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按分类查找All DSP编程(246) 

[DSP编程] Spi_8SEG

DSP2833x 芯片 CPU 定时器 0 定时 1s 中断,中断服务子程序完成数码管计数显示(秒钟实验) 这个例子主要是定时器 0 定时中断时中断服务子程序做以下 3 方面工作(a)定时器计数器加 1(b)数码管计时显示(c)中断应答寄存器相应位清 0,以响应 PIE 同组内的其他中断请求
The CPU timer 0 of dsp2833x chip is interrupted for 1s, and the interrupt service subroutine completes the nixie tube counting display (second experiment). This example is mainly the interrupt service subroutine when timer 0 is interrupted for the following three aspects: (a) timer counter plus 1 (b) nixie tube timing display (c) the corresponding bit of interrupt response register is cleared to respond to pie Other interrupt requests in the same group (2020-02-27, C/C++, 128KB, 下载0次)

http://www.pudn.com/Download/item/id/1582785481247318.html

[DSP编程] estimate-the-phase-difference

本程序可以实现两组正弦信号的相位差测量,此程序是经CCS3.3开发出来的,可以在28335上运行。程序里面包括相位差测量算法程序,FFT程序,加噪声程序。分步给出方便修改。程序中还对不同开发环境给出大量头文件和库文件,方便大家调试成功
The program can achieve two sinusoidal signals phase difference measurement, this procedure is developed by CCS3.3, you can run on 28335. Program which includes the phase difference measurement algorithm, FFT procedures, plus noise process. Step gives easy modification. The program also gives a lot of different development environments header files and libraries to facilitate the successful commissioning (2013-11-13, C/C++, 1226KB, 下载69次)

http://www.pudn.com/Download/item/id/2398719.html

[DSP编程] signalprocessing

1、产生信号,两个实正弦信号的叠加,幅度分别为2、4、1、3;混入均值为0、方差为1的白噪声。采用自适应滤波器对其进行去噪。 2.产生信号,为两个实正弦信号的叠加,其幅度均为4,混入均值为0、方差为1的白噪声。采用有限脉冲响应法设计一个维纳滤波器估计信号 ,并求最小均方误差。 3.产生高斯分布的白噪声w(n),自行给定一个5阶AR模型,让该白噪声通过这个AR模型,得到输出信号x(n),再估计x(n)的AR模型数,比较估计的结果和原来给定的AR模型的参数。
A signal to be generated, the superposition of two real sinusoidal signals, amplitude, respectively 2,4,1,3 mixed with mean 0 and variance 1 white noise. Noising using adaptive filter them. (2) generate signals as a superposition of two real sinusoidal signal whose amplitude was 4, mixed with mean 0 and variance 1 white noise. The finite impulse response method to design a Wiener filter estimate signal, and for the minimum mean square error. 3 Gaussian white noise generated w (n), given a five self-order AR model, so that the white noise through the AR model, get the output signal x (n), and then estimate x (n) the number of AR model, Compare the results and the original estimate given by the AR model parameters. (2013-07-04, matlab, 2KB, 下载25次)

http://www.pudn.com/Download/item/id/2296824.html

[DSP编程] netpid

实现了步进电机32细分驱动。为了补偿步进 电机相绕组电流与其产生磁场之间非线性引起的误差,采用了最小二乘 法对细分步距角误差曲线进行了拟合与修正,提高了细分精度。为了检 测32细分后的步距角,采用了自准直仪加精密数显转台的光学测量方法,
Subdivision driver stepper motor 32. In order to compensate for the stepper motor phase winding current produced between the magnetic field caused by the nonlinearity error, using the method of least squares subdivision step angle error curve fitting and correction to improve the subdivision precision. To step angle detection for 32 segments, using the the autocollimator plus precision digital turntable optical measurement method, (2013-04-14, Others, 20KB, 下载5次)

http://www.pudn.com/Download/item/id/2199519.html

[DSP编程] control-of-the-DC-Motor

我在德国留学电机控制实验课的直流部分,包括分别对电流环和转速换的校正,并对是否对反馈值加滤波器性能进行对比,并附上了效果图,是学习电机控制基础,程序的效果明显,实现无差控制,并对动态性能和扰动性能进行了测试。先运行初始化文件,然后加载模型,关于初始化文件的注释,都是电机的参数的设置,直接看模型就可以
I study in Germany motor control DC part of the experimental class, including the correction for the current loop and speed, and whether the feedback value plus filter performance compared, together with the effect diagram, is learning motor control based program the effect is obvious, and no difference in control and the dynamic performance and disturbance performance test. Can run the initialization file, and then load the model, and comment on the initialization file are motor parameter settings, look directly at the model (2013-04-08, matlab, 329KB, 下载27次)

