算术编码在航空侦察图像压缩中的应用

航空侦察图像原始数据量非常大,在实时回传的过程中必须进行压缩。在对图像进行有损或无损压缩时,无损压缩算法是必要的。本文研究了算术编码在系统中的应用,提出一种混合进制的算术编码,提高编码效率的同时增强了系统的抗误码扩散能力。试验验结果表明该算法明显优于单一进制的编码算法。     

基于可逆提升小波变换和上下文预测的SAR图像无损压缩算法

 结合可逆提升小波变换和上下文预测技术 ,提出基于可逆提升小波变换和上下文预测的合成孔径雷达 (SAR)图像的无损压缩算法。分析了基于提升小波的无损压缩算法和基于上下文预测编码的无损压缩算法的优缺点 ,联合两种体制 ,提出先用提升小波变换去除整幅图像的冗余 ,再用边缘保留预测器和子预测器联合技术进一步消除子带内系数间的空间冗余性 ,最后采用自适应上下文建模技术对残差进行分类自适应熵编码。除具有常规小波变换编码的优异性能外 ,该算法的压缩率优于即将推出的国际标准JPEG2000。     

遥测系统中喷泉码的低延时改进算法

针对喷泉码应用于遥测系统时存在较大系统延时的问题,在对其延时特性进行分析的基础上,提出了一种改进的编译码算法.当编码器接收到部分数据符号时,按照预先制定的度选取策略和符号选取策略,选择当前编码符号的度和生成该编码符号的数据符号进行编码.该算法以减小编码延时为目标,通过分析编码符号发送速率的稳定性和可容忍的最长信号闪断时间可确定合理的编码延时.仿真得到了不同丢符号率下的编码延时、系统延时和所需的编码冗余,结果表明,与传统方案相比,改进方案的编码延时减小了一半以上.通过理论分析和仿真验证,得出了改进算法可明显减小基于喷泉码的遥测系统延时的结论.     

一种语音数据压缩算法及其实现

本文通过对语音数据的特点进行分析，提出了一种对语音数据进行无损压缩的算法；并将其压缩效率与其它一些常用压缩软件（如ＡＲＪ，ＬＨＡ等）的压缩效率进行了比较。     

干扰环境中的跳频技术

一、引言跳频就是传输信息的载波频率或它的频率组合的周期性变化。图1是一般跳频发射机和接收机的方框图。数据调制可以用多种形式,最初采用了二进制频移键控(F. S. K.)。当采用二进制F. S. K时,每一个编码比特(符号)用两个频率中的某一个来发射,其中一个频率代表逻辑“1”(mark),而另一个频率代表逻辑“0”(space),可能出现的频率对是伪随机码作周期变化的,每一变化形成一跳。     

HFU零件分类编码系统简介

零件分类编码系统是用数字、字母或符号对零件各有关特征进行描述和标识的一套特定的规则和依据。按照零件分类编码系统对零件进行编码,并在此基础上按相似性原则分组,是实施成组技术的关键和基础,也是实现CAD/CAM时用计算机处理零件特征信息的前提条件。目前,世界上的零件分类编码系统已有上百种。国外的系统多数是专利,有些是公开的。这些系统在适用范围上大都有一定的局限性,     

调频体制载波跟踪技术研究

编码和多符号检测技术在PCM/FM系统中的应用对载波精确跟踪提出了更高的要求。本文介绍了多符号检测原理,针对以往载波跟踪的不足,提出了一种把译码和跟踪结合起来进行载波跟踪的方法,给出了其跟踪原理框图,对其进行了理论推导,并对把这种方法应用于PCM/FM体制测速功能进行了设想。     

功率约束下低相关旁瓣的通导一体化波形研究

通信导航一体化是未来无线系统的发展趋势,可以更加充分地利用频谱资源。设计了一种新的一体化波形,在实现通信符号传递的同时,利用其进行时延测距进而实现相应的定位导航功能。一种联合的优化目标函数被提出,其中以星座图上的范数误差衡量通信性能,以自相关函数的加权积分旁瓣电平衡量波形的定时性能。结合一体化系统的实际,在问题求解时同时考虑了波形总功率约束和波形恒模约束,并采用变量交替迭代优化的算法进行迭代求解。最后利用数值仿真验证了算法的有效性。     

变换域通信系统及其关键技术

变换域通信系统通过信道估计在变换域设计信号波形避开干扰,它没有采用载波调制,而是用一个类似噪声的基函数进行信息调制。论述了变换域通信系统的基本概念,对变换域通信系统的关键技术——环境干扰估计与检测、基函数序列与波形设计、调制解调与高速率数据传输进行了分析,最后指出了其应用前景。     

功率约束下低相关旁瓣的通导一体化波形研究

通信导航一体化是未来无线系统的发展趋势，可以更加充分地利用频谱资源。设计了一种新的一体化波形，在实现通信符号传递的同时，利用其进行时延测距进而实现相应的定位导航功能。一种联合的优化目标函数被提出，其中以星座图上的范数误差衡量通信性能，以自相关函数的加权积分旁瓣电平衡量波形的定时性能。结合一体化系统的实际，在问题求解时同时考虑了波形总功率约束和波形恒模约束，并采用变量交替迭代优化的算法进行迭代求解。最后利用数值仿真验证了算法的有效性。     

Speech Data Rate Reduction Part I: Applicability of Modern Estimation Theory

Efficient coding of continuous speech signals for digital representation has attracted much interest in recent years. The underlying aim of efficient coding methods is to reduce the channel capacity required to represent a signal to meet a specific reconstruction fidelity criterion. To achieve this objective, modern speech data compression techniques rely on two very similar procedures. One procedure uses predictive deconvolution which subtracts from the current signal value that portion which can be predicted from its past and thus removes redundancy in the speech by removing sequential correlation. The signal thus requires fewer bits for equivalent quantization error. The second procedure involves identification of a complete mathematical model of the speech producing mechanism. This involves determination of the characteristics of the source that drives this transfer function. Data reduction is again achieved since the rate of change of the parameters of the speech model is much smaller than the rate of change of the speech waveform. This paper develops these data reduction procedures in terms of modern estimation theory, specifically a Kalman filter model, and illustrates the utility of this model as an analysis tool by means of an example based on a uniform tube which provides a qualitative assessment of the potential of the technique for application to real speech signals.     

