• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.4
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• pp.47-54
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• 1983

The performance of an intra-frame DCT/DPCM hybrid coding is investigated with the criteria of normalized mean square error and subjective test for various system parameters. It includes the prediction coefficient in transform domain, normalization factor and bit-map in block quantizer, and adaptive coding. It is shown that the generalized covariance model of image is a convenient tool for bit-map and adaptive coding, and for a fast low bit-rate coding.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.7
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• pp.819-825
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• 1988

An 1-bit DPCM image coding method is presented. Our method is specially designed to reduce the slope overload which seems to be the major performance degradation factor in 1-bit DPCM. In the present algorithm, based on the classification of neighborhoods by its flatness, slope strength and direction, predictor and quantizer operate adaptively through switching action. Compared with some other methods by computer simulation, proposed method shows improved performance in image quality as well as in signal to noise ratio. This gain mainly comes from the reduced slope overload and seems large to compensate the increased complexity in prediction. As a post processing, Lee filter is used to reduce the granular noise subjectively annoying in flat region.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.5
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• pp.765-776
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• 1995

This paper describes a speaker adaptation method to improve the recognition performance of MLP(multiLayer Perceptron) based HMM(Hidden Markov Model) speech recognizer. In this method, we use lst-order linear transformation network to fit data of a new speaker to the MLP. Transformation parameters are adjusted by back-propagating classification error to the transformation network while leaving the MLP classifier fixed. The recognition system is based on semicontinuous HMM's which use the MLP as a fuzzy vector quantizer. The experimental results show that rapid speaker adaptation resulting in high recognition performance can be accomplished by this method. Namely, for supervised adaptation, the error rate is signifecantly reduced from 9.2% for the baseline system to 5.6% after speaker adaptation. And for unsupervised adaptation, the error rate is reduced to 5.1%, without any information from new speakers.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4E
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• pp.43-50
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• 1999

A fast converging coding algorithm that adaptively selects the modes of macroblocks is introduced. For a given frame, the optimal modes are selected based on the decision curves that minimize the overall distortion at a given bit rate. The method proposed in this paper is different from the conventional ones in that it does not manipulate the quantizer to meet the target bit rate but it satisfies the target bit rate by finding optimal modes of macroblocks which result consistent visual quality. Lagrange multiplier of the unconstrained cost function is controlled to trigger decision curves to generate appropriate modes to meet bit rate and the curve is obtained by utilizing simulated annealing optimization technique. The algorithm is implemented within H.261 video codec and simulation results demonstrate superior visual quality.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.2 s.302
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• pp.49-58
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• 2005

In this paper, we proposed a blind watermarking algerian to use characteristics of a scalar quantizer which is the recommended in the JPEG2000 and JPEG. The proposed algorithm shifts a quantization index according to the value of each watermark bit to prevent losing the watermark information during the compression by quantization. Therefore, the watermark is embedded during the process of quantization, not an additional process for watermarking, and is adaptively applied as a assigned quantizer according application areas. Before embedding process, a LFSR(Linear feedback shift register) rearranged the watermark for the security of the watermark itself and in the embedding process, a LFSR is used to hide the watermarking positions. Therefore the embedded watermark can he extracted by only the owner who knows the initial value of LFSR without the original image. The visual recognizable pattern such as a binary image was used as the watermark. The experimental results showed that the proposed algerian satisfies the robustness and imperceptibility corresponding to the major requirement of watermarking. The results showed the largest error rate to be $5.7\%$ for attack. The experimental result which compares the proposed algorithm with the Mohamed algorithm showed that the proposed algorithm was better than it, exactly $4\~5$ times for the attacks of JPEG and JPEG2000.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.2
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• pp.105-111
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• 2017

This paper compares the SE-MMA (Signed Error-MMA) and QE-MMA (Quantized Error-MMA) adaptive equalization algorithm in order to compensates the intersymbol interference due to channel in the transmission of spectral efficient nonconstant modulus signal such as 16-QAM. In the currently MMA adaptive equalizer, the error signal is needed for the updating the tap coefficient. The SE-MMA uses the polarity of error signal for reduce the computational operation in that process, the QE-MMA consider the this polarity and finite bit power-of-2 quantized component in that process, so they has different equalization performance. In order to comparing these performance, the computer simulation was performed in the same channel and environment, the output signal constellation of equalizer, residual isi and maximum distortion, MSE, SER were applied. As a result of computer simulation, the QE-MMA have more superior performance than the SE-MMA in every performance index.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.2
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• pp.11-19
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• 2003

This paper proposes an algorithm to reduce the estimation error of surface spectral-reflectance(SR) using a conventional 3-band RGB camera. In the proposed method, estimation error can be reduced by using adaptive principal components(PCs) for each color region. In order to build adaptive set of PCs, n SR populations are organized for n PC sets by using Lloyd quantizer design algorithm. Macbetch ColorCheckcer is utilized as initial representative SR values for 1485 Munsell color chips of total color population and the Munsell chips arc divided subsets and a set of corresponding adaptive PCs per each subset is organized. As a result of experiments, the proposed method showed advanced estimation performance compared to both the two 3-band PCA methods and the 5-band wiener method.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.118-125
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• 1998

We introduce an adaptive finite state vector quantization using a new side match distortion function. The conventional side match distortion function can make the gray level transition between the block bounddaries as smooth as possible and proper state codebooks in the flat areas where the spatial correlations are high. But it can't make proper codebooks in the edge areas where the spatial correlations are not high. The proposed distortion function adds the variances which represent the image characteristics to the conventional side match distortion function as weighted values. Then it can select better state codebooks than the conventional side match distortion function. Also if it predicts a wrong state, the proposed quantizer can correct the state. As a result, we can obtain the satisfiable image quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1383-1391
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• 1999

This paper presents a rate control method that minimizes global distortion under given target bit rates for video communication. This method makes the quality of reconstructed images better than that of the conventional ones based on R-D model at the same bit rates. Given a set of quantizers, a sequence of macroblocks to be quantized selects the optimal quantizer for each macroblock so that the total cost measure is minimized and the finite buffer is never in overflow. To solve this problem we provide a heuristic algorithm based on Lagrangian optimization using an operational rate-distortion framework and a quantization method follows H.263recommendation.