Intelligent video and audio applications for learning enhancement

The role of computers in school education is briefly discussed. Multimodal interfaces development history is shortly reviewed. Examples of applications of multimodal interfaces for learners with special educational needs are presented, including interactive electronic whiteboard based on video image analysis, application for controlling computers with facial expression and speech stretching audio interface representing audio modality. Intelligent and adaptive algorithms applications to the developed multimodal interfaces are discussed.     

An adaptive post-filtering method producing an artificial Lombard-like effect for intelligibility enhancement of narrowband telephone speech

Post-filtering can be used in mobile communications to improve the quality and intelligibility of speech. Energy reallocation with a high-pass type filter has been shown to work effectively in improving the intelligibility of speech in difficult noise conditions. This paper introduces a post-filtering algorithm that adapts to the background noise level as well as to the fundamental frequency of the speaker and models the spectral effects observed in natural Lombard speech. The introduced method and another post-filtering technique were compared to unprocessed telephone speech in subjective listening tests in terms of intelligibility and quality. The results indicate that the proposed method outperforms the reference method in difficult noise conditions.     

改善多媒体通信质量的覆盖层路由系统

针对近年来互联网上迅速兴起的多媒体通信应用,分析和指出了当前互联网网络层路由不适于传输多媒体数据的内在缺陷,为改善多媒体通信质量,设计并实现了一个名为CORS的覆盖层路由系统,通过构建和使用多条覆盖层路径来突破网络层单路径路由的局限性,并提供应用感知的传输层服务。利用在全球网络实验平台——PlanetLab上的真实实验验证了CORS的有效性。     

A mapping model of spectral tilt in normal-to-Lombard speech conversion for intelligibility enhancement

Environmental noise degrades the speech intelligibility when listening to the phone. Although the phone has a clean signal source, it is still difficult for the listener to get information. Intelligibility enhancement (IENH) is a type of perceptual enhancement technique for clean speech rendered in noisy environments. This study focuses on IENH by normal-to-Lombard speech conversion, which is inspired by Lombard reflex. In this conversion process, the key point is to map the spectral tilt from the normal speech (normal style) to the Lombard speech (Lombard style). For mapping the spectral tilt, we propose a mapping model combining linear-prediction-based mapping networks and tilt modification. Compared with previous studies, we use deep neural networks (DNNs) instead of Gaussian-based models for higher dimensional mapping, and inventively add a tilt modification module to reduce the mapping errors of formant magnitudes further. In this paper, we use AVS-M codec and two datasets as the benchmark platform. The valuation shows that our method gets better results than reference methods in both objective and subjective experiments.

     

An effectiveness study of a CAD system augmented by audio feedback

The question addressed is the effectiveness of man-computer communication in computer-aided design (CAD). Many physical systems that design engineers develop on interactive CAD systems, in reality, produce noises that would give information about the system's performance.

The results of a controlled man-computer study are presented here. The task for the subjects was to design the acoustical treatment for a noisy laboratory. A control group worked in a conventional manner with the CAD system. The study group received continuous audio feedback indicative of the degree of noise reduction of their design. The statistically significant results are presented.     

An adaptive website system to improve efficiency with web mining techniques

The paper proposes an adaptive web system—that is, a website that is capable of changing its original design to fit user requirements. For the purpose of improving shortcomings of the website, and also to make it much easier for users to access information, the system analyzes user browsing patterns from their access records. This paper concentrates on the operating-efficiency of a website—that is, the efficiency with which a group of users browse a website. By achieving high efficiency, users spend less operating cost to accomplish a desired user goal. Based on user access data, we analyze each user's operating activities as well as their browsing sequences. With this data, we can calculate a measure of the efficiency of the user's browsing sequences. The paper develops an algorithm to accurately calculate this efficiency and to suggest how to increase the efficiency of user operations. This can be achieved in two ways: (i) by adding a new link between two web pages, or (ii) by suggesting to designers to reconsider existing inefficient links so as to allow users to arrive at their target pages more quickly. Using this algorithm, we develop a prototype to prove the concept of efficiency. The implementation is an adaptive website system to automatically change the website architecture according to user browsing activities and to improve website usability from the viewpoint of efficiency.     

