• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.465-468
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• 2001

MPEG-4 Parametric Coding 중 저 비트율로 음성신호를 부호화하는 HVXC(Harmonic Vector excitation Ending)의 복호화 모듈인 LSP 합성필터와 무성음 합성부, 유성음 합성부를 VHDL을 이용하여 구현하였다. MPEG-4 HVXC의 복호화 과정은 코드북을 이용하여 LSP 계수, VXC signal, 그리고 Spectral Envelop이 복호화 되어 각각 LSP 역필터, 무성음과 유성음 합성단을 통과하여 LPC계수와 유,무성음 여기신호로 변환된 후 LPC 합성필터링 과정을 거쳐 최종적으로 음성신호를 출력시킨다. LSP inverse filter에서 사용되는 cosine함수값을 위하여 Table based Approximation을 이용하여 적은 양의 Table 값을 사용하여 정확하고 고속의 cosine 연산을 수행하였다. VXC 복호화 과정에서는 신호의 중복성을 제거하는 Hidden Address in LSH 방법을 사용하여 코드북의 크기를 줄였다. 유성음 합성단에서는 IFFT 모듈을 이용하여 연산속도를 증가 시켰다. 최종적으로 위와 같이 구현된 시스템을 Simulation을 통해 Software 검증을 하였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.80-84
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• 1999

In this paper, the split model analysis algorithm, which can generate the wideband speech signal from the spectral information of narrowband signal, is developed. The split model analysis algorithm deals with the separation of the 10/sup th/ order LPC model into five cascade-connected 2/sup nd/ order model. The use of the less complex 2/sup nd/ order models allows for the exclusion of the complicated nonlinear relationships between model parameters and all the poles of the LPC model. The relationships between the model parameters and its corresponding analog poles is proved and applied to each 2/sup nd/ order model. The wideband speech signal is obtained by changing only the sampling rate.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.11
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• pp.1020-1028
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• 2011

As rotating machines play an important role in industrial applications such as aeronautical, naval and automotive industries, many researchers have developed various condition monitoring system and fault diagnosis system by applying artificial neural network. Since using obtained signals without preprocessing as inputs of neural network can decrease performance of fault classification, it is very important to extract significant features of captured signals and to apply suitable features into diagnosis system according to the kinds of obtained signals. Therefore, this paper proposes a neural-network-based fault diagnosis system using AR coefficients as feature vectors by LPC(linear predictive coding) and EIV(errors-in variables) analysis. We extracted feature vectors from sound, vibration and current faulty signals and evaluated the suitability of feature vectors depending on the classification results and training error rates by changing AR order and adding noise. From experimental results, we conclude that classification results using feature vectors by EIV analysis indicate more than 90 % stably for less than 10 orders and noise effect comparing to LPC.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.3
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• pp.136-141
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• 2012

One of the main reasons for serious road accidents is driving while drowsy. For this reason, drowsiness detection and warning system for drivers has recently become a very important issue. Monitoring physiological signals provides the possibility of detecting features of drowsiness and fatigue of drivers. One of the effective signals is to measure electroencephalogram (EEG) signals and electrooculogram (EOG) signals. The aim of this study is to extract drowsiness-related features from a set of EEG signals and to classify the features into three states: alertness, drowsiness, sleepiness. This paper proposes a neural-network-based drowsiness detection system using Linear Predictive Coding (LPC) coefficients as feature vectors and Multi-Layer Perceptron (MLP) as a classifier. Samples of EEG data from each predefined state were used to train the MLP program by using the proposed feature extraction algorithms. The trained MLP program was tested on unclassified EEG data and subsequently reviewed according to manual classification. The classification rate of the proposed system is over 96.5% for only very small number of samples (250ms, 64 samples). Therefore, it can be applied to real driving incident situation that can occur for a split second.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.6
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• pp.655-660
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• 2005

