• Journal of the Institute of Electronics Engineers of Korea SD
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• v.47 no.9
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• pp.1-6
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• 2010

H.264 video coding standard is widely used due to the high compression rate and quality. H.264 decoders usually have pipeline architecture by a macroblock or a $4{\times}4$ sub-block. The period of the pipeline is usually fixed to guarantee the operation in the worst case which results in many idle cycles and the requirement of high data bandwidth and high performance processing units. We propose adaptive pipeline architecture for H.264 decoders for efficient decoding and lower the requirement of the bandwidth for the memory bus. Parameters and coefficients are delivered using hand-shaking communication through dedicated interconnections and frame pixel data are transferred using AMBA AHB network. The processing time of each block is variable depending on the characteristics of images, and the processing units start to work whenever they are ready. An H.264 decoder is designed and implemented using the proposed architecture to verify the operation using an FPGA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9C
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• pp.712-719
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• 2008

Temporal error concealment indicates the algorithm that restores the lost video data using temporal correlation between previous frame and current frame with lost data. It can be categorized into the methods of block-based and pixel-based concealment. The proposed method in this paper is for pixel-based temporal error concealment using affine transform. It outperforms especially when the object or background in lost block has geometric transform which can be modeled using affine transform, that is, rotation, magnification, reduction, etc. Furthermore, in order to maintain good performance even though one or more motion vector represents the motion of different objects, we defines a cost function. According to cost from the cost function, the proposed method adopts affine error concealment adaptively. Simulation results show that the proposed method yields better performance up to 1.9 dB than the method embedded in reference software of H.264/AVC.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.289-294
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• 2002

A valid speech-sound block can be classified to provide important information for speech recognition. The classification of the speech-sound block comes from the MRA(multi-resolution analysis) property of the DWT(discrete wavelet transform), which is used to reduce the computational time for the pre-processing of speech recognition. The merging algorithm is proposed to extract valid speech-sounds in terms of position and frequency range. It needs some numerical methods for an adaptive DWT implementation and performs unvoiced/voiced classification and denoising. Since the merging algorithm can decide the processing parameters relating to voices only and is independent of system noises, it is useful for extracting valid speech-sounds. The merging algorithm has an adaptive feature for arbitrary system noises and an excellent denoising SNR(signal-to-nolle ratio).

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.335-344
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• 2008

We propose an adaptive garbage collection technique for hybrid flash memory which has both SLC and MLC. Since SLC area is fast and MLC area has low cost, the proposed scheme utilizes the SLC area as log buffer and the MLC area as data block. Considering the high write cost of MLC flash, the garbage collection for the SLC log buffer moves a page into the MLC data block only when the page is cold or the page migration invokes a small cost. The other pages are moved within the SLC log buffer. Also it adjusts the parameter values which determine the operation of garbage collection adaptively considering I/O pattern. From the experiments, we can know that the proposed scheme provides better performance compared with the previous flash management schemes for the hybrid flash and finds the parameter values of garbage collection close to the optimal values.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.49 no.11
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• pp.933-940
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• 2021

The space-borne SAR(Synthetic Aperture Radar) system radiates the microwave signal and receives the backscattered signal. The received signal is converted to digital at the Digital Receiver, which is implemented at the end of the SAR sensor receiving chain. The converted signal is formated after signal processing such as filtering and data compression. Two quantization are conducted in the Digital Receiver. One quantization is an analog to digital conversion at ADC(Analog-Digital Converter). Another quantization is the BAQ(Block Adaptive Quantization) for data compression. The quantization process is a conversion from a continuous or higher bit precision to a discrete or lower bit precision. As a result, a quantization noise is inevitably occurred. In this paper, the impact of two quantization processes are analyzed in a view of SNR degradation.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.2
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• pp.100-106
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• 2005

