• Title/Summary/Keyword: Information Signal Process

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Implementation of Sharpness-Enhancement Algorithm based on Adaptive-Filter for Mobile-Display Apparatuses (Mobile Display 장치를 위한 Adaptive-Filter 기반형 선명도 향상 알고리즘의 하드웨어 구현)

  • Im, Jeong-Uk;Song, Jin-Gun;Lee, Sung-Jin;Min, Kyoung-Joong;Kang, Bong-Soon
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2007.10a
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    • pp.109-112
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    • 2007
  • Definition-Enhancement of the digitalized image has been being made researches continuously due to application a camera to a mobile-apparatus and the advent of a digital camera. In particular, the inputted image from a sensor goes through the process of ISP(Image Signal Process) prior to output as a visual image. The high-frequency components are offset by LPF(Low Pass Filter) that eliminates the noise of high spatial-frequency at the moment. In this paper, we propose an algorithm that outputs more vivid image by using adaptive-HPF(High Pass Filter) that has apt coefficients for diverse conditions of an image edge, nevertheless we do not employ any Edge-Detection algorithm to enhance a blurred image.

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Audio /Speech Codec Using Variable Delay MDCT/IMDCT (가변 지연 MDCT/IMDCT를 이용한 오디오/음성 코덱)

  • Sangkil Lee;In-Sung Lee
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.16 no.2
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    • pp.69-76
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    • 2023
  • A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

Fuzzy Sensor Algorithm for Measuring Traffic Information using Analytic Hierarchy Process (계층 분석방법을 이용한 교통량검지를 위한 퍼지센서 알고리즘)

  • Jin, Hyun-Soo
    • Journal of the Korean Institute of Intelligent Systems
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    • v.12 no.3
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    • pp.193-201
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    • 2002
  • For measuring a traffic symbolic confusion Quantity and symbolic air pleasantness, we use fuzzy sensor algorithm maded by symbolic information Quantity. Hut for implementation of fuzzy sensor, we use some symbolic information item, this method cannot produce precise output because we use vague fuzzy rule method and we cannot abundance fuzzy for precision of fuzzy rule method. For this reason, this paper introduce new fuzzy sensor algorithm composed of not fuzzy rule method but using Analytic Hierachy Process. To prove that new method is good, two type of fuzzy sensor applied to traffic signal controller and through much passing vehicle, two fuzzy sensor compared each other.

Low Power Dual-Level LVDS Technique using Current Source Switching (전류원 스위칭에 의한 저전력 듀얼레벨 차동신호 전송(DLVDS) 기법)

  • Kim, Ki-Sun;Kim, Doo-Hwan;Cho, Kyoung-Rok
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.44 no.1
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    • pp.59-67
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    • 2007
  • This paper presents a low power dual-level low voltage differential signaling (DLVDS) technique using current source switching for LCD driver ICs in portable products. The transmitter makes dual level signal that has two different level signal 400mVpp and 250mVpp while keeping the advantages of LVDS. The decoding circuit recovers the primary signal from DLVDS. The low power DLVDS is implemented using a $0.25{\mu}m$ CMOS process under 2.5V supply. The proposed circuit shows 800Mbps/2-line data rate and 9mW, 11.5mW power consumptions in transmitter and receiver, respectively. The proposed DLVDS scheme reduce power consumption dramatically compare with conventional one.

A Study on Nonlinear Filter for Removal of Complex Noise (복합잡음 제거를 위한 비선형필터에 관한 연구)

  • Lee, Kyung-Hyo;Ryu, Ji-Goo;Kim, Nam-Ho
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2008.10a
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    • pp.455-458
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    • 2008
  • Former times Information Technology generally has only depended on text or sound, while nowadays information is being moved through a variety of image media. Cell phone, TV and computer have been major elements of modem society as mediators using image signal. Therefore, image signal processing also has been treated importantly and done actively. The processing has been developed in many fields of digital image processing technologies as image data compression, recognition, restoration, etc. Noises are inevitably generated by using the signals during the processing, and typical types of the noise are Impulse(salt & pepper) and AWGN(Addiction White Gaussian Noise). To reduce the noise, various kinds of filters have been developed, and according to each noise, it is being used different filter each. However, the noise is not generated by one signal but by a complex. In this paper, I suggested an image filter to remove the complex noise, and compared with existing filters' methods for verification.

