数字语音信号的非均匀采样的实现

该文主要叙述数字语音信号的基于FFT的非均匀采样的技术实现问题,涉及WAV文件格式、分段、对其进行非均匀降速重采样频率的选择原则及生成和语音信号重构,并对实际的WAV格式的数字语音文件用VC++6.0编写的程序实现了基于FFT的重采样;分析方法、实现程序不仅对WAV格式的数字语音文件有效,而且也适用于其它格式的数字语音文件和非语音信号的非均匀采样的实现。     

高速非均匀采样信号的重构方法

并行交替采样中通道间的时基偏差易导致采样非均匀,且现有的信号重构方法运算量较大,容易使采集性能迅速下降.通过分析信号的频谱特征,提出了一种实时性较高的双通道并行采样重构算法;并采用FFT算法及反傅里叶变换对通道时基误差进行了实时校正.实验结果验证了该算法的可行性.     

语音PCM编码通信系统的SIMULINK仿真

脉冲编码调制(PCM)是一种将模拟语音信号变换成数字信号的编码方式。利用Simulink可视的动态的环境,文中提出从整个通信系统的角度上给出真实语音PCM编码的通信过程。系统在发送端系统对WAV文件进行采样,模拟真实语音,然后利用A律压缩、量化及编码,形成PCM编码,加上信道编码和交织后最后送入基带数字调制系统。接收端通过数字解调、信道解码、PCM解码最后恢复语音信号。仿真系统得出不同信噪比下的系统误码率和还原语音的MOS主观测试。     

基于非均匀采样的频谱分析和弱信号检测

由于采样设备和被采样信号的限制,完全的均匀采样很难实现,实际中很多情况也需要用到非均匀采样,而当采样信号是非均匀时采样定理不再适用。使用基于最小二乘法的非均匀频谱分析法,详细讨论了频谱噪声的产生原因和影响因素。由于频谱噪声会淹没一些弱信号的频率成分,使用基于最小二乘法的幅值衰减法,衰减强信号频率成分和频谱噪声,通过仿真实验成功检测出幅值为强信号幅值一亿分之一的弱信号,并对非均匀采样信号的频率成分进行高精度的参数估计。     

基于FPGA硬件实现的谐波检测方法

针对现有系统对谐波检测实时性差和精度低的问题,介绍一种基于傅立叶变换和FPGA硬件实现的谐波检测方法.分析了谐波检测中影响测量精度的关键因素,采用数字锁相环来同步被测信号,以减小由非同步采样所产生的误差.基-4FFT 处理器的硬件设计采用全并行的乘法运算单元结构和并行的存储分配方法,最大限度地提高谐波检测的速度.数字锁相环和基-4 FFT 算法用VHDL语言设计实现,并用MAX plus Ⅱ软件进行仿真,仿真结果表明,所设计的数字锁相环可以很好地跟踪被测信号,在180ms时,误差仅为0.01Hz,很好地消除了非同步采样所引起的测量误差;采用所设计的基-4FFT运算器对给定的谐波数据进行运算,得到的谐波幅值和相位误差小于0.05%,运算时间仅为8μs.     

VOC语音文件格式分析及应用

随着多媒体技术的发展,语音文件的应用越来越广泛,而语音文件的存储格式又是多种多样的,为了充分利用各种语音工具的特点,以及在开发软件中使用语音技术,就需要了解它们的格式,从而进行语音格式的相互转换,对语音数据进行处理和播放。语音文件常见的格式有VOC、WAV和AU等,而VOC语音文件是现在声霸卡和语音采集卡的主要存储格式,本文分析了VOC语音文件的特点和压解方法,并用Turbo C编制了在没有声霸卡支持下,用一般PC机的扬声器来实现VOC语音文件的播放方法。     

压缩感知理论在语音通信系统中的应用研究

本文介绍了压缩感知理论,并把这一理论应用在语音通信系统中。根据语音信号的短时平稳性、频域的可稀疏性,采用结合DCT和FFT两种不同稀疏基的OMP重构算法对语音信号进行重构,并对重构结果进行主客观评价。实验结果表明:固定帧长和压缩感知采样率M/N对语音信号重构性能有影响;以及当M/N>0.4时,DCT稀疏基表现出更好的重构性能;当M/N<0.4时,FFT稀疏基表现出更好的重构性能。     

基于自适应陷波的非均匀采样信号处理系统

非均匀采样可以突破奈奎斯特采样定理的限制,检测出超过采样频率的信号频率,但非均匀采样引起信号的频谱噪声,使得非均匀采样的小信号检测难于实现。研制了一种实时非均匀采样信号处理系统,采用自适应陷波方法计算非均匀采样信号的频率,逐步滤除幅度较大的信号,从而检测出小幅度信号。详细说明了自适应陷波方法的原理和实现方法,并介绍了基于数字信号处理器(DSP)的非均匀采样信号处理系统。     

基于Matlab的音频文件数字滤波处理

声音信号是一维连续信号,而计算机只能处理离散信号.为了从离散信号还原连续信号,根据采样定理,可以确定采样频率的最小值.wav文件是一种数字声音文件格式,本文基于Matlab分析了wav声音文件频谱与声音的关系.通过一个数字滤波实例,处理一个带有盲音的wav文件,去除其中的噪声,并给出了不同采样频率对输出声音信号的影响.     

