• Title/Summary/Keyword: 실시간 멀티미디어 스트리밍

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Real-Time Bandwidth Management Service for Effective Multiple Isochronous Streaming Transmission in IEEE1394 based Home Network (IEEE1394 기반 홈네트워크에서 효율적인 다중 등시성 스트리밍 전송을 위한 실시간 대역폭 관리 서비스)

  • Chae Hwa-Young;Jung Gi-Hoon;Kang Soon-Ju
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.9B
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    • pp.838-847
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    • 2006
  • In order to support multiple multimedia streaming services in home networks, many critical issues must be considered. In addition, handling the shortage of network bandwidth is one of the most significant and complicated issues. In this paper, real-time bandwidth management service is suggested as a solution to the problem regarding the IEEE1394-based home network. In order to handle the shortage of network bandwidth and to enhance the bus utilization rate, the proposed service combines two methods. First, the bus bandwidth management function determines the state of the network bandwidth and restores the residual bandwidth, which is excessively occupied by a streaming service, to the available free bandwidth. Second, the Isochronous Streaming (IS) Scheduler manages all streaming services according to priority. In order to test the proposed service, we implemented a prototype steaming management middleware and evaluated it by using the IEEE1394 network test-bed.

Video Transmission Technique based on Deep Neural Networks for Optimizing Image Quality and Transmission Efficiency (영상 품질 및 전송효율 최적화를 위한 심층신경망 기반 영상전송기법)

  • Lee, Jong Man;Kim, Ki Hun;Park, Hyun;Choi, Jeung Won;Kim, Kyung Woo;Bae, Sung Ho
    • Journal of Broadcast Engineering
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    • v.25 no.4
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    • pp.609-619
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    • 2020
  • In accordance with a demand for high quality video streaming, it needs high data rate in limited bandwidth and more traffic congestion occurs. In particular, when providing real time video service, packet loss rate and bit error probability increase significantly. To solve these problems, a raptor code, which is one of FEC(Forward Error Correction) techniques, is pervasively used in the application layers as a method for improving real-time service quality. In this paper, we propose a method of determining image transmission parameters based on various deep neural networks to increase transmission efficiency at a similar level of image quality by using raptor codes. The proposed neural network uses the packet loss rate, video encoding rate and data rate as inputs, and outputs raptor FEC parameters and packet sizes. The results of the proposed method present that the throughput is 1.2% higher than that of the existing multimedia transmission technique by optimizing the transmission efficiency at a PSNR(Peak Signal-to-Noise Ratio) level similar to that of the existing technique.

A Study of Performance Analysis on Effective Multiple Buffering and Packetizing Method of Multimedia Data for User-Demand Oriented RTSP Based Transmissions Between the PoC Box and a Terminal (PoC Box 단말의 RTSP 운용을 위한 사용자 요구 중심의 효율적인 다중 수신 버퍼링 기법 및 패킷화 방법에 대한 성능 분석에 관한 연구)

  • Bang, Ji-Woong;Kim, Dae-Won
    • Journal of Korea Multimedia Society
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    • v.14 no.1
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    • pp.54-75
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    • 2011
  • PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

Modifications of DCF Scheme to Alleviate Jitter Property in WLANs (무선 랜의 지터 성능 개선을 위한 DCF 개선 방안)

  • Kang, Dong-Hun;Seok, Seung-Joon
    • Journal of Internet Computing and Services
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    • v.9 no.5
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    • pp.153-163
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    • 2008
  • Almost literature has so far proposed the modifications to IEEE 802.11 DCF MAC protocol to improve system throughput and average delay. Jitter(variance in packet delays) is, however, considered as an important QoS parameter for real-time streaming service which is popularized increasingly. Unfortunately, legacy access protocol for WLAN can't support the jitter. This paper proposes two schemes modifying the DCF MAC protocol to reduce the average value of jitter for multimedia frames arrived at received nodes. The one scheme, referred to as DCW-DCF(Decreasing Contention Window-DCF), reduces CW(Contention Window) size by half whenever a transmitted frame collided against other frames and back-off happens. Also, it increase initial CW size according as the link utilization and the collision probability increase. The other scheme, referred to as D-DIFS(Differential-DIFS), assigns different values to DIFS. That is, the DIFS interval is deceased as the number of back-off for a frame increases. In this paper, the two schemes are evaluated through simulation using ns2 and simulation results show that the proposed schemes are effective for improving the jitter property of standard WLAN.

