哮喘,即支气管哮喘,是一种异质性疾病,常以慢性气道炎症为特征,表现为随时间变化和加剧的呼吸系统相关症状,如喘息、胸闷、气短和咳嗽,同时有可变的呼气气流受限.气流受限可能之后变为持续存在.

Aimed at a certain type of a single stage solid sounding rocket,traj ectory,power and aerodynam-ic models are established in this paper.Taking the maximum height of rocket theory traj ectory peak as op-timization goal,the aerodynamic/power integrated optimization are realized by using the method of se-quential approximate optimization to obtain a set of better parameters in conception design.In the whole process of optimization,the number of times for shifting model is less and the convergence is quick in ve-locity.

MP3已经成为数字音频的主流压缩编码.该文从实验出发探讨了MP3编解码对时域和频域两方面影响,基于此提出了一种MP3编解码的水印算法:针对时域变化采用载体信号和重采样划分来实现嵌入与提取同步,通过量化相邻子帧低频能量比来嵌入信息以消除频域失真.并基于嵌入帧平均信噪比和提取准确率估计给出了该算法的参数参考值.实验证明该算法不仅能够用于抗低码率低采样高压缩比的MP3编解码,且具有良好的不可感知性和嵌入容量,同时还进行了AAC编解码测试.

A NLMB-DM (Nonlinear Multi-bit Dither Modulation) method which can embed information into BMP color images with high capacity is proposed in this paper. On the basis of NLMB-DM method, two adaptive methods which separately use the idea of RDM (Rational Dither Modulation) method and Watson visual model are brought forward. The results show that these 3 methods both have the property of high capacity, good imperceptibility and low bit error rate.

数字音频编解码系统采用MDCT/IMDCT实现时间-频率域的互换,以消除音频分帧引起的时间域混叠效应.本文提出一种新的MDCT/IMDCT快速实现方案,基于N/8点FFT变换核,采用奇偶双路并行和蝶型单元技术,与现有快速算法相比,运算速度和吞吐能力均提高一倍,并且该方案既可以实现MDCT正变换,也可以实现相应的反变换.为了验证方案的正确性,在Altera FPGA开发板上完成了N=256点MDCT的实验.结果表明,该实现方案在运算速度和数据吞吐率等方面取得很大的改进.

In low rate speech processing,conventional synchronous overlap and add(SOLA) method for time-scale modification encounters the problem when the modification rate is higher,the modified speech signal is less intelligible because the neglect of perceptually sensitive portions damages speech articulation.This paper proposes an improved time-scale modification method based on the knowledge.Both transient portions and signal energy play a critical role in speech perception.After identifying transient portions,steady portions and quiet portions by Mel-frequency cepstral coefficient(MFCC) and energy,the proposed method uses time-scale modification to separate portions with different modification rates.Experimental results indicate that the approach is perceptually superior to the conventional SOLA method,thus improving the efficiency in low rate speech coding.

An efficient audio encoder for MPEG AAC is exhibited in this paper,by carrying out the algorithm level optimizations of three critical modules.In psychoacoustic model,a block switching method in time domain is presented instead of PE-based algorithm,in order to resolve the quality degrade caused by erroneous decision of block type.In addition,an analysis filter calculating the N-point MDCT transform employs a parallel scheme basing on N/8-point FFT,improving the computation rate and the data throughput.Finally,to meet the bit rate/distortion critical,a novel technique through pre-processing and adjustment of scale factor in quantization block is used.According to the objective evaluation results,the encoder proposed in this paper gets the performances of both low complexity and high audio quality.

Given the observation under repeated embedding,the audio distortion is the fiercest for the very first time and drops significantly subsequently,a steganalysis method of DSSS(Direct Sequence Spread Spectrum)was proposed,which is implemented by support vector machine(SVM)as a classifier,using various distortion measures sensitive to DSSS stego-method as features,avoiding estimation of the original signal.The experiment results verify that the proposed method can detect the DSSS embedding effectively and can also work with different embedding strength,and two methods of detection with different embedding capacities were suggested.

The conventional SOLA method of time-scale modification encounters the problem that the hi-gher the modification rate,the less intelligible the time-scale modified speech signal,because of the negl-ect of different contributions to articulation of different speech signal parts. This paper proposes a parti-tion time-scale modification method based on the knowledge that how fast spectrum changes and how much energy the signal contains,and both play a critical role in speech perception. After identifying por-tions with different spectrum and energy of a speech signal,the proposed method applies timescale modification to different portions with different modification rate. The result of subjective preference test indicates that the performance of the proposed method is superior to that of the conventional SOLA method.

Watermarking has a broad range of applications for multimedia copyright protection.This paper discusses the traditional approach of spread-spectrum audio watermarking and the correlation detection method.By exploring the deviation of mean and variance of original audio signals in practice from the correlation detection model,we present a novel mechanism with adaptive encoding intensity and using the psycho-acoustic model.The purpose is to improve detection convergence and robustness and the water-mark imperceptivity.The proposed mechanism is tested on a benchmark suite including several attacks.

提出了一种基于时钟相位抖动的混沌随机数产生器,并在XilinxFPGA开发板上实现并进行了测试。测试结果表明,最高输出速率达8Mbps,实现了设计要求。硬件结构简单,并且该RNG的输入源不受电路噪声源的影响,易于今后的IC实现。