• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.227-238
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• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.3
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• pp.48-54
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• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.4
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• pp.67-75
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• 2009

Multi-layered video coding that provides scalability across the visual content has emerged for easily adaptive service over current heterogeneous network. However, the network is still error prone environment so that video service may suffer packet loss or erroneous decoding of the video. Especially distortion caused by the burst error may propagate to several pictures until intra refreshing, which will raise a terrific degradation of picture quality. To overcome the problem at terminal independently, we propose a new error concealment algorithm for the multi-layered video coding. The proposed method uses the similarity of between layers in the multi-layered video coding and POCS (Projections Onto Convex Sets) which is a powerful error concealment tool, but heavily dependent on initial values. To find adequate initial value which can reduce iteration times as well as achieve high performance, we took consideration into both features of layered approach coding and the correlation in neighbor blocks. The simulation results show that the proposed concealment method works well.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.5
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• pp.21-31
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• 2002

As the MPEG streaming traffics are transmitted over the Internet, MPEG packet losses cause error propagation in both spatial and temporal domains which in turn leads to severe degradation in image quality. This image degradation is become worse by the bursty packet loss at the network. In this paper, we propose a new error spreading method to reduce the effects of bursty packet losses over the Internet. The proposed error spreading method spreads out the not utilize packet losses widely, and can be used for videocasting and video-broadcasting that do not utilize the ARQ method. The proposed method can not only reduce the consecutive losses of image blocks, but also improve the performance of error concealment. Experimental results show that the deterioration of MPEG video quality can be reduced by spreading out the MPEG packet losses.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.45-48
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• 1999

인터넷에서 전송 지연에 의해 발생한 패킷 손실은 인터넷이라는 공유망을 사용함으로써 불가피하게 발생하게 된다. 이러한 패킷 손실은 음성과 같이 실시간 처리가 필요한 미디어를 전송할 경우 매우 심각한 문제를 야기시킨다. 본 논문에서는 인터넷을 통한 음성통신에서 전송 지연으로 인해 발생한 패킷 손실을 time scale modification(TSM)기번을 이동하여 복원하는 방법들을 검토하였다. 청취평가 결과, 송수신단 양쪽에서 TSM을 적용할 경우 기존의 가장 우수한 접근방식인 재생성(regeneration) 방식과 거의 대등한 음질을 얻을 수 있다.

• ETRI Journal
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• v.34 no.4
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• pp.645-648
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• 2012

An incremental subspace learning scheme to recover lost speech segments online is presented. Our contributions in this work are twofold. First, the recovery problem is transformed into an interpolation problem of the time-varying gains via nonnegative matrix factorization. Second, incremental nonnegative matrix factorization is employed to allow online processing and track the evolution of speech statistics. The effectiveness of the proposed scheme is confirmed by the experiment results.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.307-309
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• 2003

VoIP를 이용한 인터넷 전화에서 패킷 손실로 인한 음질 저하는 큰 문제이다. 본 논문에서는 VoIP에서 가장 많이 쓰이는 코덱 중 하나인 G.729 에서의 기존 패킷 손실 복구 방법들을 살펴보고 성능을 개선하고자 한다. 제안한 방법은 보간법의 적용을 위해 LSP 파라미터와 FCB 이득 파라미터의 미래값을 추정하며, 연속된 패킷 손실에서의 파라미터 자연성을 높이기 위해 LSP 파라미터와 ACB 길이 파라미터를 선택적으로 3차 spline 보간한다. TMIT 등의 환경에서 기존의 방법과 제안한 방법의 SD를 비교하여 객관적 평가를 수행하였고, 실험 결과 제안한 방법으로 성능이 개선됨을 확인하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.139-142
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• 2002

본 연구에서는 VoIP와 같은 패킷망에서 G.729 CS-ACELP 음성부호화기에 패킷 손실 은닉 알고리즘을 적용하여, 패킷 손실로 인한 음질 저하의 완화에 관한 실험을 수행하였다. 패킷 손실 은닉은 수신된 패킷으로부터 복호된 파형을 저장해두었다가, 손실이 발생하면 피치 동기가 맞도록 선택한 파형을 손실된 패킷자리에 대체하는 방법과 연속적인 손실 이후에 음성부호화기의 메모리를 초기화하는 방법에 기반하고 있다. 실제 VoIP 통화 실험에서 측정한 패킷 손실 분포에 대해 패킷 손실로 인한 음질 저하를 완화할 수 있음을 확인하였다.