• Journal of KIISE:Computing Practices and Letters
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• v.14 no.2
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• pp.206-210
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• 2008

The most noticeable aspect of progressive streaming is the media playback during its download through TCP to avoid a lengthy wait for a content to finish downloading. By employing TCP, it is usually possible to detect lost packets by using the checksum and sequence numbering functions of TCP Thereafter, we can recover the lost packets by the retransmission function of TCP. However, there must remain enough amount of media data in the recipient buffer in order to guarantee seamless media playback even during retransmission. In this paper, we propose an efficient algorithm for determining the initial buffering time before start of playback to guarantee seamless playback during retransmission considering the probability of client buffer under-flow. The effectiveness of the proposed algorithm will be proved through extensive simulation results.

• Journal of Communications and Networks
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• v.10 no.4
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• pp.430-436
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• 2008

Recent studies have convinced that network coding can improve the performance of live media streaming in terms of startup delay, resilience to peer dynamics, as well as reduced bandwidth cost on dedicated streaming servers. However, there still exist some strategy drawbacks and neglected problems which need to be further researched. In addition to the commonly used evaluation parameters of the network and user experiences mentioned above, we focus on additional key factors, playback lag and switch lag, which have not been fully explored in previous work. In this paper, we present SonicStream, a novel and fully implemented live peer to peer (P2P) media streaming system with consideration of rich user experiences, including startup delay, playback continuity, playback lag, switch lag, etc. In pursuit of a further enhanced user experience, we revise traditional peer selection/data scheduling methods. Through a series of experimental evaluations and a cautious comparison with the latest similar work $R^2$, the superior performance of SonicStream has been preliminarily verified.

• ETRI Journal
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• v.43 no.3
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• pp.447-458
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• 2021

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.

• Journal of the Korean Society of Industry Convergence
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• v.25 no.2_2
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• pp.299-307
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• 2022

Wireless earphones, along with smart devices, are the most sought-after products by consumers. Compared to general earphones, wireless earphones do not have twisted wires and are easy to use, so various types of products are currently on the market. However, information on quality is somewhat lacking, so consumers tend to purchase products according to brand awareness, and manufacturers are delivering information to consumers using different standards for each product because there is no standard for quality control. In particular, the playback time of wireless earphones is a factor that directly affects consumers' purchases, so a standard measurement method is needed to properly measure it. In this paper, we present a method for measuring the audio playback time of wireless earphones derived from domestic wireless earphone status survey, commercial product measurement test, and research analysis, and a developed standard measurement method. In addition, this paper proposes a measurement result reporting format to provide accurate information to consumers and induce a fair competitive environment for each product to manufacturers.

• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.111-116
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• 2013

Because LEO (Low Earth Orbit) satellite has very limited contact time between satellite and ground station, all telemetry data generated on satellite are stored in a mass memory and transmitted to the ground during the contact time. There are two downlink modes, real-time mode and playback mode. Only real-time data frames are transmitted to the ground in real-time mode, real-time and playback data frames stored into mass memory are transmitted to the ground in playback mode. In accordance with the data transmission rate, there are two downlink rates, low downlink rate and high downlink rate. This paper briefly introduces downlink interfaces and flight software of a LEO satellite developed in KARI. And it presents the telemetry storage method, real-time and playback downlink management method, and downlink channel and rate control method.

• The Journal of the Korea Contents Association
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• v.18 no.3
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• pp.34-40
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• 2018

In this study, the optimal buffer size is calculated for seamless video playback on a mobile device. Buffer means the memory space for multimedia packet which arrives in mobile device for video play such as VOD service. If the buffer size is too large, latency time before video playback can be longer. However, if it is too short, playback service can be paused because of shortage of packets arrived. Hence, the optimal buffer size insures QoS of video playback on mobile devices. We model the process of buffering into a discret-time queueing model. Mean busy period length and mean waiting time of Geo/G/1 queue with N-policy is analyzed. After then, we uses the main performance measures to present numerical examples to decide the optimal buffer size on mobile devices. Our results enhance the user satisfaction by insuring the seamless playback and minimizing the initial delay time in VOD streaming process.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.1
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• pp.69-76
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• 2014

In this paper, we propose a buffering scheme to improve the performance in P2P live streaming systems by adjusting the ratio of caching and prefetching portion of each peer. To this end, we assign all the peers into many groups depending on their playback periods. We then determine the ratio of caching and prefetching portion in each peer depending on its playback time position relative to those of other peers within the same group. In other words, as the playback position of a peer gets later, we increase the ratio of its caching portion. On the contrary, as the playback position of a peer gets eariler, we increase the ratio of its prefetching portion. This can significantly increase the degree of data duplication among peers that belong to each specific group. By simulation experiments, we show that our proposed an integrated prefetching/caching scheme can improve the performance considerably in terms of jitter ratio, initial playback delay and shared buffermap ratio when compared to the existing fixed portion buffering scheme.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.105-112
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• 2011

We propose an adaptive load balancing scheme to transport real-time video streams efficiently in this paper. The playback buffer level of a video requesting client is high or low temporarily in mobile environment. This scheme attempts to allocate more network bandwidth to serve a video request with the lower buffer level preferentially. In this scheme, the amount of network bandwidth is dynamically allocated to the requesting clients according to their playback buffer levels in a distributed mobile system. In order to improve the quality of service and real-time performance of individual video playback, the proposed load balancing scheme tries to maximize the number of frames that are transported successfully to the client prior to their playback times. Fair services can also be provided to all the concurrent clients by making their playback situation more adaptive. The performance of this load balancing scheme is compared with that of other static load balancing scheme through extensive simulation experiments, resulting in the higher ratio of frames transmitted successfully within given deadlines.

• International Journal of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.166-172
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• 2020

Video streaming application such as YouTube is one of the most popular mobile applications. To adjust the quality of video for available network bandwidth, a streaming server provides multiple representations of video of which bit rate has different bandwidth requirements. A streaming client utilizes an adaptive bit rate scheme to select a proper video representation that the network can support. The download behavior of video streaming client player is governed by several parameters such as maximum buffer size. Especially, the size of the maximum playback buffer in the client player can greatly affect the user experience. To tackle this problem, in this paper, we propose the maximum buffer size optimization according to available network bandwidth and buffer status. Our simulation study shows that our proposed buffer size optimization scheme successfully mitigates playback stalls while preserving the similar quality of streaming video compared to existing ABR schemes.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.31-39
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• 1993

In the playback signal processing of a digital magnetic recording system, the major signal processing processes consist of pulse equalization. pulse detection, clock recovery, and data recovery. Equalizer which compensates interference occurrde between pulses recorded in high density on a magnetic media is realized by pulse slimming method, and pulse detection by a integrating detector. Clock recovery from the detector output was accomplished by using PLL. and data recovery to reduce noise effects was carried out by utilizing the three sampling clocks recovered in clock recovery process. In this paper these processes are implemented in hardware and its performance is evaluated by experimenting with a commercial DAT. It was found that the playback signal processor proposed is suitable to the practical high density magnetic recording system.