Near lossless data compression onboard a hyperspectral satellite

To deal with the large volume of data produced by hyperspectral sensors, the Canadian Space Agency (CSA) has developed and patented two near lossless data compression algorithms for use onboard a hyperspectral satellite: successive approximation multi-stage vector quantization (SAMVQ) and hierarchical self-organizing cluster vector quantization (HSOCVQ). This paper describes the two compression algorithms and demonstrates their near lossless feature. The compression error introduced by the two compression algorithms was compared with the intrinsic noise of the original data that is caused by the instrument noise and other noise sources such as calibration and atmospheric correction errors. The experimental results showed that the compression error was not larger than the intrinsic noise of the original data when a test data set was compressed at a compression ratio of 20:1. The overall noise in the reconstructed data that contains both the intrinsic noise and the compression error is even smaller than the intrinsic noise when the data is compressed using SAMVQ. A multi-disciplinary user acceptability study has been carried out in order to evaluate the impact of the two compression algorithms on hyperspectral data applications. This paper briefly summarizes the evaluation results of the user acceptability study. A prototype hardware compressor that implements the two compression algorithms has been built using field programmable gate arrays (FPGAs) and benchmarked. The compression ratio and fidelity achieved by the hardware compressor are similar to those obtained by software simulation     

音频信号矢量编码算法

为了在保证译码恢复的声音质量良好的前提下,减小编码的压缩率,以减小声音信号的存储空间,提出了一种将线性预测编码、SOM神经网络矢量编码以及Huffman编码相结合的声音信号编码算法,将1列声音信号转换为2列信号,这样就可以进行后续的矢量编码。实现了预测编码和矢量编码的结合。利用Matlab软件编程进行了声音信号编解码实验。实验结果表明,在保证声音质量的前提下,该编码方法的码率小于MEPG-1 Layer3的最低的64kbps标准码率,且算法简单。文章提出的编码算法在音频压缩编码方面将具有较高的研究价值和很好的应用前景。     

Chirp Waveform Generation Using Digital Samples

This paper describes two methods of generating an analog frequency-modulated waveform by the use of a small number of digital samples of the ?chirp? waveform. The number of digital samples required is a function of the time-bandwidth product. For certain values of time-bandwidth product, this type of signal generation becomes extremely efficient. Several proofs are offered which show how to select ?optimum? values of time-bandwidth products. Two hardware implementations are suggested. One is based on the use of modulo arithmetic and a small stored memory table. The second method utilizes the inherent signal symmetries available if ?optimum? time-bandwidth products are selected. The symmetrical signal patterns are stored in recirculating reversible shift registers which can be read out at high speeds.     

Hardware-Constrained Hybrid Coding of Video Imagery

The design, implementation, and performance of a video bandwidth compression system is described. In this system, compression is obtained by several methods including the use of DCT/DCPM hybrid coding, frame rate reduction, and resolution reduction. The overall compression ratio is up to 1000:1. The hardware-constrained design of the DCT and the DPCM is described and a new method is derived to solve the optimum integer bit-assignment problem associated with the block quantization process in the DPCM. Computer simulation results are presented which predict that the performance of the system using the derived optimal bit assignment method is superior to those obtained by other bit assignment methods. The real-time hybrid coding system design is optimized for a set of ?modified? average statistics to compress a wide variety of input video images. This approach eliminates the problem of nonzero dc mean value which could otherwise cause serious degradations in the system performance. The compression system is fully implemented and the quality of the reconstructed video as predicted by computer simulation has been demonstrated by the actual hardware performance. The PSNR of the reconstructed imagery is in excess of 36 dB at 2 bits per pixel.     

Nonredundant error correction DQPSK for the aeronautical-satellitechannel

The performance of differentially encoded quadrature phase-shift keying (DQPSK) system employing nonredundant error correction (NEC) receivers with single- and double-error correction capability is analyzed and evaluated for the aeronautical satellite channel. The NEC is an attractive coding technique which employs differential detectors with more than one symbol delay elements and which does not introduce any redundancy as other coding schemes do. As typical for aeronautical satellite communications, a Rician fading channel with Gaussian power spectrum has been considered. Unlike the additive, uncorrelated from symbol to symbol interference such as additive white Gaussian noise (AWGN) or static cochannel interference (CCI) which has been investigated in the past, analysis of the performance in a fading channel is much more difficult. The difficulty arises from the multiplicative and correlative nature of the fading interference. Bit error rate (BER) performance evaluation results have been obtained by means of computer simulation for various channel conditions, including different values of the K-factor and the fading B_DT. These results have indicated that considerable performance gains as compared with conventional differentially detected systems are achieved for high values of K and for very fast fading. Both of these conditions are encountered in typical aeronautical communication systems. Wherever possible, heuristic explanations of the trend of the obtained BER performance evaluation results are also given