• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.11-20
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• 2005

In previous video conference system, when the number of participants in video conference increases by n, the bandwidth and memory of n2 is required. And also, it brings about increase in traffic and problem of a say during a conference in aspect of transmission of voice data. In this paper, we propose an algorithm of remote video conference using silence detection algerian to resolve the questions such as buffering method of video data in server and heavy traffic detection algorithm to the increase in participants. Video data buffering algorithm is not a method of broadcasting to other client in the server, but this algorithm uses two other methods; the buffering method of receiving compressed video data from clients and the indexing method for acquiring the video data of other participants in clients according to clients' bandwidth and network transmission speed. We apply a voice transmission algerian and a channel management algorithm to the remote video conference system. The method used in the voice transmission algorithm is a silence detection algorithm which does not send silent participants' voice data to the server. The channel management algorithm is a method allocating a say to the participants who have priority. In consideration of average 20 frames and 30ms regardless of a number of participants, we can safely conclude that the transmission of video and voice data is stable.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.12
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• pp.539-546
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• 2002

In this paper, we consider a wireless multimedia environment to service both real-time video traffic and non-real-time WWW-application traffic In our suggested new packet scheduling algorithm, we consider both the accumulation counter and SIR to reduce delay in real-time traffic. In addition, our packet scheduling algorithm gives priority first to real-time video traffic service and then to non-real-time internet Packet service when real-time traffic service is absent. From the simulation results, we find that the AC (Accumulation Counter) scheme has much smaller delay than the conversional M-LWDF scheme for real-time video data users, which has a special quality sensitive to its own delay. We also consider the transmission structure of using both the frame period in the time-axis and the OVSF codes in the code-axis at the same time, which is similar to the structure of HSDPA system.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.39 no.2
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• pp.7-13
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• 2002

When a USB digital camera is used for PC video-conference applications, motion picture data need to be transferred to the PC through the USB port. Due to the mismatch between the data rates of the USB and the motion picture, data compression should be performed before the transmission from the USB. While many motion picture compression algorithms require large intermediate memory space, the JPEG algorithm does not need to store an entire frame for the compression. Instead, a relatively small buffer is required at the input of the JPEG compression engine to resolve the inconsistency between the orders of the inputted data and the consumed data. Data reordering can be easily implemented using a double buffering scheme, which still requires a considerable size of memory. In this paper, a novel memory management scheme is proposed to avoid the double buffering. The proposed memory architecture requires a small amount of memory and a simple address generation scheme, resulting in overall cost reduction.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.1
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• pp.21-26
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• 2002

Video stream invoke a variety of traffic with the structure of compression algorithm and image complexity. Thus, it is difficult to allocate the resource on the both sides of sender and receiver, and playout on the Internet such as a packet switched network. Thus, in this paper we proposed video stream smoothing using multistream for the effective transmission of video stream. This method specifies the type of LDU(logical data unit) according to the type of original stream, and then makes a large number of streams as a fixed size, and transfers them. So, the proposed method can reduce the buffering time which occurs during the process of the smoothing and prefetch be robust to the jitter on network, as well. Consequently, it has the effective transmission characteristics of fully utilizing the clients bandwidth.