一种小波域音频信息隐藏方法

提出了一种基于量化的小波域音频隐藏算法，将保密语音隐藏到载体音频中．为提高隐藏重和保密语音传输的安全性，对保密语音进行了小波域压缩编码和m序列的扩频调制，生成待隐藏的比特序列；通过量化方法，将编码和调制后的保密语音隐藏到载体音频的小波系数中；保密语音的恢复过程不需要使用原始音频、仿真结果表明，隐藏有保密语音的载体音频听觉质量没有明显下降，提取的保密语音感知质量较好；该算法对重量化、加噪、低通滤波等攻击均有良好的鲁棒性．     

Distance rendering and perception of nearby virtual sound sources with a near-field filter model

Headphone rendering of nearby virtual sound sources represents to date an open issue in 3-D audio, due to a number of technical challenges and temporal requirements involved in the measurement of individual Head-Related Transfer Functions (HRTFs). In order to tackle this problem, we propose a filter model of near-field effects based on the Distance Variation Function (Kan et al., 2009). Thanks to its simple structure and low order, the model can be applied to any far-field virtual auditory display to yield a realistic and computationally efficient near-field compensation of spectral and binaural effects. The model is subjectively evaluated in two psychophysical experiments where the relative distance of pairs of virtually rendered sound sources is judged. Results show that even though sound intensity overshadows subtler near-field effects when it is available as a cue for distance, the model is capable of offering relative distance information of near lateral virtual sources when intensity cues are removed. Furthermore, performances of the model in relative distance rendering are compared to those of alternative near-field rendering methods available in the literature.     

LINUX下一种改进的音频会议系统的实现方法

提出了一种计算缓存区大小的改进方案．根据RTCP包中反馈回的丢包率,动态地改变音频数据的发送策略,有区分地采取差错控制手段,提高了网络带宽的利用率．在此基础上,设计了一个以RTP协议为基础的音频会议系统的模型,并在Linux平台上用多线程的方法实现了各个模块．模拟实验表明,该系统能自适应地调整缓存区大小和数据包的发送策略,在实时性及带宽资源的利用上都达到了较理想的效果．     

一类单参数假设检验拒绝域的直观构造法

本文提出直观法的理论 ,举出用直观法构造单参数假设检验拒绝域的应用实例 ,并得出构造形如 ( * )式的检验拒绝域的一种方法     

关于超音速多普勒效应的思考

超音速多普勒效应由于其不便于测量,基本停留在纯理论分析阶段,导致人们产生了一些错误的观点.本文通过逻辑分析、计算机软件模拟等手段,对超音速多普勒效应的音频时序进行了分析研究,找出了错误观点产生的原因,作出了相应的解释,并对错误观点进行了修正,提供了解决方案.     

多路音频信号的数字化光纤传输系统

在信息传输的各个领域中，光纤作为传输信号媒介的应用得以迅速发展。光纤的音频多路数字化传输也是光纤领域目前开发的热点课题之一，它充分利用光纤本身所具有的各种特点，使音频信号的传输更有效、更可靠。本文利用串行信号的时分复用技术，完成了多路音频数字化信号的光纤传输。提出信号传输中,数据信号和控制信号的复用及解复用解决方法。     

Statistical and Visual Analysis of Audio,Text, and Image Features for Multi-Modal Music Genre Recognition

We present a multi-modal genre recognition framework that considers the modalities audio, text, and image by features extracted from audio signals, album cover images, and lyrics of music tracks. In contrast to pure learning of features by a neural network as done in the related work, handcrafted features designed for a respective modality are also integrated, allowing for higher interpretability of created models and further theoretical analysis of the impact of individual features on genre prediction. Genre recognition is performed by binary classification of a music track with respect to each genre based on combinations of elementary features. For feature combination a two-level technique is used, which combines aggregation into fixed-length feature vectors with confidence-based fusion of classification results. Extensive experiments have been conducted for three classifier models (Naïve Bayes, Support Vector Machine, and Random Forest) and numerous feature combinations. The results are presented visually, with data reduction for improved perceptibility achieved by multi-objective analysis and restriction to non-dominated data. Feature- and classifier-related hypotheses are formulated based on the data, and their statistical significance is formally analyzed. The statistical analysis shows that the combination of two modalities almost always leads to a significant increase of performance and the combination of three modalities in several cases.     

基于时间序列模型分析的ADHD儿童听觉持续性注意研究

摘要：通过建立时间序列模型,对ADHD儿童与正常儿童的听觉持续性注意水平进行拟合,并分析2组被试间的差异,动态地观测ADHD儿童的注意缺陷水平,并实现对ADHD儿童注意水平的预测.被试选取ADHD儿童30名,正常儿童10名,对被试进行听觉持续性操作测试,对每个序列的平均正确率进行时间序列模型分析,比较2组被试间的差异.根据数据的特征选用Holt模型对ADHD儿童的持续性注意水平进行建模及拟合,而正常儿童组则选用简单模型（Simple）.与正常儿童相比,ADHD儿童存在显著的注意缺陷特征,持续性注意水平随时间变化表现出明显的下降趋势.     

Speech-Music-Noise Discrimination in Sound Indexing of Multimedia Documents

Sound indexing and segmentation of digital documents especially in the internet and digital libraries are very useful to simplify and to accelerate the multimedia document retrieval. We can imagine that we can extract multimedia files not only by keywords but also by speech semantic contents. The main difficulty of this operation is the parameterization and modelling of the sound track and the discrimination of the speech, music and noise segments. In this paper, we will present a Speech/Music/Noise indexing interface designed for audio discrimination in multimedia documents. The program uses a statistical method based on ANN and HMM classifiers. After pre-emphasis and segmentation, the audio segments are analysed by the cepstral acoustic analysis method. The developed system was evaluated on a database constituted of music songs with Arabic speech segments under several noisy environments.