http://www.pudn.com/Download/item/id/2190118.html

[DSP编程] timer

医疗设备的实时运行系统的实时控制。用于TI数字信号处理器。它可以被用于使用得克萨斯仪器5515 dsp芯片的数据采集系统。非常有用的。需要TI芯片组库。医学上的应用开发工具,它已经推出了一套完整的信号链设计和软件的心电图,数字听诊器和脉冲血氧计等产品。每个三款医疗开发套件(MDKs)是由购买模拟前端(AFE)模块的具体电路设计,为每个最终产品加上TMS320C5515 DSP评估模块(EVM)。随着硬件和软件的设计信息,包括原理图,应用程序的源代码,医疗专用算法和技术文件,每个MDK降低客户的开发时间由6至8个月。此外,MDK适用提供了极大的评估平台,帮助医疗设备制造商专注于产品的差异化,如算法开发和增强功能。他们还为新手开发谁想要进入医疗行业的迅速降低门槛。
Medical equipment, real-time operating system real-time control. For TI digital signal processor. (2012-12-17, C++ Builder, 1KB, 下载12次)

http://www.pudn.com/Download/item/id/2085971.html

[DSP编程] haimingchuang

海明窗  语音信号一般在10ms到30ms之间,我们可以把它看成是平稳的。为了处理语音信号,我们要对语音信号进行加窗,也就是一次仅处理窗中的数据。因为实际的语音信号是很长的,我们不能也不必对非常长的数据进行一次性处理。明智的解决办法就是每次取一段数据,进行分析,然后再取下一段数据,再进行分析。
Hamming window voice signal is generally between 10ms to 30ms, we can see it as is stationary. To handle voice signals, we add window to the speech signal, that is, once a process only the data in the window. Because the actual speech signal is very long, we can not do not have a very long data one-time processing. The sensible solution is to always take a piece of data, analysis, and then remove the piece of data, and then analyzed. (2012-10-20, matlab, 1KB, 下载21次)

http://www.pudn.com/Download/item/id/2021457.html

[DSP编程] ddspphomeworri

数字信号处理的应用之一是从含有加性噪声的信号中去除噪声。现有被噪声污染的信号x[k]=s[k]+d[k],式中: 为原始信号d[k]为均匀分布的白噪声。(1)分别产生50点的序列s[k]与白噪声序列ddd[k],将二者叠加生成x[k],并在同一张图上绘出x0[k],d[k]与x[k]的序列波形。(2)均值滤波能有效去除叠加在低频信号上的噪声。已知3点滑动平均数字滤波器的单位脉冲响应为h[k]=[1,
One of the applications of digital signal processing to remove noise from the signal with additive noise. Existing noise pollution signal x [k] = s [k] the+d [k], where: the original signal d [k] uniformly distributed white noise. (1), respectively, to produce a 50-point sequence s [k] white noise sequences ddd [k], the two superimposed to generate x [k], and plotted on the same graph x0 [k], d [k] and x [k] sequence waveform. (2) the mean filter can effectively remove noise superimposed on the low-frequency signals. Known 3-point moving average digital filter unit impulse response is h [k] = [1, (2012-07-10, matlab, 28KB, 下载13次)

http://www.pudn.com/Download/item/id/1934950.html

[DSP编程] clock

基于DSP的数字模拟钟实现 左半屏显示模拟钟,右半屏显示数字钟,下方红绿灯 每1秒亮一位,实现循环彩灯效果。 通过键盘可该日期、星期和时间 具体实现方法是: 1.先按数字键选择要修改的数据位 (234567分别对应月日周时分秒) 2.按 + , - 对应的数据执行加1或减1
DSP-based digital and analog clock analog clock display to achieve left and right screen display digital clock, every 1 seconds off the bottom of a traffic light to achieve circulating lantern effect. Through the keyboard of the date, time of week and specific method is: 1. Press the number key you want to modify the data bits (234 567 corresponding to every second day of week) 2 Press' +' ,' -' perform the corresponding data plus 1 or minus 1 (2011-06-17, Others, 471KB, 下载4次)

