USB多路音频采集系统的设计与实现

提出了一种基于USB1.1的多路音频采集系统的实现方案。实际测试表明,该系统克服了现有系统中音频数据传输影响视频质量的缺陷,具有音频保真度高、成本低等突出优点。     

面向高级音频编码的通用隐写分析方法

针对面向高级音频编码(AAC)音频压缩标准的内容安全和隐写分析算法相对滞后的问题,提出一种面向AAC压缩域的通用隐写分析方法。该算法利用相邻的修正的离散余弦变换(MDCT)系数之间的相关性,构建基于帧间帧内多阶差分相关性的隐写分析子特征,结合AAC编码特性对子特征进行加权融合,得到用于通用隐写分析的特征集合,并采用随机森林组合分类器,实现了面向AAC MDCT系数修改的通用隐写分析。实验结果表明,所提算法对现有隐写算法能够实现有效的通用检测,在相对嵌入率为50%的条件下,各种隐写算法的检测率都能达到80%以上。     

AVS-P10音频解码器算法分析与定点化

AVS-P10是我国第一部应用于移动环境的音频编解码国家标准。在分析AVS-P10解码算法的基础上,对参考代码进行了精简和封装。针对定点处理器应用需求,对精简后的代码进行了定点化实现和优化。分别采用CMOS评分与SNR指标,对定点解码器的解码质量进行了主、客观测试,并对优化前后的定点解码器的运算效率进行了比对测试。结果表明,提出的AVS-P10定点解码器的解码音质达到与浮点解码信号的音质相当,且运算复杂度明显下降。     

Full-sharing: efficient bandwidth scheduling for video streaming over broadband cable networks (BCNs)

Broadband Cable Networks (BCNs) bring high-speed Internet access to home and make emerging multimedia streaming applications feasible. However, bandwidth contention is still a challenging problem in providing efficient IP-based Video-On-Demand (VOD) service on BCNs, due to the lack of effective approaches to exploit the unique characteristics of BCNs. To address the bandwidth contention issue, we propose an efficient video scheduling technique, called full-sharing scheduling in this paper. This technique fully exploits the unique characteristics of BCNs to reduce the bandwidth consumption of video sessions sharing a cable channel of fixed capacity, thereby maximizing the number of simultaneous video sessions on the single channel. Furthermore, we analyze the expected bandwidth and the session blocking probability of a video under the full-sharing scheduling. Based on this analysis, we design an efficient video assignment mechanism for maximizing the profit of a VOD system in scheduling videos on BCNs. Through both analysis and simulation, we show that our approach minimizes the bandwidth consumption of video sessions compared with the previous approaches and has significant advantages on BCNs. The proposed approach is also directly applicable on other broadcast/multicast networks in which clients have sufficient buffer and downstream bandwidth, e.g., satellite broadband networks. This work was supported in part by the National Science Foundation under the grants ANI-0073819, ITR-0085824, and CAREER Award NCR-9734428. Any opinions, findings, and conclusions or recommendations expressed in this paper are those of the authors and do not necessarily reflect the views of the National Science Foundation.     

An efficient scheduling algorithm for scalable video streaming over P2P networks

During recent years, the Internet has witnessed rapid advancement in peer-to-peer (P2P) media streaming. In these applications, an important issue has been the block scheduling problem, which deals with how each node requests the media data blocks from its neighbors. In most streaming systems, peers are likely to have heterogeneous upload/download bandwidths, leading to the fact that different peers probably perceive different streaming quality. Layered (or scalable) streaming in P2P networks has recently been proposed to address the heterogeneity of the network environment. In this paper, we propose a novel block scheduling scheme that is aimed to address the P2P layered video streaming. We define a soft priority function for each block to be requested by a node in accordance with the block’s significance for video playback. The priority function is unique in that it strikes good balance between different factors, which makes the priority of a block well represent the relative importance of the block over a wide variation of block size between different layers. The block scheduling problem is then transformed to an optimization problem that maximizes the priority sum of the delivered video blocks. We develop both centralized and distributed scheduling algorithms for the problem. Simulation of two popular scalability types has been conducted to evaluate the performance of the algorithms. The simulation results show that the proposed algorithm is effective in terms of bandwidth utilization and video quality.     

