On variable rate frame independent predictive speech coding

Re-engineering ILBC

Christopher M. Garrido, Manohar Murthi, Søren Vang Andersen

Research output: Chapter in Book/Report/Conference proceedingConference contribution

9 Citations (Scopus)

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 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.

Original languageEnglish
Title of host publicationICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings
Volume1
StatePublished - Dec 1 2006
Event2006 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2006 - Toulouse, France
Duration: May 14 2006May 19 2006

Other

Other2006 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2006
CountryFrance
CityToulouse
Period5/14/065/19/06

Fingerprint

Speech coding
coding
engineering
Internet
coders
Packet loss
Internet protocols
Entropy
Channel coding
Redundancy
entropy
linear prediction
congestion
state vectors

ASJC Scopus subject areas

  • Electrical and Electronic Engineering
  • Signal Processing
  • Acoustics and Ultrasonics

Cite this

Garrido, C. M., Murthi, M., & Andersen, S. V. (2006). On variable rate frame independent predictive speech coding: Re-engineering ILBC. In ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings (Vol. 1). [1660121]

On variable rate frame independent predictive speech coding : Re-engineering ILBC. / Garrido, Christopher M.; Murthi, Manohar; Andersen, Søren Vang.

ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. Vol. 1 2006. 1660121.

Research output: Chapter in Book/Report/Conference proceedingConference contribution

Garrido, CM, Murthi, M & Andersen, SV 2006, On variable rate frame independent predictive speech coding: Re-engineering ILBC. in ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. vol. 1, 1660121, 2006 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2006, Toulouse, France, 5/14/06.
Garrido CM, Murthi M, Andersen SV. On variable rate frame independent predictive speech coding: Re-engineering ILBC. In ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. Vol. 1. 2006. 1660121
Garrido, Christopher M. ; Murthi, Manohar ; Andersen, Søren Vang. / On variable rate frame independent predictive speech coding : Re-engineering ILBC. ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. Vol. 1 2006.
@inproceedings{e8a0c891334543feb02ae8bd560694a1,
title = "On variable rate frame independent predictive speech coding: Re-engineering ILBC",
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 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.",
author = "Garrido, {Christopher M.} and Manohar Murthi and Andersen, {S{\o}ren Vang}",
year = "2006",
month = "12",
day = "1",
language = "English",
isbn = "142440469X",
volume = "1",
booktitle = "ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings",

}

TY - GEN

T1 - On variable rate frame independent predictive speech coding

T2 - Re-engineering ILBC

AU - Garrido, Christopher M.

AU - Murthi, Manohar

AU - Andersen, Søren Vang

PY - 2006/12/1

Y1 - 2006/12/1

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.

UR - http://www.scopus.com/inward/record.url?scp=33947707326&partnerID=8YFLogxK

UR - http://www.scopus.com/inward/citedby.url?scp=33947707326&partnerID=8YFLogxK

M3 - Conference contribution

SN - 142440469X

SN - 9781424404698

VL - 1

BT - ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings

ER -