基于Matlab的虚拟信号发生器设计

Matlab是一个数据分析和处理功能十分强大的工程实用软件。为输出提供了十分方便的函数和命令并且Matlab简单直观的图形用户接口易于掌握和使用。本文介绍了一种使用GUI工具箱。用Matlab实现应报信号发生器的设计。文章设计的信号发生器可以加深对信号处理概念的理解,具有一定的借鉴应用价值。     

Matlab在基音周期检测教学中的应用研究

基音周期是《语音信号处理》中的重点和难点,基音周期作为语音信号的重要参数,在语音编码、语音合成和语音识别等方面,有着非常重要的作用。本文采用Matlab对基因周期进行仿真分析,从仿真图上查看语音信号的基音周期,让学生更形象的理解语音信号的基音周期。     

基于Matweb的信号与系统自主学习系统的实现

采用Matlab作为开发工具,结合Web技术及其相应功能,设计并实现了信号与系统自主学习系统。该系统涵盖了信号与系统课程教学中的所有重难点,包括信号的发生、信号分析、系统分析、滤波器设计等。系统具有良好的可视化交互界面,用户只需通过浏览器来输入数据,递交给服务器,调用Matlab进行计算,返回给用户结果。该系统的实现,在远程教学仿真和虚拟实验领域都有广阔的应用前景。     

基于麦克风阵列的语音增强系统设计

针对目前在近场语音环境中语音增强系统噪声干扰大、难以清晰还原信号等局限性,设计并实现一种基于麦克风阵列的新型语音增强系统。系统硬件设计新颖,采用低功耗设计,最大支持12路语音信号同时采集,同时对噪声的处理采用自适应差分脉冲编码调制和卡尔曼滤波算法相结合分开处理的新型方式。通过Matlab仿真和封闭近场环境下的实际测试,结果表明该系统性能稳定可靠,可以提高语音清晰度,实现短距离的多路语音增强。     

基于Gnuradio与Hackrf的无线通信收发系统实现

随着无线通信技术的高速发展,需要对大量的信号及数据进行高效、快速、可操作性更高并且相对低成本的处理,这便对无线通信系统提出了更高的要求。Gnuradio是一种免费的开源软件无线电开发平台工具,在其图形化操作界面中集成了丰富的信号源与信号处理模块,通过功能强大的信号处理模块可以在PC端对无线通信中发送与接收端的信号进行编解码、调制与解调,时域和频域同步等处理。以Hackrf作为硬件平台,通过搭建GRC流图在射频段与PC端对信号及数据进行处理并发送与接收,实现了基于软件无线电平台的无线通信收发系统。     

基于Matlab COM组件的信号处理平台设计

设计了一种基于Matlab组件对象模型(component object model,COM)的信号处理平台.阐述Visual Basic和Matlab的接口方法,提出系统的体系结构,详细讨论COM组件在信号处理中应用.该系统在Visual Basic环境中引用COM组件,COM组件对信号进行分析处理后将结果返回给Visual Basic显示.系统充分利用Matlab丰富的函数库,能够脱离Matlab环境,信号处理效率高.     

某型雷达信号处理故障诊断系统设计与应用

为快速、准确定位某型雷达信号处理系统故障部位,需要设计一套故障诊断系统;在深入研究信号处理系统工作原理的基础上,通过分类总结信号处理系统故障类型,依托信号处理系统硬件开发平台,采用精确单模式串匹配算法和VC++可视化编程环境,设计开发了信号处理故障诊断系统;在后续试验和设备日常维护中,该诊断系统定位数十次信号处理系统故障;结果表明,该诊断系统具有故障定位快速准确等优点,能够实现信号处理系统故障的快速维修。     

脉冲多普勒雷达信号处理仿真系统研究

为了提高雷达信号处理系统性能,研究信号处理系统模型算法,构建了一个雷达信号处理系统仿真模型.在对雷达信号处理模型算法详细分析的基础上,利用Matlab工具实现对该算法模型的仿真验证.通过对仿真结果的分析,可以看出该算法能够准确提取目标的距离、速度信息,对杂波和噪声的抑制能力较强,验证了信号处理仿真系统模型的可行性和算法的可靠性.     

