Playback Time Research Articles

P2P 라이브 스트리밍 시스템을 위한 적응적 버퍼링 기법

In this paper, we propose a buffering scheme to improve the performance by adaptively adjusting the length of the buffering period according to the degree of data duplication among peers that belong to each specific playback period in P2P live streaming systems. To do so, we first assign all the peers into many groups depending on their playback time positions. We then determine the length of their buffering periods according to the number of peers belonging to each group. That is, we increase the length of the buffering periods when much buffered data are duplicated among peers within a group while we decrease it when few buffered data are duplicated. By simulation experiments, we show that our proposed buffering scheme can improve the performance significantly compared to the fixed length buffering scheme that have been employed in the existing mesh-based P2P systems.

Managing Digital Rights for P2P Live Broadcast and Recording on the Internet

Live broadcast over a peer-to-peer (P2P) network imposes a unique set of challenges to a digital rights management (DRM) system. Highly correlated service request arrivals at the start of a live event require peak-load provisioning if clients acquire licenses at playback time. Distributing the license management load across a P2P network requires the digital rights management system to ensure the integrity of both digital rights, the protection of client privacy and, at the same time, system scalability. In this paper we describe the requirements imposed on a digital rights management system in distributing live broadcast over a P2P network and present our design of such a system to meet the above challenges. We discuss the system's operation under a number of threat models and how to extend the system to further improve scalability and support network digital video recording (DVR). We close the paper after presenting some scalability results collected from a production P2P live broadcast network using our DRM design.

A 60 mW Class-G Stereo Headphone Driver for Portable Battery-Powered Devices

A 60 mW 1.15 mA/channel Class-G stereo headphone driver primarily designed for demanding applications in mobile phones and other portable communication devices is described. The architecture of the driver has been chosen to overcome the various design challenges for audio amplifiers in a wireless environment. A high-order feed-forward loop topology provides high immunity to battery disturbance with extended correction range. In addition, a Class-G amplifier with a Class-AB/B driving stage improves the small-signal efficiency and extends music playback time. Implemented in 0.18 μm CMOS technology, the stereo headphone driver achieves a DR of 111 dB and PSRR of 120 dB at the GSM TDMA frequency of 217 Hz while occupying an area of 2.3 mm <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> .

VBR 비디오 스트림 전송을 위한 모바일 클라이언트 버퍼 수준 기반 스케쥴링 알고리즘

모바일 통신망에서 클라이언트들의 재생 버퍼 수준을 이용하여 VBR 비디오 스트림을 전송하는 스케쥴링 알고리즘들이 본 논문에서 제시된다. 이 알고리즘들은 모바일 통신망에서 집중식 비디오 서버와 클라이언트들 간에 제한된 통신 대역폭을 최대한 활용할 수 있다. 비디오 서버는 동시에 다수의 비디오 요청을 서비스하기 때문에 통신망 대역폭을 공평하고 효율적으로 할당하여 활용하는 것이 중요하다. 각 비디오 재생시 서비스 품질과 실시간적 성능을 향상시키기 위하여 비디오 서버는 일시적으로 버퍼 수준이 낮은 비디오 요청 작업을 우선적으로 서비스하고자 더 많은 대역폭을 할당하게 된다. 버퍼 수준이 상이한 모바일 클라이언트들에게 공평한 서비스와 부하 균형이 제공되는 시뮬레이션 결과가 나타났으며 각 클라이언트의 재생 시간전까지 성공적으로 전송되는 프레임의 수가 최대화되었다. In this paper, we propose scheduling algorithms for transporting variable-bit-rate video stream using playback buffer level of the clients over wireless communication networks. The proposed algorithms attempt to maximize the utilization of limited bandwidth between the central video server and the clients over a mobile network. Since a video server may serve several video request at the same time, it is important to allocate and utilize network bandwidth to serve them fairly and efficiently. In order to improve the quality of service and real-time performance of individual video playback, the video server attempts to allocate temporarily more network bandwidth to serve a video request with the lower buffer level preferentially. The simulation results prove the fair service and load balancing among the mobile concurrent clients with different buffer levels and hence maximizing the number of frames that are transported successfully to the client prior to their playback times.