http://www.pudn.com/Download/item/id/1572366.html

[DSP编程] dm642

1\colorbar_ 是一个向左滚动的彩条,主要验证视频输出。DSP负载90 多。 2\colorbar 是一个经过优化的向左滚动的彩条,DSP负载9.5 左右。 3\loopback 是一个一路视频输入同时一路视频输出的例程 4\loopback_pip 是一个两路视频输入同时一路视频输出的例程,把这两路视频以画中画的形式叠加在一起。 5\h263_loopback :H263编解码例程。 6\jpeg_loopback :MJPEG 编解码例程。 7\mpeg2_loopback :MPEG2编解码例程。 8\video_networking :MJPEG编解码例程同时网络传输。演示了jpeg加网络传输的一个网络视频服务器的例子,在PC上用IE浏览器可以看到图像。这个工程的IP地址是动态分配的。 9\video_networking_ip :MJPEG编解码例程同时网络传输。这个工程和上一个基本一样,只是这个工程的IP地址是固定的:192.168.0.253. 10\是声音实验例程,从线路输入,从线路输出。也可以通过改变audio.c中的myaic23_init()这个函数改成从话筒输入。 void myaic23_init() { reSetAic23() SetAic23Sample_rate(AIC23_REG8_44_1KHZ) SetAic23Line() // SetAic23Mic() } 11\loopback_uart是一个图像输入到输出,还有串口收发的例程
1\colorbar_ 是一个向左滚动的彩条,主要验证视频输出。DSP负载90 多。 2\colorbar 是一个经过优化的向左滚动的彩条,DSP负载9.5 左右。 3\loopback 是一个一路视频输入同时一路视频输出的例程 4\loopback_pip 是一个两路视频输入同时一路视频输出的例程,把这两路视频以画中画的形式叠加在一起。 5\h263_loopback :H263编解码例程。 6\jpeg_loopback :MJPEG 编解码例程。 7\mpeg2_loopback :MPEG2编解码例程。 8\video_networking :MJPEG编解码例程同时网络传输。演示了jpeg加网络传输的一个网络视频服务器的例子,在PC上用IE浏览器可以看到图像。这个工程的IP地址是动态分配的。 9\video_networking_ip :MJPEG编解码例程同时网络传输。这个工程和上一个基本一样,只是这个工程的IP地址是固定的:192.168.0.253. 10\是声音实验例程,从线路输入,从线路输出。也可以通过改变audio.c中的myaic23_init()这个函数改成从话筒输入。 void myaic23_init() { reSetAic23() SetAic23Sample_rate(AIC23_REG8_44_1KHZ) SetAic23Line() // SetAic23Mic() } 11\loopback_uart是一个图像输入到输出,还有串口收发的例程 (2011-05-04, Windows_Unix, 1589KB, 下载67次)

http://www.pudn.com/Download/item/id/1516435.html

[DSP编程] LabVIEW-data-acquisition-system

基于LabVIEW的多通道数据采集系统的设计,江苏科技大学硕士毕业论文。全文分为2部分第一部分数据采集硬件的设计中,采用DSP 作为采集卡的CPU,利用USB 接口实现了采集卡与PC 机的通信,整个外围逻辑时序部分用CPLD 控制。第二部分数据管理与分析,作者利用LabVIEW 编写了基于Windows XP的上位机应用程序以及上位机与采集卡之间的通信程序;利用LabSQL 工具包建立了ACCESS 数据库,实现对采集到的数据的存储,并可以查询历史数据。同时系统可以生成报表以及可以对所需要的数据进行打印,并可以对得到的数据进行一些必要的分析,如加窗函数,逐点分析。本文版权归作者所有!下载后产生后果下载者个人负责
LabVIEW-based multi-channel data acquisition system, Jiangsu University of Science Thesis. Paper is divided into two parts the first part of data acquisition hardware design, acquisition card with DSP as CPU, USB interface using a capture card and PC-communication, the entire outer part of the logic timing control with the CPLD. The second part of the data management and analysis, the authors prepared using LabVIEW Windows XP-based PC applications and PC and the communication between the acquisition card program use LabSQL Kit ACCESS database was established to achieve the collected data storage and can query the historical data. At the same time the system can generate the required reports and data can be printed, and can get the necessary data analysis, such as windowing function, point by point analysis. 将中文译成英语 All copyright of the article! Consequences for those who download download individual responsibility (2011-04-22, PDF, 766KB, 下载135次)