谱减法语音信号增强在Labview环境下的实现

谱减法是语音信号处理中一种非常重要的方法，使用该方法的目的是降低语音信号中噪声所占比例，是语音识别中一个非常重要的环节。Matlab 软件是语音信号处理的常用软件，Labview 软件是一款功能强大的数据采集处理软件。本文介绍了如何将Labview和Matlab结合起来，完成谱减法语音信号的采集及语音信号的增强处理在Labview环境下的实现。     

语音信号时频特征显示系统的设计和仿真

语音信号处理算法众多,但用于语音处理算法验证和开发的可视化研究平台极少。基于MATLAB GUI技术,完成语音信号典型时频特征参数提取和显示系统仿真平台的设计。可实现多种格式音频文件的载入和播放、波形和频谱显示、以及线性预测倒谱系数和美尔倒谱系数的计算、存储和显示等功能。系统界面友好、操作方便,可实现参数的交互输入并控制显示结果。仿真结果验证了相关时频特征参数提取算法的正确性,提高了对算法或数据处理效果的直观认识。     

一种新的基于多小波的图像去噪算法

图像去噪是最基本的图像处理技术，小波分析是目前国际上最新的时间．频率分析工具，是信号去噪的强有力处理工具。通过小波变换可以将交织在一起的混合信号分解成不同频率的块信号。多小波所拥有的对称性、正交性、有限支撑等重要特性弥补了单小波的不足。本文提出了一种新的图像去噪方法，该方法以多小波变换为基础。实验证明，具有很好的去噪效果。     

A new method of speech transmission over space time block coded co-operative MIMO–OFDM networks using time and space diversity

Reliable and good quality of service for speech transmission over wireless network has been a major challenge for the communication engineers and researchers. In this paper a new technique of speech compression and transmission using different Daubechies wavelets in a space time block coded co-corporative MIMO–OFDM networks using time and space diversity has been proposed. The main focus has been laid on design and development of wavelet based compression of multimedia signals for cooperative MIMO–OFDM system. We tried to find out various major issues regarding the wavelet compression of a speech signal. These issues include choice of a wavelet, decomposition level and thresholding criteria suitable for speech compression and transmission in co-operative MIMO–OFDM systems. A wavelet based speech compression technique using hard and soft thresholding algorithm has been proposed. The work shows that wavelet compression with QPSK modulation is a promising compression technique in a cooperative MIMO–OFDM system which makes use of the elegant theory of wavelets. The performance has been evaluated using mean square error, peak signal to noise ratio, compression ratio, bit error rate, and retained signal energy. It has been found that the transmitted speech signal is retrieved well under noisy conditions at higher order Daubechies wavelets. From the results it is clear that proposed technique aims at a radio access technology that can provide service performance comparable to that of current fixed Line accesses. To evaluate the performance of the proposed method, various performance parameters have been compared with previously implemented techniques and it has been found that the proposed work shows better performance as compared to the already existing techniques.     

A PC-based digital speech spectrograph

The author describes how a TMS32010-based Tl-speech PC board from Texas Instruments was combined with an off-the-shelf, PC graphics board to produce a real-time speech spectrograph system. The Realtime Spectral Lab (RSL) is a relatively inexpensive, PC-based, commercial-quality system. It produces high-quality spectrograms of up to 2 s of speech within 5 s of speech input. Also discussed are the algorithms chosen and the software structures evolved to implement the system. The resulting instrument is flexible. Dual spectrograms can show either simultaneous wideband and narrowband analyses of the same utterance, or equal frequency-resolution versions with differing time resolution. In addition, users can compare a target spectrogram to changing, real-time input. They can also mark a wideband spectrogram with cursors and expand the selected time segment into another, fine time-resolution spectrogram. The discussion is a case study of nontrivial DSP (digital signal processing) systems design and implementation by hardware selection and DSP programming alone     

