• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.3
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• pp.6-12
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• 1984

The basic binary quantized first-order digital phase locked loop (DPLL) is modified in order to reduce the aquisition time and steadyftate phase error. Adding the loop that corrects the phase difference by detecting the falling zero-crossing time, an effort for the improving the performance is performed and the performance compared with that of the basic DPLL. Using a graphical method, the phase locking processes of the modified DPLL for a phase step and a frequency step input are depicted visually in the absence of noise. The performance of the modified DPLL for a sinusoidal input added narrow band random noise is evaluated using the Chapman-Kolmogorov equation. This approach is verified by direct computer simulation. The steady-state phase error and the average aquisition time of the modified DPLL are compared with those of the basic DPLL, It is shown that the aquisition time of the modified DPLL is shortened about twice, also, as signal to noise ratio increases, the effect of the modification increases and the steady-state phase error approaches to zero.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.3
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• pp.488-493
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• 2008

This paper represents a dual-mode type transmission technique for a high reliable narrow-band power line communication(PLC) modem, and its design and implementation of a system-on-chip(SoC). The proposed transmission technique is based on a Chirp modulation for the purpose of overcoming time variations of power line channel environments in the narrow-bandwidth of the frequency range of 95-145.5 kHz. The designed modem is fabricated utilizing a mixed 0.18 ${\mu}m$ CMOS technology. Especially, according to the power line channel environments the data transmission rate can be selectively changed into 2.5 kbps and 480 bps. The total hardware complexity of the implemented chip is about 50,000 gates, the power consumption is about 26mW, and the operating frequency is up to 5.12 MHz.

• MALSORI
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• no.46
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• pp.37-50
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• 2003

In this paper, we evaluate and compare the performance of speech/music discrimination based on various feature parameters and classifiers. As for feature parameters, we consider High Zero Crossing Rate Ratio (HZCRR), Low Short Time Energy Ratio (LSTER), Spectral Flux (SF), Line Spectral Pair (LSP) distance, entropy and dynamism. We also examine three classifiers: k Nearest Neighbor (k-NN), Gaussian Mixure Model (GMM), and Hidden Markov Model (HMM). According to our experiments, LSP distance and phoneme-recognizer-based feature set (entropy and dunamism) show good performance, while performance differences due to different classifiers are not significant. When all the six feature parameters are employed, average speech/music discrimination accuracy up to 96.6% is achieved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.2
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• pp.172-179
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• 1991

In this paper we have proposed an efficient coding algorithm using directional filtering. First an image is transformed by using DCT which has better energy compaction and then the transformed image is divided into a low frequency component and several high frequency components. The transformed coefficients of each parts are transmitted respectively by using huffman code and these are transformed inversely at receiver. For the directional components total edge images are reconstructed at zero crossing points. We are able to reduce the amount of data by getting of complex component and making directional angles 90. As a results, this proposed method is better than that of Kunt in respect of processing time and memories. We have 38dB of image quality with objective measurement of PSNR and 0.26bpp of compression ratio which is acceptable.

• Proceedings of the Korea Multimedia Society Conference
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• 1998.10a
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• pp.226-231
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• 1998

본 연구에서는 짧은 시간에 계산이 가능하며, 음성을 특징 지울 수 있는 파라미터로서 영 교차율(zero crossing rate), 단 구간 에너지(short-term, energy) 그리고 포만트(formant)를 사용하였다. 특정 화자의 음성을 입력 받아서 단모음인 'ㅏ, ㅐ, ㅓ, ㅔ, ㅗ, ㅜ, ㅡ. ㅣ'에 대한 인식을 위해 위의 세가지 파라미터를 측정하였다. 영 교차율과 단 구간 에너지 파라미터는 유성음과 무성음의 구별과 음성인지 아닌지를 판별하는데 사용하였다. 포만트 파라미터는 10차 켑스트럼(cepstrum)을 이용하여 구하였으며, 각 단모음을 판별하기 위해서 사용하였다. 하나의 단모음을 입력받아 처리하여 텍스트로 출력하는데 평균 0.065sec에 처리하며, 각각의 단모음에 대해 93%, 10개의 테스트 문장에 대해 72%의 인식률을 보이고 있다.

• Proceedings of the KIPE Conference
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• 2008.06a
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• pp.154-156
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• 2008

본 논문은 대용량 충 방전 커패시터 부하를 갖는 CCPS (Capacitor Charging Power Supply)의 충전시간 제어방법을 제안한다. 기존의 대용량 Capacitor charger들은 LC공진탱크 및 스위칭 주파수를 개 루프 방식으로 제어한다. 이와 같은 제어 방법은 LC 공진탱크의 파라미터 변화로 인해 충전시간 제어가 불가능하다. 본 논문은 이러한 문제점을 해결하기 위해 Zero-crossing 방법을 이용 전류의 공진주파수를 검출한 뒤 스위칭 주파수를 가변 함으로서 원하는 시간과 안정성을 동시에 만족 할 수 있는 피드백 제어 방법을 제안한다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.12
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• pp.1635-1643
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• 1995

This paper proposes an algorithm for machine recognition of phonemes in continuously spoken Korean. The proposed algorithm is a static strategy neural network. The algorithm uses, at the stage of training neurons, features such as the rate of zero crossing, short-term energy, and either PARCOR or auditory-like perceptual linear prediction(PLP) but not both, covering a time of 171ms long. Numerical results show that the algorithm with PLP achieves approximately the frame-based phoneme recognition rate of 99% for small vocabulary recognition experiments. Based on this it is concluded that the proposed algorithm with PLP analysis is effective in phoneme recognition.

• 제어로봇시스템학회:학술대회논문집
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• 1987.10b
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• pp.473-478
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• 1987

The purpose of this study is to design a phoneme-class recognition system for Korean language using filter-bank analysis and zero crossing rate method. First, the speech signals are separated in 16 bandpass filters to obtain short-time spectrum of speech signals, and digitized by 16-ch A/D converter. And then, with the set of features which extracted from patterns of ratios of each channel energy level to overall energy level, the decision rules are made for recognize unknown speech signal. In this experiment, the recognition rate was about 93.1 percent for 7 vowels under multitalker environment and 74.4 percent for 10 initial sounds at single speaker.

• Speech Sciences
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• v.9 no.4
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• pp.265-274
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• 2002

In this paper we propose an automatic segmentation system that outputs the time alignment information of phoneme boundary using Viterbi search with HMM (Hidden Markov Model) and corrects these results by an UVS (unvoiced/voiced/silence) classification algorithm. We selecte a set of 39 monophones and a set of 647 extended phones for HMM models. For the UVS classification we use the feature parameters such as ZCR (Zero Crossing Rate), log energy, spectral distribution. The result of forced alignment using the extended phone set is 11% better than that of the monophone set. The UVS classification algorithm shows high performance to correct the segmentation results.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.33-36
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• 1988

In this paper, the starting point of affricate is detected from the first predictor coefficient of a 12-pole linear predictive coding (LPC) analysis and phonemic segmentation is done through measuring short time energy and zero crossing rate. By this segmentation method, the duration of an aspirate can be mearsured in order to detect an aspirate or not.