やり投げ競技を対象とした運動観察におけるスーパースロー映像の有効性に関する基礎的検討

Recently, video footage has been used for motion observation of primary players for sports instruction. In this paper, we analyze motion observation of an expert javelin thrower at several frame rates of attempt movie. We obtained an interview with an expert javelin instructor to investigate the effectiveness of high speed movies on motion observation with frame rates of 30, 60, 120, and 240 fps. As a result, (1) playback times, (2) pause times, (3) fast-forward or rewind times, and (4) playback order were measured. In addition, 57 of the meaningful units for instruction were obtained from the interview analysis. The results also show that the high speed movie is effective for figuring out a player's conscious of bodymotion and setting an agenda to adjust motion in dynamic postures for motor learning. The differences of the tendency of motion observation between each frame rate were also discussed.

Reducing Perceptual Discontinuity P2P Live Video Streaming Using Adaptive Scheme Based on Frame Loss Visibility

When a live streaming video is delivered over a peer-to-peer (P2P) network, the video is divided into several chunks and distributed to peers. Each chunk has its own playback time deadline. A chunk delay can be a problem of great concern because a serious chunk delay can produce obvious discontinuity of the streaming video, resulting in low satisfaction for a viewer. The playback time of a delayed chunk is overlapped with that of the next chunk. Traditionally, fast forwarding and uniform frame skipping are the two methods used to deal with the chunk delay problem. However, they may easily produce the effect of perceptual discontinuity and perceived by the viewer. This study proposes a frame dropping method based on frame loss visibility information in order to maintain visual continuity of the video. The frame with lower loss visibility will be dropped first. In a previous work, the idea of loss visibility was applied by a router to develop a frame dropping strategy when network congestion occurs. In this work, we apply it to the chunk-level playback controller in P2P network. The number of frames to be dropped is dependent on the value of chunk delay. Given a fixed chunk delay, the main difference between the proposed approach and the uniform frame skipping method is the way of choosing frames for dropping (fast forwarding method does not drop any frame but play all the frames at a faster speed). A viewer's satisfaction is evaluated by using the way of subjective video quality assessment. The proposed method can produce higher MOS (mean opinion score) than that of the traditional methods (fast forwarding and uniform frame skipping), demonstrating the effectiveness of the proposed approach.

IEEE 802.21-based seamless multicast streaming with dynamic playback control

With the explosion of video streaming available on the Internet, online multimedia applications become more popular in our days and the video quality of the Internet multimedia applications is directly affected by the network transmission state, which will be worse while switching the ongoing network connection from one wireless interface to another heterogeneous wireless technology, such as IEEE 802 families, UMTS (universal mobile telecommunications system) network and 3GPP LTE (3rd generation partnership project-long term evolution). In order to perform seamless handover between heterogeneous wireless networks, IEEE group proposed the “IEEE 802.21 standard” and defined a middleware function called “media independent handover function (MIHF)” to smooth the handover. In this paper, a dynamic playback control for multicasting streaming based on IEEE 802.21 is proposed to reduce the influence of handover between heterogeneous networks. In addition, we evaluate three different rate modes for seamless multicasting streams. The simulation results show that different playback frame rate modes, including the usual, incremental and linear modes, can achieve different video quality and can extend the playing time for handover video playback.

