Description: Title: MMSE Receiver for DS-SS in AWGN Channel
Author: Panson Tantikovit
Summary: An adaptive receiver for DS-SS systems
MATLAB Release: R12.1
Required Products: Communications Toolbox,Signal Processing Blockset
Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain.
-Title: MMSE Receiver for DS-SS in AWGN Channel Author: Panson Tantikovit Summary: An adaptive receiver for DS-SS systems MATLAB Release: R12.1 Required Products: Communications Toolbox,Signal Processing Blockset Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain. Platform: |
Size: 19456 |
Author:zzp |
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Description: 设信号 ，用 对x(t)采样得x(n)，是否会发生频谱混叠？现利用FFT分析其频谱。
1．编程绘制该信号的波形。
2．若令N=16，编程对x(n)做FFT运算，并绘制其幅频特性曲线。
3．令N=1024，编程对x(n)做FFT运算，并绘制其幅频特性曲线。
4．分析2、3的运算结果。
设计调试报告要求：
1．工作原理简述；
2．设计思路；
3．难点及解决方法；
4．设计、调试结果及分析；
5．程序文本及操作步骤。-based signal used to x (t) in the sample x (n), whether there will be a spectrum aliasing? FFT analysis is the use of their spectrum. 1. Programming mapping of the signal waveform. 2. If so N = 16, the programming of x (n) do FFT, and the mapping of its amplitude-frequency characteristic curve. 3. Order N = 1024, the programming of x (n) do FFT, and the mapping of its amplitude-frequency characteristic curve. 4. Analysis of two, three computational results. Design debugging reporting requirements : 1. Principle outlined; 2. Design; 3. Difficulties and solutions; 4. Design, debugging and analysis of results; 5. Text and operating procedures steps. Platform: |
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Author:魏臻 |
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Description: 录制一段个人自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Record a person s own voice signal, and recording the signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum given filter performance indicators, using a window function method and the bilinear transform filter design and draw the frequency response filter then use the filter of their own design for the signal acquisition filtering, after filtering the signal to draw the time-domain waveform and spectrum, and filtering of signals before and after contrast, analysis of signal changes playback of voice signal Finally, the design of a signal processing system interface. Platform: |
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Author:yim |
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Description: 录制一段自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Section of their own voice recording signal, and recording the signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum given filter performance indicators, using a window function method and bilinear transformation design of filters, and draw the frequency response filter then use the filter of their own design for the signal acquisition filtering, after filtering the signal to draw the time-domain waveform and spectrum, and filtering of signals before and after contrast, analysis of signal changes playback voice signal Finally, the design of a signal processing system interface. Platform: |
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Author:joe |
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Description: 小波变换的一级分解过程是，原始信号分别进行低通、高通滤波，再分别进行二元下抽样，就得到低频、高频（也称为平均、细节）两部分系数；而多级分解则是对上一级分解得到的低频系数再进行小波分解，是一个递归过程。-Wavelet decomposition level is, the original signal were low-pass, high pass filter, and then carried out under the binary sample, we obtained low-frequency, high-frequency (also known as the average, details) coefficient of two parts and multi-level decomposition it is the decomposition level to be low-frequency coefficients further wavelet decomposition is a recursive process. Platform: |
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Author:王正 |
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Description: Using the function to sample the voice signal and achieve fast Fourier transform in MATLAB, and then get the signal characteristics of the spectrum.Filtering the signal from the filter,and then playback the signal of voice Platform: |
Size: 72704 |
Author:林霞 |
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Description: 使用matlab对正弦信号采样 以及对其进行DFT变换-Matlab sinusoidal signal on the use of sampling, as well as its DFT transform Platform: |
Size: 38912 |
Author:史惠 |
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Description: 录制一段个人自己的语音信号。对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；对语音信号进行加噪和去噪处理，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；实现快录慢放、慢录快放等功能。-Record a person' s own voice signal. Of the recorded signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum of voice signals, noise and de-noising processing, draw Filtered time-domain signal waveform and spectrum, and filtering the signal before and after comparative analysis of signal changes playback voice signal realize quickly recorded slow release, slow release recorded faster functions. Platform: |
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Author:或或 |
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Description: this file has codes that describes how to ccmpute the signal spectrum , the power spectrum, how to calculate the autocorrelation sequence of a signal, how to calculate the autoregressive coeffecients of a signal,and how to reduce the noisy elements in a speech sample. Platform: |
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Author:pulkit sharma |
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Description: This package contains script to calculate aliasing for given time domain signal. Script to plot in time domain the function: x(t)=A0cos(2pif0t)+A1cos(2pif1t)
where fs1=300 Hz and fs2=150 Hz (sampling rate)
A0 and A1 are amplitude s signal (armónicos).
f0 is the maximun component in frecuency of the signal to sample,
when it shows X(t) signal with a sampling frecuency 300 Hz,
we will see Aliasing :)
It s because we aren t meetting with the Nyquist Theorem, because we don t choice Fs>=2FNy Fs(Sampling Frec) FNy(Frec Nyquist) to sample aplying fft(fast fourier transform) with at least 600 Hz.
