• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2009.01a
• /
• pp.49-54
• /
• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• ETRI Journal
• /
• v.30 no.3
• /
• pp.441-450
• /
• 2008

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real-time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter-layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter-media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter-media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.

• Journal of Korea Multimedia Society
• /
• v.12 no.5
• /
• pp.653-663
• /
• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio. In the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Journal of Electrical Engineering and information Science
• /
• v.2 no.3
• /
• pp.41-50
• /
• 1997

In this paper, we describe the design, implementation and evaluation for a multimedia conferencing system with intramedia and intermedia synchronization support between audio and video. The synchronization mechanism proposed here is capable of dynamically adapting to various network conditions thus providing an optimized QoS. In realizing the system based on this mechanism, NeVoT on Mbone is used for audio and VIC for video. Furthermore a synchromization controller is designed and realized with a unique process in supporting intermedia synchronization. Each media agents handling its media stream are modified with intramedia synchronization function. And a communicative function between media agents and synchronization controller is added as well for intermedia synchronization function. Each media agents function reports its buffering status to the synchronization control process which in turn send out optimized buffering delay value thus supporting intermedia synchronization. The realized system is configured and tested on Ethernet and ATM network where performance measurements were performed and its effective synchronization support has been assured.

• Journal of Broadcast Engineering
• /
• v.23 no.2
• /
• pp.197-205
• /
• 2018

Panoramic image is one of the technologies that are commonly used today. However, technical difficulties still exist in panoramic video production. Without a special camera such as a 360-degree camera, making panoramic video becomes more difficult. In order to make a panoramic video, it is necessary to synchronize the timeline of multiple videos shot at multiple locations. However, the timeline synchronization method using the internal clock of the camera may cause an error due to the difference of the internal hardware. In order to solve this problem, timeline synchronization between multiple videos using visual information or auditory information has been studied. However, there is a problem in accuracy and processing time when using video information, and there is a problem in that, when using audio information, there is no synchronization when there is sensitivity to noise or there is no melody. Therefore, in this paper, we propose a timeline synchronization method between multiple video using audio waveform. It shows higher synchronization accuracy and temporal efficiency than the video information based time synchronization method.

• Journal of Information Processing Systems
• /
• v.15 no.4
• /
• pp.890-903
• /
• 2019

The single carrier frequency domain equalization (SC-FDE) technology is an important part of the broadband wireless access communication system, which can effectively combat the frequency selective fading in the wireless channel. In SC-FDE communication system, the accuracy of timing synchronization directly affects the performance of the SC-FDE system. In this paper, on the basis of Schmidl timing synchronization algorithm a timing synchronization algorithm suitable for FPGA (field programmable gate array) implementation is proposed. In the FPGA implementation of the timing synchronization algorithm, the sliding window accumulation, quantization processing and amplitude reduction techniques are adopted to reduce the complexity in the implementation of FPGA. The simulation results show that the algorithm can effectively realize the timing synchronization function under the condition of reducing computational complexity and hardware overhead.

• Proceedings of the IEEK Conference
• /
• 1999.11a
• /
• pp.1045-1048
• /
• 1999

In this paper, we report on a set of new algorithms to realize a nonlinear compressed-domain video/audio editor that overcomes various realization problems. For efficiency, the underlying algorithm, which uses a central data structure in the form of doubled linked lists, performs soft edits of cut and paste (which, in turn, involves soft implementations of frame type conversion) and addresses problems relating to video/audio synchronization and random access, and decoder buffer control.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.19 no.8
• /
• pp.1612-1627
• /
• 1994

There is a synchronization function which deals with only single media of text in the OSI Session Layer. So new synchronization schem and synchronization protocol are required for multimedia communications which include audio, video and graphic as well as text information. In this paper, conceptional Multmedia Synchronization Layer(MS layer) environment is composed and its service primitives and protocols based on 'multi-channel, base media scheme' are designed and proposed for multimedia synchronization services. This MS layer Manager (MSM) establishes the MS layer connection to the peer MS layer and manages each media channel which is created in MS layer media by media. The MSM also finds the synch-position through the media frame number by utilizing it like the time stamp to provide inter-media synchronization services as well as intra-media synchronization services.

• Journal of Korea Multimedia Society
• /
• v.10 no.8
• /
• pp.1039-1051
• /
• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.41 no.3
• /
• pp.155-162
• /
• 2004

This paper concentrated on lip synchronization (lip sync) test for DTV with respect to audio and video signals using a finite digital bitstream In this paper, we propose a new lip sync test method which does not effect on the current program by use of the transient effect area test signals (TATS) and audio-video time index lip sync test signals (TILS).the experimental result shows that the time difference between audio and video signal can be easily measured by captured oscilloscope waveform at any time.