High-Quality P2P Video Streaming System Considering the Cooperation of Constitution Information and Delivery Status

Recently, video streaming services using P2P (Peer-to-Peer) have attracted attention to solve the problem of load concentration on servers and to reduce large latency. Many P2P streaming systems, like Coolstreaming, however, take a complicated approach to control playback timing severely. This leads to less churn resiliency and less adaptability to fluctuation of network traffic. Therefore, we focus on a simple and robust approach to realize “pseudo” streaming with high quality, which is based on BitTorrent. In the existing methods with the simple approach, peers download pieces just closer to playback timing to decrease the playback discontinuity. However, these methods do not consider the constitution of video structure in sophisticated manner. P2P streaming system must consider several important metrics for high-quality and fair distribution. Therefore, in this paper, we propose a new P2P video streaming system considering the cooperation of three important metrics; video structure, playback timing, and piece dispersion on network. In this system, users vary three piece selections to suit the delivery status. Specifically, users preferentially download pieces which affect the video quality, which are closer to playback timing, and which improve the delivery efficiency. Moreover, we show the effectiveness of the proposed method by computer simulation.

Adaptive Prefetching Scheme for Peer-to-Peer Video-on-Demand Systems with a Media Server

In this paper, we consider Peer-to-Peer Video-on-Demand (P2P VoD) systems based on the BitTorrent file sharing protocol. Since the Rarest First policy adopted in the original BitTorrent protocol frequently fails to collect pieces corresponding to a video file by their playback time, we need to develop a new piece selection rule particularly designed for P2P VoDs. In the proposed scheme, we assume the existence of a media server which can upload any piece upon request, and try to bound the load of such media server with two techniques. The first technique is to estimate pieces which are not held by any peer and prefetch them from the media server. The second technique is to switch the mode of each peer according to the estimated size of the P2P network. The performance of the proposed scheme is evaluated by simulation.

분산 모바일 멀티미디어 시스템에서 재생 버퍼 수준에 기반한 동적 대역폭 할당 기법

In this paper, we propose a scheme for dynamic allocating network bandwidth based on the playback buffer levels of the clients in a distributed mobile multimedia system. In this scheme, the amount of bandwidth allocated to serve a video request depends on the buffer level of the requesting client. If the buffer level of a client is low or high temporarily, more or less bandwidth will be allocated to serve it with an objective to make it more adaptive to the playback situation of this client. By employing the playback buffer level based bandwidth allocation policy, fair services can also be provided to the clients. In order to support high quality video playbacks, video frames must be transported to the client prior to their playback times. The main objectives in this bandwidth allocation scheme are to enhance the quality of service and performance of individual video playback such as to minimize the number of dropped video frames and at the same time to provide fair services to all the concurrent video requests. The performance of the proposed scheme is compared with that of other static bandwidth allocation scheme through extensive simulation experiments, resulting in the 4-9% lower ratio of frames dropped according to the buffer level.

Evaluation and Analysis of Buffer Requirements for Streamed Video Data in Video on Demand Applications

The integration of computers and communications is able to automate storage, processing and transportation of information. Video information is considered most bulky as it needs high space and consequently high bandwidth for real time playback. Thereby, Video on Demand (VOD) systems are inherently resource intensive and demanding with respect to the performance for continuous playback. The data stream is to be maintained between server and consumer for longer periods once playback of video begins. The restricted bandwidth of all contemporary media and the need to maintain long duration connectivity have been major bottleneck for jitter and breaks free VOD services. In this paper, an analysis and evaluation of buffer requirements for jitter and break free playback of video from a streaming server has been presented. In practice, the ordered delivery of data packets through public domain networks is not guaranteed, which can break the sequential playback of streamed video data. The performance of VOD system thus is critically dependent upon buffer available with the user. The limits and bounds for continuous playback versus buffer and real transmission plan have been illustrated, which have been identified by the authors during their recent research work cited under references. This paper also overviews the video streaming and playback processes for VOD systems and provides for graphical and formal representation of limits and bounds for buffers.

