• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.10
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• pp.1561-1570
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• 1989

In this paper, we propose an image coding scheme which combines the Laplacian pyramid structure and a hierarchical finite state classified vector quantizer in the DCT domain, namely FSDCT-CTQ. First, an optimal bit allocation problem for fixed rates DCT-CVQ on the Laplacian pyramid structure is described. In an asymptotic case, with an optimal bit allocation, a coding gain over scalar quantization of each Laplacian plane is derived. Second, it is experimentallhy shown that the Laplacian pyramid structure provides a considerable codng gain in the sense of total MMSE (minimum mean squared error). Finally, we propose an FS-DCT-CVQ which exploits the hierarchicla correlation between the Laplacian planes. Simulation results on real images show that the proposed coding scheme can reconstruct an image with 30.33 dB at 0.192 bpp, 32.45 dB at 0.385 bpp, respectively.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.18 no.5
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• pp.1208-1222
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• 2024

Music resources are characterized by quantization, diversification and complication. With the rapid increase of the demand for music resources, the storage of music resources is very large. In order to improve the retrieval effect of music resources, a massive music resources retrieval method based on ant colony algorithm is proposed to effectively use music resources. This paper constructs autocorrelation function to extract pitch feature of music resource, classifies the music resource information by calculating feature similarity. Using ant colony algorithm to correlate the feature of music resource, gain the result of correlative, locate the result of detection and get the result of multi-module. Simulation results show that the proposed method has high precision and recall, short retrieval time and can effectively retrieve massive music resources.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.4
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• pp.772-780
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• 1994

Vector quantization with two-channel conjugate codebook has been researched as an efficient coding technique that can reduce the computational complexity and codebook storage. This paper proposes FSVQ using two-channel conjugate codebook in order to reduce the number of state codebooks. Input vector in the two-channel conjugate FSVQ is coded with state codebook of a seperated state according to each codebook. In addition, LOT is adopted to obtain to obtain a high coding gain and to reduce blocking effect which appears in the block coding. As a result, although FSVQ can achieve higher data compression ratio than general vector quantization, it has a disadvantage of having a very large number of state codebooks. However FSVQ with two-channel conjugate codebooks can employ a significantly reduced number of state codebooks, even though it has a small loss in the PSNR compared with the conventional FSVQ using one codebook. Moreover FSVQ in the LOT domain can reduce blocking effect and high coding gain compared with FSVQ in the spatial domain.

• Journal of Advanced Navigation Technology
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• v.22 no.3
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• pp.228-232
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• 2018

Dealing with quntization error in a control system properly becomes much more important as many devices are connected through network and controlled. Thus, in this paper, we study a data rate condition on quantizer to achieve practical stability in a discrete time linear time invariant system with state feedback control. First, required data rate is shown to depend on eigenvalue of the closed loop system, the size of the initial state vector, the magnitude of initial quantization error, and control gain in the absence of process noise. It additionally depends on the maximum magnitude of process noise when noise is not zero. Asymptotic analysis shows that a new design method may be needed to reduce the date rate for a networked control in the presence of quantization error and noise.. We provide a simple numerical evaluation of uniform quantizer and logarithmic qunatizer to assess their characteristics of practical stability depending on data rate in the presence of noise.

• Journal of IKEEE
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• v.27 no.4
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• pp.573-578
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• 2023

This paper introduces a new locking technique applicable to high-performance digital Phase-Locked Loops (DPLL). The study employs additional thermometer codes to reduce quantization errors in LC-based Digital Controlled Oscillators (DCO). Despite not implementing the entire DCO codes in thermometer mode, this method effectively reduces quantization errors through enhanced linearity. In the initial locking phase, binary codes are used, and upon completion of locking, the system transitions to thermometer codes, achieving high frequency linearity and reduced jitter characteristics. This approach significantly reduces the number of switches required and minimizes the oscillator's area, especially in applications requiring low DCO gain (Kdco), compared to the traditional method that uses only thermometer codes. Furthermore, the jitter performance is maintained at a level equivalent to that of the thermometer-only approach. The efficacy of this technique has been validated through modeling and design at the RTL level using SystemVerilog and Verilog HDL.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.931-936
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• 2006

This paper presents different issues of the real-time compression algorithms without compromising the video quality in the distributed environment. The theme of this research is to manage the critical processing stages (speed, information lost, redundancy, distortion) having better encoded ratio, without the fluctuation of quantization scale by using IP configuration. In this paper, different techniques such as distortion measure with searching method cover the block phenomenon with motion estimation process while passing technique and floating measurement is configured by discrete cosine transform (DCT) to reduce computational complexity which is implemented in this video codec. While delay of bits in encoded buffer side especially in real-time state is being controlled to produce the video with high quality and maintenance a low buffering delay. Our results show the performance accuracy gain with better achievement in all the above processes in an encouraging mode.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.33-38
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• 1996

This paper provides an approach for representing an optimum vector quantizer by a scalar nonlinear gain-plus-additive noise model. The validity and accuracy of this analytic model is confirmed by comparing the calcuated model quantization errors with actual simulation of the optimum Linde-Buzo-Gray(LBG) vector quantizer. Using this model we frm MSE measure of an M-band filter bank codec in terms of the equivalent scalar quantizatin model and find the optimum FIR filter coefficients for each channel in the M-band structure for a given bit rate, given filter length, and given input signal correlation model. Specific design examples are worked out for 4-tap filters in the two-band paraunitary case. These theoretical results are confirmed by extensive Monte Carlo simulation.

• The Journal of the Acoustical Society of Korea
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• v.17 no.7
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• pp.12-21
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• 1998

본 논문은 음성 부호화기인 MPMLQ(Multi Pulse Maximum Likehood Quantization)를 고정 소숫점 범용 DSP에 실시간으로 구현할 때 발생되는 계산량을 줄이기 위한 변형된 형태의 MPMLQdp 관한 것이다. MPMLQ는 음성 신호에서 선형 예측 계수와 피치 정보를 추출하고 남은 잔여 신호와 가장 유사한 여기 신호를 표현할 때 상관법을 이용 한다. 상관법은 DSP상에 구현할 때 계수 승산 오버플로우를 발생시킬 수 있으므로 연산후 항상 점검하여야 한다. 이것은 MPMLQ 구현시 전체 계산량의 많은 부분을 차지한다. 본 논 문은 이러한 문제점에 착안하여 계수 승산 오버플로우가 발생하지 않도록 입력 음성신호의 크기를 2비트 만큼 줄이고, 이로 인하여 같은 크기로 줄어든 잔여 신호를 고려하여 MPMLQ에서 여기신호의 크기를 표현하는 고정 코드북 이득표를 적절히 조절하였다. 실험 결과 변형된 MPMLQ의 SSNR은 0.040325dB(실험data기준) 향상되었으며, 계산량에 있어서 도 17.7%의 처리속도 향상되었다. 따라서 고정 소숫점 범용 DSP에 실시간 구현이 가능하였다.

• Journal of Advanced Navigation Technology
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• v.14 no.1
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• pp.102-107
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• 2010

This paper presented a rate-distortion estimation method for effective mode decision in H.264/AVC. In this approach, in order to decide a mode, laplacian distribution modeling of DCT coefficients is utilized, which do not need to such process as quantization, entropy coding. From the simulation results, proposed a method showed that rate-distortion between proposed scheme and practical value was almost the same and performed 0.02dB of PSNR gain.