; combined source-channel coding; entropy codes; speech coding; Internet low bit-rate coder; VoIP; adaptive codebook; adaptive pulse gain quantization scheme; entropy coding; inter-frame coding; joint source channel coding; network congestion; perceptual evaluation of speech quality; predictive speech coding; variable rate frame; voice over Internet protocol; Channel coding; Codecs; Counting circuits; Entropy coding; Internet telephony; Quantization; Robustness; Source coding; Speech analysis; Speech coding;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech and Signal Processing, 2006. ICASSP 2006 Proceedings. 2006 IEEE International Conference on
  • Conference_Location
    Toulouse
  • ISSN
    1520-6149
  • Print_ISBN
    1-4244-0469-X
  • Type

    conf

  • DOI
    10.1109/ICASSP.2006.1660121
  • Filename
    1660121