• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.10
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• pp.130-139
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• 1994

Video compression algorithms are based on removing spatial and/or temproal redundancy inherent in image sequences by predictive(DPCM) encoding, transform encoding, or a combination of predictive and transform encoding. In this paper, each 8$\times$8 DCT coefficient of DFD(displaced frame difference) is adaptively quantized by one of the four quantizers depending on total distortion level, which is determined by characteristics of HVS(human visual system) and buffer status. Therefore, the number of possible quantizer selection vectors(patterns) is 4$^{64}$. If this vectors are coded, toomany bits are required. Thus, the quantizer selection vectors are limited to 2048 for Y and 512 for each U, V by the proposed method using SWAD(sum of weighted absolute difference) for discriminating vectors. The computer simulation results, using the codebook vectors which are made by the proposed method, show that the subjective and objective image quality (PSNR) are goor with the limited bit allocation. (17Mbps)

• Proceedings of the KSPS conference
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• 1996.10a
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• pp.448-455
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• 1996

In this paper, we have proposed a VQ algorithm which uses a generating order to make quantize feature vector of speech signal. The proposed algorithm inspects what codeword follows a(ter present codeword and adds new index to established codebook, when mapping speech signal. We present a variable bit rate for new codebook, and propose an efficient compressed way of information. In this way, the number of computation and the number of codewords to be searched are reduced considerably. The performance of the proposed VQ algorithm is evaluated by spectrum distortion measure and bit rate. The obtained spectrum distortion is reduced about 0.22 [db], and the bit rate is saved over 0.21 bit/frame.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2287-2290
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• 2003

In this paper, an image vector quantization method is proposed not only to improve the compression ratio but also to reduce the computation cost. The proposed method could save the computation cost of codebook generation and encoding by using the modified LBG algorithm of Partial Search Partial Distortion (PSPD) in wavelet domain, by which the code book was constructed together with the partial codebook search, the partial code vector elements, and the interruption criterion. We have designed and implemented the vector quantizer to verify the improvement in reducing compression ratio in encoding processing and reducing the computation cost.

• Journal of Industrial Technology
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• v.19
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• pp.209-216
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• 1999

In this paper, we proposed a modofied genetic algorithm to minimize misclassification rate for determining the codebook. Genetic algorithms are adaptive methods which may be used solve search and optimization problems based on the genetic processes of biological organisms. But they generally require a large amount of computation efforts. GAVQ can choose the optimal individuals by genetic operators. The position of individuals are optimized to improve the recognition rate. The technical properties of this study is that prevents us from the local minimum problem, which is not avoidable by conventional VQ algorithms. We compared the simulation result with Matlab using phoneme data. The simulation results show that the recognition rate from GAVQ is improved by comparing the conventional VQ algorithms.

• Proceedings of the KSPS conference
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• 2006.11a
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• pp.131-134
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• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.10-13
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• 1988

An isolated-word recognition method using adaptively partitioned multisection codebooks is proposed. Each training utterance was divided into several sections according to its pattern extracted by labeling technique. For each pattern, reference codebooks were generated by clustering the training vectors of the same section. In recognition procedure, input speech was divided into the sections by the same method used in codebook generation procedure, and recognized to the reference word whose codebook represented the smallest average distortion. The proposed method was tested for 100 Korean words and attained recognition rate about 96 percent.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.276-285
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• 2011

In this paper, we propose new cooperative precoder selection technique for interference limited MIMO networks. Our proposed method gives weighting to precoders in the codebook according to each precoder's performance priority. By applying our proposed method to precoder selection sequence, performance of entire system can be improved in terms of sumrate, stability, and feedback rate.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.827-830
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• 2000

In this paper, we propose the method using PCA(principal component analysis) algorithm when proposed algorithm performs multimedia information indexing. After we extract DC coefficients of DCT from MPEG video stream which is an international standard of moving picture compression coding, we apply PCA algorithm to image made of DC coefficients and extract the feature of each DC image. Using extracted features, we generate codebook and perform multimedia information indexing. The proposed algorithm Is very fast when indexing and can generate optimized codebook because of using statistical feature of data

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.7-10
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• 2000

본 논문은 이동통신상에서 송신측의 송신 채널 에러정보를 수신측에서 송신측으로 전달하는 음성패킷을 이용하여 송신측에 알려주어 압축과정에 이용하게 할 수 있는 방법을 제안한다. ACELP(Algebraic CELP)구조 방식을 가지고 있는 음성압축기들 중 G.729을 사용하며 음성 패킷정보 중 Pitch Delay와 Fixed Codebook를 이용하여 전송음성 패킷안에 상대방의 송신 채널정보를 싣는다. 수신측에서 받은 패킷이 Erasure로 판단되었을 때 패킷정보들 중 Fixed Codebook Index를 만들게 되는 4개의 Optimal 펄스 중 2개의 펄스만 사용하며 나머지 2개의 펄스는 약속된 임의의 위치에 위치시킨 후 송신측에 전송시킨다. 상대방에서 약속에 맞는 위치의 펄스를 보내왔을 때를 체크함으로써 자신의 송신채널 상태를 알 수 있게 된다 송신채널에$5\%$ Erasure 채널 에러가 발생했을 때 채널정보를 가진 패킷의 음질은 약 0.1dB 정도 떨어지게 된다. 하지만 음성압축 전송 시 송신채널의 정보를 이용하여 무선채널에러에 강하게 할 수 있다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1993.06a
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• pp.1321-1325
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• 1993

This paper investigates the use of Fuzzy vector quantizer(FVQ) in speech synthesis. To compress speech data, we employ K-means algorithm to design codebook and then FVQ technique is used to analysize input speech vectors based on the codebook in an analysis part. In FVQ synthesis part, analysis data vectors generated in FVQ analysis is used to synthesize the speech. We have fined that synthesized speech quality depends on Fuzziness values in FVQ, and the optimum fuzziness values maximized synthesized speech SQNR are related with variance values of input speech vectors. This approach is tested on a sentence, and we compare synthesized speech by a convensional VQ with synthesized speech by a FVQ with optimum Fuzziness values.