A novel server-side proxy caching strategy for large-scale multimedia applications

Nowadays, server-side Web caching becomes an important technique used to reduce the User Perceived Latency (UPL). In large-scale multimedia systems, there are many Web proxies, connected with a multimedia server, that can cache some most popular multimedia objects and respond to the requests for them. Multimedia objects have some particular characteristic, e.g., strict QoS requirements. Hence, even some efficient conventional caching strategies based on cache hit ratio, meant for non-multimedia objects, will confront some problems in dealing with the multimedia objects. If we consider additional resources of proxy besides cache space, say bandwidth, we can readily observe that high hit ratios may deteriorate the entire system performance. In this paper, we propose a novel placement model for networked multimedia systems, referred to as the H k / T model, which considers the combined influence of arrival rate, size, and playback time to select the objects to be cached. Based on this model, we propose an innovative Web caching algorithm, named as the ART-Greedy algorithm, which can balance the load among the proxies and achieve a minimum Average Response Time (ART) of the requests. Our experimental results conclusively demonstrate that the ART-Greedy algorithm outperforms the most popular and commonly used LFU (Least Frequently Used) algorithm significantly, and can achieve a better performance than the byte-hit algorithm when the system utilization is medium and high.

Multimedia IP architecture trends in the mobile multimedia consumer device

A dedicated media processor is used in many mobile consumer devices to accelerate video, image, graphics, and display processing. Increased demand for higher pixel resolution, higher quality image and video processing, more graphics performance necessitates dramatically increased signal processing capabilities. To provide the increased performance at acceptable cost and power requires redesign of the traditional architecture. By wisely partitioning algorithms across programmable and fixed-function blocks, the performance goals can be met while still maintaining flexibility for new feature requirements and new standards. For a better than acceptable user experience and playback time, all IPs (display, graphics, video, and imaging) have to be optimized as an “end to end” system. In this paper, an overview of the future trends of multimedia IP processor architectures is provided that describes the implications on system architecture, power, and performance.

A transmission control SCTP for real-time multimedia streaming

Multimedia streams over the Internet have a strict playback delay time, i.e., multimedia data arriving after the playback time cannot be played by the receiver and they are discarded. In this paper, we introduce a transmission control SCTP (TC-SCTP) which has a transmission control sub-layer (TCSL) in which the multimedia streaming server determines whether data can be played in the receiver before sending the data and decides whether to send data messages or not. In addition, TCSL employs differentiated retransmission policy depending on the type of multimedia. We evaluate the performances of SCTP, partial reliability (PR)-SCTP, and TC-SCTP via ns-2 simulator. The simulation results demonstrate that the TC-SCTP protocol can decrease the amount of transmissions and increase the video decodable ratio, compared with other protocols.

무선 LAN에서 MPEG-4 비디오 전송을 위한 대역폭 할당과 실시간 통신 스케쥴링 기법

Network bandwidth is one of the major factors that impact the cost of a video service. In this paper we propose approaches to reducing the bandwidth requirement for transporting MPEG-4 video traffic over wireless LANs while guaranteeing a required level of quality of service(QoS). To support high quality video playbacks, video frames must be transported to the client prior to their playback times. A real-time transmission scheduling is used for this purpose, which transmits each frame assigned with a priority according to its importance. It addresses the challenge for a scheduling algorithm that efficiently handles the changing workloads of MPEG-4 video traffic. The goal of our research is to maximize the number of frames that are transported within their deadlines while minimizing the tardiness of frames that missed their deadlines. The performance of the proposed method is compared with that of similar service mechanisms through extensive simulation experiments.

Scheduling Techniques for Media-on-Demand

Broadcasting popular media to clients is the ultimate scalable solution for media-on-demand. Recently, it was shown that if clients can receive data at a rate faster than what they need for playback and if they can store later parts of the media in their buffers, then much higher scalability may be obtained. This paper addresses scheduling problems arising from these new systems for media-on-demand. For given amount of bandwidth, we reduce the maximal start-up delay time for an uninterrupted playback. We achieve our results by introducing two techniques. In the first, the media is arranged on the channels such that clients gain from buffering later parts of the transmission before the actual start of the playback. In the second, segments of different media may be mixed together on the same channel. We introduce a simple class of recursive round-robin scheduling algorithms that implement both techniques. Our results improve the best known asymptotic results. Moreover, our scheduling algorithms outperform known results for practical values for number of media and number of broadcasting channels. For some specific small values, we present solutions that are better than those achieved by our algorithms. Finally, we show that our techniques are useful for models in which clients may not receive data from all the channels, and are applicable to media with different lengths and popularities.

