基于DSP的数字预失真系统设计

为提高无线通信系统的通信质量与效率,提出一种基于DSP自适应数字预失真技术的系统设计。采用TI公司的低功耗、高性能数字信号处理器TMS320VC5502,有效提高了信号处理速度,减少了回路延时。根据信号的幅度,数控衰减器工作在不同的控制模式,以满足不同系统输出功率的要求。结果表明,该系统能有效改善功率放大器的线性度,并将功放的邻道功率比（ACPR）降低了10 dB左右。     

基于因子图的自适应波束形成

图模型具有广泛的应用,它为许多问题提供了一种新的表达方式和研究思路。因子图作为一类重要的图模型,尤其适用于多变量的复杂统计模型。因子图的引入可以使复杂的多变量问题得到简化。因子图理论在系统建模以及信号检测和估计算法中有着重要的应用。国内外不少学者将因子图理论应用于复杂的通信信号处理,但目前很少见到将因子图理论应用在雷达信号处理中。为了将因子图理论作为一种有效的工具应用于雷达信号处理,提出了用因子图理论实现雷达信号处理中的自适应波束形成技术（ADBF）的方法,这为用图模型研究雷达信号处理提供了一个很好的思路。     

A novel smart antenna system implementation for broad-band wirelesscommunications

A novel design of smart antenna system with adaptive beamforming capability is introduced for broad-band wireless communication. To achieve high data throughput of the multi-antenna system, a parallel analog-digital (A/D) signal processing scheme is proposed. The essential idea is to realize the real-time beamforming using heterodyne RF and IF circuitry. The bottleneck of digital signal processor (DSP) I/O and processing speed is thus relieved, while the advanced signal processing capability of the DSP chip is utilized. Based on this idea, a 5.8 GHz smart antenna receiver is implemented. Various experiments are carried out to examine the direction of arrival (DOA) estimation, beam synthesis, and bit error rate (BER) performances of the system. A 20-Mb/s data throughput using binary phase shift keying (BPSK) modulation is demonstrated for this eight-element adaptive array     

A high-speed adaptive antenna array with simultaneous multibeam-forming capability

A new type of adaptive beamforming antenna system architecture is proposed for multichannel wireless communications. Multibeam communication with high data throughput is accomplished using the proposed beamformer architecture. The system consists of analog mixers, a multitone direct digital synthesizer (DDS), and a digital signal processor (DSP) controller. The essential idea of multibeam forming is based on a multitone weighting scheme combined with analog-digital hybrid signal processing. While the real-time multibeam construction is realized by the analog mixer circuits and a DDS, the complicated adaptive beamforming and direction-of-arrival estimation algorithms are carried out by the DSP. In this architecture, only one beamformer circuit is required to handle multiple beams, leading to significant reduction in hardware counts. A 5.8-GHz eight-element adaptive beamforming array successfully demonstrates two-beam simultaneous beamforming with less than three degrees of peak and steering errors and more than 20-dB interference suppression. The test-bed exhibits successful two-channel data recovery at 25-Mb/s data throughput in each channel with binary phase-shift keying modulation, for simultaneous dual-beam reception. The bit-error-rate measurement validates the robustness of the communication quality under strong interferences.     

短波自适应选频控制器的DSP实现

短波自适应选频控制器是改善短波通信质量的系统。语言中结合短波自适应选频的工作原理，给出了利用ＤＳＰ芯片实现短波自适应选频的硬件结构和软件实现方法。最后给出了室内及实际线路的试验结果。多次实际线路试验表明；该控制器能够在预置的频率中选出最佳频道，且符合美国军标ＭＩＬ－ＳＴＤ－１８８－１４１Ａ。     

Waveform coding for low-power digital filtering of speech data

This paper evaluates waveform coding techniques known from low bit-rate communication for their usefulness in low-power digital FIR filtering of speech signals. The encodings considered include linear PCM, PCM with adaptive and logarithmic quantization, and differential PCM, combined with two's-complement and sign-magnitude number representation. Selected implementation aspects for each alternative are discussed. Experimental results are presented to quantify potential power savings subject to statistical signal properties and operating conditions. Guidelines for the choice of encoding in application-specific digital signal processing of speech data are provided.     

