Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP

During last two decades various speech coding algorithms have been developed. The range of toll speech frequency is from 300 Hz- 3400 Hz. Generally, human speech signal could be classified as non-stationary signal because of its fluctuation randomly over the time axis. One important assumption made...

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Main Author:	Talib Alshammari, Wissam
Format:	Thesis
Language:	English English
Published:	2012
Subjects:	T Technology (General) TJ Mechanical engineering and machinery
Online Access:	http://eprints.utem.edu.my/id/eprint/24122/1/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP%20-%20Wissam%20Talib%20Alshammari%20-%2024%20Pages.pdf http://eprints.utem.edu.my/id/eprint/24122/2/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP.pdf
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id	my-utem-ep.24122
record_format	uketd_dc
spelling	my-utem-ep.241222022-03-16T11:49:59Z Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP 2012 Talib Alshammari, Wissam T Technology (General) TJ Mechanical engineering and machinery During last two decades various speech coding algorithms have been developed. The range of toll speech frequency is from 300 Hz- 3400 Hz. Generally, human speech signal could be classified as non-stationary signal because of its fluctuation randomly over the time axis. One important assumption made to make the analysis of such signal even easier by assuming the speech signal is quasi-stationary over short range (frame). The frames of speech signal can be classified further into Voiced or Unvoiced, where the voiced part is quasi-stationary while the unvoiced part as an AWGN. The quality of the synthesized signal is degraded significantly due to the excitation of voiced part not equally spaced within the frame and the excitation of the unvoiced part is not exact AWGN. This assumption produced a non-natural speech signal but with high intelligible level. One more reason is that the frame could have voiced plus unvoiced parts within the same frame, and by classifying this frame as voiced or unvoiced due to rigid decision would drop the level of quality significantly. Speech compression commonly referred to as speech coding, where the amount of redundancies is reduced, and represent the speech signal by set of parameters in order to have very low bit rates. One of these speech coding algorithms is linear predictive coding (LPC-10). This thesis implements LPC-10 analysis and synthesis using Matlab and C coding. LPC-10 have been compared with some other speech compression algorithms like pulse code modulation (PCM), differential pulse code modulation (DPCM), and code excited linear prediction coding (CELP), in term of segmental signal to quantization noise ratio SEG-SQNR and mean squared error MSE using Matlab simulation. The focus on LPC-10 was implemented on the DSP board TMS320C6713 to test the LPC-10 algorithm in realtime. Real-time implementation on TMS320C6713 DSP board required to convert the Matlab script into C code on the DSP Board. Upon successfully completion, comparison of the results using TMS320C6713 DSP against the simulated results using Matlab in both graphical and tabular forms were made. 2012 Thesis http://eprints.utem.edu.my/id/eprint/24122/ http://eprints.utem.edu.my/id/eprint/24122/1/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP%20-%20Wissam%20Talib%20Alshammari%20-%2024%20Pages.pdf text en public http://eprints.utem.edu.my/id/eprint/24122/2/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP.pdf text en validuser http://plh.utem.edu.my/cgi-bin/koha/opac-detail.pl?biblionumber=93414 mphil masters Universiti Teknikal Malaysia Melaka Faculty of Electronics and Computer Engineering
institution	Universiti Teknikal Malaysia Melaka
collection	UTeM Repository
language	English English
topic	T Technology (General) TJ Mechanical engineering and machinery
spellingShingle	T Technology (General) TJ Mechanical engineering and machinery Talib Alshammari, Wissam Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
description	During last two decades various speech coding algorithms have been developed. The range of toll speech frequency is from 300 Hz- 3400 Hz. Generally, human speech signal could be classified as non-stationary signal because of its fluctuation randomly over the time axis. One important assumption made to make the analysis of such signal even easier by assuming the speech signal is quasi-stationary over short range (frame). The frames of speech signal can be classified further into Voiced or Unvoiced, where the voiced part is quasi-stationary while the unvoiced part as an AWGN. The quality of the synthesized signal is degraded significantly due to the excitation of voiced part not equally spaced within the frame and the excitation of the unvoiced part is not exact AWGN. This assumption produced a non-natural speech signal but with high intelligible level. One more reason is that the frame could have voiced plus unvoiced parts within the same frame, and by classifying this frame as voiced or unvoiced due to rigid decision would drop the level of quality significantly. Speech compression commonly referred to as speech coding, where the amount of redundancies is reduced, and represent the speech signal by set of parameters in order to have very low bit rates. One of these speech coding algorithms is linear predictive coding (LPC-10). This thesis implements LPC-10 analysis and synthesis using Matlab and C coding. LPC-10 have been compared with some other speech compression algorithms like pulse code modulation (PCM), differential pulse code modulation (DPCM), and code excited linear prediction coding (CELP), in term of segmental signal to quantization noise ratio SEG-SQNR and mean squared error MSE using Matlab simulation. The focus on LPC-10 was implemented on the DSP board TMS320C6713 to test the LPC-10 algorithm in realtime. Real-time implementation on TMS320C6713 DSP board required to convert the Matlab script into C code on the DSP Board. Upon successfully completion, comparison of the results using TMS320C6713 DSP against the simulated results using Matlab in both graphical and tabular forms were made.
format	Thesis
qualification_name	Master of Philosophy (M.Phil.)
qualification_level	Master's degree
author	Talib Alshammari, Wissam
author_facet	Talib Alshammari, Wissam
author_sort	Talib Alshammari, Wissam
title	Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
title_short	Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
title_full	Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
title_fullStr	Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
title_full_unstemmed	Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP
title_sort	real-time implementation of lpc-10 codec on tms320c6713 dsp
granting_institution	Universiti Teknikal Malaysia Melaka
granting_department	Faculty of Electronics and Computer Engineering
publishDate	2012
url	http://eprints.utem.edu.my/id/eprint/24122/1/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP%20-%20Wissam%20Talib%20Alshammari%20-%2024%20Pages.pdf http://eprints.utem.edu.my/id/eprint/24122/2/Real-Time%20Implementation%20Of%20LPC-10%20Codec%20On%20TMS320C6713%20DSP.pdf
_version_	1747834068688437248

Real-Time Implementation Of LPC-10 Codec On TMS320C6713 DSP

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