• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2007.04a
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• pp.309-312
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• 2007

수학, 물리학, 전자전기공학에서 취급하는 수치 자료는 매우 크거나 정밀도가 높으므로 계산과정에서 흔히 오버플로우나 언더플로우가 발생할 수 있다. 본 연구는 이런 현상을 최소화하기 위하여 겁수(Kalpa Number) 표현방법을 제안한다. 겁수는 고정소수점수와 부동소수점수를 저장할 수 있는 자료형을 제공하며, 정수형 및 실수형의 지수와 가수부분을 문자형 배열에 저장함으로써 매우 큰 수를 취급할 수 있다. 이를 이용하여 프로그램을 작성하면 베르누이 수나 전자의 움직임과 관련된 계산 프로그램에서 오버플로우나 언더플로우를 최소화할 수 있다.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.527-539
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• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.188-198
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• 2000

As MPEG-2 is introduced, a new editing system that works at the compressed domain is required. It is efficient to paste the two different video sequences in the compressed domain, but after repeated pasting, the result sequence can violate MPEG-2 rate control algorithm. In other words, it causes the underflow of the buffer during the decoding time. The earlier study inserted an additional fade-in sequence but it is not natural when we make a program. In this paper, a nobel method using requantization of the underflow sequence is presented.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10a
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• pp.135-138
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• 2006

본 논문에서는 텍스처 매핑을 처리하기 위한 텍스처 유닛 하드웨어 설계에 효율적인 새로운 유동형 소수점 포맷을 제안한다. 기존 고정 소수점 포맷은 하드웨어가 간단한 반면 고품질 텍스처 처리를 수행할 경우 오버플로우/언더플로우가 발생하며 부동 소수점 포맷은 이를 해결할 수 있으나 하드웨어가 복잡하다. 제안한 방식은 오버플로우/언더플로우를 해결하면서 부동소수점보다 하드웨어 크기를 줄여서 본 포맷을 적용한 가산기는 부동소수점보다 26% 작으며 곱셈기는 고정/부동 소수점보다 절반 이상으로 작다. 따라서 제안한 포맷은 100Mhz 이상의 빠른 동작이 가능하며 모바일 3차원 그래픽 가속기의 텍스처 유닛 설계에 효과적이다.

• Journal of KIISE
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• v.44 no.12
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• pp.1296-1306
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• 2017

Recently, as the number of Internet of Things users has increased, IETF (Internet Engineering Task Force) has released the CoAP (Constrained Application Protocol). So Internet of Things have been researched actively. However, existing studies are difficult to adapt to streaming service due to low transmission rate that result from buffer underflow. In other words, one block is transmitted one block to client's one request according to the internet environment of limited resources. The proposed scheme adaptively adjusts the rate of CON(Confirmable) message among all messages for predicting the exact network condition. Based on this, the number of blocks is determined by using buffer occupancy rate and content download rate. Therefore it improves the quality of user experience by mitigating playback interruption. Experimental results show that the proposed scheme solves the buffer underflow problem in Internet of Things streaming environment by controlling transmission rate according to the network condition.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.135-142
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• 2016

When video service is performed in simulation using NS-2(Network Simulation-2), the video replay is performed as the received frame order. In the existing video replay method based on the received frame order, as the frame orders of receiver and transmitter are different, the receiver buffer does not have the effect that the packets between the frames of transmitter buffer holds a regular size and packet dense and sparsity phenomenon in the receiver buffer is made by the irregular packet size due to the unpredictable reversed order of received partial frames. The above dense and sparsity phenomenon increases the probability of buffer overflow and underflow generation. To prevent these problems, the proposed frame receive order relocation method adds an extra replay buffer which rearranges the order of receive frame as the order of transmit frame, so it has the effect that the packets between the transmit frames keeps a regular size. Through the simulation using NS-2 and JSVM(Joint Scalable Video Model), the generation number of buffer overflow and underflow, and PSNR(Required Peak Signal to Noise Ratio) performance between the existing method and proposed method were compared. As a result, it was found that the proposed method would have better performance than the existing method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.2
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• pp.32-40
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• 2011

Reversible data hiding, which can recover the original image without any distortion after the extraction of the hidden data, has drawn considerable attention in recent years. However, underflow and overflow problems have occurred occasionally in the embedded image. To overcome these problems, we propose a new reversible data hiding algorithm which embeds a compressed location map used to identify these underflow and overflow points. In addition, the proposed algorithm allows for multilevel data hiding to increase the hiding capacity. The simulation results demonstrate that the proposed algorithm generates good performances in the PSNR, the embedding capacity, and the size of side information.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.5
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• pp.451-457
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• 1991

Four computational algorithms are discussed and compared in terms of complexity, required memory, and the number of channels which calculate the blocking probabilities in the two-dimensional multi-slot connection traffic. The overflow and underflow problems can be circumvented, and thus blocking probabilities can be calculated even up to 10000 channels, by the proposed last algorithm. This algorithm is utilized in calculating the blocking probabilities for I-slot and 6-slots calls. And the results of the simulation by the SLAM-II are compared with those of the analytic calculation.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.250-252
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• 2001

QoS(Quality of Service)측면에서 멀티미디어 응용 서비스를 제공하기 위해서 멀티미디어 통신은 여러 가지 요구사항을 지원하여야 한다. 연속적으로 생성된 미디어의 시간적인 관계를 유지하면서 재생시킬 수 있는 동기화 메커니즘은 무엇보다도 중요한 문제이다. 전달되는 매체를 공동을 사용하기 때문에 트래픽의 폭주시, 지역적으로 멀리 떨어져 있는 경우에는 지연의 현상이 발생하게 된다. 본 논문에서는 지연에 의해 발생되는 지연지터에 대하여 완충역할을 할 수 있는 최소 버퍼크기를 제시하였다. 주문형멀티미디어서버(MOS)의 경우 각 미디어 별로 서버에 저장하여 서비스하는 연구가 현재 활발히 진행되고 있으며 이것은 네트워크의 부하를 줄일 수 있다는 장점을 가지고 있다. 또한 이러한 데이터의 경우 클라이언트에서 어느 정도 지터를 예상하여 버퍼의 크기를 갖고 있어야 하는 문제는 아직도 큰 문제점으로 나타나고 있다. 본 논문은 버퍼의 크기 및 재생에 관한 동기화 시점을 조정함으로서 버퍼의 오버플로우나 언더플로우의 현상을 방지하고자 한다. 제안한 기법은 기존의 기법보다 향상된 동기화 기법임을 시뮬레이션을 통해 검증하였다.