By actinio --vjgonzalezg@gmail.com-This package contains script to calculate aliasing for given time domain signal. Script to plot in time domain the function: x(t)=A0cos(2pif0t)+A1cos(2pif1t)
where fs1=300 Hz and fs2=150 Hz (sampling rate)
A0 and A1 are amplitude s signal (armónicos).
f0 is the maximun component in frecuency of the signal to sample,
when it shows X(t) signal with a sampling frecuency 300 Hz,
we will see Aliasing :)
It s because we aren t meetting with the Nyquist Theorem, because we don t choice Fs>=2FNy Fs(Sampling Frec) FNy(Frec Nyquist) to sample aplying fft(fast fourier transform) with at least 600 Hz.
By actinio --vjgonzalezg@gmail.com-- Platform: |
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Author:actinio |
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Description: DPCM编码,简称差值编码，是对模拟信号幅度抽样的差值进行量化编码的调制方式.-DPCM coding, referred to as difference coding is to sample the analog signal amplitude to quantify the difference between the modulation coding. Platform: |
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Author:强强 |
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Description: 数字通信系统中采用基带传输和频带传输[ 1, 2 ] ,在抽样判决器之前,信号叠加了信道噪声. 本文利用
Matlab软件仿真[ 3, 4 ]分析了在判决器前叠加了高斯噪声后,信噪比与系统的误码率之间的关系.-Digital communication system using baseband transmission and band transmission [1, 2], in a sample before the decision device, the signal superimposed on the channel noise. In this paper, Matlab software simulation [3, 4] of the device in the sentence before the superposition of the Gaussian noise After the SNR and BER relationship. Platform: |
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Author:zhujianbo |
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Description: 脉冲编码调制(PCM)实现
编程实现PCM技术的三个过程：采样、量化与编码。
采样：低通连续信号采样，以 x=sin(200*t) m=x./(200*t) m=m.*m 为例说明低通采样定理，绘出信号时、频图形；带通连续信号采样，以x=sin(20*t) m=x./t 为例说明带通采样定理，绘出信号时、频图形。
量化：均匀量化，以幅度 的正弦信号为例实现为64级电平的均匀量化；非均匀量化，输入A律PCM编码器的正弦信号 ，采样序列为 ，将其进行PCM编码，给出编码器的输出码组序列
编码：以上述信号为例，实现A律的13折线近似法及国际标准PCM对数A律量化编码。-Pulse code modulation (PCM) to achieve
PCM technology programming three processes: sampling, quantization and coding.
Sample: low-pass continuous signal sampling to x = sin (200* t) m = x./(200* t) m = m.* m an example low-pass sampling theorem, draw the signal, frequency graphics bandpass sampling continuous signals to x = sin (20* t) m = x./t an example bandpass sampling theorem, draw the signal, frequency graph.
