• 전자공학회논문지B
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• 제29B권11호
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• pp.130-137
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• 1992

This paper presents the word recognition method using a neural-fuzzy pattern matching, in order to make a proper speech pattern for a spectrum sequence and to improve a recognition rate. In this method, a frequency variation is reduced by generating binary spectrum patterns through associative memory using a neural network, and a time variation is decreased by measuring the simillarity using a fuzzy pattern matching. For this method using binary spectrum patterns and logic algebraic operations to measure the simillarity, memory capacity and computation requirements are far less than those of DTW using a conventional distortion measure. To show the validity of the recognition performance for this method, word recognition experiments are carried out using 28 DDD city names and compared with DTW and a fuzzy pattern matching. The results show that our presented method is more excellent in the recognition performance than the other methods.

• 대한의용생체공학회:의공학회지
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• 제22권3호
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• pp.259-267
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• 2001

The visual evoked potentials(VEPs) is used to assist in the diagnosis of specific disorders associated with involvement of the sensory visual pathways. The P100 latency is an important parameter which is diagnosis of optic nerve disorders. There are characteristics of latency delay, wave distortion, amplitude deduction in abnormal subjects. It is difficult to diagnose in the case of producing peak at the P100 latency. In this paper, difference of pattern between normal VEPs and abnormal VEPs using the Choi-Williams distribution method is studied. We observed the relationship about time and spectrum. The result shown that normal VEPs had maximum spectral value at 20Hz~26.7Hz and abnormal VEPs had maximum spectral value at 16.7Hz~20Hz. Also normal VEPs spectrum is higher than abnormal VEPs spectrum.

• 전기학회논문지
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• 제56권3호
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• pp.623-628
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• 2007

This paper presents the new power line coupler which is applicable to spread spectrum power line communications. The proposed coupler maintains the adequate value of a capacitor between the transmitter mode and the receiver mode using a switch. In the transmit mode, the relatively high value of the capacitor is chosen to minimize the attenuation of transmitted signals. In the receiver mode, the value of the capacitor is chosen to be small enough so that the coupler attenuates power line noises. This coupler reduced the magnitude distortion due to having a high Q value and the power consumption caused by the AC current flowing into the capacitor. The simulation and measurement results show the improved performance in the transmitter and receiver mode, respectively.

• The Journal of the Acoustical Society of Korea
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• 제27권1E호
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• pp.1-10
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• 2008

A noise suppression algorithm is proposed for nonstationary noisy environments. The proposed algorithm is different from the conventional approaches such as the spectral subtraction algorithm and the minimum statistics noise estimation algorithm in that it classifies speech and noise signals in time-frequency bins. It calculates the ratio of the variance of the noisy power spectrum in time-frequency bins to its normalized time-frequency average. If the ratio is greater than an adaptive threshold, speech is considered to be present. Our adaptive algorithm tracks the threshold and controls the trade-off between residual noise and distortion. The estimated clean speech power spectrum is obtained by a modified gain function and the updated noisy power spectrum of the time-frequency bin. This new algorithm has the advantages of simplicity and light computational load for estimating the noise. This algorithm reduces the residual noise significantly, and is superior to the conventional methods.

• 정보통신설비학회논문지
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• 제10권2호
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• pp.46-52
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• 2011

This paper presents adaptive spread-spectrum watermarking scheme on the H.264/AVC video. The H.264/AVC has become popular video coding since it's compression performance has been improved greatly. That is why there is not enough space for embedding watermark using the spread-spectrum method. To overcome this problem, we propose selective block embedding scheme. The blocks which don't contain non-zero DCT (Discrete Cosine Transform) coefficients are selected by value. So we have enough space to embed watermarks. Experimental result shows that proposed scheme improves recovery rates of watermark about 12% and reduces distortion of watermarked video.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1995년도 제12회 음성통신 및 신호처리 워크샵 논문집 (SCAS 12권 1호)
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• pp.123-126
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• 1995

The waveform coding are concerned with simply preserving the wave shape of speech signal through a redundancy reduction process. In the case of speech synthesis, the wave form coding with high quality are mainly used to the synthesis by analysis. However, because the parameters of this coding are not classified as either excitation and vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In this paper, we proposed a new pitch alteration method that can change the pitch period in waveform coding by using scaling the time-axis and compensating the spectrum. This is a time-frequency domain method that is preserved in the phase components of the waveform and that has a little spectrum distortion with 2.5% and less for 50% pitch change.

• 한국자기공명학회논문지
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• 제4권2호
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• pp.103-115
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• 2000

Wavelet transforms are introduced as a new tool to distinguish real peaks from the noise contaminated NMR data in this paper. New algorithms of two wavelet transforms including Daubechies wavelet transform as a discrete and orthogonal wavelet transform (DWT) and Morlet wavelet transform as a continuous and nonorthogonal wavelet transform(CWT) were developed fer noise elimination. DWT and CWT method were successfully applied to the noise reduction in spectrum. The inevitable distortion of NMR spectral baseline and the imperfection in noise elimination were observed in DWT method while CWT method gives a better baseline ahape and a well noise suppressed spectrum.

• 한국통신학회논문지
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• 제21권2호
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• pp.333-339
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• 1996

In this paper, an effiicent quatization method of line spectrum pairs(LSP) with cascaded structure of vector quantizer and scalar quantizer is proposed. First, input LSP parameters is vector-quantized using a codebook with a moderate number of entries. In the second stage of quantization, the components of residual vector are individution improve the quantizer by the scalar quantizer. The utilization of ordering property and the inclusion of interframe prediction improve the quantizer performance and remove the stability check routine. The new vector-scalar cascaded quantizer using 27 bits/frame shows a transparent quality that an average specytural distortion is 1 dB and the frame proportion with above 2 dB spectral distion is less than 2%.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.405-408
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• 2002

LSP(Line Spectrum Pairs) Parameter is used for speech analysis in vocoders or recognizers since it has advantages of constant spectrum sensitivity. low spectrum distortion and easy linear interpolation. However the method of transforming LPC(Linear Predictive Coding) into LSP is so complex that it takes much time to compute. Among conventional methods, the real root method is considerably simpler than others, but nevertheless, it still suffers from its jndeterministic computation time because the root searching is processed sequentially in frequency region. We suggest a method of reducing the LSP transformation time using voice characteristics The proposed method is to apply search order and interval differently according to the distribution of LSP parameters. in comparison with the conventional real root method, the proposed method results in about 46.5% reduction. And, the total computation time is reduce to about 5% in the G.723.1 vocoder.

• 한국산학기술학회:학술대회논문집
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• 한국산학기술학회 2010년도 춘계학술발표논문집 1부
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• pp.277-280
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• 2010

LSP(Line Spectrum Pairs) Parameter is used for speech analysis in vocoders or recognizers since it has advantages of constant spectrum sensitivity, low spectrum distortion and easy linear interpolation. However the method of transforming LPC(Linear Predictive Coding) into LSP is so complex that it takes much time to compute. Among conventional methods, the real root method is considerably simpler than others, but nevertheless, it still suffers from its indeterministic computation time because the root searching is processed sequentially in frequency region. We suggest a method of reducing the LSP transformation time using voice characteristics.