http://www.pudn.com/Download/item/id/1501438.html

[DSP编程] downloadpaper

基于DSP 的电吉他数字音效器设计,、设计了DSP 为核心处理器的数字音效处理系统,研究了数字失真类音效算法的建 模和实现。该系统包含延时、合唱、失真、布鲁斯、混响、均衡等17 种音响效果
DSP-based digital audio electric guitar design, and design of the DSP core processors, digital audio processing system, digital distortion kind of sound modeling and algorithm implementation. The system consists of delay, chorus, distortion, blues, reverb, equalization and other 17 kinds of sound effects (2010-11-03, PDF, 394KB, 下载117次)

http://www.pudn.com/Download/item/id/1335402.html

[DSP编程] F2812-FFT

FFT 并不是一种新的变换,它是离散傅立叶变换(DFT)的一种快速算法。由于我们在计算DFT 时一次复数乘法需用四次实数乘法和二次实数加法;一次复数加法则需二次实数加 法。每运算一个X(k)需要4N 次复数乘法及2N+2(N-1)=2(2N-1)次实数加法。所以 整个DFT 运算总共需要4N^2 次实数乘法和N*2(2N-1)=2N(2N-1)次实数加法。如此一来,计算时乘法次数和加法次数都是和N^2 成正比的,当N 很大时,运算量是可观的,因而需要 改进对DFT 的算法减少运算速度。
FFT is not a new transformation, it is the discrete Fourier transform (DFT) of a fast algorithm. Since we have a complex multiplication when calculating DFT Xuyong four real multiplications and real additions secondary a complex quadratic addition Zexu real addition. Each operator of a X (k) requires 4N times complex multiplication and 2N+2 (N-1) = 2 (2N-1) times the real number addition. Therefore, the DFT computation requires 4N ^ 2 times in total real multiplications and N* 2 (2N-1) = 2N (2N-1) times the real number addition. Thus, when calculating the number of multiplications and additions and are proportional to N ^ 2, when N is large, the computation is considerable and, therefore, need to improve on the DFT algorithm to reduce computational speed. (2010-09-13, C/C++, 395KB, 下载33次)

http://www.pudn.com/Download/item/id/1295442.html

[DSP编程] DSP_Exp1

1、 学习Matlab最有效的方式是自己动手以及经常查Help。 2、 遇到不很确定的函数或者操作,最好的办法就是自己建一个测试矩阵然后试一下,不会对数据产生不好的影响,也十分直观,因为很多细节老师也不会很清楚。 3、 养成用Editer编程的好习惯,最好结构化编程,经常使用自建的函数,使得代码清晰易查错。 4、 自己给自己加注释,运行程序时可以先试着一行一行运行,看一下每一行的输出是否与预期相同等等,出错的时候自上而下慢慢查。
1, the most effective way to learn Matlab their own hands, as well as regular check Help. 2 has encountered are not sure of the function or operation, the best way is to build their own and then try a test matrix will not produce a bad effect on the data, but also very intuitive, because many details of the teachers are not very clear. 3, to develop a good habit to use Editer programming, the best structured programming, frequent use of self-built functions, making the code clear and easy troubleshooting. 4, add their own notes to yourself, run the program can first try to run line by line, take a look at the output of each line is the same as expected and so on, an error check when the top-down slowly. (2010-03-17, Others, 201KB, 下载4次)

http://www.pudn.com/Download/item/id/1089779.html

[DSP编程] 51djv

j-v.cn 51djv 系统全源码提供。所有演示功能请查看 www.dj-v.cn,商用请告之,任何修改后发布,请发一份COPY给我,万分感谢。 现在由于本人对此程序的后续发展感到很茫然,不知道如何发展了,有兴趣的朋友加Q 5207269,Q群 6446358。
j-v.cn 51djv system to provide full source code. All presentation features, please view the www.dj-v.cn, commercial please to announce any revised release, please send a COPY to me most grateful. I am now as a result of the development of procedures for follow-up was very confused, I do not know how to develop, and friends are interested in additional Q 5207269, Q group 6446358. (2009-06-03, Java, 17801KB, 下载110次)