A novel quantum steganography-Steganalysis system for audio signals

Multimedia Tools and Applications - As a substitute for classical solutions, quantum information hiding techniques have become an essential issue in the field of quantum communications by utilizing...     

Tensor semantic model for an audio classification system

Audio resources are a very important part of multimedia information.The classification effect of audio is directly related to the service mode of personal resource management systems.At present,vector features have been widely used in audio classification systems.However,some semantic correlations among different audio information can not be completely expressed by simple vector representation.Tensors are multidimensional matrices,and their mathematical expansion and application can express multi-semantic information.The tensor uniform content locator(TUCL) is proposed as a means of expressing the semantic information of audio,and a three-order tensor semantic space is constructed according to the semantic tensor.Tensor semantic dispersion(TSD) can aggregate some audio resources with the same semantics and,at the same time,its automatic classification can be accomplished by calculating the TSD.In order to effectively utilize TSD classification information,a radial basis function tensor neural network(RBFTNN) is constructed and used to train an intelligent learning model.Experimental results show that the tensor model can significantly improve the classification precision under multi-semantic classification requests within an information resource management system.     

Extending the notion of a window system to audio

With audio's increasing importance in computer applications, users will soon need presentation, management and organizational capabilities similar to visual window systems to avoid a confusing cacophony of multiple audio sources sounding at once. The ways in which an audio window system could be used are described. These include multimedia documents, spatial data management systems, and teleconferencing. The signal processing methods used to create hierarchical and spatial distribution among nearly arbitrary (not pure sine wave) audio sources are discussed. A prototype system, combining hierarchical and spatial processing functions with a computer-controlled switch, software and human input devices, is presented. Two envisioned implementations, a terminal-based system and a network-based server, are described. Preliminary work suggests that an effective audio window system needs much less complexity and fewer levels of digital signal processing precision than the current prototype     

A flexible and scalable audio information retrieval system for mixed‐type audio signals

The content‐based classification and retrieval of real‐world audio clips is one of the challenging tasks in multimedia information retrieval. Although the problem has been well studied in the last two decades, most of the current retrieval systems cannot provide flexible querying of audio clips due to the mixed‐type form (e.g., speech over music and speech over environmental sound) of audio information in real world. We present here a complete, scalable, and extensible content‐based classification and retrieval system for mixed‐type audio clips. The system gives users an opportunity for flexible querying of audio data semantically by providing four alternative ways, namely, querying by mixed‐type audio classes, querying by domain‐based fuzzy classes, querying by temporal information and temporal relationships, and querying by example (QBE). In order to reduce the retrieval time, a hash‐based indexing technique is introduced. Two kinds of experiments were conducted on the audio tracks of the TRECVID news broadcasts to evaluate the performance of the proposed system. The results obtained from our experiments demonstrate that the Audio Spectrum Flatness feature in MPEG‐7 standard performs better in music audio samples compared to other kinds of audio samples and the system is robust under different conditions. © 2011 Wiley Periodicals, Inc.     

音响系统回波抵消研究

针对音响系统在室内产生的回波干扰,采用基于频域回波抵消算法,结合DSP芯片TMS320VC5416硬件的处理方法,设计了RS232串口通讯的人机界面控制程序,从而有效地消除了干扰噪音。该方法可用于多种音频设备、会议系统,以抵消回波产生的干扰。     

K类音频功放的防破音控制系统

设计实现了一种可用于K类音频功放的防破音控制系统,通过自动检测输出削波失真调整系统增益,确保K类音频功放在较大的输入动态范围内,输出音频信号光滑无失真。确保K类音频功放在整个工作电压内保持低谐波失真,且保持恒定输出功率。采用0.5μm CMOS工艺实现了集成这种防破音控制系统的2.0 W单声道K类音频功放。测试结果显示,在3.3 V~4.2 V电源电压范围、增益设置为24 d B、4Ω喇叭负载下,该K类音频功放能够在0~1.2 V_(rms)的动态输入范围内保持低谐波失真(THD+N)<0.5%,恒定无削波输出功率2.0 W。     