Adaptive interface selection over cloud-based split-layer video streaming via multi-wireless networks

As mobile devices such as tablet PCs and smartphones proliferate, the online video consumption over a wireless network has been accelerated. From this phenomenon, there are several challenges to provide the video streaming service more efficiently and stably in the heterogeneous mobile environment. In order to guarantee the QoS of real-time HD video services, the steady and reliable wireless mesh is necessary. Furthermore, the video service providers have to maintain the QoS by provisioning streaming servers to respond the clients’ request of different video resolution. In this paper, we propose a reliable cloud-based video delivery scheme with the split-layer SVC encoding and real-time adaptive multi-interface selection over LTE and WiFi links. A split-layer video streaming can effectively scale to manage the required channels on each layer of various client connections. Moreover, split-layer SVC model brings streaming service providers a remarkable opportunity to stream video over multiple interfaces (e.g. WiFi, LTE, etc.) with a separate controlling based on their network status. Through the adaptive interface selection, the proposed system aims to ensure the maximizing video quality which the bandwidth of LTE/WiFi accommodates. In addition, the system offers cost-effective streaming to mobile clients by saving the LTE data consumption. In our system, an adaptive interface selection is developed with two different algorithms, such as INSTANT and EWMA methods. We implemented a prototype of mobile client based on iOS particularly by using iPhone5S. Moreover, we also employ the split-layer SVC encodes in streaming server-side as the add-on module to SVC reference encoding tool in a virtualized environment of KVM hypervisor. We evaluated the proposed system in an emulated and a real-world heterogeneous wireless network environments. The results show that the proposed system not only achieves to guarantee the highest quality of video frames via WiFi and LTE simultaneous connection, but also efficiently saves LTE bandwidth consumption for cost-effectiveness to client-side. Our proposed method provides the highest video quality without deadline misses, while it consumes 50.6% LTE bandwidth of ‘LTE-only’ method and 72.8% of the conventional (non-split) SVC streaming over a real-world mobile environment.     

FESORIA: An integrated system for analysis,management and smart presentation of audio/video streaming services

Due to the elevated consumption of resources, the high cost of the production of contents and the quality of service required in audio/video streaming services, it is extremely important to optimize all the elements involved in the deployment of these services. With this goal in mind, provider companies have developed their management and presentation tools. At the same time, some specific tools for audio/video streaming analysis have appeared. Data are collected from servers and proxies by analyzing their log files in order to generate different types of reports. In spite of their utility, there is a disconnection between these types of tools. In this way, several important relationships between collected data are lost and the influence of other important aspects such as the behaviour of the users and their relationship with the subject or the length of the contents is not considered. This generates inaccurate analyses and the impossibility to improve the presentation, for example by generating recommendations using the information gathered from the analysis tool. Fesoria is a system which combines both characteristics. It is an analysis tool and, at the same time, a system to manage the whole audio/video service. Fesoria is able to process the logs gathered from the streaming servers and proxies, and combine the extracted information with other types of data, such as content metadata, content distribution networks architecture, user preferences, etc. All this information is analyzed in order to generate reports on service performance, access evolution and users’ preferences, and thus to improve the presentation of the services. The system has been used in real audio/video services since 2001 with satisfactory results.

MPEG-4流媒体系统中的视音频同步

流媒体系统中视音频同步是非常重要的。首先简要地介绍了MPEG 4的相关概念,分析了流媒体系统中影响视音频同步的因素,提出了一套相应的解决方案,最后,给出了一个基于MPEG 4的视频监控系统的实现。     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

无线Mesh网流媒体传输速率控制策略及模型*

基于无线Mesh网（WMN）的网络特征,提出适合WMN流媒体传输的速率控制策略,并给出其相应的模型描述。提出的策略和模型充分利用链路的多样性,降低由重传机制和节点冲突造成的流媒体传输时延增大并改善流媒体传输的性能,同时兼顾WMN接入有线网络的TCP友好性等特征。     

基于FFMPEG解码的音视频同步实现

为实现音视频同步播放,针对音视频数据同时被采集,但编码和存储独立的情况,提出了将音频播放时钟作为同步时钟,采用时间戳技术实现历史音视频同步播放.该方法使用FFMPEG对历史音视频文件分别进行解码,将解码后计算得到的音频播放时钟作为同步时钟,控制视频播放速度同步到音频播放时钟上,保证了音视频数据流畅播放,同步无滞后,无延迟.通过实验设计,验证了提出的基于音频播放时钟的时间戳同步方法是有效的.     

