• 융합신호처리학회논문지
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• 제7권4호
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• pp.178-183
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• 2006

As a lot of researches on the speech signal processing are performed due to the recent rapid development of the information-communication technology. the pitch period is used as an important element to various speech signal application fields such as the speech recognition. speaker identification. speech analysis. or speech synthesis. A variety of algorithms for the time and the frequency domains related with such pitch period detection have been suggested. One of the pitch detection algorithms for the time domain. AMDF (average magnitude difference function) uses distance between two valley points as the calculated pitch period. However, it has a problem that the algorithm becomes complex in selecting the valley points for the pitch period detection. Therefore, in this paper we proposed the modified AMDF(M-AMDF) algorithm which recognizes the entire minimum valley points as the pitch period of the speech signal by using the rotation transform of AMDF. In addition, a threshold is set to the beginning portion of speech so that it can be used as the selection criteria for the pitch period. Moreover the proposed algorithm is compared with the conventional ones by means of the simulation, and presents better properties than others.

• 대한의용생체공학회:의공학회지
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• 제6권2호
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• pp.15-22
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• 1985

We have developed a microprogramir!able signal processor for real-time ultrasonic signal processing. Processing speed was increased by the parallelism in horizontal microprogram using 104bits microcode and the Pipelined architecture. Control unit of the signal processor was designed by microprogrammed architec- ture and writable control store (WCS) which was interfaced with host computer, APPLE- ll . This enables the processor to develop and simulate various digital signal processing algorithms. The performance of the processor was evaluated by the Fast Fourier Transform (FFT) program. The execution time to perform 16 bit 1024 points complex FF7, radix-2 DIT algorithm, was about 175 msec with IMHz master Clock. We can use this processor to Bevelop more efficient signal processing algorithms on the biological signal processing.

• Journal of Information Processing Systems
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• 제18권1호
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• pp.97-114
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• 2022

Interest is increasing in electrocardiogram (ECG) signal analysis for embedded devices, creating the need to develop an algorithm suitable for a low-power, low-memory embedded device. Linear approximation of the ECG signal facilitates the detection of fiducial points by expressing the signal as a small number of vertices. However, dynamic programming, a global optimization method used for linear approximation, has the disadvantage of high complexity using memoization. In this paper, the calculation area and memory usage are improved using a linear approximated template. The proposed algorithm reduces the calculation area required for dynamic programming through local optimization around the vertices of the template. In addition, it minimizes the storage space required by expressing the time information using the error from the vertices of the template, which is more compact than the time difference between vertices. When the length of the signal is L, the number of vertices is N, and the margin tolerance is M, the spatial complexity improves from O(NL) to O(NM). In our experiment, the linear approximation processing time was 12.45 times faster, from 18.18 ms to 1.46 ms on average, for each beat. The quality distribution of the percentage root mean square difference confirms that the proposed algorithm is a stable approximation.

• 융합신호처리학회논문지
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• 제14권2호
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• pp.95-98
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• 2013

LMS(Least Mean Square)알고리즘은 강인성, 높은 추적성, 구현의 단순성 때문에 아직도 많이 사용되는 알고리즘이다. 하지만, LMS알고리즘은 비균일적인 수렴율과 EMSE(Excess mean square error)사이에 trade-off를 가진다. 이러한 단점을 극복하기 위해, 많은 가변 스텝 사이즈 알고리즘이 연구되고 있다. 빠른 수렴속도를 위하여 복잡한 가변스텝 방식을 사용 하는데 많은 계산량을 필요로 한다. IGC(Instantaneous Gain Control) 알고리즘은 원신호와 잡음신호의 순시이득값을 사용하여 계산량은 줄이고 높은 성능을 유지한다. 하지만 IGC 알고리즘은 이득값 계산시 log함수에 의해 실행시간이 오래 걸리는 단점을 가진다. 제안하는 변형 IGC알고리즘은 실행시간을 지연하는 log 함수를 제거하여, 실시간 소음제거 처리에 맞게 변형하여 성능 개선을 이룰 수 있었다.

