• The Transactions of the Korean Institute of Electrical Engineers
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• v.41 no.5
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• pp.533-540
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• 1992

The Self Creating and Organizing (SCO) is a new architecture and one of the unsupervized learning algorithm for the artificial neural network. SCO begins with only one output node which has a sufficiently wide response range, and the response ranges of all the nodes decrease automatically whether adapting the weights of existing node or creating a new node. It is compared to the Kohonen's Self Organizing Feature Map (SOFM). The results show that SCONN has lots of advantages over other competitive learning architecture.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37A no.12
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• pp.1043-1053
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• 2012

In distributed video coding (DVC) systems, the complexity of encoders is greatly reduced by removing the motion estimation operations in encoders, since the correlation between frames is utilized in decoders. The transmission of parity bits is requested through the feedback channel, until the related errors are corrected to decode the Wyner-Ziv frames. The requirement to use the feedback channel limits the application of DVC systems. In this paper, we propose an efficient method to remove the feedback channel in DVC systems. First, a simple side information generation method is proposed to calculate the amount of parity bits in the encoder, and it is shown that the proposed method yields good performance with low complexity. Then, by calibrating the theoretical entropy with three parameters, we can calculate the amount of parity bits in the encoder and remove the feedback channel. Moreover, an adaptive method to determine quantization parameters for key frames is proposed. Extensive computer simulations show that the proposed method yields better performance than conventional methods.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.1
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• pp.323-343
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• 2020

With the rapid development of computer and communication techniques, copyright protection of vector geographic data has attracted considerable research attention because of the high cost of such data. A novel adaptive watermark detection algorithm is proposed for vector geographic data that can be used to qualitatively analyze the robustness of watermarks against data addition attacks. First, a watermark was embedded into the vertex coordinates based on coordinate mapping and quantization. Second, the adaptive watermark detection model, which is capable of calculating the detection threshold, false positive error (FPE) and false negative error (FNE), was established, and the characteristics of the adaptive watermark detection algorithm were analyzed. Finally, experiments were conducted on several real-world vector maps to show the usability and robustness of the proposed algorithm.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.2
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• pp.63-68
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• 2015

This paper proposes a robust multiple audio watermarking algorithm using improved QIM(quantization index modulation) with adaptive stepsize for different signal power and SVM(support vector machine) decoding model. The proposed algorithm embeds watermarks into both frequency magnitude response and frequency phase response using QIM. This multiple embedding method can achieve a complementary robustness. The SVM decoding model can improve detection rate when it is not sure whether the extracted data are the watermarks or not. To evaluate robustness, 11 attacks are employed. Consequently, the proposed algorithm outperforms previous multiple watermarking algorithm, which is identical to the proposed one but without SVM decoding model, in PSNR and BER. It is noticeable that the proposed algorithm achieves improvements of maximum PSNR 7dB and BER 10%.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.4
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• pp.67-76
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• 2012

Expanding services and productions for screen content videos recently, necessity of new compression techniques is emerging. The next-generation video coding standard is also considering specified coding tools for screen content videos, but it is still preliminary stage. In this paper, we investigate the characteristics of screen content videos for which we propose the quantization in spatial domain to improve coding efficiency. The proposed method directly employs quantization for residual signal without any transformations. The proposed method also applies adaptive coefficients prediction and in-loop filter for quantized residual signals in spatial domain based on the characteristics of screen content videos. As a results, the proposed method for the random access, the low-delay and the all-intra modes achieve bit-saving about 4.4%, 5.1%. and 4.9%, respectively.

• The Transactions of the Korea Information Processing Society
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• v.4 no.1
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• pp.149-160
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• 1997