This paper studied the feature parameters less affected by the emotional variation for the development of the robust speech recognition technologies. For this purpose, the effect of emotional variation on the speech recognition system and robust feature parameters of speech recognition system were studied using speech database containing various emotions. In this study, LPC cepstral coefficient, met-cepstral coefficient, root-cepstral coefficient, PLP coefficient, RASTA met-cepstral coefficient were used as a feature parameters. And CMS and SBR method were used as a signal bias removal techniques. Experimental results showed that the HMM based speaker independent word recognizer using RASTA met-cepstral coefficient :md its derivatives and CMS as a signal bias removal showed the best performance of $7.05\%$ word error rate. This corresponds to about a $52\%$ word error reduction as compare to the performance of baseline system using met - cepstral coefficient.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2012.05a
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• pp.887-890
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• 2012

본 논문에서는 켑스트럼 계수의 변화에 따른 남성화자와 여성화자의 음향학적인 특징벡터를 비교하여 분석하는 기초적인 연구를 수행한다. 특히 FFT 켑스트럼 및 LPC 켑스트럼에 대한 남녀의 음향학적인 특징벡터의 차이점을 나타낸다. 향후 이러한 차이점을 기초로 하여 신경회로망 등에 의한 성별 인식에 대한 연구를 수행함으로써 남성화자 및 여성화자를 분리할 수 있는 근거를 마련하는 기초연구이다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.2
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• pp.23-28
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• 1991

This paper is a study on isolated word recognition by independent speaker, we propose DTW speech recognition system by modified dynamic averaging method as reference pattern. 57 city names are selected as recognition vocabulary and 2th LPC cepstrum coefficients are used as the feature parameter. In this paper, besides recognition experiment using modified dynamic averaging method as reference pattern, we perform recognition experiments using causal method, dynamic averaging method, linear averaging method and clustering method with the same data in the same conditions for comparison with it. Through the experiment result, it is proved that recogntion rate by DTW using modified dynamic averaging method is the best as 97.6 percent.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.243-249
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• 2007

We propose an effective voice-based gender identification using group delay(GD) Generally, features for speech recognition are composed of magnitude information rather than phase information. In our approach, we address a difference between male and female for GD which is a derivative of the Fourier transform phase. Also, we propose a novel way to incorporate the features fusion scheme based on a combination of GD and magnitude information such as mel-frequency cepstral coefficients(MFCC), linear predictive coding (LPC) coefficients, reflection coefficients and formant. The experimental results indicate that GD is effective in discriminating gender and the performance is significantly improved when the proposed feature fusion technique is applied.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.85-88
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• 2001

Line spectrum pair(LSP) 계수는 양자화 오류에 강하고. 선형 릴간에 효율적이며, 필터의 안정성 판정이 용이하므로 LPC를 대신하여 음성 부호화에 널리 사용되고 있다. 일반적으로 LSP 계수간에는 일정한 상관관계가 나타나고, 이 특성을 이용하면 LSP 계수의 부호량을 줄일 수 있는 가능성이 있나. 본 논문에서는 LSP 계수를 압축하기 위해 principal component analysis(PCA)를 사용한 방법을 제안한다. 제안된 방법에서는 LSP 계수를 Karhunen-Loeve(KL) 변환해 에너지가 집중되는 고유치(eigenvalue)와 고유벡터(eigenvector)를 찾고 값을 양자화 한다. 성능 평가를 위해 2.4kbps MELP(mixed excitation linear prediction)와 8kbps QCELP(qualcumn code excited linear prediction) 음성 부호화기를 사용해 결과 값을 비교했고, 압축률이 증가하는 것을 확인했다.

• Journal of the Korea Computer Industry Society
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• v.5 no.4
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• pp.491-502
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• 2004

In this paper, we propose a HMM model using FSVQ(First Section VQ), fuzzy theory and doubly spectral feature, as study on the isolated word recognition system of speaker-independent. In the proposed paper, LPC cepstrum coefficients and regression coefficients of LPC cepstrum as doubly spectral feature be used. And, training data are divided several section and first section is generated codebook of VQ, and then is obtained multi-observation sequences by order of large propabilistic values based on fuzzy nile from the codebook of the first section. Thereafter, this observation sequences of first section is trained and is recognized a word to be obtained highest probaility by same concept. Besides the speech recognition experiments of proposed method, we experiment the other methods under the equivalent environment of data and conditions. In the whole experiment, it is proved that the proposed method is superior to the others in recognition rate.