Code excited linear predictive speech coding implementation on DSP

CELP or Code Excited Linear Prediction is capable of coding a relatively good quality speech at low bit rates, mainly having 4800 bps, for the FS1016 Standard. Since CELP algorithm involves complex mathematical computations, familiarization of different theories is done through MATLAB simulations. T...

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Main Authors: Anunciacion, Amiel, Casquero, Mark Anthony, Ching, Carolyn Grace, Onan, Jaquelyn
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Language:English
Published: Animo Repository 2000
Online Access:https://animorepository.dlsu.edu.ph/etd_bachelors/11882
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Institution: De La Salle University
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spelling oai:animorepository.dlsu.edu.ph:etd_bachelors-125272021-09-08T03:55:53Z Code excited linear predictive speech coding implementation on DSP Anunciacion, Amiel Casquero, Mark Anthony Ching, Carolyn Grace Onan, Jaquelyn CELP or Code Excited Linear Prediction is capable of coding a relatively good quality speech at low bit rates, mainly having 4800 bps, for the FS1016 Standard. Since CELP algorithm involves complex mathematical computations, familiarization of different theories is done through MATLAB simulations. This thesis incorporates the CELP algorithm into two DSP processors. The Matlab code is translated to C code which is compiled into the TMS320C54X speech into 144 bits per frame. The encoded bits are transmitted to the other EVM via connecting wires. The bits received by the second EVM are decoded and processed into a synthesized speech. After processing, the final decompressed speech values are monitored using the C source debugger. These values are manually obtained and are made play in Matlab. The main purpose of this thesis is to implement the CELP system, with encoder and decoder onto two DSP Evaluation Modules. The efficiency of the system/quality of output speech is determined quantitatively by obtaining the signal to quantizing noise ratio of the input versus the output speech. the result is audible speech. The analyzer part of the CELP system is not implemented on the DSP chip due to the processor's fixed point and program complexity limitations. Moreover, module interface failed because of technical problems in synchronization. However, manual processing is successful since the sample by sample speech values obtained from the debugger is played audibly when transferred to Matlab. The main purpose of this thesis is to implement the CELP system, with encoder and decoder onto two DSP Evaluation Modules. The efficiency of the system/quality of output speech is determined quantitatively by obtaining the signal to quantizing noise ratio of the input versus the output speech. The result is audible speech. The analyzer part of the CELP system is not implemented on the DSP chip due to the processor's fixed point and program complexity limitations. Moreover, module interface failed because of technical problems in synchronization. However, manual processing is successful since the sample by sample speech values obtained from the debugger is played audibly when transferred to Matlab. 2000-01-01T08:00:00Z text https://animorepository.dlsu.edu.ph/etd_bachelors/11882 Bachelor's Theses English Animo Repository
institution De La Salle University
building De La Salle University Library
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content_provider De La Salle University Library
collection DLSU Institutional Repository
language English
description CELP or Code Excited Linear Prediction is capable of coding a relatively good quality speech at low bit rates, mainly having 4800 bps, for the FS1016 Standard. Since CELP algorithm involves complex mathematical computations, familiarization of different theories is done through MATLAB simulations. This thesis incorporates the CELP algorithm into two DSP processors. The Matlab code is translated to C code which is compiled into the TMS320C54X speech into 144 bits per frame. The encoded bits are transmitted to the other EVM via connecting wires. The bits received by the second EVM are decoded and processed into a synthesized speech. After processing, the final decompressed speech values are monitored using the C source debugger. These values are manually obtained and are made play in Matlab. The main purpose of this thesis is to implement the CELP system, with encoder and decoder onto two DSP Evaluation Modules. The efficiency of the system/quality of output speech is determined quantitatively by obtaining the signal to quantizing noise ratio of the input versus the output speech. the result is audible speech. The analyzer part of the CELP system is not implemented on the DSP chip due to the processor's fixed point and program complexity limitations. Moreover, module interface failed because of technical problems in synchronization. However, manual processing is successful since the sample by sample speech values obtained from the debugger is played audibly when transferred to Matlab. The main purpose of this thesis is to implement the CELP system, with encoder and decoder onto two DSP Evaluation Modules. The efficiency of the system/quality of output speech is determined quantitatively by obtaining the signal to quantizing noise ratio of the input versus the output speech. The result is audible speech. The analyzer part of the CELP system is not implemented on the DSP chip due to the processor's fixed point and program complexity limitations. Moreover, module interface failed because of technical problems in synchronization. However, manual processing is successful since the sample by sample speech values obtained from the debugger is played audibly when transferred to Matlab.
format text
author Anunciacion, Amiel
Casquero, Mark Anthony
Ching, Carolyn Grace
Onan, Jaquelyn
spellingShingle Anunciacion, Amiel
Casquero, Mark Anthony
Ching, Carolyn Grace
Onan, Jaquelyn
Code excited linear predictive speech coding implementation on DSP
author_facet Anunciacion, Amiel
Casquero, Mark Anthony
Ching, Carolyn Grace
Onan, Jaquelyn
author_sort Anunciacion, Amiel
title Code excited linear predictive speech coding implementation on DSP
title_short Code excited linear predictive speech coding implementation on DSP
title_full Code excited linear predictive speech coding implementation on DSP
title_fullStr Code excited linear predictive speech coding implementation on DSP
title_full_unstemmed Code excited linear predictive speech coding implementation on DSP
title_sort code excited linear predictive speech coding implementation on dsp
publisher Animo Repository
publishDate 2000
url https://animorepository.dlsu.edu.ph/etd_bachelors/11882
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