命名数据网络中的视频传输研究综述

互联网目前已经发展为一个由实时视频和视频点播等内容分发服务主导的网络传统IP网络对于视频分发类任务的支持存在组播的部署复杂且开销大,不能有效利用多路径获取内容、对移动性的支持差和难以同时满足可靠性以及低延时需求等问题命名数据网络(named data networking,NDN)作为新型的下一代互联网体系结构,支持网...     

基于MFCC和短时能量混合的异常声音识别算法

针对现行异常声音识别算法复杂度高和特征识别率低的问题,将梅尔频率倒谱系数（MFCC）与短时能量混合特征应用到异常声音识别系统中。该混合特征使得高斯混合模型（GMM）分类器可获得比使用MFCC特征及其差分MFCC更好的分类性能。给出了系统实现的具体步骤,并通过仿真实验证明了该算法的有效性,分类器的平均识别率可达到90%以上,并且计算复杂度小。     

Time-, wavelength-, and code-domain optical reflection monitoring for next-generation access-metro networks

Distributed optical reflectors are proposed to implement essential fault management operations such as fault detection, localization, and notification, in next generation all-optical access-metro networks in the optical layer. Fixed time-slots, wavelengths, or optical codes are assigned to selected key network locations along the path of a monitoring signal where corresponding mirrors are placed. The reflection received at a transmitting node enables online all-optical monitoring of the selected locations. After detailing the network architecture, we explore fundamental system design issues such as real-time fault localization algorithms, fault notification delay upperbounds, and delay line calculation algorithms for synchronous operation. Our simulation results showcase standard and long-reach passive optical networks, and demonstrate that our algorithms achieve fault notification at light speed using off-the-shelf components.     

Object extraction scheme and protocol for energy efficient image communication over wireless sensor networks

To date, wireless sensor networks lack the most powerful human sense – vision. This is largely due to two main problems: (1) available wireless sensor nodes lack the processing capability and energy resource required to efficiently process and communicate large volume of image data and (2) the available protocols do not provide the queue control and error detection capabilities required to reduce packet error rate and retransmissions to a level suitable for wireless sensor networks. This paper presents an innovative architecture for object extraction and a robust application-layer protocol for energy efficient image communication over wireless sensor networks. The protocol incorporates packet queue control mechanism with built-in CRC to reduce packet error rate and thereby increase data throughput. Unlike other image transmission protocols, the proposed protocol offers flexibility to adjust the image packet size based on link conditions. The proposed processing architecture achieves high speed object extraction with minimum hardware requirement and low power consumption. The system was successfully designed and implemented on FPGA. Experimental results obtained from a network of sensor nodes utilizing the proposed architecture and the application-layer protocol reveal that this novel approach is suitable for effectively communicating multimedia data over wireless sensor networks.     

P2P流媒体服务方案及其关键技术研究

根据近年来P2P流媒体服务方案的研究成果,归纳并总结P2P流媒体的关键技术,包括覆盖网络构建技术、数据调度技术以及各种优化技术。由此可知,P2P流媒体是一种性价比高、扩展性好的流媒体服务体系,并指出P2P流媒体可进一步的研究方向为P2P流媒体的应用与测量、理论分析与建模、体系架构以及与视频编码、移动网络技术的结合等。     

Lossless data transmission for Internet of things application over wireless multimedia sensor networks using enhanced and optimal path scheduling approach to maximizing the quality of service

The congestion of packet forwarding between a source and destination is challenging on downlink transmission in the entire file (ex. Audio and Video). Whenever file is been uploaded to the server, a user requests for file where server transmits it without knowledge of user's bandwidth, which is a major, cause of packet loss or time duration in the receiver end. To accumulate the better solution, Enhanced and Optimal Path Scheduling Approach (EOPSA) designs to find optimal path scheduling for multimedia data transmission in multimedia sensor network over cloud server using IoT devices. EOPSA studied the multisource video-on-demand streaming in multimedia sensor networks. The method introduced a heuristic distributed protocol to find optimal route for multimedia data transmissions. Efficient way to identify the bandwidth before the transmission ensures link establishment between sender and receiver. Here, the capture of bandwidth helps to check user's system capability to forward requested media data. Based on experiment evaluation, EOPSA improves 0.20 packet delivery ratio, 130 throughput, 0.20 second average delay and 14 communication overhead for 15, 25, 50, 75, and 100 nodes compared than conventional methods.