• Proceedings of the KIEE Conference
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• 2007.04a
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• pp.140-142
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• 2007

This paper presents an effective method of precise velocity control at low speed with a low resolution encoder. Multirate observer to estimate the velocity at every DSP control period is used except a constant velocity mode. The observer corrects the estimation error when detects pulse signal. Unlike the conventional methods, the multirate estimator is stable at a low speed. However, the multirate estimator shows ripples at a constant velocity. Thus, in this paper we use a velocity prediction method which uses the present velocity from the previous average velocity to reject the ripple. In a summary, at a constant speed mode, the predicted velocity is used. Otherwise, the estimated velocity by the multirate obvserver is used. The effectiveness of the multirate observer and ripple rejection at low speed is verified through various simulations.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.52-57
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• 2005

The adaptive multirate wideband (AMR-WB) speech coder has an extended audio bandwidth from 50 Hz to 7 kBz and operates on nine speech coding bit-rates from 6.6 to 23.85 kbit/s. In this Paper, we present the real-time implementation of AMR-WB speech coder using 16bit fixed-point TMS320C5509 that has dual MAC units. Firstly, We implemented AMR-WB speech coder in C 1anguage level using intrinsics, and then performed optimization in assembly language. The computational complexity of the implemented AMR-WB coder at 23.85 kbit/s is 42.9 Mclocks. And this coder needs the program memory of 15.1 kwords, data ROM of 9.2 kwords and data RAM of 13.9 kwords.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1337-1344
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• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.43-50
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• 2010

DDC(Digital Down Converter) is a conversion technology to decimate to a lower sampling rate and DDC for the future development of communications technology has the necessary skills. So, it has been recognized in the wireless and the SDR(Software Defined Radio) system as essential components. In addition, research is underway on spectrum sensing for efficient communications environment due to the shortage of frequency resources. In this paper, the DDC systems were analyzed for CIC(Cascaded Integrator Comb) Filter, WDF(Wave Digital Filter), SRC(Sample Rate Conversion) each module. Moreover, we proposed a linkage effectively between DDC system and Spectrum Sensing for improve the efficiency of use of frequency by computer simulations. The simulation results of the DDC system was applied to the spectrum sensing capabilities. Also, performance and complexity of the results were derived and proposed system was the result of the check.