专用语音信箱数据采集与处理

介绍的专用语音信箱由２ 条中继线、７ 台分机、１ 个语音处理单元以及呼叫处理程序和语音信箱管理程序组成。内外线电话能够互相呼叫或拨号访问信箱,并拥有多种程控业务新功能。语音信箱具有查询、留言和播放公众信息等功能,在被叫忙音或无应答时自动进入信箱。呼叫处理程序使用状态迁移法解决呼叫信号采集、处理的多重性问题;使用时间调度技术解决多用户的实时处理问题;使用ＶｉｓｕａｌＢａｓｉｃ的ＭＳＣＯＭＭ 控件实现了串行口交互通信;将多媒体ＭＩＣ控件用于语音信息的记录、储存和重放过程。该语音信箱有明显的实用价值     

盲人用计算机软件系统中的语音和自然语言处理技术

本文介绍了智能技术与系统国家重点实验室开发的“北极光”盲人用计算机软件系统中涉及的语音和语言处理技术。该系统能够获取和分析需要反馈的屏幕信息,通过语音合成平台将其内容朗读出来,对用户进行语音提示;与汉语自动分词、语言模型等自然语言处理技术的结合,使系统能够进行汉字和盲文的转换,反馈信息可以通过盲文点显器输出,使用户能够摸读盲文点字来获取所需要的信息,用户也可以采用盲文输入法进行输入,输入结果可转换为汉字文本形式。     

工业以太网确定性调度的设计与实现

工业控制网络对网络的实时性要求很高,不仅要求传输速度快,数据传输还要具有确定性。以太网的通信存在不确定性,不能满足实时性要求,成为以太网应用于工业控制领域的主要障碍。本设计实现的实时以太网就是网络中每个节点的通信被严格地限定在规定时间内,确保在同一网段内同时只有一台在发送数据。     

Speech encryption using hybrid-hyper chaotic system and binary masking technique

In this paper, a novel speech encryption algorithm based on hybrid-hyper chaotic system has presented. Instead of using normal chaotic system a hybrid-hyper chaotic system has used for improving the security level of speech communication models. Hyper-chaotic system is highly complex and dynamic system than normal chaotic system where it has more than one positive Lyapunov exponents. Hybrid chaotic system has designed by a disturbed discrete system by another one discrete system. In this algorithm, the input speech signal has compressed by Discrete Cosine Transform (DCT) to reduce the residual intelligibility. The compressed speech signal has permuted by hybrid chaotic system, which has designed using Zaslavsky and Zigzag maps. For substitution process, a reference speech signal has generated by Hidden Markov Model (HMM) speech synthesizer and permuted by using hyper-chaotic system. Masking of encryption signal has done by a masking sequence, which has obtained from the hyper-chaotic system. Our proposed work provides high security for the audio and speech signal over an insecure public network than other traditional speech encryption algorithms based on normal chaotic systems. The betterment of proposed algorithm is proven using the following metrics: key space analysis, key sensitivity analysis, information entropy measure, correlation coefficient analysis, Signal to Noise Ratio (SNR) analysis, subjective evaluation of speech quality, Perceptual Evaluation of Speech Quality (PESQ) analysis, NSCR (Number of Samples Changing Rate) and UACI (Unified Averaged Changed Intensity) analysis have carried out from cryptographic point of view and presented in this paper. The results proof that the proposed speech encryption algorithm ensures appreciable security system with robust encryption and decryption quality.

     

语音信号的MATLAB分析与处理

MATLAB是十分强大的用于数据分析和处理的工程实用软件,利用其来进行语音信号的分析、处理和可视化十分便捷。文中介绍了在MATLAB环境中如何驱动声卡采集语音信号和语音信号采集后的文档处理方法,并介绍了FFT 频谱分析原理及其显示、MATLAB中相关函数的功能、滤波器的设计和使用。在此基础上,对实际采集的一段含噪声语音信号进行了相关分析处理,结果表明利用MATLAB处理语音信号十分简单、方便且易于实现。     

基于多传感器的组合导航接口子系统

针对多传感器组合导航系统对数据通讯的同步性和实时性的要求，以单片机和DSP双CPU导航计算机构成的嵌入式导航系统为对象，开展了相应的接口子系统设计和实现研究。完成了组合导航计算机的接口电路，并以此为基础，设计了整个导航接口子系统的时序控制流程，以实现多传感器数据采集的同步性和实时性。同时给出了数据采集周期和导航解算周期的时序控制，以及双CPU之间的实时数据通讯的实现流程，最后通过实物联调验证了该方案设计的可行性。