scholarly journals Video and Audio Fusion for Streaming Applications

Author(s):  
Wei Lu

This thesis explores a technique for the fusion of streamed audio and video services for real-time applications. It discusses certain novel techniques used to overcome the problems with video and audio synchronization over the Internet of a tele-bot. We have developed a demonstration called the WAX, at the Network-Centric Applied Research Team (N-CART) laboratory located within the School of Computer Science at Ryerson University. WAX is equipped with an onboard camera and a microphone, as well as a 2.4 GHz wireless transceiver for transmitting video and audio feeds, and at the same time receiving commands from the WAX robot server. By launching a web browser and loading the Java client applets, a user can see as well as hear what is around WAX in near real-time, while being able to move the robot around its environment.

2021 ◽  
Author(s):  
Wei Lu

This thesis explores a technique for the fusion of streamed audio and video services for real-time applications. It discusses certain novel techniques used to overcome the problems with video and audio synchronization over the Internet of a tele-bot. We have developed a demonstration called the WAX, at the Network-Centric Applied Research Team (N-CART) laboratory located within the School of Computer Science at Ryerson University. WAX is equipped with an onboard camera and a microphone, as well as a 2.4 GHz wireless transceiver for transmitting video and audio feeds, and at the same time receiving commands from the WAX robot server. By launching a web browser and loading the Java client applets, a user can see as well as hear what is around WAX in near real-time, while being able to move the robot around its environment.


Author(s):  
Zhonghua Yang ◽  
Yanyan Yang ◽  
Yaolin Gu ◽  
Robert Gay

The Internet has gone from near invisibility to near ubiquity and penetrated into every aspect of society in the past few years (Department of Commerce, 1998). The application scenarios have also changed dramatically and now demand a more sophisticated service model from the network. In the early 1990s, there was a large-scale experiment in sending digitized voice and video across the Internet through a packet-switched infrastructure (Braden, Clark, & Shenker, 1994). These highly visible experiments have depended upon three enabling technologies: (a) Many modern workstations now come equipped with built-in multimedia hardware, (b) IP multicasting, which was not yet generally available in commercial routers, is available, and (c) highly sophisticated digital audio and video applications have been developed. It became clear from these experiments that an important technical element of the Internet is still missing: Real-time applications often do not work well across the Internet. The Internet, as originally conceived, offers only a very simple quality-of-service (QoS), point-to-point, best-effort data delivery. However, for a real-time application, there are two aspects of the problem with using this service model. If the sender and/or receiver are humans, they simply cannot tolerate arbitrary delays; on the other hand, if the rate at which video and audio arrive is too low, the signal becomes incomprehensible. To support real-time Internet applications, the service model must address those services that relate most directly to the time of delivery of data. Real-time applications like video and audio conferencing typically require stricter guarantees on throughput and delay. The essence of real-time service is the requirement for some service guarantees in terms of timing. In response to these demands of real-time multimedia applications, the Internet Engineering Task Force (IETF) has significantly augmented the Internet protocol stack based on the Internet integrated-services model, which is the focus of this article.


2008 ◽  
pp. 809-817
Author(s):  
Z. Yang ◽  
Y. Yang ◽  
Y. Gu ◽  
Robert Gay

The Internet has gone from near invisibility to near ubiquity and penetrated into every aspect of society in the past few years (Department of Commerce, 1998). The application scenarios have also changed dramatically and now demand a more sophisticated service model from the network. In the early 1990s, there was a large-scale experiment in sending digitized voice and video across the Internet through a packet-switched infrastructure (Braden, Clark, & Shenker, 1994). These highly visible experiments have depended upon three enabling technologies: (a) Many modern workstations now come equipped with built-in multimedia hardware, (b) IP multicasting, which was not yet generally available in commercial routers, is available, and (c) highly sophisticated digital audio and video applications have been developed. It became clear from these experiments that an important technical element of the Internet is still missing: Real-time applications often do not work well across the Internet. The Internet, as originally conceived, offers only a very simple quality-of-service (QoS), point-to-point, best-effort data delivery. However, for a real-time application, there are two aspects of the problem with using this service model. If the sender and/or receiver are humans, they simply cannot tolerate arbitrary delays; on the other hand, if the rate at which video and audio arrive is too low, the signal becomes incomprehensible. To support real-time Internet applications, the service model must address those services that relate most directly to the time of delivery of data. Real-time applications like video and audio conferencing typically require stricter guarantees on throughput and delay. The essence of real-time service is the requirement for some service guarantees in terms of timing. In response to these demands of real-time multimedia applications, the Internet Engineering Task Force (IETF) has significantly augmented the Internet protocol stack based on the Internet integrated-services model, which is the focus of this article.