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A Markov Decision Process (MDP) based Load Balancing Algorithm for Multi-cell Networks with Multi-carriers

  • Yang, Janghoon
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.8 no.10
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    • pp.3394-3408
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    • 2014
  • Conventional mobile state (MS) and base station (BS) association based on average signal strength often results in imbalance of cell load which may require more powerful processor at BSs and degrades the perceived transmission rate of MSs. To deal with this problem, a Markov decision process (MDP) for load balancing in a multi-cell system with multi-carriers is formulated. To solve the problem, exploiting Sarsa algorithm of on-line learning type [12], ${\alpha}$-controllable load balancing algorithm is proposed. It is designed to control tradeoff between the cell load deviation of BSs and the perceived transmission rates of MSs. We also propose an ${\varepsilon}$-differential soft greedy policy for on-line learning which is proven to be asymptotically convergent to the optimal greedy policy under some condition. Simulation results verify that the ${\alpha}$-controllable load balancing algorithm controls the behavior of the algorithm depending on the choice of ${\alpha}$. It is shown to be very efficient in balancing cell loads of BSs with low ${\alpha}$.

A Semi-Markov Decision Process (SMDP) for Active State Control of A Heterogeneous Network

  • Yang, Janghoon
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.10 no.7
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    • pp.3171-3191
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    • 2016
  • Due to growing demand on wireless data traffic, a large number of different types of base stations (BSs) have been installed. However, space-time dependent wireless data traffic densities can result in a significant number of idle BSs, which implies the waste of power resources. To deal with this problem, we propose an active state control algorithm based on semi-Markov decision process (SMDP) for a heterogeneous network. A MDP in discrete time domain is formulated from continuous domain with some approximation. Suboptimal on-line learning algorithm with a random policy is proposed to solve the problem. We explicitly include coverage constraint so that active cells can provide the same signal to noise ratio (SNR) coverage with a targeted outage rate. Simulation results verify that the proposed algorithm properly controls the active state depending on traffic densities without increasing the number of handovers excessively while providing average user perceived rate (UPR) in a more power efficient way than a conventional algorithm.

A Study on the Estimation of Glottal Spectrum Slope Using the LSP (Line Spectrum Pairs) (LSP를 이용한 성문 스펙트럼 기울기 추정에 관한 연구)

  • Min, So-Yeon;Jang, Kyung-A
    • Speech Sciences
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    • v.12 no.4
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    • pp.43-52
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    • 2005
  • The common form of pre-emphasis filter is $H(z)\;=\;1\;- az^{-1}$, where a typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of filter and equals R(1)/R(0) in Auto-correlation method. This paper proposes a new flattening algorithm to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency. After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed algorithm flattened the weaked high frequency components effectively. That is, we could improve the flattened characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.

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Comparison of TDC Circuit Design Method to Constant Delay Time

  • Choi, Jin-Ho
    • Journal of information and communication convergence engineering
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    • v.8 no.4
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    • pp.461-465
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    • 2010
  • This paper describes the design method of Time-to-Digital Converter(TDC) to obtain the constant delay time and good reliability. The reliability property is described with delay elements. In TDC the time signal is converted to digital value which is based on delay elements for the time interpolation. To obtain the constant delay time, the first and the last delay elements have different structure compared to the middle delay elements. In the first and the last delay elements, the driving ability could be controlled for the different delay time. The delay element can be designed by analog and digital devices. The delay time of the element using analog devices is not sensitive to process parameters than that of the element using digital devices. And the TDC circuit by the elements using analog devices shows better reliability than that by the elements using digital devices also.

On the design of 64bit CLSA adder using the optimized algorithm (최적 알고리즘을 이용한 64비트 CLSA 가산기 설계)

  • 이영훈;김상수
    • Journal of the Korea Society of Computer and Information
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    • v.4 no.3
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    • pp.47-52
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    • 1999
  • The efficiency of an adder which plays an important role in micro-process and DSP greatly depends on the kinds of carry generation method. So in this paper. I used both CLA excellent in the speed and CSA best in the chip-size. The 64bit adder is designed with high speed which is two optimum combination. Therefore this paper suggested the way of CLSA improving both speed and chip-size. and proved the excellence of the designed circuit.