On kernel support for real-time multimedia applications

A scalable framework for mobile real-time group communication services

<div class="htmlview paragraph">The new generation of drive-by-wire systems currently under development has demanding requirements on the electronic architecture. Functions such as brake-by-wire or steer-by-wire require continued operation even in the presence of component failures. The electronic architecture must therefore provide fault-tolerance and real-time response. This in turn requires the operating system and the communication layer to be predictable, dependable and composable. It is well known that this properties are best supported by a time-triggered approach. A consortium consisting of German and French car manufacturers and suppliers, which aims at becoming a working group within the OSEK/VDX initiative, the OSEKtime consortium, is currently defining a specification for a time-triggered operating system and a fault-tolerant communication layer.<span class="xref"><sup>1</sup></span> The operating system and the communication layer are based on applicable interfaces of the OSEK/VDX standard. This paper describes the basic concepts of these two service layers and the electronic architecture.</div>

Service Oriented Computing (SOC) is a computing paradigm broadly pushed by vendors, utilizing services to support the rapid development of distributed applications in heterogeneous environments. The visionary promise of SOC is a world of cooperating services being loosely coupled to flexibly create dynamic business processes and agile applications that may span organizations and computing platforms and can nevertheless adapt quickly and autonomously to changes of requirements or context. Consequently, the subject of Service Oriented Computing is vast and enormously complex, spanning many concepts and technologies that find their origins in diverse disciplines like Workflow Management Systems, Component Based Computing, "classical" Web applications, and Enterprise Application Integration (EAI) including Message Oriented Middleware. In addition, there is a strong need to merge technology with an understanding of business processes and organizational structures, a combination of recognizing an enterprise's pain points and the potential solutions that can be applied to correct them. Middleware, on the other hand, is defined by the ObjectWeb consortium as the software layer in a distributed computing system that lies between the operating system and the applications on each site of the system. Middleware is the enabling technology of system and enterprise application integration (EAI) and therefore it clearly and evidently plays a key role for SOC. While the immediate need of middleware support for Service Oriented Architectures (SOA) is evident, current approaches and solutions mostly fall short by primarily providing support for the EAI aspect of SOC only and do not sufficiently address composition support, service management and monitoring. Moreover, quality properties (in particular dependability and security) need to be addressed not only by interfacing and communication standards, but also in terms of integrated middleware support. But what makes these issues so different in a SOA setting? Why - for instance - is traditional middleware support for transaction processing different to transaction processing in SOA, reflecting different types of atomicity needs? One answer lies in the administrative heterogeneity, the loose coupling between coarsegrained operations and long-running interactions, high dynamicity, and the required flexibility during run-time. Recently, massive-scale and mobility were added to the challenges for Middleware for SOC. However, loose coupling is not always the best approach to solve a particular problem. In order to temporarily allow for a stronger (traditional) form of coupling (like group membership agreement or atomic transaction), the middleware also has to provide explicit and configurable means to change between different strengths of coupling and various communication paradigms. This will enable servicebased applications to take the best from both worlds by dynamically adjusting the interactions between the composed services. The highly dynamic modularity and need for flexible integration of services (e.g. Web service implementations) may require new middleware architectures, protocols, and services. These considerations also lead to the question to what extent service-orientation at the middleware layer itself is beneficial (or not). Recently emerging "Middleware as service" offerings, from providers like Amazon or from the open source community, support this trend towards "infrastructure services" that can be purchased and consumed over the Internet. However, this model may not be suitable for all kinds of middleware functions, including those addressing dependability (availability, reliability, integrity, safety, maintainability). Providing end-to-end properties and addressing cross-cutting concerns in a crossorganizational SOA is a particular challenge and the limits and benefits thereof have still to be investigated.

The paper proposes a transmission system for Motion-JPEG2000 video in mobile communications. In Motion JPEG2000, video data are compressed in frames according to JPEG2000 part1. In JPEG2000 part1 encoding, an input image is coded in the order of significant sub-bands, which are low-frequency bands. The proposed transmission system therefore assigns each sub-band data suitable weights according to their significance by using orthogonal multi-code CDMA so that bit errors are concentrated in the high-frequency sub-bands which are of little significance on image reconstruction. Concretely speaking, copies of significant sub-band data are made and the copied data are orthogonally multiplexed. The multiplexed data are averaged in the receiver so that the bit errors of significant sub-bands are reduced. On the other hand, the other sub-band data are not copied in order to increase transmission rates. Furthermore, audio data are transmitted with high quality by orthogonal multiplexing. The proposed system adaptively performs the above process for video and audio data. Moreover, these proposed systems are effective in the progressive order of LRCP (layer - resolution level - component - position) and of RLCP (resolution level - layer - component - position) on JPEG2000. Simulation results demonstrate that the proposed system can improve image quality compared with the conventional transmission system under the same BER.

