• MALSORI
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• no.53
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• pp.157-169
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• 2005

This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.1
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• pp.1-8
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• 1982

Many researches on the parametric representation of speech ,signals using the adaptive linear prediction method have been studied for the past few years. In this paper, we used the LPC(Linear Predictive Coding)method to analyae the parameters of Korean vowels and by using those parameters we studied the affinity between every pair of Korean vowels. As a result of our study, it is found that each pair of Korean vowels that has a greater phonetic affinity also has a greater affinity of vocal tract parameters than other pairs.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2455-2458
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• 2003

In this paper, the Classification of Energy Labeling has been Proposed. Energy Parameters of input signal which is extracted from each phoneme is labelled. And groups of labelling according to detected energies of input signals are detected. Next, DTW processes in a selected group of labeling. This leads to DTW processing faster than a previous algorithm. In this Method, because an accurate detection of parameters is necessary on the assumption in steps of a detection of speeching duration and a detection of energy parameters, variable windows which are decided by pitch period is used. Extract algorithms don't search for exact frame energy, because 256 frame window-sizes is fixed. For this reason, a new energy extraction method has been proposed. A pitch period is detected firstly; next window scale is decided between 200 frames and 300 frames. The proposed method make it possible to cancel an influence of windows.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.27 no.11
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• pp.1864-1872
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• 2003

Hidden Markov Model(HMM) has been widely used in speech recognition, however, its use in machine condition monitoring has been very limited despite its good potential. In this paper, HMM is used to recognize rotor fault pattern. First, we set up rotor kit under unbalance and oil whirl conditions. Time signals of two failure conditions were sampled and translated to auto power spectrums. Using filter bank, feature vectors were calculated from these auto power spectrums. Next, continuous HMM and discrete HMM were trained with scaled forward/backward variables and diagonal covariance matrix. Finally, each HMM was applied to all sampled data to prove fault recognition ability. It was found that HMM has good recognition ability despite of small number of training data set in rotor fault pattern recognition.

• The Journal of the Acoustical Society of Korea
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• v.10 no.5
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• pp.11-19
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• 1991

음성 신호의 합성기법들 중에서 파형코딩법은 음질이 우수하기 때문에 분석에 의한 합성법으로 많이 사용하고 있다. 그렇지만 음원과 성도의특성을 분리하지 않고 파형의 잉여분만을 제거한 후에 파 형자체를 저장하기 때문에 규칙에 의한 합성기법으로 사용하기에는 어려움이 많다. 본 논문은 파형코딩 법 중 선형 PCM 코딩법으로 저장된 음성파형에 대해 피치를 양분할 수 있는 주기반분법을 제안하여 파형자체의 음원을 분리하지 않고 피치 주기를 변경시킬 수 있는 새로운 피치 변경법을 제안하였다. 따 라서 음질이 우수한 파형코딩 합성법으로 규칙에 의한 합성을 수행할 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.6
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• pp.642-649
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• 1989

In this paper the theory of digital filters which can directly process the ADM encoded signal, and their structures are studied. In order to investigate the frequency characteristics of DM filters with the strucutures presented in this paper, a sampled speech signal is used for the input data. The result of computer simulation shows that the presented DM filter structures can be used effectively for the direct process of ADM encoded signals, ven though they do not posses sufficiently sharp cutoff characteristics.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.2
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• pp.86-88
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• 2002

An estimation method for the characteristic parameters in the low frequency signal is proposed in this paper. A low frequency signal is assumed to be modulated or distorted by high frequency terms. The algorithm proposed in this paper is designed to select set of local maximums in a successive manner, hence it is denoted as the iterative peak picking(IPP) algorithm. The IPP algorithm is operating in the time domain and is using only the comparison operation between two neighboring samples. Therefore, its computational complexity is very low and the delay caused by the computation is negligible, which make the real-time operation possible with economic hardware. The proposed algorithm is verified on the pitch estimation of speech signal and blood pulse estimation.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.77-82
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• 2011

In this paper, we investigate source separation algorithms for stereo audio mixed using amplitude panning method. This source separation algorithms can be used in various applications such as up-mixing, speech enhancement, and high quality sound source separation. The methods in this paper estimate the panning angles of individual signals using the principal component analysis being applied in time-frequency tiles of the input signal and independently extract each signal through directional filtering. Performances of the methods were evaluated through computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.30-35
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• 2001

The most important characteristic of Emile Bell's sound is a beating. It is modulation phenomenon which appears as a result of interference multiplication in time domain. This modulation phenomenon can be modeled as DSB-SC which suppress carrier and signals distributed both sides. The beatiog wave is observed in Laman distribution signal for polyvinyl speech signal, water vein wave, tide wave. The beating wave is caused by asymmetry Property of the bell.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.2
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• pp.423-430
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• 2011

In adaptive filtering, the sparseness of impulse response and input signal characteristics are very important factors of it's performance. This paper presents a subband improved proportionate normalized least square (SIPNLMS) algorithm which combines IPNLMS for impulse response sparseness and subband filtering for prewhitening the input signal. As drawing and combining the advantage of conventional approaches, the proposed algorithm, for impulse responses exhibiting high sparseness, achieve improved convergence speed and tracking ability. Simulation results, using colored signal(AR(4)) and speech input signals, show improved performance compared to fullband structure of existing methods.