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IP Over USB for Improved QoS of UDP/IP Messages (UDP/IP 메시지 전송의 QoS 성능 향상을 위한 IP Over USB)

  • Jang, Byung-Chul;Park, Hyeon-Hui;Yang, Seung-Min
    • The KIPS Transactions:PartA
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    • v.14A no.5
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    • pp.295-300
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    • 2007
  • The Linux-based embedded systems such as mobile telephones. PDAs and MP3 players are widely in use. USB(Universal Serial Bus) is the interface for data communication between the computers and these peripheral devices. Some embedded systems like intelligent home networking and multimedia streaming require guaranteed QoS(Quality of Service), which is needed for real time transmission of UDP/IP messages through USB. Although USB Ethernet driver is supported by USB Gadget API in Linux, it is unable to provide the desirable QoS required by each type or small embedded systems due to the unpredictability or TCP/IP Stack in Linux. This paper proposes IP-Over-USB to improve QoS of UDP/IP message transmission in the embedded systems using USB in Linux system.

Hybrid Error Concealment Algorithm for Intra-Frame in H.264 (H.264의 인트라 프레임을 위한 하이브리드 에러 은닉 알고리즘)

  • Yim Chang-Hoon;Kim Won-Jung;Lim Hye-Sook
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.8C
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    • pp.777-785
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    • 2006
  • H.264 is the prominent video coding standard in various applications such as real-time video streaming and digital multimedia broadcasting, since it provides enhanced compression performance, error resilience tools, and network adaptation. Since compressed video stream is vulnerable to packet loss, error resilience and error concealment(EC) tools are essential for the transmission of video over the Internet. In this paper, we first propose a simple temporal EC method that improves the EC performance for intra-frame in H.264 when the amount of motion is relatively small. Then we propose a new hybrid EC method for intra-frame in H.264, which combines the spatial EC and temporal EC adaptively. The simulations are performed in packet-lossy environments, and the proposed hybrid EC method shows about 0.5-4dB PSNR improvement compared to the conventional spatial EC method that is used for intra-frame in H.264.

Audio Quality Enhancement at a Low-bit Rate Perceptual Audio Coding (저비트율로 압축된 오디오의 음질 개선 방법)

  • 서정일;서진수;홍진우;강경옥
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.6
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    • pp.566-575
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    • 2002
  • Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

Resolution Scheme of CRP Reservation Conflicts for Supporting QoS in ECMA-392-Based Ship Area Networks (ECMA-392 기반 선박 내 네트워크에서 QoS를 지원하는 CRP 예약 충돌 해결 방안)

  • Lee, Seong Ro;Oh, Joo-Seong;Kim, Beom-Mu;Lee, Yeonwoo;Jeong, Min-A;Lee, Seung Beom
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.39C no.12
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    • pp.1298-1306
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    • 2014
  • In this paper, we use the ECMA-392 standard, the first cognitive radio networks to meet the demand for IT services in the ship. And, we investigate about the conflicts of devices that support multi-hop network based on the ECMA-392. Therefore, we propose the resolution scheme of CRP reservation conflicts to solve this. The current CRP reservation approaches does not solve the CRP reservation conflicts of ECMA-392 between devices that move at a distance of three hops. Therefore, it is not possible to guarantee the QoS of real time services such as multimedia streaming. So, we provide a new CRP reservation scheme to avoid conflict through a new 2-hop CRP Availability IE and the change of CRP Control field. From the simulation results, we know that the proposed method of 3 hops CRP reservation conflicts resolution improves the throughput of device.