Adaptive multimedia synchronization in a teleconference system

Xie, Yong; Liu, Changdong; Lee, Myung J.; Saadawi, Tarek

doi:10.1007/s005300050134

Cited by 23 publications

(8 citation statements)

References 27 publications

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“…Also the calculation complexity increases as window size increases while calculating network statistics. At the same time, as the window size decreases, the adaptation is more responsive to the volatile network behavior [7] [14]. So it is important to select the window size which adapts and estimates accurate network characteristics.…”

Section: Results and Analysismentioning

confidence: 99%

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A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

Chandran¹,

Lingam²

2018

IJECE

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Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms.

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Section: Results and Analysismentioning

confidence: 99%

“…B. H. Kim et al [12] used a static buffer size of 200ms to hold packets. Event counting algorithm proposed in [14] relies on the measure of relative times in handling the object-related events. Normal approximation is used to identify the delay of network in [15].…”

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confidence: 99%

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

Chandran¹,

Lingam²

2018

IJECE

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“…They are: 1) "Cushion size". In order to counteract the delay introduced by the operating system of the Destination Terminal, the RAT tool [16] estimates a "cushion 6 Root Mean Square Error size", i.e. a lower bound for the playout delay so as to prevent sound card buffer from emptying.…”

Section: Resultsmentioning

confidence: 99%

“…For each speech/video couple of MDUs to be played out simultaneously, the stream with lower buffering time becomes the slave and increases the delay in order to achieve the lip-sync. The master/slave strategy proposed in [5] is an improvement of the implementation proposed in [6], and it has been further refined in [7] and applied to Wi-Fi technology in [8]. Briefly, [5] defines a set of thresholds related to the loss ratio and to the Intra-Sync error for both speech and video streams.…”

Section: Introductionmentioning

confidence: 99%

A Synchronization Control Scheme for Videoconferencing Services

Bartoli¹,

Iacovoni²,

Ubaldi³

2007

JMM

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We propose a synchronization control scheme which achieves both speech/video Intra-Stream synchronizations and Inter-Stream synchronization for videoconferencing services over IP networks. The driving principle of our scheme is to guarantee the Intra-Sync speech timing relationships (hence the speech quality) and to adjust the video Intra-Sync and the Inter-Sync accordingly. Towards this aim we use a preventive control for the speech stream and a reactive control for the video stream. More precisely, we use an adaptive playout algorithm that keeps the Intra- Sync constraints within the talkspurts, while the network jitters are compensated by modifying only the silence period lengths on the basis of both speech and video packet delays. We implemented our scheme in a prototype, which allowed us to test the effectiveness of our solution. Actually, we could appreciate both perfect speech intelligibility and very satisfactory user perceived lip-sync. The latter is because the Inter-Sync error is concentrated only at the beginning of the talkspurts, where known experimental tests have shown that it is not detectable.

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“…A second method proposes that the generating time of each object is time-stamped by the local Sample Clock (SC), and a set of PlayOut Clocks (POCs)-one for each incoming stream-is provided at the sink. In this method, an algorithm is employed to calibrate the PlayOut Clocks (POCs), which manages the presentations of multimedia streams [8]. The limitation of these synchronization mechanisms based on temporal relationships is that they do not provide the appropriate synchronization under different delay and jitter conditions.…”

Section: Advances In Multimediamentioning

confidence: 99%

Real-Time Adaptive Content-Based Synchronization of Multimedia Streams

Elhajj

Dargham

et al. 2011

Advances in Multimedia

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Traditional synchronization schemes of multimedia applications are based on temporal relationships between inter- and intrastreams. These schemes do not provide good synchronization in the presence of random delay. As a solution, this paper proposes an adaptive content-based synchronization scheme that synchronizes multimedia streams by accounting for content in addition to time. This approach to synchronization is based on the fact that having two streams sampled close in time does not always imply that these streams are close in content. The proposed scheme primary contribution is the synchronization of audio and video streams based on content. The secondary contribution is adapting the frame rate based on content decisions. Testing adaptive content-based and adaptive time-based synchronization algorithms remotely between the American University of Beirut and Michigan State University showed that the proposed method outperforms the traditional synchronization method. Objective and subjective assessment of the received video and audio quality demonstrated that the content-based scheme provides better synchronization and overall quality of multimedia streams. Although demonstrated using a video conference application, the method can be applied to any multimedia streams including nontraditional ones referred to as supermedia like control signals, haptic, and other sensory measurements. In addition, the method can be applied to synchronize more than two streams simultaneously.

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Adaptive multimedia synchronization in a teleconference system

Cited by 23 publications

References 27 publications

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

A Synchronization Control Scheme for Videoconferencing Services

Real-Time Adaptive Content-Based Synchronization of Multimedia Streams

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