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TMS320C6713 DSK Implementation of G.711 Coded VoIP Signal

by Imran Ghous, Habibullah Jamal, Tahir Muhammad
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International Journal of Computer Applications
Foundation of Computer Science (FCS), NY, USA
Volume 65 - Number 18
Year of Publication: 2013
Authors: Imran Ghous, Habibullah Jamal, Tahir Muhammad
10.5120/11028-6434

Imran Ghous, Habibullah Jamal, Tahir Muhammad . TMS320C6713 DSK Implementation of G.711 Coded VoIP Signal. International Journal of Computer Applications. 65, 18 ( March 2013), 45-53. DOI=10.5120/11028-6434

@article{ 10.5120/11028-6434,
author = { Imran Ghous, Habibullah Jamal, Tahir Muhammad },
title = { TMS320C6713 DSK Implementation of G.711 Coded VoIP Signal },
journal = { International Journal of Computer Applications },
issue_date = { March 2013 },
volume = { 65 },
number = { 18 },
month = { March },
year = { 2013 },
issn = { 0975-8887 },
pages = { 45-53 },
numpages = {9},
url = { https://ijcaonline.org/archives/volume65/number18/11028-6434/ },
doi = { 10.5120/11028-6434 },
publisher = {Foundation of Computer Science (FCS), NY, USA},
address = {New York, USA}
}
%0 Journal Article
%1 2024-02-06T21:20:35.056133+05:30
%A Imran Ghous
%A Habibullah Jamal
%A Tahir Muhammad
%T TMS320C6713 DSK Implementation of G.711 Coded VoIP Signal
%J International Journal of Computer Applications
%@ 0975-8887
%V 65
%N 18
%P 45-53
%D 2013
%I Foundation of Computer Science (FCS), NY, USA
Abstract

The quality of speech signal over a VoIP system is degraded by various network layer problems which include jamming, jitter, packet loss. Different types of noises also degrade the quality of speech signal such as external noise and quantization noise. This paper improves the quality of VoIP speech signal affected by these noises and network layer problems. The quality of degraded VoIP speech signal coded by using ITU-T G. 711 audio coding standard and implemented on the TMS320C6713 DSK has been compared with the quality of VoIP speech signal coded by using G. 729 audio data compression standard which uses code-excited linear prediction speech coding (CS-ACELP) for coding purpose and is implemented on TMS320C6713 DSK. Speech Enhancement Algorithm has been proposed for the improvement in the quality of VoIP signal. In order to evaluate the performance PESQ (ITU-T P. 862, Perceptual Evaluation of Speech Quality) is used.

References
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Index Terms

Computer Science
Information Sciences

Keywords

Noise Filtering G. 711 Speech Enhancement Algorithm TMS320C6713 DSK VoIP DSP Implementation FIR Hanning Window Filter

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