基于ARM+DSP的钢琴调律器设计

针对钢琴调律工作的特点,结合现代数字信号处理技术和嵌入式技术,提出了基于ARM和DSP双处理器的钢琴调律器的设计思路.选用音频编解码芯片TLV320AIC23对琴音信号采集;采用以TMS320VC5402为核心的信号处理模块完成信号处理算法实现,包括基频的提取、基频范围的确定、采样率自适应变换以及调音判断等;利用以S3...     

信号处理课程中的应用实例展示平台

信号处理课程是电子信息类专业的基础课。本文设计并实现了一个信号处理课程的应用实例展示平台,该平台应用MATLAB GUI进行开发,包含了数字信号处理、通信原理、数字图像处理、语音信号处理、水下通信五个方面的15个实例,且每个实例制作成子菜单的形式,方便用户使用。设计的应用实例展示平台有助于相关课程的辅助教学,可帮助学生理解信号处理在各领域的应用。     

Digital signal processing. 1. Digital filters and the DFT

The wide availability of inexpensive and relatively fast digital signal processing chips has given rise to a great expansion in the application of these devices in many areas, particularly consumer electronics. The ease with which digital processing operations can now be performed has allowed the realisation of highly sophisticated signal processing algorithms. The paper introduces the first of two main areas of digital signal processing, namely digital filtering and Fourier transformation. An overview is presented of both finite impulse response and infinite impulse response digital filter types. These designs are later extended to cover the concept of an adaptive filter, which automatically trains or adapts its characteristics or frequency response to compensate for signal distortions or interference. Finally, the discrete Fourier transform is introduced for measuring the frequency spectrum of a signal     

通信信号处理系统概率计算的应用

随着现代化技术的不断发展,人们对移动通信和无线接入技术也产生了很大程度的依赖性.现代化的通信信号处理系统需要寻求一种新的模式来满足当代社会的需求,融入更加简便、精准和实用的元素.概率计算在通信信号处理系统中起着很大的作用,为通信信号处理系统打开了一个新的突破口,能够提高通信信号处理系统的效率和精准度.通过对通信信号处理系统概率计算的具体体现、应用前景和意义进行分析和探讨,总结归纳出了通信信号处理系统概率计算的应用,以供相关人员进行参考.     

多采样率信号处理的发展

数字通信系统中,为适应传输、降低资源消耗、适于处理操作,常需要变换信号的采样率。多采样率信号处理理论从语音信号处理中发展起来,在应用中不断丰富。随着软件无线电的应用,多采样率变换在数字信号领域占据越来越重要的地位。多采样率信号处理技术与小波分析、分数阶傅里叶变换等其他信号处理技术相结合是未来发展的方向。     

Signal processing for underwater acoustic communications

The performance and complexity of signal processing systems for underwater acoustic communications has dramatically increased over the last two decades. With its origins in noncoherent modulation and detection for communication at rates under 100 b/s, phase-coherent digital communication systems employing multichannel adaptive equalization with explicit symbol-timing and phase tracking are being deployed in commercial and military systems, enabling rates in excess of 10 kb/s. Research systems have been shown to further dramatically increase performance through the use of spatial multiplexing. Iterative equalization and decoding has also proven to be an enabling technology for dramatically enhancing the robustness of such systems. This article provides a brief overview of signal processing methods and advances in underwater acoustic communications, discussing both singlecarrier and emerging multicarrier methods, along with iterative decoding and spatial multiplexing methods.     

自适应LMS算法的DSP实现

自适应LMS算法是一种很有用且很简单的估计梯度的方法,在图像处理中得到广泛应用。而DSP是一种高速、高性能特别擅长进行数字信号处理运算的微处理器,用DSP实现数字图像信号的处理有很好的实时性及处理精度。本文完成了一种基于DSP的自适应LMS维纳滤波器的设计,从而实现了对数字图像进行初步降噪处理。     

数字振荡器的实现原理与仿真编程

鉴于数字信号处理中的实时应用问题，探讨了数字滤波器的实时实现原理，包括离散时间网络结构的作用和离散时间信号处理的特点，重点介绍了实时应用中实现数字滤波器的基本思想，运用MATLAB和所导出的通用数字振荡器的系统函数对一个非正弦数字振荡器进行了仿真编程，还讨论了有关实时处理的几个重要的系统性质，表达了作者对于当前学习和应用数字信号处理方法的观点。     

