• Journal of Korea Multimedia Society
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• v.8 no.1
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• pp.56-61
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• 2005

In this thesis, propose speech period and pitch detection for continuous speech recognition. This mathod is distinguishes between vowel and consonant to frame unit in continuous speech, for distinguishable voice. Powerful extraction of speech period could threshold energy make use of input signal to real noise environment. Also algorithm of this method distinguish between vowel and consonant at the same time in voice make use of zero crossing rate and short time energy to extractible speech period.

• 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.

• Proceedings of the KIEE Conference
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• 1997.07e
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• pp.1824-1826
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• 1997

In this paper, in order to obtain statistical informations on lightning electromagnetic waveforms, electric field waveforms produced by lightning return strokes were measured and analyzed. The electric field measuring system consists of hemisphere antenna 30[cm] in diameter, integrator and data acquisition. system. The frequency bandwidth of the measuring system is 200[Hz] to 1.56[MHz], and the sensitivity is 0.96[mV/V/m]. The mean value of front time of electric field waveforms produced by positive lightning return strokes is 5.87[${\mu}s$], and that of negative is 4.12[${\mu}s$]. The mean values of zero-crossing time for positive or negative electric field waveforms are 35.00 and 26.61[${\mu}s$], respectively. The mean value of percentage dip-depth for positive electric field waveforms is 33.68[%], and that for negative is 28.36[%].

• The Transactions of the Korean Institute of Electrical Engineers
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• v.40 no.1
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• pp.31-38
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• 1991

A direct digital control technique of a current source using the phase-controlled rectifier is presented. A digital firing technique without sensing the line voltage is proposed. This scheme generates firing pulses directly from error signal between command and output voltage. Thus the phase detection transformers filters and zero-crossing detector are unnecessary. The synchronism is modeled and analized. Also a software synchronization algorithm is presented without a look up table and controls the system in real time with fast dynamic characteristics. Using the single-chip microprocessor 8097BH, the direct digital control is implemented with minimal hardware structure. Using the time-weighted performance index, the optimal discrete IPM control technique is also proposed to control the current of the PCR.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.6-9
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• 1996

In this paper, we attempted to analyze the contraction patterns of elbow flexor muscle during isometric, concentric and eccentric contraction. The analysis parameters are consisted of Sequency domain parameters (mean frequency, median frequency, skewness, kurtosis) and time domain parameters (zero crossing, positive maxima, integrated EMG). As a results, the analysis parameters have specific trends for muscles, muscle contraction patterns, muscle contraction angles. Especially, at the time domain analysis, IEMG is a dominant parameter for analysis of activation patterns, and the skewness, kurtosis are useful parameters for functional recognition.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.46 no.5
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• pp.8-14
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• 2009

In the research field of speech recognition, pinpointing the endpoints of speech utterance even with the presence of background noise is of great importance. These noise present during recording introduce disturbances which complicates matters since what we just want is to get the stationary parameters corresponding to each speech section. One major cause of error in automatic recognition of isolated words is the inaccurate detection of the beginning and end boundaries of the test and reference templates, thus the necessity to find an effective method in removing the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two linear time-domain measurements: the short-time energy, and short-time zero-crossing rate. They perform well for clean speech but their precision is not guaranteed if there is noise present, since the high energy and zero-crossing rate of the noise is mistaken as a part of the speech uttered. This paper proposes a novel approach in finding an apparent threshold between noise and speech based on Lyapunov Exponents (LEs). This proposed method adopts the nonlinear features to analyze the chaos characteristics of the speech signal instead of depending on the unreliable factor-energy. The excellent performance of this approach compared with the conventional methods lies in the fact that it detects the endpoints as a nonlinearity of speech signal, which we believe is an important characteristic and has been neglected by the conventional methods. The proposed method extracts the features based only on the time-domain waveform of the speech signal illustrating its low complexity. Simulations done showed the effective performance of the Proposed method in a noisy environment with an average recognition rate of up 92.85% for unspecified person.

• 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.

• 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.

• 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.

• 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.