Scalable On-Demand Streaming of Nonlinear Media

A conventional video file contains a single temporally-ordered sequence of video frames. Clients requesting on-demand streaming of such a file receive (all or intervals of) the same content. For popular files that receive many requests during a file playback time, scalable streaming protocols based on multicast or broadcast have been devised. Such protocols require server and network bandwidth that grow much slower than linearly with the file request rate. This paper considers ldquononlinearrdquo video content in which there are parallel sequences of frames. Clients dynamically select which branch of the video they wish to follow, sufficiently ahead of each branch point so as to allow the video to be delivered without jitter. An example might be ldquochoose-your-own-endingrdquo movies. With traditional scalable delivery architectures such as movie theaters or TV broadcasting, such personalization of the delivered video content is very difficult or impossible. It becomes feasible, in principle at least, when the video is streamed to individual clients over a network. For on-demand streaming of nonlinear media, this paper analyzes the minimal server bandwidth requirements, and proposes and evaluates practical scalable delivery protocols.

Piecewise Patching for Time-shifted TV Over HFC Networks

In video-on-demand (VOD) systems, a number of sophisticated architectures have been proposed to provide instantaneous services through the use of multicast technology. Most of the architectures focus on minimizing bandwidth requirements under the assumption that clients play from beginning to end, but this assumption is not suitable for time-shifted TV. Moreover, they have been proposed on an IP network, and failed to make full use of the broadcasting property of hybrid fiber coaxial (HFC) network. In this paper, we propose a novel delivery scheme, named as piecewise patching (PP), to tackle these problems. The piecewise patching scheme provides the video data for the users in a piecewise mode instead of a whole patching stream which includes all missing data. The former user can utilize the data from the later users' patching stream whose playback time is in advance of the former user's patching stream. In the proposed scheme, each client receives the data simultaneously from arbitrary channels which broadcast the same content. Simulation results show that the proposed scheme improves the performance of time-shift TV service significantly in terms of total server bandwidth requirement.

Cross-layer optimized rate adaptation and scheduling for multiple-user wireless video streaming

We present a cross-layer optimized video rate adaptation and user scheduling scheme for multi-user wireless video streaming aiming for maximum quality of service (QoS) for each user,, maximum system video throughput, and QoS fairness among users. These objectives are jointly optimized using a multi-objective optimization (MOO) framework that aims to serve the user with the least remaining playback time, highest delivered video seconds per transmission slot and maximum video quality. Experiments with the IS-856 (1timesEV-DO) standard numerology and ITU pedestrian A and vehicular B environments show significant improvements over the state-of- the-art wireless schedulers in terms of user QoS, QoS fairness, and the system throughput.

Synchronization modeling and its application for SMIL2.0 presentations

A novel synchronization model namely Extended Real-Time Synchronization Model (E-RTSM) for modeling SMIL2.0 temporal behaviors is proposed in this paper. E-RTSM deals with event-based/non-deterministic synchronization as well as schedule-based synchronization in SMIL2.0. Converting of the temporal relationship of a SMIL2.0 document to E-RTSM is presented. Moreover, design of the E-RTSM-based data-retrieving engine for SMIL2.0 presentations is also proposed in the paper. The data-retrieving engine estimates the worst-case playback time of each object at the parsing stage and applying an error compensation mechanism at run-time to adjust the estimated playback time as well as the schedule of the fetching requests for data retrieval. Performance measurements from the real implementation of the E-RTSM-based data-retrieving engine for SMIL2.0 presentations have demonstrated the efficiency of the proposed technique.