• Journal of IKEEE
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• v.13 no.2
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• pp.232-240
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• 2009

This paper presents a robust vehicle detection and tracking algorithm for supervision of illegal parking. The proposed algorithm is composed of four parts. First, a vehicle detection algorithm is proposed using the improved codebook object detection algorithm to segment moving vehicles from the input sequence. Second, a preprocessing technique using the geometric characteristics of vehicles is employed to exclude non-vehicle objects. Then, the detected vehicles are tracked by an object tracker which incorporates histogram tracking method with Kalman filter. To make the tracking results more accurate, histogram tracking results are used as measurement data for Kalman filter. Finally, Real Stop Counter (RSC) is introduced for trustworthy and accurate performance of the stopped vehicle detection. Experimental results show that the proposed algorithm can track multiple vehicles simultaneously and detect stopped vehicles successfully in the complicated street environment.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.410-415
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• 2005

Recently, most of the speaker verification systems are based on the pattern recognition approach method. And performance of the pattern-classifier depends on how to classify a variety of speakers' feature parameters. In order to classify feature parameters efficiently and effectively, it is of great importance to enlarge variations between speakers and effectively measure distances between feature parameters. Therefore, this paper would suggest the positively mixed model scheme that can enlarge inter-speaker variation by searching the individual model with world model at the same time. During decision procedure, we can maximize inter-speaker variation by using the proposed mixed model scheme. We also make use of a symbol probability weighting function in this system so as to reduce vector quantization errors by measuring symbol probability derived from the distance rate of between the world codebook and individual codebook. As the result of our experiment using this method, we could halve the Detection Cost Function (DCF) of the system from $2.37\%\;to\;1.16\%$.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.190-196
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• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.37 no.2
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• pp.15-20
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• 2000

In this paper, we presents a new MCDI(motion compensated difference image) coding method using CVQ(classifled vector quantization) whoes MCD(motion compensated difference) block is classified by proposed classifier using motion vector and compensated block The variance of MCD block is closely related with the magnitude of motion vector as well as the variance of compensated block, so using this property, we propose a new classifier. This scheme has no side information of the classifier what sub-codebook is selected, and simulation results show that the proposed method exhibits a good performance even when compared with a conventional method that requires classification bits.

• The Journal of the Acoustical Society of Korea
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• v.10 no.3
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• pp.37-43
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• 1991

In this paper, we compare the performances of speaker adaptation which consist of two stages of processing for an HMM-based speech recognition system. We compare three kinds of VQ adaptation methods which may be used in the first stage to reduce the distortion error for a new speaker : label prototype adaptation, adaptation with a codebook from adaptation speech itself, and adaptation with a mapped codebook. We then compare the performance of four kinds of HMM parameter adaptation methods which may be used in the second stage to transform HMM parameters for a new speaker : adaptation by the Viterbi algorithm, that by the DTW algorithm, that by the iterative alignment algorithm. The results show that adaptation based on the fuzzy histogram algorithm yields the highest accuracy in an HMM-based speech recognition system.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.5-11
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• 1996

This paper is the study for recognizing single vowels of speaker-independent, and we suppose a method of speech recognition using VQ(Vector Quantization)/NN(Neural Network). This method makes a VQ codebook, which is used for obtaining the observation sequence, and then claculates the probability value by comparing each codeword with the data, finally uses these probability values for the input value of the neural network. Korean signle vowels are selected for our recognition experiment, and ten male speakers pronounced eight single vowels ten times. We compare the performance of our method with those of fuzzy VQ/HMM and conventional VQ/NN According to the experiment result, the recognition rate by VQ/NN is 92.3%, by VQ/HMM using fuzzy is 93.8% and by VQ/NN using fuzzy is 95.7%. Therefore, it is shown that recognition rate of speech recognition by fuzzy VQ/NN is better than those of fuzzy VQ/HMM and conventional VQ/HMM because of its excellent learning ability.

• Journal of Satellite, Information and Communications
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• v.11 no.3
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• pp.133-137
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• 2016

Millimeter wave wireless communication systems, especially those targeting indoor high rate data transfer, have a strong requirement for high quality wireless link. Unfortunately, in this frequency band, the electromagnetic wave has to sustain the high propagation loss caused by the smaller wavelengths. In this scenario, beamforming technique, which enhances the link quality by focusing the radiation power on a direction, becomes one of the most important techniques in millimeter wave band wireless communication. In recent year, there been conducted many research on beamforming to improve the performance of wireless system. In this paper, we evaluate the performance of a simplified codebook-based beamforming scheme which is based on multiple-procedure and three-state beam selection. The simplified scheme significantly reduces beamforming setup time, comparing to the exhaustive searching, two-level searching adopted in IEEE 802.15.3c standard, and also conventional multi-level scheme.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1
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• pp.22-34
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• 1992

There exists a transform trellis code that is optimal for stationary Gaussian sources and the squared-error distortion measure at all rates. In this paper, we train an asymptotically optimal version of such a code to obtain one which is matched better to the statistics of real world data. The training algorithm uses the M algorithm to search the trellis codebook and the LBG algorithm to update the trellis codebook. We investigate the trained transform trellis coding scheme for the first-order AR(autoregressive) Gaussian source whose correlation coefficient is 0.9 and actual speech sentences. For the first-order AR source, the achieved SNR for the test sequence is from 0.6 to 1.4 dB less than the maximum achievable SNR as given by Shannon's rate-distortion function for this source, depending on the rate and surpasses all previous known results for this source. For actual speech data, to achieve improved performance, we use window functions and gain adaptation at rate 1.0 bits/sample.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.1
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• pp.70-76
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• 2011

LTE Re1-10 aims at peak. data rates of 1Gbits/s for the downlink and 500 Mbits/s for the uplink, which can be accomplished by not only wide spectrum but also advanced MIMO techniques such as precoded MIMO and cooperative relays. Considering some relays can have more direct signal paths than mobile stations do, LoS components are examined to build more efficient codebooks for Rician channels. The proposed codebooks perform better than the existing LTE codebooks as the criterium of LoS, K-factor increases. Conserving the advantages and max-min chordal distance of the existing LTE codebooks, the proposed ones also maximize the minimum chordal distances between codewords over Rician fading channels. Link-level simulation with LTE system parameters confirm the performance improvements as the value of K increases.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.299-307
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• 2003

There have been many studies on speaker normalization which aims to minimize the effects of speaker's vocal tract length on the recognition performance of the speaker independent speech recognition system. In this paper, we propose a simple vector quantizer based linear warping speaker normalization method based on the observation that the vector quantizer can be successfully used for speaker verification. For this purpose, we firstly generate an optimal codebook which will be used as the basis of the speaker normalization, and then the warping factor of the unknown speaker will be extracted by comparing the feature vectors and the codebook. Finally, the extracted warping factor is used to linearly warp the Mel scale filter bank adopted in the course of MFCC calculation. To test the performance of the proposed method, a series of recognition experiments are conducted on discrete HMM with thirteen mono-syllabic Korean number utterances. The results showed that about 29% of word error rate can be reduced, and that the proposed warping factor extraction method is useful due to its simplicity compared to other line search warping methods.