2011 ◽  
Vol 338 ◽  
pp. 796-799
Author(s):  
Wei Chang Feng

E-Yuan multimedia system is developed for the rich audio and video resource on the Internet and on its server side, it can automatically search and integration of network video and audio resources, and send to the client side for the user in real-time broadcast TV viewing, full use of remote control operation, Simply it’s a very easy to use multimedia system. This article introduces its infrastructure, main technical ideas and you can also see some details about server side and client side. At the same time, the improvement on how to collect and integrate video resources is comprehensively elaborated.


1998 ◽  
pp. 354-388 ◽  
Author(s):  
S. Rudkin ◽  
A. Grace ◽  
M. W. Whybray

1993 ◽  
Vol 2 (3) ◽  
pp. 244-258 ◽  
Author(s):  
Michael Zyda ◽  
David Pratt ◽  
John Falby ◽  
Paul Barham ◽  
Kristen Kelleher

The Naval Postgraduate School Networked Vehicle Simulator IV (NPSNET-IV) is a low-cost, student-written, real-time networked vehicle simulator that runs on commercial, off-the-shelf workstations (the Silicon Graphics IRIS family of computers). NPSNET-IV has been developed at the Naval Postgraduate School's (NPS) Department of Computer Science in the Graphics and Video Laboratory. It utilizes Simulation Network (SIMNET) databases and SIMNET and Distributed Interactive Simulation (DIS) networking formats. The DIS networking format is flexible enough to allow multiple players to game over the Internet. The availability of NPSNET-IV lowers the entry costs of researchers wanting to work with SIMNET, DIS, and follow-on systems. Without the contributions of the department's M.S. and Ph.D. candidates, the NPSNET project would be impossible to maintain and continue. The diversity of their interests accounts for the broad range of research areas within the project.


2021 ◽  
Author(s):  
Colin Xialin Huang

There are increasing demands for real-time streaming video applications over the Internet. However, the current generation Internet was not originally designed for real-time streaming applications and only provides best-effort services, so there are many challenges in the deployment of video streaming applications over the Internet. This thesis investigates a hybrid end-to-end rate adaptation framework that provides application-level enhancements to achieve Quality of Service (QoS) for MPEG-4 FGS-Encoded video bandwidth on the path and the terminal process capabilities based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the proportion of load status feedbacks from the receivers. The sender and the receivers act together to minimize the possibility of network congestion by adjusting the transmission rate to match the network conditions. This framework achieves inter-receiver fairness in a heterogeneous multicast environment and improves QoS stability for MPEG-4 FGS video streaming over the Internet.


2008 ◽  
pp. 1781-1788
Author(s):  
Christos Bouras ◽  
Apostolos Gkamas ◽  
Dimitris Primpas ◽  
Kostas Stamos

The heterogeneous network environment that Internet provides to real time applications as well as the lack of sufficient QoS (Quality of Service) guarantees, many times forces applications to embody adaptation schemes in order to work efficiently. In addition, any application that transmits data over the Internet should have a friendly behaviour towards the other flows that coexist in today’s Internet and especially towards the TCP flows that comprise the majority of flows. We define as TCP friendly flow, a flow that consumes no more bandwidth than a TCP connection, which is traversing the same path with that flow (Pandhye 1999).


2003 ◽  
Vol 91 (7) ◽  
pp. 1086-1104 ◽  
Author(s):  
M.A. El-Gendy ◽  
A. Bose ◽  
K.G. Shin

2011 ◽  
Vol 148-149 ◽  
pp. 7-10
Author(s):  
Wei Chang Feng ◽  
Xiao Meng Chen

E-Yuan multimedia system is developed for the rich audio and video resource on the Internet and on its server side, it can automatically search and integration of network video and audio resources, and send to the client side for the user in real-time broadcast TV viewing, full use of remote control operation, Simply it’s a very easy to use multimedia system. This article introduces its infrastructure, main technical ideas and you can also see some details about server side and client side.


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