• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2019.06a
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• pp.114-115
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• 2019

This paper introduces an adaptive scalar quantization scheme for video coding technology. The method utilizes the property of the coefficient groups (CG) inside each transform block so that the dead-zone interval of the scalar quantizer is adaptively set up for different CGs. Its experimental results show that our proposed quantization scheme can achieve BDBR reduction of 4.75%, 5.93, and 5.16% for Y, Cb, and Cr channel respectively when encoding with HEVC.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.2
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• pp.11-19
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• 2010

Most of deblocking filters assumes that the block boundaries are accurately known and the coding information like quantization parameters are available. In some applications such as commercial television, however, many external video inputs without known block boundary and coding information arc given. If a decompressed video sequence heavily degraded with blocking artifacts is given through the external video port, it is absolutely necessary to detect block boundaries and control the strength of deblocking filtering by analysing the given images. This paper presents an efficient method to find the block boundaries and estimate the strength of the blocking artifacts without the knowledge of coding information. In addition, the confidence of the estimated blocking artifact information is also evaluated to control the adaptive deblocking filter robustly. Experiment results show that the estimated block boundary locations and strength relative strength and confidence information are practically good enough to reduce the blocking artifacts without prior knowledge.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.11
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• pp.88-96
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• 1994

This paper proposes a new vector quantization scheme which exploits high correlations among indexes in vector quantization. An optimal vector quantizer in the rate-distortion sense can be obtained, if it is designed so that the average distortion can be minimized under the constraint of the conditional entropy of indes, which is usually much smaller than the entropy of index due to the high correlations among indexes of neighboring vectors. The oprimization process is very similar to that in ECVQ(entropy-constrained vector quanization) except that in the proposed scheme the Viterbi algorithm is introduced to find the optimal index sequence. Simulations show that at the same bitrate the proposed method provides higher PSNR by 1.0~3.0 dB than the conventional ECVQ when applied to image coding.

• Journal of Broadcast Engineering
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• v.17 no.5
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• pp.876-890
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• 2012

In this paper, we investigated the performances of the distributed video codec with adaptive Gray code to apply for frame-difference signal. That is, the best cases and the worst cases were analyzed and compared by considering the statistical characteristics of the frame difference signal in view of the Gray code allocation. Through computer simulations, if 9-bit data for frame difference signal is generated for luminance signal with 8-bit definition and so n-bit is allocated to the quantized coefficient, we were able to find the best method to reduce the virtual channel noise by adding $256+2^{9-n-1}$ to the frame difference signal. Through computer simulation with test video sequences, it was shown that the performance difference between the best cases and the worst cases is larger than about 1.5dB at same rate. It is expected that the results in this paper are applicable for the transform-domain scheme as well as the pixel-domain scheme.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.5
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• pp.66-79
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• 2009

In this paper, we propose a new adaptive intra coding technique using 1-D and 2-D integer transforms for improving coding efficiency of H.264/AVC. Proposed technique selects the most effective transform and prediction mode for each block after processing 1-D and 2-D transforms of all prediction modes. In case of using 1-D transform, $4{\times}4$ block is divided into four $1{\times}4$ or $4{\times}1$ subblocks and then each subblock is predicted and subtracted by using the decoded subblock located at the nearest position in the direction of prediction. After prediction error subblock is processed by 1-D transform and quantization, four subblocks are merged back into original $4{\times}4$ block and then, reordered as 1-D signal by a DC biased zigzag scanning pattern according to the prediction mode. Finally, comparing the coding efficiency between bitstreams based on 1-D transform and conventional 2-D transform, prediction mode and quantized coefficients for each block are decided and corresponding quantized coefficients are transmitted. Experimental results show that the proposed adaptive technique increases 0.34dB in BD-PSNR and decreases 4.03% in BD-Bitrate on the average compared with H.264/AVC.

• The Journal of the Acoustical Society of Korea
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• v.4 no.3
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• pp.33-41
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• 1985

이 논문에서는 frequency-weighted MSE를 최소화하는 적응 pre-filter와 post-filter를 음성파형 부호화기에 적용했을 때의 성능을 분석한다. 먼저 여러 다양한 pre-filter와 post-filter에 의한 noise shaping 효과를 이론적으로 보여준다. 그리고 frequency-weighted SNR 척도를 사용하여 적응 pre-filter 와 post filter에 의한 성능면에서의 이득을 이론적으로 유도한다. 적응 pre-filter와 post-filter를 ADM과 ADPCM 부호화기에 적용해본 결과에 의하면 음성파형 부호화기의 성능을 FWSNR\sub SEG\ 척도로 약 3dB 정도 개선할 수 있음을 알 수있다. 또한 pre-filter와 post-filter를 사용하면 청각적으로 중요한 영향을 미치는 1kHz에서 3kHz 사이의 양자화 잡음을 효과적으로 줄일 수 있다.

• Journal of Broadcast Engineering
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• v.9 no.2
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• pp.142-150
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• 2004

MPEG-4 AVC is an international video coding standard that is being developed. The standardization is mainly targeted at a very high compression ratio. MPEG-4 AVC is a general video coding technology that may be used in various applications. Therefore, MPEG-4 AVC needs an efficient rate control method to be applied in various communication environments. This paper is based on the rate control method that considers human visual characteristic. In this paper, we propose the constant bit rate control method using adaptive quantization and scene change detection. Experiments performed using various test images and experimentation results exhibits a favorable performance over existing rate control methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.315-323
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• 2004

In this paper we present a method to generate the bit stream of static video test patterns for testing the picture quality in DTV system. The proposed user-defined quantization table is suitable for the static video test pattern and for minimizing the deterioration of picture quality by quantization, the underflow or overflow of video buffer generated on the process of coding the static video test pattern is compensated by a adaptive zero stuffing algorithm so that optimal picture quality is implemented. Experimental result showed that the test pattern stream encoded by MPEG-2 software with the proposed algorithm had a stable bit rate and good video quality during the decoding process, which is about 3 dB higher than that of the conventional case.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.5
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• pp.7-14
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• 2012

The quantized beamforming systems always need the channel state information, which must be quantized into a finite set of vectors (named codebook), and feedback only sends the index representing the desired vector. Thereby it minimized the impact of feedback errors, caused by feedback overhead and delay. In this regard, index assignment (IA) methods, an exhaustive-search and group-based schemes, have been presented for minimizes the performance degradation without additional feedback bits. In this paper, we proposed enhanced group-based IA method, which used the optimal codebook design with chordal distance, having the adaptive properties in application of the existing IA methods. When the number of transmit antennas is 4 and LTE codebook is used, Monte-Carlo simulations verify that the proposed scheme has a power advantage of 0.5~1dB to obtain the same bit error rate than methods without IA, and it has 0.1~0.2 dB better performance compared with the existing IA methods over same environment.