Quantization: uniform quantization, in order to realize the magnitude of sinusoidal signal as an example for the 64 level uniform quantization non-uniform quantization, input A law PCM encoder sine signal, the sampling sequence, to be PCM encoded, the encoder is given The output code sequence
Code: A Case Study of the signal to achieve the 13 A line approximation law and international standards on the number of A law PCM coding quantization. Platform: |
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Author:马芳 |
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Description: 基于C开发的三个隐层神经网络,包括
1）初始化权、阈值子程序；
2）第m个学习样本输入子程序；
3）第m个样本教师信号子程序；
4）隐层各单元输入、输出值子程序；
5）输出层各单元输入、输出值子程序；
6）输出层至隐层的一般化误差子程序；
7）隐层至输入层的一般化误差子程序；
8）输出层至第三隐层的权值调整、输出层阈值调整计算子程序；
9）第三隐层至第二隐层的权值调整、第三隐层阈值调整计算子程序；
10）第二隐层至第一隐层的权值调整、第二隐层阈值调整计算子程序；
11）第一隐层至输入层的权值调整、第一隐层阈值调整计算子程序；
12）N个样本的全局误差计算子程序。
-The three development based on C hidden layer neural networks, including
1) initialization, threshold subroutines,
2) first m a learning samples input subroutines,
3) the first m a sample teachers signal subroutines,
Each unit 4) hidden layer input and output value subroutines,
Each unit 5) output layer of input and output value subroutines,
6) output layer to hidden layer of generalization error subroutines,
7) hidden layer to input layers of general error subroutines,
8) output layer to the value of the hidden layer, the output layer threshold adjustment subroutines,
9) third hidden layer to the second hidden layer, the value of the hidden layer threshold adjustment subroutines,
10) second hidden layer to the value of the hidden layer, the second hidden layer threshold adjustment subroutines,
11) first hidden layer to the input value adjustment, the first hidden layer threshold adjustment subroutines,
12) N samples of global error subroutines. Platform: |
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Author:kison |
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Description: 对RFID读写器接收模块的解调以及FM0解码进行了MATLAB仿真。首先我们对读写器接收到的信号进行仿真，里面附带着标签信息；其次对接收到的ASK信号进行相干解调；接着解调后的信号经过抽样判决，进行FM0解码。仿真结果可表明接收到的信号与发射的信号一致。-On the RFID reader receiver module FM0 decoding and demodulation and a MATLAB simulation. First of all we readers received signal simulation, which included the tag information followed by the received signal coherent ASK demodulation then demodulate the signal through the sample after the verdict, for FM0 decoding. Simulation results show that the received signal and the signal line emission. Platform: |
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Author:朵朵 |
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Description: 块自适应处理算法先有采样快拍数据计算采样协方差矩阵，再来计算自适应权矢量。典型的块自适应处理算法为采样矩阵求逆(SMI)算法。仿真了采样矩阵求逆法(SMI)的波束形成方向图旁瓣的高低受信号快拍数的影响。-Block adaptive processing algorithms prior sample snapshot data of the sampling covariance matrix, again computing adaptive weight vector. A typical block adaptive processing algorithms for the sample matrix inversion (SMI) algorithm. Simulation of the sampling matrix inversion method (SMI) beamforming pattern sidelobe level of the signal by the number of snapshots. Platform: |
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Author:Billy |
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Description: 该文档详细讲述了关于matlab在通信仿真中的具体应用，包括模拟调制中的信号处理以及数字信号的抽样量化过程的matlab实例-This document describes in detail the communication matlab simulation on the specific applications, including analog modulation of the signal processing and digital signal sample quantization process of matlab examples Platform: |
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Author:郭悦 |
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Description: 利用SIMULINK和M函数相结合的方式仿真BFSK 调制在多路径瑞丽衰落信道中的传输性能。其中 source产生速率为10Kbit/s、帧长度为1秒的二进制数据源data,并且通过BFSK产生调制信号。BFSK调制的频率间隔为24KHz, BPSK调制符号的样点数为2，调制信号通过多径瑞利衰落信道，移动终端相对运动速率为40公里/小时，接收端对信号进行解调，并把解调后的信号和原始数据信号相比较计算误比特率。最后Sink模块根据SNR与误比特率的关系绘制曲线。-And M functions using SIMULINK combination of simulation BFSK modulation multipath fading channel transmission performance in Ruili. Wherein the source generation rate of 10Kbit/s, the frame length is 1 second binary data source data, and generates a modulated signal BFSK. BFSK modulation frequency interval of 24KHz, BPSK modulation symbols for the two sample points, a modulated signal multipath Rayleigh fading channel, the rate of relative movement of the mobile terminal is 40 km/h, the receiver demodulates the signal and supplies the demodulated after the raw data signal and the signal calculated by comparing the bit error rate. Finally Sink module drawing curves based on SNR and BER relationship. Platform: |
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Author:陈思远 |
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Description: The main procedure of the experiment was to sample a signal with different sampling frequencies and observe the differences between them. This experiment is important for us because it aims to teach how to use matlab commands to reconstruct a sampled signal from its frequency components by using interpolation techniques. Platform: |
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Author:herrberk |
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Description: 对信号进行频谱分析时，数据样本应有足够的长度，一般FFT程序中所用数据点数与原含有信号数据点数相同，这样的频谱图具有较高的质量，可减小因补零或截断而产生的影响。-The signal spectrum analysis, data sample should be of sufficient length, the general FFT program used in the data points with the same number of data containing the signal spectrum, that has high quality, can decrease the influence caused by zero padding or truncation. Platform: |
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Author:Mrqi |
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Search - sample the signal matlab