• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.93-99
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• 2011

This paper proposes a novel network jitter estimation algorithm for robust VoIP communication system. The proposed method computes the current network environment mode using the differences of arrival time and generation time from sequential received packets. According to the current network environment mode, the jitter variance weights is adjusted to minimize the error for estimating the network jitter. The jitter average and variance are calculated by the autoregressive estimated algorithm, and then the network jitter is estimated by applying the jitter variance weights.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.133-138
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• 2014

Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.353-354
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• 2012

광대역 VoIP 네트워크 환경에서는 불안정한 네트워크 환경으로 인해 음성패킷이 불규칙적으로 수신되어 음성데이터의 재생이 원활하지 못하다. 이러한 문제점을 해결하기 위해 본 논문에서는 네트워크 상태에 따라 원활하게 음성패킷을 재생시키는 스케줄링 방식을 제안한다. 제안하는 방식은 수신단에 도착한 패킷 헤더정보를 이용해 네트워크 지터를 추정하고, 추정된 지터와 지터버퍼와 음성프레임버퍼에 존재하는 패킷수 및 음성프레임 개수, 음성클래스정보에 따라 음성프레임의 길이를 변화시켜 재생시킴으로써 수신단의 버퍼링 지연을 줄이고 출력신호의 음성왜곡을 최소화한다. 제안하는 스케줄링 방식의 성능측정을 위해 버퍼링 지연과 PESQ를 기존 음성패킷 재생 스케줄링 방식과 비교한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2005.11a
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• pp.109-112
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• 2005

ATSC 지상파 DTV 시스템에서 심볼 타이밍 동기 성능 개선을 위한 알고리즘을 제안한다. 일반적으로 심볼 타이밍 동기를 위해 사용되는 가드너 방법은 다중 경로 페이딩 환경에서 성능이 좋지만 지터에 의해 성능 열화가 발생한다. 지터량는 루프 필터 대역폭이 작을수록 작아지지만, 수렴속도는 느려지게 된다. 수렴속도는 빠르면서 수렴 후 지터량를 감소시키기 위해 일정시간마다 루프필터의 출력 값을 평균하고 이 평균값을 이용하여 옵셋량을 추정한 후 추정된 옵셋의 변화율에 따라 루프 필터의 대역폭을 줄여 지터의 크기를 줄이는 알고리즘을 제안한다.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.512-517
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• 2018

Improving voice quality is a major concern in telecommunications. In this paper, we propose a robust speech quality enhancement against background noise and packet loss at VoIP (Voice-over-IP) receiver. The proposed method combines network jitter estimation based on hybrid Markov chain, adaptive playout scheduling using the estimated jitter, and speech enhancement based on restoration of amplitude and phase to enhance the quality of the speech signal arriving at the VoIP receiver over IP network. The experimental results show that the proposed method removes the background noise added to the speech signal before encoding at the sender side and provides the enhanced speech quality in an unstable network environment.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.15 no.4
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• pp.237-242
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• 2022

Acquiring high-quality images in control and measurement systems is one of the important factors. Among image acquisition technologies, SFF (Shape from Focus) is a technology for recovering a 3D shape by acquiring 2D images with different focus levels by moving an object at a predetermined step size along the optical axis. For SFF, when an object is moved at a constant step size, mechanical vibration, referred as jitter noise, occurs in each step along the optical axis. In this paper, a new state estimation filter is designed and applied for reducing the jitter noise. For the application of the proposed method, the jitter noise and focus curves are modeled as Gaussian function. Experimental results demonstrate the effectiveness of proposed method.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.89-101
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• 2015

This paper proposes a statistical model of 3-D positions when tracking moving targets using the uncooled infrared (IR) stereo camera system. The proposed model is derived from two errors. One is the position error which is caused by the sampling pixels in the digital image. The other is the timing jitter which results from the irregular capture-timing in the infrared cameras. The capture-timing in the IR camera is measured using the jitter meter designed in this paper, and the observed jitters are statistically modeled as Gaussian distribution. This paper derives an integrated probability distribution by combining jitter error with pixel position error. The combined error is modeled as the convolution of two error distributions. To verify the proposed statistical position error model, this paper has some experiments in tracking moving objects with IR stereo camera. The 3-D positions of object are accurately measured by the trajectory scanner, and 3-D positions are also estimated by stereo matching from IR stereo camera system. According to the experiments, the positions of moving object are estimated within the statistically reliable range which is derived by convolution of two probability models of pixel position error and timing jitter respectively. It is expected that the proposed statistical model can be applied to estimate the uncertain 3-D positions of moving objects in the diverse fields.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 2002.06a
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• pp.76-79
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• 2002

Ghost cancelers need to accurately estimate the distortions suffered by transmitted signals on their way to receivers by capturing ghost canceling reference signals in the vertical blanking interval. As ghosts become much severer, sync separation tends to malfunction, making it impossible to acquire measurement data for channel estimation. This paper presents a robust method of acquiring ghost data using their correlation properties, and evaluates its performance through computer simulation. The reference signals are inserted in such a manner that the polarity alternates from one frame to another to remove color bursts and sync signals, and to increase the signal-to-noise ratio as well. As a result, however, they are prone to timing jitters. So a simple yet effective method is proposed that can correct for even fractional time delays. The timing errors are found to be less than about 4% of the sample spacing.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.10 s.340
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• pp.1-8
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• 2005

This Paper Proposes the algorithm for improving the Performance or symbol timing synchronization in hoc terrestrial DTV system. The Gardner algerian is used for symbol timing synchronization has good performance in multipath fading environment but degradation of performance is caused by jitter. Though the amount of jitter becomes more little as narrow bandwidth of loop Inter, convergence speed becomes slower. This paper propose the algorithm that averages out values of loop filter every certain time and gradually reduces the bandwidth of loop filter after estimating offset using this average for the high speed of convergence and reducing the met of jitter. The proposed algorithm has better performance with high speed of convergence and the amount of jitter than conventional method.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.943-944
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• 2013

모바일 VoIP 음성통화는 지연, 지터 그리고 패킷 손실과 같은 네트워크 장애요소로 인해 품질저하가 발생된다. 본 논문에서는 이러한 네트워크 장애요소로 인한 음성통화 품질이 저하되는 문제를 해결하기 위하여 저런력 및 고품질 음성통화를 위한 지능형 모바일 VoIP 시스템을 제안한다. 제안된 방식은 손실은틱, 지터추정, 플레이아웃 스케줄링 등의 모듈로 구성되어 있다. 성능 측정 결과, 제안된 알고리즘은 기존 알고리즘에 비해 높은 PESQ와 낮은 버퍼링 지연을 보여주었다.