TY - GEN
T1 - On variable rate frame independent predictive speech coding
T2 - 2006 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2006
AU - Garrido, Christopher M.
AU - Murthi, Manohar N.
AU - Andersen, Søren Vang
PY - 2006
Y1 - 2006
N2 - 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 Muiti-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.
AB - 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 Muiti-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.
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M3 - Conference contribution
AN - SCOPUS:33947707326
SN - 142440469X
SN - 9781424404698
T3 - ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings
SP - I717-I720
BT - 2006 IEEE International Conference on Acoustics, Speech, and Signal Processing - Proceedings
Y2 - 14 May 2006 through 19 May 2006
ER -