• Journal of the Korea Society of Computer and Information
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• v.11 no.2 s.40
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• pp.35-42
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• 2006

In this paper we implement a new simulator for performance analysis of a parallel digital signal processing distributed shared memory multiprocessor systems. using Parsec The key idea of this simulator is suitable in simulation of system that uses DMA function of TMS320C6701 DSP chip and local memory which have fast access time. Also, because correction of performance parameter and reconfiguration for hardware components are easy, we can analyze performance of system in various execution environments. In the simulation, FET, 2D FET, Matrix Multiplication. and Fir Filter, which are widely used DSP algorithms. have been employed. Using our simulator, the result has been recorded according to different the number of processor, data sizes, and a change of hardware element. The performance of our simulator has been verified by comparing those recorded results.

• Proceedings of the IEEK Conference
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• summer
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• pp.391-394
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5kbytes for program code. Maximum required time of 29.2ms for processing a frame of 32ms of speech validates real-time operation of the implemented system.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.23-26
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• 2000

본 논문에서는 국제 통신 표준화기구인 ITU-T에서 인터넷폰과 화상회의를 목적으로 채택된 G.723.1 음성 부호화 시스템을 TMS320C5409를 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 G.723.1 음성 부호화기는 6.3kbps 전송률일 때 인코더 25.75MIPS이고 디코더 1.99MIPS의 최대 복잡도를 나타내고, 5.3kbps 전송률일 때 인코더 17.69MIPS이고 디코더 1.9Ml PS의 최대 복잡도를 나타낸다. 사용된 메모리는 program ROM llkwords, data ROM(table) 9.45kwords, RAM 2.8kwords 정도이며, 실시간 처리된 출력음성은 C simulation결과와 같은 음질을 보였다. 구현된 G.723.1 음성 부호화기는 ITU-T에서 제공되는17개의 테스트 벡터를 모두bit-exact하게 통과하였다.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• Physical Therapy Rehabilitation Science
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• v.13 no.1
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• pp.86-94
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• 2024

Objective: To evaluate the changes in bladder capacity and storage through non-invasive neuromodulation by application of repetitive Trans magnetic stimulation (rTMS) and genital nerve stimulation (GNS) in traumatic spinal cord survivors. Design: A Single Case Study. Method: The Patient was registered in trail with the clinical trial registry of India (CTRI/2022/05/042431). The Patient was interposed with rTMS on lumbar area, from T11-L4 vertebrae with 1 Hz and the intensity was 20% below that elicited local paraspinal muscular contraction for 13 minutes. GNS was placed over dorsum of the penis with the cathode at the base and anode 2 cm distally at 20 Hz, 200 microseconds, Continuous and biphasic current was delivered and amplitude of stimulation necessary to elicit the genito-anal reflex. For assessment, Neurological examination was done for peri-anal sensation (PAS), voluntary anal contraction (VAC) and bulbocavernous reflex (BCR), deep anal pressure (DAP), and American Spinal Injury Association Impairment Scale (ASIA scale). Outcome assessment was done using Urodynamics, Spinal Cord Independence Measure Scale Version-III (SCIM-III), American Spinal Injury Association Impairment Score (ASIA Score), Beck's Depression Inventory Scale (BDI). The baseline evaluation was taken on Day 0 and on Day 30. Results: The pre-and post-data were collected through ASIA score, SCIM-III, BDI and Urodynamics test which showed significant improvement in bladder capacity and storage outcomes in the urodynamics study across the span of 4 weeks. Conclusion: rTMS along with GNS showed improvement in bladder capacity & storage, on sensory-motor score, in functional independence of individual after SCI.

• Journal of international Conference on Electrical Machines and Systems
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• v.3 no.4
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• pp.428-432
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• 2014

Rotating machines are an integral part of large electrical power machinery in most of the industries. Any degradation or outages in the rotating electric machinery can result in significant losses in productivity. It is critical to monitor the equipment for any degradation's so that it can serve as an early warning for adequate maintenance activities and repair. Prior research and field studies have indicated that the rotating machines have a particular type of signal structure during the initial start-up transient. A machine performance can be studied based on the effect of degradation in signal parameters. In this paper a data-acquisition system and the FFT algorithm has been design and model using the MATLAB and Simulink. The implementation had been carried out on the TMS320 DSP Processor and various testing and verification of the machine performance had been carried out. The results show good agreement with expected results for both simulated and real-time data. The real-time data from AC water pumps which have rotating motors built-in were collected and analysed. The FFT algorithm provides frequency response and based on this frequency response performance of the machine had been measured.The FFT algorithm provides only approximation about the machine performances.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.1
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• pp.129-136
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• 2004

In this paper, we proposed quad variable rates ADPCM coding method for efficient speech transmission and real time porcessing is implemented on TMS320C6711-DSP. The modified ADPCM with four variable coding rates, 16[kbps], 24[kbps], 32[kbps] and 40[kbps] are used for speech window samples for good quality speech transmission at a small data bits and real time encoding and decoding is implemented using DSP. ZCR is used to identify the influence of the noise on the speech signal and to decide the rate change threshold. For noise superior signals, low coding rates are applied to minimize data bit and for noise inferior signals, high coding rates are applied to enhance the speech quality. In most speech telecommunications, silent period takes more than half of the signals, speech quality close to 40[kbps] can be obtained at comparabley low data bits and this is shown by simulation and experiments. TMS320C6711-DSK board has 128K flash memory and performance of 1333MIPS and has meets the requirements for real time implementation of proposed coding algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.2
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• pp.86-94
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• 2010

In this paper, we propose spatially adaptive color demosaicing of noisy Bayer data. When sensor noises are not considered in demosaicing, they may degrade result image. In order to obtain high resolution image, sensor noises are considered in the color demosaicing step. We identify flat, edge and pattern regions at each pixel location to improve the performance of the algorithm and to reduce complexity. Based on the pre-classified regions, the demosaicing of the G channel is performed using the local statistics to reduce the interpolation error. The sensor noise is simultaneously removed by a modified version of non-local mean filter in the green and in the color difference domain. The R and B channels are interpolated easily using fully interpolated and denoised G and color difference values. Experimental results show that the proposed method achieves a significant improvement in terms of visual and numerical criteria, when compared to conventional methods.

• MALSORI
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• no.52
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• pp.111-120
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.

• Journal of the Korea Society of Computer and Information
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• v.21 no.5
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• pp.99-109
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• 2016

Creative performance has been regarded as the key to the success of an organization in recent years, and is considered essential for the survival of an organization. Organizations must find and develop creative solutions to deal with a variety of business issues. How can organizations become more creative? To develop creativity, organizations must make it easier to connect the knowledge and perspectives of its various members, who may be scattered around the world, by developing a virtual team. Drawing from the transactive memory systems (TMS), which include expertise location, credibility, and coordination, this study investigates how the positive affect of team members influences the development of creative performance during virtual collaboration where face-to-face team activities are limited. The proposed structured model was empirically tested with cross-sectional data from 322 individuals. Results indicated that the positive affect of team members was found to moderate the relationship between TMS and creativity. Through this study, we expect to provide an understanding of the mechanisms involved in developing creativity among team members in a virtual work environment.