Embedded system for contrast enhancement in low-vision

This paper presents a real-time contrast enhancement system, implemented in FPGA and adapted to display the processed images on a Head Mounted Display (HMD). A novel visual processing scheme is proposed which combines a version of the algorithm known as Contrast Limited Adaptive Histogram Equalization (CLAHE) with a spatial filtering based on a bio-inspired retina model. The system is designed so that visually impaired people can improve their functionality in environments with non-uniform lighting or with abrupt changes in lighting conditions. The parallelism offered by FPGA devices allow to achieve real-time processing with VGA-resolution images, reaching up to 60 frames per second. This system, developed on a FPGA of reduced complexity, has been compared in performance with a parallel implementation on a portable platform based on GPU.     

An efficient system using implicit feedback and lifelong learning approach to improve recommendation

The Journal of Supercomputing - This paper presents&nbsp;a new technique for contextual item-to-item Collaborative Filtering-based Recommender System, an improved version popularised by...     

可用于电声测试系统的专业音频接口设计

针对电声测试系统对低成本、高性能音频接口的需求,设计了一种由下位机嵌入式系统和上位机计算机软件组成的电声测试系统专用音频接口。音频接口采用高速USB2.0接口传输音频数据,通过音频编解码器对音频信号进行D/A和A/D转换,实现播放和录音功能。经测试音频接口生成正弦信号的总谐波失真低于专业声卡;信号采集通道的幅频特性优于专业声卡;同步录音功能能够实现录音和播放完全同步。     

一种用于专业音频传输系统的精确时钟同步方法

在专业网络音频传输系统中,由于晶振制造工艺、环境温度等因素导致主从节点音频时钟产生差异,引起系统失真率串升。而系统中以太网按照音频采样时钟的节拍来传输音频数据包。为此提出一种在物理层和MAC层的MII接口处进行音频采样时钟恢复的方案,同时设计相应的时钟调整算法进行晶振频率补偿,以提高主从节点音频时钟的同步性。在Xilinx FPGA平台上进行实际测试验证,结果表明,传输系统的失真度(包含模数和数模转换产生的失真)小于0.005%,长期运行的结果也表明了系统时钟同步的稳定性。     

D类音频功放的可集成Pop-Click噪声抑制系统

设计实现了一种可集成于D类音频功放芯片内部的Pop-Click噪声抑制系统,通过全新的输出级软启动控制以及辅助反馈环路的采用,保证了软启动过程中的环路稳定性和输出驱动级的完整性,实现良好稳定的PopClick噪声抑制。采用0.5μm CMOS工艺实现了集成Pop-Click噪声抑制系统的2.0 W单声道D类音频功放。测试结果显示,在单位增益、8Ω喇叭负载下,该D类音频功放的Pop-Click噪声能被有效抑制至1.5 mV内。     

An information supply chain system view for managing rare infectious diseases: The need to improve timeliness

Timely reporting of rare infectious disease cases to the public health system, especially after identification at laboratories, is essential to initiate quick and effective public health response. To ensure that the public health reporting system is appropriately monitoring the rare infectious diseases under surveillance, it is recommended to have a regular assessment of timeliness, especially after the rare infectious case is confirmed. This study aimed to evaluate the timeliness of data reported to the Ohio Disease Reporting System (ODRS), a public health reporting system in Ohio, for managing rare infectious diseases. In a cross-sectional analysis of rare infectious disease reporting data in four local health jurisdictions (LHJs) in the state of Ohio, wide delays were found between various reporting steps, particularly when the laboratories were not using the electronic method of reporting, and the delay observed was mainly at the hospital level and at the LHJ level. This study highlights the supply chain nature of information transfer and calculates the delay at various interacting points of the information supply chain system. The results establish that a centralized approach with an electronic disease reporting system conveys information faster than traditional reporting channels (decentralized approach). Delays of the decentralized approach are isolated at various stakeholder levels and with respect to various types of rare infectious diseases for better understanding of the information supply chain system for managing rare infectious diseases.