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  • GC-CCA 2011
  • 16.Cnf-142
Papers Published at GC-CCA 2011
All 9 Papers
IDAuthors and TitlePages
16.Cnf-64 Prof. Ahmed Dooguy Kora
Mr. Octave Ringar
Dr. Samuel Ouya
Prof. Jean-Pierre Cances
Wavelength division multiplexing of 1D-code division multiple access systems limitations in optical network using large spectrum sources
1-5
16.Cnf-87 Dr. Musaria Mahmood
Prof. Fawzi Al-Naima
Prof. Nikolaos Uzunoglu
OPNET simulation of data transmission network: a case study for control system in a petroleum refinery complex in Iraq
6-11
16.Cnf-118

GS Citations

Ms. Imane Benkhelifa
Dr. Samira Moussaoui
APPL: Anchor path planning based localization for wireless sensor networks
12-17
16.Cnf-124 Dr. Omar Batarfi
Mrs. Arwa Diwali
A Proposed Model for Certificate Validation in WPKI
18-22
16.Cnf-126 Mr. Badoui Yamine
Prof. Rahim Tafazolli
Dynamic Storage Method for the UMTS Radio Resource Control
23-27
16.Cnf-131 Mr. Amin Ebrahimzadeh
Dr. Javad Musevi Niya
Energy Efficient Centralized Clustering Hierarchical Routing Protocol for Wireless Sensor Networks
28-31
16.Cnf-142 Dr. Abdul_bary Suleiman
Dr. Mahmod Mahmod
Mr. Omar W. Hamdon
Real-Time TMS3206713 DSP-Based Adaptive Noise Cancellation for Speech Signal Processing
32-36
16.Cnf-145 Dr. Rafeeq A. Al-Hashemi
Prof. Hilal Saleh
Dr. Ahmed Alobaidi
Data Mining Implementation for Keywords Extraction
37-42
16.Cnf-147 Mr. Leonardo O. Iheme
On the information content of an image transmitted over fading channels using CC versus RS-CC coding
43-48
16.Cnf-142 Paper View Page
Title Real-Time TMS3206713 DSP-Based Adaptive Noise Cancellation for Speech Signal Processing
Authors Dr. Abdul_bary Suleiman, University of Mosul, Mosul, Iraq
Dr. Mahmod Mahmod, University of Mosul, Mosul, Iraq
Mr. Omar W. Hamdon, University of Mosul, Mosul, Iraq
Abstract Implementation of Real-time adaptive noise cancellation for strongly corrupted speech signal using the TMS3206713 DSP starter kit (DSK) is achieved. LMS algorithm is used for adaptation. The program code for the system is made in a MATLAB Simulink environment. From MATLAB the model is converted to DSP through Code Composer Studio,an integrated development environment for TI's digital signal processors. The analog input channel of the DSK is adjusted so as to satisfy the requirement of having two inputs for the adaptive noise canceller. The LMS parameters are chosen to give higher performance and improvement in signal to noise ratio of 4dB is obtained.
Track ISKE: Intelligent Systems and Knowledge Engineering
Conference 4th Mosharaka International Conference on Communications, Computers and Applications (MIC-CCA 2011)
Congress 2011 Global Congress on Communications, Computers and Applications (GC-CCA 2011), 22-24 July 2011, Istanbul, Turkey
Pages 32-36
Topics Learning Algorithms and Training
Knowledge-Based Systems
ISSN 2227-331X
DOI
BibTeX @inproceedings{142CCA2011,
title={Real-Time TMS3206713 DSP-Based Adaptive Noise Cancellation for Speech Signal Processing},
author={Abdul_bary Suleiman, and Mahmod Mahmod, and Omar W. Hamdon},
booktitle={2011 Global Congress on Communications, Computers and Applications (GC-CCA 2011)},
year={2011},
pages={32-36},
doi={}},
organization={Mosharaka for Research and Studies} }
Paper Views 65 Paper Views Rank 135/524
Paper Downloads 27 Paper Downloads Rank 182/524
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