• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.21 no.1
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• pp.55-60
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• 2021

This paper relates with the performance evaluation of QE-MMA (Quantized Error-MMA) adaptive equalization algorithm based on the stepsize and quantizer bit number in order to reduce the intersymbol interference due to nonlinear distortion occurred in the time dispersive channel. The QE-MMA was proposed using the power-of-two arithmetic for the H/W implementation easiness substitutes the multiplication and addition into the shift and addition in the tap coefficient updates process that modifies the SE-MMA which use the high-order statistics of transmitted signal and sign of error signal. But it has different adaptive equalization performance by the step size and quantizer bit number for obtain the sign of error in the generation of error signal in QE-MMA, and it was confirmed by computer simulation. As a simulation, it was confirmed that the convergence speed for reaching steady state depend on stepsize and the residual quantities after steady state depend on the quantizer bit number in the QE-MMA adaptive equalization algorithm performance.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2C
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• pp.172-181
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• 2010

In this paper, we propose an adaptive block quantization method. The propose method perfrect reconstructs side information without high complexity in the encoder side, as transmitting motion vectors from a decoder to an encoder side. Also, at the encoder side, residual signals between reconstructed side information and original frame are adaptively quantized to minimize parity bits to be transmitted to the decoder. The proposed method can effectively allocate bits based on bit error rate of side information. Also, we can achieved bit-saving by transmission of parity bits based on the error correction ability of the LDPC channel decoder, because we can know bit error rate and positions of error bit in encoder side. Experimental results show that the proposed algorithm achieves bit-saving by around 66% and delay of feedback channel, compared with the convntional algorithm.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.77-83
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• 1999

H.263 is an international standard of ITU-T that can makes the service such as video phone, video conference in the transmission line less than 64Kbps. This recommendation draft has used motion estimation/compensation, transform coding and quantizing methods. TMN5 used for the performance estimation of H.263 has fundamentally used DCT in transform coding method and presented quantizer for quantizing the DCT transform coefficient. This paper is presenting adaptive quantizer effectively able to quantize DCT coefficient considering the human visual sensitivity while the structure of TMN5 is maintaining. As quantizer that proposed DCT-based H.263 could make transmit more frame than TMN5 in a same transfer speed, it could lower the frame drop effect. And the luminance signal appeared the difference of -0.3 ～ +0.7dB in the average PSNR for the estimation of objective image quality and the chrominance signal appeared the improvement in about 1.5dB in comparision with TMN5. As a result it can attain the better image quality compared to TMN5 in the estimation of subjective image quality.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.164-169
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• 2002

칼라 정보를 이용하여 영상을 정합하기 위해서는 적은 수의 칼라 집합으로 영상을 표현하는 영상 양자화 과정이 필요하다. 적응적 양자화를 사용하는 경우에는 균일 양자화에 비해 높은 정합 성능을 기대할 수 있지만 상이한 칼라 집합의 생성으로 인해 영상 정합 과정이 힘들게 된다. 이에 본 논문에서는 상이한 칼라 집합을 갖는 영상을 정합할 수 있는 기초적인 연구를 수행하였다. 영상 정합을 위해 우선 STR(sort-tile-recursive) 방법[1]을 응용하여 질의 영상의 각 칼라에 대한 유사 칼라를 DB 영상으로부터 빠르게 선정할 수 있는 방법을 개발하였다. 질의 칼라와 유사 칼라간의 유사도를 정의하고 이를 기반으로 영상간의 유사도를 계산함으로써 영상 정합에 이용할 수 있도록 하였다. 칼라간의 유사도는 칼라 차이가 고려되어 정의되는데 칼라 차이는 칼라 공간에서의 칼라 거리로 계산된다. 칼라 거리를 계산하기 위해 유클리디언 거리를 이용할 경우 많은 계산량이 요구되므로 기존의 시티블록 거리나 체스보드 거리에 비해 유클리디언 거리를 좀더 유사하게 근사화하면서 빠른 계산이 가능한 거리 계산 방법을 개발하였다.

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.157-165
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• 2020

In this paper. we propose an effective compensation scheme to the quantization error arisen from quantized learning in a machine learning on an embedded system. In the machine learning based on a gradient descent or nonlinear signal processing, the quantization error generates early vanishing of a gradient and occurs the degradation of learning performance. To compensate such quantization error, we derive an orthogonal compensation vector with respect to a maximum component of the gradient vector. Moreover, instead of the conventional constant learning rate, we propose the adaptive learning rate algorithm without any inner loop to select the step size, based on a nonlinear optimization technique. The simulation results show that the optimization solver based on the proposed quantized method represents sufficient learning performance.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.1
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• pp.57-62
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• 2019

This paper evaluates the QE-MMA (Quantized Error-MMA) adaptive equalization algorithm by the number of quantizer in order to compensates the intersymbol interference due to channel in the transmission of high spectral efficient nonconstant modulus signal. In the adaptive equalizer, the error signal is needed for the updating the tap coefficient, the QE-MMA uses the polarity of error signal and correlation multiplier that condered nonlinear finite bit power-of-two quantizing component in order to convinience of H/W implementation. The different adaptive equalization performance were obtained by the number of quantizer, these performance were evaluated by the computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion, MSE, SER were applied as a performance index. As a result of computer simulation, it improved equalization performance and reduced equalization noise were obtained in the steady state by using large quantizer bit numbers, but gives slow in convergence speed for reaching steady state.

• Journal of Broadcast Engineering
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• v.2 no.2
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• pp.194-205
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• 1997

In video compression standards such as MPEG and H.263. rate control is one of the key components for good coding performance. This paper presents a simple adaptive rate control scheme using a piecewise linear approximation model. While conventional buffer control approach is performed by adjusting the quantization parameter linearly according to the buffer fullness. the proposed approach uses a piecewise linear approximation model derived from logarithmic relation between the quantization parameter and bitrate in data compression. In addition. a forward analyzer performed in the spatial domain is used to improve image quality. Simulation results demonstrate that the proposed method provides better performance than the conventional one and reduces the fluctuation of the PSNR per frame while maintaining the quality of the reconstructed frames at a relatively stable level.

• The Journal of the Acoustical Society of Korea
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• v.9 no.6
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• pp.5-13
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• 1990

본 논문에서는 16Kbps 및 32 Kbps 전송속도에서 ADM의 음질을 개선하기 위하여 두 가지 방 식을 적용한다. 첫째로, 고차 예측기 또는 적응 예측기를 ADM에 활용한다. ADM의 경우에 2차 또는 3 차 예측기를 사용하면 16Kbps 전송속도에서는 별로 개선이 없지만 32Kbps 전송속도에서는 SQNR\sub SEG\척도로 약 3-4dB의 상당한 이득이 얻어진다. 또한 ADM에 적응 예측기를 활용하면 최대 성능은 SZNR\sub SEG\ dir 2dB 정도 개선되지만 양자화 잡음의 축적 때문에 동작 범위가 매우 좁아진다. 둘 째로, 지연 결정 방식을 ADM에 이용한다. 지연 결정 방식을 2차 예측기를 갖고 있는 ADM에 적용하면 약 2dB 정도 개선되지만 양자화 잡음의 축적 때문에 동작 범위가 매우 좁아진다. 둘째로 지연 결정 방 식을 ADM 에 이용한다. 지연 결정 방식을 2차 예측기를 갖고 있는 ADM에 적용하면 1차 예측기를 갖 고 있는 ADMDP 적용했을 때 보다 16또는 32Kbps일 때 SQNR\sub SEG\척도로 재래의 ADM 보다 약 5dB 정도의 성능 개선이 얻어진다.