• Journal of KIISE:Software and Applications
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• v.33 no.5
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• pp.499-507
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• 2006

Recently human gait has been considered as a useful biometric supporting high performance human identification systems. This paper proposes a view-based pedestrian identification method using the dynamic silhouettes of a human body modeled with the Hidden Markov Model(HMM). Two types of gait models have been developed both with an endless cycle architecture: one is a discrete HMM method using a self-organizing map-based VQ codebook and the other is a continuous HMM method using feature vectors transformed into a PCA space. Experimental results showed a consistent performance trend over a range of model parameters and the recognition rate up to 88.1%. Compared with other methods, the proposed models and techniques are believed to have a sufficient potential for a successful application to gait recognition.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1
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• pp.13-21
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• 1995

In this paper, we improve an image classifier algorithm based on neural network learning. It consists of two steps. The first is input pattern generation and the second, the global neural network implementation using an improved back-propagation algorithm. The feature vector for pattern recognition consists of the codebook data obtained from self-organization feature map learning. It decreases the input neuron number as well as the computational cost. The global neural network algorithm which is used in classifier inserts a control part and an address memory part to the back-propagation algorithm to control weights and unit-offsets. The simulation results show that it does not fall into the local minima and can implement easily the large-scale neural network. And it decreases largely the learning time.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.9
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• pp.1-7
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• 2009

LTE-A (long term evolution advanced) uplink selected SC-FDMA (single carrier frequency division multiple access) to maintain low PAPR for a low price and tiny terminal. Recently, the application of codebook-based preceding to LIE-A uplink is discussed. This paper shows that PAPR can be increased when various preceding schems are applied to the SC-FDMA as an LTE-A uplink. When, considering nonlinear power amplifier model to the transmitter, BER performances of preceded schemes degrade. But equal gain transmission scheme does show very low PAPR. So the link-level performance of EGT is superior to those of any other preceding schemes. Computer simulations also confirm the anticipated link-level performances.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.4
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• pp.685-690
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• 1994

A problem associated with vector quantization(VQ) is the computational complexity incurred in searching for a codevector with the closet to a given input vector, where the complexity increases exponentionally with proportion to codebook size and then limits practical application. In this paper, a simple and fast, but efficient, VQ encoding algorithm is presented using a reference codevector as start codevector of premature exit condition, which eliminates distance claculation of unlikely codevectors. The algorithm is to find reference codevector having the possibility to be the nearest vector to input vector first and then to incorporate premature exit condition. The proposed algorithm needs only 10~15% of mathematical operations compared with the conventional full search VQ. Algorithm the number of additions and comparsions of the proposed algorithm is not reduced greatly, the number of multiplication is reduced up to 70~80% compared with other fast VQ encoding methods.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.2
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• pp.150-155
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• 2004

In this paper, an emotion recognition method using speech signal is presented. Six basic human emotions including happiness, sadness, anger, surprise, fear and dislike are investigated. The proposed recognizer have each codebook constructed by using the wavelet transform for the emotional state. Here, we first verify the emotional state at each filterbank and then the final recognition is obtained from a multi-decision method scheme. The database consists of 360 emotional utterances from twenty person who talk a sentence three times for six emotional states. The proposed method showed more 5% improvement of the recognition rate than previous works.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.2 s.332
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• pp.83-88
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• 2005

We propose the design method for the VLSI architecture of FNNPDS combined PDS(partial distance search) and FNNS(fast nearest neighbor search), which are used to fast encoding in vector quantization, and obtain the results that FNNPDS(fast nearest neighbor partial distance search) is faster method than the conventional methods by simulation. In simulations, we investigate timing diagrams described searching time of the nearest codevector for an input vector, and compare the average clock cycles per input vector for Lena and Peppers images. According to the result of simulations, the number of the clock cycle of FNNPDS was reduced to $79.2\%\~11.7\%$ as compared with the number using the conventional techniques.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.293-298
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• 2003

In this paper. we propose a fast search algorithm for nearest neighbor vector quantization (NNVQ). The proposed algorithm rejects those codewords which can not be the nearest codeword and reduces the search range of codebook. Hence it reduces computational time and complexity in encoding process, while it provides the same SD performance as the conventional full search algorithm. We apply the proposed algorithm to the adaptive multi-rate (AMR) speech coder and a general vector quantizer designed by LBG. algorithm. Simulation results show effectiveness of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.110-114
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• 1997

When applying the nonuniform sampling to noisy speech signal, the required data rate increases to be comparable to or more than that by uniform sampling such as PCM. To solve this problem, we have proposed the waveform coding method, multiband nonuniform waveform coding(MNWC), applying the nonuniform sampling to band-separated speech signal[7]. However, the speech quality is deteriorated when it is compared to the uniform sampling method, since the high band is simply modeled as a Gaussian noise with average level. In this paper, as a good method to overcome this drawback, the high band is modeled as one of 16 codewords having different center frequencies. By doing this, with maintaining high speech quality as MOS score of average 3.16, the proposed method achieves 1.5 times higher compression ratio than that of the conventional nonuniform sampling method(CNSM).

• Journal of IKEEE
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• v.17 no.3
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• pp.241-247
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• 2013

This paper presents the design of a System on Programmable Chip (SoPC) based on Field Programmable Gate Array (FPGA) for speech recognition in which Mel-Frequency Cepstral Coefficients (MFCC) for speech feature extraction and Vector Quantization for recognition are used. The implementing process of the speech recognition system undergoes the following steps: feature extraction, training codebook, recognition. In the first step of feature extraction, the input voice data will be transformed into spectral components and extracted to get the main features by using MFCC algorithm. In the recognition step, the obtained spectral features from the first step will be processed and compared with the trained components. The Vector Quantization (VQ) is applied in this step. In our experiment, Altera's DE2 board with Cyclone II FPGA is used to implement the recognition system which can recognize 64 words. The execution speed of the blocks in the speech recognition system is surveyed by calculating the number of clock cycles while executing each block. The recognition accuracies are also measured in different parameters of the system. These results in execution speed and recognition accuracy could help the designer to choose the best configurations in speech recognition on SoPC.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.107-113
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• 2017

This paper provides a new approach for modeling of vector quantizer(VQ) followed by analysis and design of subband codecs with imbedded VQ's. We compute the mean squared reconstruction error(MSE) which depend on N the number of entries in each codebook, k the length of each codeword, and on the filter bank(FB) coefficients in subband codecs. We show that the optimum M-band filter bank structure in presence of pdf-optimized vector quantizer can be designed by a suitable choice of equivalent scalar quantizer parameters. Specific design examples have been developed for two different classes of filter banks, paraunitary and the biorthogonal FB and the 2 channel case. These theoretical results are confirmed by Monte Carlo simulation.