Advanced digital receiver principles and technologies for PCS

The synergy between digital radio communications and VLSI signal processing is revolutionizing the design of wireless terminals. Driving this synergy are certain fundamental paradigms in modern communication theory, digital signal processing, and VLSI design. The authors discuss the modern centers-of-gravity model, which they believe is emerging as the basis for the successful design and implementation of advanced digital communication systems. Central to this model are design principles that enable engineers to systematically derive digital receiver structures and explore algorithm and architecture trade-offs using sophisticated tools. Digital signal processing technology is critical in the implementation of these digital receiver structures efficiently. Finally, CAD tools for digital communications system design and design space exploration are shown to be of crucial importance in the efficient execution of these designs     

数字信号处理中的加窗问题探究

随着科学技术水平的不断提高,我国的数字信号处理技术也得到了较快的发展。尤其是在模拟信号向数字信号转化的具体过程里,利用加窗技术能够较好的提高实施的效率和质量。针对现今比较繁杂的信号分析处理方式以及相关要求来看,不仅要保证信号的精准还要确保其不失真。本文将针对数字信号处理中的加窗问题进行简要的探讨和研究,促进其更好的发展。     

Fully static multiprocessor array realizability criteria forreal-time recurrent DSP applications

The paper considers real time implementation of recurrent digital signal processing algorithms on an application-specific multiprocessor system. The objective is to devise a periodic, fully static task assignment for a DSP algorithm under the constraint of data sampling period by assuming interprocessor communication delay is negligible. Toward this goal, the authors propose a novel algorithm unfolding technique called the generalized perfect rate graph (GPRG). They prove that a recurrent algorithm will admit a fully static multiprocessor implementation for a given initiation interval if and only if the corresponding iterative computational dependence graph of this algorithm is a GPRG. Compared with previous results, GPRG often leads to a smaller unfolding factor α_GPRG     

A New Blind Beamforming Procedure Based on The Conjugate Gradient Method for CDMA Mobile Communications

The objective of this paper is to present an adaptive algorithm for computing the weight vector which provides a beam pattern having its maximum gain along the direction of the mobile target signal source in the presence of interfering signals within a cell. The conjugate gradient method (CGM) is modified in such a way that the suboptimal weight vector is produced with the computational load of O(16N), which has been found to be small enough for the real-time processing of signals in most land mobile communications with the digital signal processor (DSP) off the shelf, where N denotes the number of antenna elements of the array. The adaptive procedure proposed in this paper is applied to code division multiple access (CDMA) mobile communication system to show its excellent performance in terms of signal to interference plus noise ratio (SINR), bit error rate (BER), and capacity, which are enhanced by about 7 dB, times, and 7 times, respectively, when the number of antenna elements is 6 and the processing gain is 20 dB.     

The TMS320 family of digital signal processors

This paper begins with a discussion of the characteristics of digital signal processing, which are the driving force behind the design of digital signal processors. The remainder of the paper describes the three generations of the TMS320 family of digital signal processors available from Texas Instruments. The evolution in architectural design of these processors and key features of each generation of processors are discussed. More detailed information is provided for the TMS320C25 and TMS320C30, the newest members in the family. The benefits and cost-performance tradeoffs of these processors become obvious when applied to digital signal processing applications, such as telecommunications, data communications, graphics/image processing, etc.     

基于矩阵优化理论的自适应数字波束形成算法研究

自适应数字波束形成作为天线技术和数字信号处理融合的产物,近年来成为跨领域的研究热点,并且广泛应用到新一代的相控阵雷达和移动通信中。文中提出一种基于矩阵流形理论的自适应波束形成算法。不同于传统的波束形成算法,该算法将波束形成问题转化为高维优化问题,并引入矩阵流形优化理论,在Stiefel流形上求解该问题。由于降低了求解问题的维度,限制了最优解的范围,该算法相比于传统的自适应波束形成算法,具有收敛速度快、运算量小、鲁棒性好的优点。