http://www.pudn.com/Download/item/id/791398.html

[DSP编程] dspjtag

J T AG 接口插座与DSP芯片的距离:为了保证JTAG信号不受干扰,需 要注意两者之间的距离不超过六英寸(15甲24厘米),超过这个距离,就需要在中 间加缓冲芯片。本设计中使用了244作为缓冲芯片,但其原因不是由于器件之间 距离过长,而是考虑到仿真器工作在5V电压,DSP引脚为3.3V,为了电平兼容 性而进行的电压转换功能。
JT AG socket interface with DSP chips Distance: JTAG signals in order to guarantee uninterrupted, need to pay attention to the distance between the two is not more than 6 inches (15 A 24 centimeters), more than this distance, we need to increase the buffer between chips. This design was used as a buffer chip 244, but its not because the distance between devices is too long, but taking into account the work in the simulator 5V Voltage, DSP-pin to 3.3V, in order to conduct the compatibility level of the voltage conversion function. (2008-05-23, Others, 291KB, 下载65次)

http://www.pudn.com/Download/item/id/471124.html

[DSP编程] DC_Brushless_Motor_Servo_control

直流无刷电机的伺服控制,DSP控制C程序愿代码加说明文档
Brushless DC motor servo control, DSP control C program is ready to add code documentation (2007-10-30, C/C++, 569KB, 下载314次)

http://www.pudn.com/Download/item/id/352613.html

[DSP编程] IMAG_c

CCS编程环境 使用的是汇编加C的混合编程方法: The programme of the Correlation Algorithm. Using INT2 to get the input signal. Array x, in first step, is the input signal produced by programme, in next step, is the input signal get from A/D, the length is 128, 32-bit floating point. Array y, in first step, is the input signal produced by programme, in next step, is the input signal get from A/D, the length is 128, 32-bit floating point. Array cor is the Correlation result, the length is 255, 32-bit floating point.
CCS programming environment used in the compilation and C mixed programming : The program of the correlation algorithm. Us ing INT2 to get the input signal. Array x, in first step, is the input signal produced by program, in next step, is the input signal get from A/D, the length is 128, 32-bit floating point. Array y, in first step, is the input signal produced by program, in next step, is the input signal get from A/D, the length is 128, 32-bit floating point. Array cor is the Correla tion result, the length is 255, 32-bit floating point. (2007-04-23, Others, 1KB, 下载6次)

http://www.pudn.com/Download/item/id/272130.html

[DSP编程] dsphomework1

数字信号处理的应用之一是从含有加性噪声的信号中去除噪声。现有被噪声污染的信号x[k]=s[k]+d[k],式中: 为原始信号d[k]为均匀分布的白噪声。 (1)分别产生50点的序列s[k]和白噪声序列d[k],将二者叠加生成x[k],并在同一张图上绘出x0[k],d[k]和x[k]的序列波形。 (2)均值滤波可以有效去除叠加在低频信号上的噪声。已知3点滑动平均数字滤波器的单位脉冲响应为h[k]=[1,1,1 k=0,1,2],计算y[k]=x[k]*h[k],在同一张图上绘出前50点y[k],s[k]和x[k]的波形,比较序列y[k]和s[k]。
digital signal processing applications from one containing additive noise signal denoising. Existing noise pollution was a signal x [k] = s [k] d [k], where : of the original signal d [k] uniform distribution of white noise. (1) have 50 points each sequence s [k] and white noise sequence d [k] superposition of the two generation x [k] and the same is likely to draw a map x0 [k] d [k] and x [k] waveform sequence. (2) mean filter can effectively remove superimposed on the low-frequency noise on the signal. 3:00 known moving average digital filter unit for pulse response h [k] = [k = 0,1,2 1,1,1] Calculation y [k] = x [k]* h [k], the same is likely to draw a map before 50 y [k] s [k] and x [k] waveform sequence comparison y [k] s and [k]. (2007-01-03, matlab, 28KB, 下载104次)

http://www.pudn.com/Download/item/id/239100.html

[DSP编程] tf3

这是一个时频分析中关于gabor变换的程序,此程序分离频率的效果不错。 Gabor变换: 式中a,b为常数,a代表栅格的时间长度,b代表栅格的频率长度 式中的 是一维信号x(t)的展开系数,h(t)是一母函数,展开 基函数是h(t)由作移位和调制生成的,
This in a time frequency analysis about the gabor transformation procedure, this procedure separation frequency effect is good. Gabor transformation: In the formula a, b is the constant, a represents the grid the time length, b on behalf of the grid frequency length in the formula is the unidimensional signal x (t) launches the coefficient, h (t) is a female function, launches the primary function is h (t) by makes shifting and the modulation production, (2006-07-04, matlab, 1KB, 下载234次)

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