This paper is concerned with the problem of speaker-adaptive speech synthes is method using a mapped codebook designed by fuzzy mapping on FLVQ (Fuzzy Learning Vector Quantization). The FLVQ is used to design both input and reference speaker's codebook. This algorithm is incorporated fuzzy membership function into the LVQ(learning vector quantization) networks. Unlike the LVQ algorithm, this algorithm minimizes the network output errors which are the differences of clas s membership target and actual membership values, and results to minimize the distances between training patterns and competing neurons. Speaker Adaptation in speech synthesis is performed as follow;input speaker's codebook is mapped a reference speaker's codebook in fuzzy concepts. The Fuzzy VQ mapping replaces a codevector preserving its fuzzy membership function. The codevector correspondence histogram is obtained by accumulating the vector correspondence along the DTW optimal path. We use the Fuzzy VQ mapping to design a mapped codebook. The mapped codebook is defined as a linear combination of reference speaker's vectors using each fuzzy histogram as a weighting function with membership values. In adaptive-speech synthesis stage, input speech is fuzzy vector-quantized by the mapped codcbook, and then FCM arithmetic is used to synthesize speech adapted to input speaker. The speaker adaption experiments are carried out using speech of males in their thirties as input speaker's speech, and a female in her twenties as reference speaker's speech. Speeches used in experiments are sentences /anyoung hasim nika/ and /good morning/. As a results of experiments, we obtained a synthesized speech adapted to input speaker.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.19-30
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• 2012

For digital data compression, the quantization is the main part of the lossy source coding. In order to improve the performance of quantization, the vector quantizer(VQ) can be employed. The encoding complexity, however, exponentially increases as the vector dimension or bit rate gets large. Much research has been conducted to alleviate such problems of VQ. Especially for high bit rates, a constrained VQ, which is called the sample-adaptive product quantizer(SAPQ), has been proposed for reducing the hugh encoding complexity of regular VQs. SAPQ has very similar structure as to the product VQ(PQ). However, the quantizer performance can be better than the PQ case. Further, the encoding complexity and the memory requirement for the codebooks are lower than the regular full-search VQ case. Among SAPQs, 1-SAPQ has a simple quantizer structure, where each product codebook is symmetric with respect to the diagonal line in the underlying vector space. It is known that 1-SAPQ shows a good performance for i.i.d. sources. In this paper, a study on designing 1-SAPQ for the first-order Markov process. For an efficient design of 1-SAPQ, an algorithm for the initial codebook is proposed, and through the numerical analysis it is shown that 1-SAPQ shows better quantizer distortion than the VQ case, of which encoding complexity is similar to that of 1-SAPQ, and shows distortions, which are close to that of the DPCM(differential pulse coded modulation) scheme with the Lloyd-Max quantizer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.12
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• pp.708-717
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• 2014

In this paper, we present a secret key distribution protocol without resorting to a key management infrastructure targeting at providing a low-complexity distributed solution to wireless network. The proposed scheme extracts a secret key from the random fluctuation of wireless channels. By exploiting time division duplexing transmission, two legitimate users, Alice and Bob can have highly correlated channel gains due to channel reciprocity, and a pair of random bit sequences can be generated by quantizing the channel gains. We propose a novel adaptive quantization scheme that adjusts quantization thresholds according to channel variations and reduces the mismatch probability between generated bit sequences by Alice and Bob. BCH codes, as a low-complexity and pratical approach, are also employed to correct the mismatches between the pair of bit sequences and produce a secret key shared by Alice and Bob. To maximize the secret key extraction rate, the parameters, quantization levels and code rates of BCH codes are jointly optimized.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12C
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• pp.1273-1279
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• 2006

In this paper, we address the design of adaptive handover schemes based on the signal strength measurement for cellular communications systems. Conventional handover algorithms, which are based only on the downlink measurement, cannot guarantee the required uplink quality because uplink channel can differ greatly from downlink channel quality. Therefore, we proposes a new suboptimal adaptive handover algorithm that considers both the uplink and downlink channel quality in order to achieve the best cell selection gains when there is a wide difference between the uplink and downlink signal quality. Simulation results show that the proposed scheme achieves better performance compared to conventional handover schemes.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.11-18
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• 2012

In this paper, an entropy constrained vector quantizer for high bit-rates is proposed. The sample-adaptive product quantizer (SAPQ), which is based on the product codebooks, is employed, and a design algorithm for the entropy constrained sample adaptive product quantizer (ECSAPQ) is proposed. The performance of the proposed ECSAPQ is better than the case of the entropy constrained vector quantizer by 0.5dB. It is also shown that the ECSAPQ distortion curve, which is based on the scalar quantizer, is lower than the high-rate theoretical curve of the entropy constrained scalar quantizer, where the theoretical curve have 1.53dB difference from Shannon's lower bound.