基于自适应重采样的雷达信号压缩算法

针对实际应用中雷达采集的信号数据量较大、难以实时传输和存储的问题,提出一种基于自适应重采样的雷达信号压缩算法。对雷达信号进行分段处理,根据每段信号频谱的特点自适应地确定一个重采样率,通过对信号的重采样实现对雷达信号的压缩,利用零插值和低通滤波实现对信号的重构。实验结果证明了该算法的有效性。     

Tamper recovery algorithm for digital speech signal based on DWT and DCT

It is a challenging work to design tamper recovery schemes for digital speech signal. Briefly, there are two problems need to be solved. One is that the signals used to tamper recovery are difficult to generate and embed, and the second is that it’s hard to tamper location precisely for attacked speech signal. In this paper, compression and reconstruction method based on discrete wavelet transform (DWT) and discrete cosine transform (DCT) is given, to obtain the compressed signals used to tamper recovery. And then frame number and compressed signals are embedded based on block-based method. Attacked signal can be located by frame number, and compressed signals are extracted and used to reconstruct the attacked signal. Theory analysis and experimental results indicate that the scheme proposed not only improves the accuracy of tamper localization, but also can reconstruct the attacked signals.     

A study on multimedia file carving method

File carving is a method that recovers files at unallocated space without any file information and used to recover data and execute a digital forensic investigation. In general, the file carving recovers files using the inherent header and footer in files or the entire file size determined in the file header. The largely used multimedia files, such as AVI, WAV, and MP3, can be exactly recovered using an internal format in files as they are continuously allocated. In the case of the NTFS, which is one of the most widely used file system, it supports an internal data compression function itself, but the NTFS compression function has not been considered in file carving. Thus, a large part of file carving tools cannot recover NTFS compressed files. Also, for carving the multimedia files compressed by the NTFS, a recovery method for such NTFS compressed files is required. In this study, we propose a carving method for multimedia files and represent a recovery plan for deleted NTFS compressed files. In addition, we propose a way to apply such a recovery method to the carving of multimedia files.     

基于小波能双重自恢复的音频数字水印算法

本文设计了一种在音频信息中加入双重图像作为数字水印,这些水印经过多种攻击能相互自恢复的算法。此算法实现了将双重图像作为数字水印嵌入到WAV音频载体中的过程,在实现过程中通过对载体进行3层小波分解,将数字水印的信息嵌入到第三层小波解的低频区,即CA3中,并对双重数字水印分别设置密钥,建立起两者间稳定的映射关系,能较好地以完成自恢复的功能。     

Speech authentication system using digital watermarking and pattern recovery

The objective of this paper is to detect speech forgery using digital audio watermarking and pattern recovery techniques. A digital watermark pattern has been attached with the speech signal to detect three kinds of alterations or forgeries such as substitution, insertion, and removal. The watermark pattern will be modified if some changes have been made to the speech contents. Modification and forgery can be measured and detected by pattern recovery. The proposed method uses the cyclic pattern embedding to overcome synchronizing problems of previous detection techniques. In addition, pattern recovery enhances the robustness to compression. This method has been tested and verified using six recording devices, which was used for collecting verbal data. The speech signals were sampled at the rate of 8 kHz and digitized at 16 bits resolution. Randomly chosen regions were substituted, removed, and compressed in MP3 at the rate of 16 kbps as well as in CELP at the rate of 11.5 kbps. The experiment shows the perfect detection for three kinds of forgeries and it proved the validity of the proposed method.     

基于WAV文件的独奏乐曲信号中的乐谱识别

大多数乐谱识别工作都是在MIDI格式文件下进行的,而音乐数据大多采用音频格式存储,为此讨论一种从WAV音频文件的独奏乐曲信号中识别出乐谱的方法。采用离散余弦变换（DCT）结合能量比对的手段,并提出了一种变化音最少法则：以十六分音符对应的时值作为步长,将乐曲信号分为若干个子段;对每个子段进行一维DCT操作,提取基音频率;对相邻子段进行能量比对,查找音符的端点;利用变化音最少法则,生成乐谱。实验表明此方法具有良好的识别效果。     

远程监听监管“96121”电话的实现方法

运用Delphi语言编程,把＂96121＂气象声讯自动答询系统中每个信箱所对应的PCM格式文件转换成WAV格式文件,实现在局域网内的所有计算机用户都能监听＂96121＂信箱内容,及时更新信箱内容,提高了气象业务服务的效率和效益。     

An overview of digital speech watermarking

Digital speech watermarking is a robust way to hide and thus secure data like audio and video from any intentional or unintentional manipulation through transmission. In terms of some signal characteristics including bandwidth, voice/non-voice and production model, digital speech signal is different from audio, music and other signals. Although, various review articles on image, audio and video watermarking are available, there are still few review papers on digital speech watermarking. Therefore this article presents an overview of digital speech watermarking including issues of robustness, capacity and imperceptibility. Other issues discussed are types of digital speech watermarking, application, models and masking methods. This article further highlights the related challenges in the real world, research opportunities and future works in this area, yet to be explored fully.     

基于短时傅里叶变换的敏感语音提取

在纯净语音条件下,提出了一种基于短时傅里叶变换的敏感语音提取,该提取利用3种FIR数字滤波器对纯净语音在不同频率范围的特征语音进行选取,从而产生不同的音效.待处理的声音文件可以是自己录制的一段语音,也可以是任意格式的音频文件,通过DSP和编译软件实现敏感语音提取,并在数字处理芯片TMS320C5402上实现.     

基于MCS-51的JJY低频电波授时接收机的设计

本文针对中国制式（BPC）低频电波授时技术保密,难以深入研究的问题,提出基于51单片机的日本制式（JJY）低频电波授时接收机,给出该系统的原理、设计、实现和验证过程。该系统能以低廉的成本获取高精度的标准授时信号,具有实现简单,技术公开的优点,对提高中国的低频授时技术有一定的参考意义。本文提出的系统能够实现＂秒＂的严格同步。