Constructing an embedded device with a real-time and a general-purpose operating system has attracted attention as a promising approach to let the device balance real-time responsiveness and rich functionalities. This paper introduces our methodology for constructing such multi-OS platform with minimal engineering cost by assuming asymmetric OS combinations unique to embedded systems. Our methodology consists of two parts. One is a simple hypervisor for multiplexing resources to be shared between operating systems. The other is modifying operating systems to allow them to be aware of each other. We constructed an experimental system executing TOPPERS and Linux simultaneously on a hardware equipped with an SH-4A processor. The modification to each operating system kernel limited to a few dozen lines of code and do not introduce any overhead that would compromise real-time responsiveness or system throughput.KeywordsEmbed SystemEngineering CostEmbed DeviceVirtual Machine MonitorVirtual ProcessorThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

Field Programable Gate Array (FPGA) implementation of audio and video processing based on Zedboard as a target platform. The Design for the audio system was implemented by verifying various audio files. Verification was done by hearing it on a speaker accordingly by connecting the stereo cable(male-to-male) to a line in (blue plug) of the board to the audio port of the laptop or phone and speaker to line out (green plug) of the board to the USB port of the laptop. And design for the video/monitoring system was implemented by interfacing the OV7670 CMOS camera module through peripheral modules of the board, VGA monitor screen through the VGA port of the board. The algorithms for the audio processing system were implemented in Verilog, a Hardware Description Language, and the video processing system was implemented in VHSIC (Very High Scale Integrated Circuits) Hardware Description Language (VHDL) in VIVADO 2017.4 as a software platform. Step by step approach to design real-time audio & video processing systems has been discussed in this paper.

Providing a customer with tailored location-based services (LBSs) is a fundamental problem. For location-estimation techniques with radio-based measurements, LBS applications are widely available for mobile devices (MDs), such as smartphones, enabling users to run multi-task applications. LBS information not only enables obtaining the current location of an MD but also pro- vides real-time push-pull communication service. For indoor environments, localization technologies based on radio frequency (RF) pattern-matching approaches are accurate and commonly used. However, to survey radio information for pattern-matching approaches, a considerable amount of time and work is spent in indoor environments. Consequently, in order to reduce the system-deployment cost and computing complexity, this article proposes an indoor positioning approach, which involves using Asus Xtion to facilitate capturing RF signals during an offline site survey. The depth information obtained using Asus Xtion is utilized to estimate the locations and predict the received signal strength (RF information) at uncertain locations. The proposed approach effectively reduces not only the time and work costs but also the computing complexity involved in determining the orientation and RF during the online positioning phase by estimating the user�s location by using a smartphone. The experimental results demonstrated that more than 78% of time was saved, and the number of samples acquired using the proposed method during the offline phase was twice as much as that acquired using the conventional method. For the on- line phase, the location estimates have error distances of less than 2.67 m. Therefore, the proposed approach is beneficial for use in various LBS applications.

UMTS is designed to offer high bandwidth radio access with QoS assurances for multimedia communications. In particular, real-time video communications services are expected to become a successful experience under UMTS networks. In this context, a video transmission service can be designed over the basis that UMTS can provide either a constant bit rate data channel or a dynamic variable bit rate data channel adapted to load conditions. In this latter approach, which is more efficient for both the user and the service provider, multimedia sources have to be timely designed in order to adapt their output rate to the instantaneous allowed channel rate. The target of this paper is to define an analytical model of adaptive real-time video sources in a UMTS network where system resources are dynamically shared among active users.

Existing and upcoming distributed multimedia applications require highly diverse services to satisfy their communication needs. Service integrated communication systems should be capable of providing high-performance real-time multipoint communication service with guaranteed quality of service (QoS). Existing communication systems and known strategies for resource reservation face increasing difficulties in fulfilling these requirements, in particular in high-speed wide area networks. Therefore, new concepts are required to support the variety of emerging applications in a heterogeneous internetworking environment. In this paper, a framework for real-time multipeer services is presented. It is based on the separation of service requirements into network bearer and transfer service requirements. The transfer service enhances the network bearer service in order to meet the service requirements of the applications. Applications and transport systems interact using an enhanced service interface, which offers several QoS parameters. The transfer service is supported by transfer components (layer 2b-4 protocol functions in end and intermediate systems), resource management functions and the integration of specialized VLSI modules for time-critical processing tasks. The hardware components can be parametrized and selected individually dependent on the required service. The transfer system guarantees specified service qualities by assigning processing resources to specific connections. Selection of the appropriate combination of components and their parametrization enhances the bearer service of the underlying network in order to provide the required multimedia service at the transport service interface. Guaranteed services are realized by the reservation of resources both in the network (for guaranteeing network bearer service) and in transfer components (for guaranteeing transfer service). The paper describes a general approach towards high performance multipeer services, and dedicated parts in more detail. Preliminary performance results of VLSI components are presented and compared with measurements of typical software implementations.KeywordsPacket LossSchedule AlgorithmForward Error CorrectionMulticast GroupMulticast ConnectionThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

In this letter, we address the problem of resource allocation with efficiency and quality of service (QoS) support in uplink for a wireless CDMA network supporting real-time (RT) and non-realtime (NRT) communication services. For RT and NRT users, there are different QoS requirements. We introduce and describe a new scheme, namely, traffic aided uplink opportunistic scheduling (TAUOS). While guaranteeing the different QoS requirements, TAUOS exploits the channel condition to improve system throughput. In TAUOS, the cross-layer information, file size information, is used to improve fairness for NRT users. Extensive simulation results show that our scheme can achieve high system throughput in uplink wireless CDMA systems, while guaranteeing QoS requirements.