• Journal of Electrical Engineering and Technology
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• 제9권5호
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• pp.1753-1761
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• 2014

Due to the inherent inhomogeneous and anisotropy nature of the composite materials, the detection of internal defects in these materials with non-destructive techniques is an important requirement both for quality checks during the production phase and in service inspection during maintenance operations. The estimation of the time-of-arrival (TOA) and/or time-of-flight (TOF) of the ultrasonic echoes is essential in ultrasonic non-destructive testing (NDT). In this paper, we used split-spectrum processing (SSP) combined with matching pursuit signal decomposition (MPSD) to develop a dedicated ultrasonic detection system. SSP algorithm is used for Signal-to-Noise Ratio (SNR) enhancement, and the MPSD algorithm is used to decompose backscattered signals into a linear expansion of chirplet echoes and estimate the chirplet parameters. Therefore, the combination of SSP and MPSD (SSP-MPSD) presents a powerful technique for ultrasonic NDT. The SSP algorithm is achieved by using Gaussian band pass filters. Then, MPSD algorithm uses the Maximum Likelihood Estimation. The good performance of the proposed method is experimentally verified using ultrasonic traces acquired from three specimens of carbon fibre reinforced polymer multi-layered composite materials (CFRP).

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 하계종합학술대회논문집
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• pp.627-630
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• 1998

In this study, we derived stop-and-go normalized DD, dual-mode normalized sato, dual-mode NCMA blind equalization algorithm for complex data. and then, the convergence characteristics of the proposed SG-NDD, dual-mode NSato blind equalization algorithms are compared with those of SG-DD, dual-mode sato algorithm. In genral, the normalized blind equalization algorithms have better convergence characteristics than the conventional algorithms.

• 한국정밀공학회:학술대회논문집
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• 한국정밀공학회 2003년도 춘계학술대회 논문집
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• pp.1046-1050
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• 2003

In this paper, we describe study on the development or multi-path ultrasonic gas flowmeter using a transit time method. This system includes 5 pairs of ultrasonic transducers. ultrasonic signal processing unit using switch matrix method, computation algorithm of gas flow rate, spool piece type multi-path pipe unit. We have developed enhanced type of main ultrasonic signal processing unit using switch matrix method fer multi-path ultrasonic gas flowmeter. Also, we have developed the new transmitting ＆ receiving method of ultrasonic waves and the new signal processing algorithm for the computation of ultrasonic transit time from received ultrasonic waves. In this study, we have designed more compact signal processing unit compared with the conventional hardware system of multi-path ultrasonic gas flowmeter. We have confirmed its reliability for multi-path ultrasonic gas flowmeter through the laboratory test using calibration system. In the future. we will perform the field test for the developed system in the POSCO gas line.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.663-664
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• 2006

This paper presents a reconfigurable digital signal processing(rDSP) architecture that is effective for implementing adaptive digital signal processing in the applications of smart health care system. This rDSP architecture employs an evolution capability of FIR filters using genetic algorithm. Parallel genetic algorithm based rDSP architecture evolves FIR filters to explore optimal configuration of filter combination, associated parameters, and structure of feature space adaptively to noisy environments for an adaptive signal processing. The proposed DSP architecture is implemented using Xilinx Virtex4 FPGA device and SMIC 0.18um CMOS Technology.

• 대한전기학회논문지:시스템및제어부문D
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• 제53권4호
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• pp.293-299
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• 2004

This paper is about the extraction of basis function for ECG signal processing. In the first step, it is assumed that ECG signal consists of linearly mixed independent source signals. 12 channel ECG signals, which were sampled at 600sps, were used and the basis function, which can separate and detect source signals - QRS complex, P and T waves, - was found by applying the fast fixed point algorithm, which is one of learning algorithms in independent component analysis(ICA). The possibilities of significant point detection and classification of normal and abnormal ECG, using the basis function, were suggested. Finally, the proposed method showed that it could overcome the difficulty in separating specific frequency in ECG signal processing by wavelet transform. And, it was found that independent component analysis(ICA) could be applied to ECG signal processing for detection of significant points and classification of abnormal beats.

• 한국전자통신학회논문지
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• 제3권2호
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• pp.65-70
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• 2008

Digital signal processing (DSP) is the process of taking a signal and performing an algorithm on it to analyze, modify, or better identify that signal.[1] To take advantage of DSP advances, one must have at least a basic understanding of DSP theory along with an understanding of the hardware architecture designed to support these new advances. There are several programming techniques that maximize the efficiency of the DSP hardware, as well as a few fundamental concepts used to implement DSP software. This article introduced some of these underlying functions that are the building blocks of complex signal processing functions, and It will touch on the fundamental concepts of DSP theory and algorithms and also provide an overview of the implementation and optimization of DSP software, and discuss the development of DSP.