OFDM is one of the promising modulation candidates for a fourth generation broadband mobile communication system because of its robustness against inter-symbol inteference(ISI). The OFDM transmission technique has the inherent flexibility to adapt the modulation scheme on each subcarrier according to the signal-to-noise ratio(SNR). In this paper, the adaptive modulation scheme of OFDM system is proposed in broadband mobile channel. The proposed algorithm allocates initially the number of bits in each subcarrier using the threshold SNR according to the target BER and controls the total number of target bits using the well-known algorithm of Chow et al. In addition, the subcarriers of the proposed algorithm are clustered in blocks, that the signalling overhead can be reduced for the bit allocation of subcarriers. For the simulated experiments, this paper presents an evaluation for the effects of target BER, the various numbers of subcarriers in the subband block, and Doppler frequency on the performance of adaptive OFDM in broadband mobile channel.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.7
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• pp.1287-1294
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• 2006

In many current wireless packet data system, the short-burst transmissions are used, and training overhead is very significant for such short burst formats. So, the availability of the short training sequence and the fast converging algorithm is essential in the adaptive equalizer. In this paper, the new equalizer algorithm is proposed to improve the performance of a MTLMS (multiple-training least mean square) based DFE (decision feedback equalizer)using the short training sequence. In the proposed method, the output of the DFE is fed back to the LMS (least mean square) based adaptive DEF loop iteratively and used as an extended training sequence. Instead of the block operation using ML (maximum likelihood) estimator, the low-complexity adaptive LMS operation is used for overall processing. Simulation results show that the perfonnance of the proposed equalizer is improved with a linear computational increase as the iterations parameter in creases and can give the more robustness to the time-varying fading.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4C
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• pp.386-394
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• 2010

In this paper, we propose an adaptive coordinated Tx-Rx beamforming scheme for inter-user interference cancellation, when a BS communicates with multiple users that each has multiple receive antenna. The conventional coordinated Tx-Rx beamforming scheme uses a fixed multi-stream per user regardless of the instantaneous channel states, that is, both user with ill-conditioned and well-conditioned channels have the same number of data streams. However, in the proposed adaptive coordinated Tx-Rx beamforming scheme, we select the number of streams per user to solve the inefficient problem of the conventional coordinated Tx-Rx beamforming. As a result of applying the adaptive coordinated Tx-Rx beamforming scheme, the BER performance is improved. Simulation results show that the proposed algorithm outperforms the conventional coordinated Tx-Rx beamforming algorithm by 2.5dB at a target BER of $10^{-2}$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2369-2381
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• 1999

In this paper, we propose an iterative algorithm for reducing the blocking artifact in block transform-coded images by using a wavelet transform. In the proposed method, an image is considered as a set of one-dimensional horizontal and vertical signals and one-dimensional wavelet transform is utilized in which the mother wavelet is the first order derivative of a Gaussian like function. The blocking artifact is reduced by removing the blocking component, that causes the variance at the block boundary position in the first scale wavelet domain to be abnormally higher than those at the other positions, using a minimum mean square error (MMSE) filter in the wavelet domain. This filter minimizes the MSE between the ideal blocking component-free signal and the restored signal in the neighborhood of block boundaries in the wavelet domain. It also uses local variance in the wavelet domain for pixel adaptive processing. The filtering and the projection onto a convex set of quantization constraint are iteratively performed in alternating fashion. Experimental results show that the proposed method yields not only a PSNR improvement of about 0.56-1.07 dB, but also subjective quality nearly free of the blocking artifact and edge blur.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.3
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• pp.113-121
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• 2013

In this paper, we propose a sub-pixel rendering algorithm using learning-based 2D FIR filters. The proposed algorithm consists of two stages: the learning and synthesis stages. At the learning stage, we produce the low-resolution synthesis information derived from a sufficient number of high/low resolution block pairs, and store the synthesis information into a so-called dictionary. At the synthesis stage, the best candidate block corresponding to each input high-resolution block is found in the dictionary. Next, we can finally obtain the low-resolution image by synthesizing the low-resolution block using the selected 2D FIR filter on a sub-pixel basis. On the other hand, we additionally enhance the sharpness of the output image by using pre-emphasis considering RGB stripe pattern of display. The simulation results show that the proposed algorithm can provide significantly sharper results than conventional down-sampling methods, without blur effects and aliasing.