A new approach to reducing delay jitter that uses multipath routing and redundant packets is described. Delay jitter causes data loss at the application layer and prevents the smooth provision of real-time multimedia communication services. In this paper, a policy of multipath routing for reducing delay jitter is discussed and an algorithm is proposed that solve the problem of selecting two paths for reducing delay jitter. The algorithm outputs a set of paths that satisfy a specified delay constraint; they share a few nodes and have a small delay difference. Simulation results showed that the set of paths is more suitable for reducing delay jitter than those selected using the routing method of a path diversity scheme.

Task-oriented semantic communication offers a promising solution for providing real-time computer vision services. However, existing research on semantic communication ignores the connection between the key performance indicators (KPIs) used to measure semantic encoding-decoding networks (i.e., inference accuracy) and wireless semantic communication networks (i.e., transmission latency). Therefore, the designed semantic communication schemes are difficult to simultaneously meet the requirements of low latency and high accuracy of emerging intelligent applications. In this paper, by deeply exploring the relationship and interdependencies between the two kinds of KPIs, we propose a real-time and efficient task-oriented end-to-end semantic communication scheme enabled by deep reinforcement learning (DRL) and information bottleneck to improve both inference and communication efficiency. Specifically, we initially use information bottleneck theory to model the optimal tradeoff between inference accuracy and communication latency, which is subsequently reformulated by variational inference to be differentiable and tractable. Then, we introduce DRL to address the non-differentiability of dynamic stochastic fading channels and channel mismatch between the training phase and deployment phase, enabling accurate selection of the most task-relevant semantic feature dimensions for transmission under dynamic fading channels. Finally, extensive experiments show that our proposed scheme achieves better performance in latency and accuracy than comparison methods.

Modern broadband access networks increasingly carry heterogeneous, delay-sensitive traffic generated by cloud-assisted applications, real-time communication services, and short-form multimedia exchanges. These workloads have bursty ON/OFF packet dynamics, creating dense arrival clusters that conventional GPON/XGS-PON upstream schedulers struggle to handle. This work proposes an AI-driven Dynamic Bandwidth Allocation (AI-DBA) framework based on a Deep Q-Network (DQN) that observes global queue states for video, voice, and data traffic and performs adaptive admission/prioritization/scheduling decisions. The DQN agent learns an adaptive scheduling policy through experience replay and reward-driven interactions to anticipate burst formations, drain high-pressure queues, and smooth resource allocation. Simulation results show that DQN-DBA maintains near-zero buffer queues and eliminates observed packet loss across all traffic classes, and stablize the delay and jitter profiles, whereas TCP and QCT-ARED experience long queue buildup, increased loss rates, and high delay/jitter variability under the same bursty traffic conditions. Across both experiments, the proposed approach consistently outperforms all baselines. Against the Non-AI scheme, the AI-based model reduces packet loss by 72.84% and average waiting time (latency) by 41.73%. In the second experiment, across video, voice, and data traffic, DQN-DBA achieves a 100% reduction in observed packet loss compared to both TCP and QCT-ARED, reduces steady-state delay by approximately 74.7% relative to TCP and 66.3% relative to QCT-ARED, and improves throughput by about 558–634% over TCP and 343–384% over QCT-ARED.The results confirm that AI-driven upstream scheduling provides a robust, scalable, and highly adaptive solution for managing bursty multi-traffic loads in next-generation PON architectures.

Interactive Live streaming is the process of transferring real-time data over the internet. The data can be of type audio, video or any common data type. Interactive live streaming is an important feature of applications and platforms in which the actions of the audience affect the content of the communication. Simultaneously, Podcasting is one of the types of contents that people are attracted to. The idea of podcasting includes recording the audio or video file and uploading it to a platform where people can access it. Now, with the thought of mixing these two, the idea of an interactive podcast was born. We analyze that low delay is a restriction for Web Interactive Streaming. Interactive communication over the internet with the restrictions of real-time boundaries requires a framework that will allow the development of the proposed project. Real Time Communication (RTC) has wide-industry uses. RTC is the standard and it also increases the browsing model which allows access to live streaming systems which consist of social media, television, chatting applications, as well as the communication media. Users are allowed to read comments, write comments, record the sessions/audio/video, and edit audio/videowhich is done within a Cloud Infrastructure which also provides services of quality. The development of the application is the most important part where using different technologies a system can stream audio and video data over the internet, handle a number of users who can make audio rooms to discuss a topic among themselves, where several people can hear them, and they also have a chance to add value to the conversation with their inputs.

With the rise of the Internet, computer systems appear to be more vulnerable than ever from security attacks. Much attention has been focused on the role of the network in security attacks, but evidence suggests that the computer server and its operating system deserve closer examination since it is ultimately the operating system and its core defense mechanisms of authentication and authorization which are compromised in an attack. This chapter provides an exploratory and evaluative discussion of the authentication and authorization features of two widely used server operating systems: Windows and Linux.