搜索资源列表
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谱估计中的Esprit算法，利用matlab进行仿真找出信号的频率，精度较高-Spectral estimation algorithm in the Esprit, the simulation using matlab to find the signal frequency, high precision
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求一个给的的信号输入的平均频率的matlab程序-to find average frequency of a signal
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本程序用于测量增强语音与纯净语音的对数谱距离，用于语音增强。对数谱距离越小，表明增强语音与纯净语音越接近，增强语音质量越好；反之，则增强语音质量越差。-Calculates the average log-spectral distance between CLEAN and NOISY
signals. Frames are 25ms long with 60 percent (15ms) overlap, hamming
windowed. RS is the remove
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关于脑电信号中机械噪声，工频噪声，以及基线漂移的处理，以及提取脑电信号中的特质点，应用matlab中的小波变换来进行消噪找特征点-On the EEG machine noise, frequency noise, and baseline drift of the processing, as well as the characteristics of EEG in point, application of wavelet transform in matlab to find featu
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Here,we want to estimate a system with ARX and ARMAX.
the euation of system is : 2(s+1)/s^2+2s+2
so first, we find the discrete equation that sampling frequency is 100ms.
Here,in our sample,na=nb=2,nc=1
also ,this algorithm find the poles and
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这是一个简单的语音基音检测方法，通过对语音信号进行滤波、分帧、求自相关函数，可得到浊音的基音周期，从而实现基频检测。-This is a simple voice pitch detection, voice signals by filtering, sub-frame, find the autocorrelation function, voiced the pitch period available to achieve the fundamental frequency testin
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利用Ansys进行板建模、划分网格，并初步计算，将Ansys的板的M、K导入到Matlab中，进行振动控制分析，并根据阻尼比自动生成阻尼阵C，绘制频率响应曲线，找到最大值，进行比例控制，并给出控制效果-The board using Ansys modeling, meshing, and preliminary calculations, the Ansys board of M, K imported into Matlab, vibration control analysis, damp
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求一个系统的幅频响应相频响应单位冲击响应以及其它-Find the frequency response of a system unit impulse response of the phase-frequency response and other
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matlab实现，验证奈奎斯特采样定理，对信号求傅里叶变换的数值解，数值求解，画出时域图，频域图-matlab implementation, verification Nyquist sampling theorem, Fourier transform of the signal find the numerical solution, numerical solution, draw the graph in time domain, frequency domain graph
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method to find frequency response of a signal using matlab
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how to scovert speech to matrix
how to convert matrix form to speech
how to find sample frequency for speech
how to play an sound signal in matlab
mf-how to scovert speech to matrix
how to convert matrix form to speech
how to find sample
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程序1：周期信号（方波）的展开
程序2：用图形表示DSB调制波形及其包络线
程序3：通过傅里叶变换找出被随机信号淹没的信号频率
程序4：DSB-AM 双边带幅度调制和解调
程序5：AM调制和解调-Procedure 1: periodic signal (square wave)
Program 2: graph DSB modulation waveform and its envelope
Application 3: through the Fourier tran
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该段代码能实现找出隐藏或淹没在噪声时域信号中信号的频率成分。用FFT对试验数据进行谱分析，指出数据包含的频率成份。-One use FFT is to find hidden or submerged in the time domain signal noise frequency components of the signal. Test data with FFT spectrum analysis, pointed out that the data contained in freq
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既然要得到的是频率玫瑰图，就要求出每个方向上在这个月中出现的次数，然后除上整个月的风向次数。通过对比，发现利用极坐标比直角坐标更简便些，故利用 find（）寻找方向值，length计算方向的长度，polar绘制玫瑰图(Since you want a frequency rose diagram, ask for the number of times per month in each direction, and then count the number of winds in the w
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对非平稳信号进行分段、截取，再对处理过后的短时信号做短时傅里叶变换，分析其频谱特性，并找到各个时间节点的频率特性，以便分析不同事件段内的声音特征。(The nonstationary signals are segmented, and then the interception, the short-time signal after processing the short-time Fourier transform, the analysis of the spectrum charac
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对非平稳信号进行分段、截取，再对处理过后的短时信号做短时傅里叶变换，分析其频谱特性，并找到各个时间节点的频率特性，以便分析不同事件段内的声音特征，达到区分不同声音的目的。(The nonstationary signals are segmented, and then the interception, the short-time signal after processing the short-time Fourier transform, the analysis of the spe
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对非平稳信号进行分段、截取，再对处理过后的短时信号做短时傅里叶变换，分析其频谱特性，并找到各个时间节点的频率特性，以便分析不同事件段内的声音特征，一图像的形式表达出，最终达到区分不同声音的目的。(The nonstationary signals are segmented, and then the interception, the short-time signal after processing the short-time Fourier transform, the analysi
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1下载：
对非平稳信号进行分段、截取，再对处理过后的短时信号做短时傅里叶变换，分析其频谱特性，并找到各个时间节点的频率特性，以便分析不同事件段内的声音特征，以图像的形式表达出，最终达到区分不同声音的目的。(The nonstationary signals are segmented, and then the interception, the short-time signal after processing the short-time Fourier transform, the analysi
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透過信號處理找出差分方程式跟他們的頻率響應(用DTFT的方式)(Find the difference equation and frequency response of DTFT in Digital Signal Processing)
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随机信号通过平方根升余弦滤波器滤波后，进入AWGN信道，接收机采用的是LMS算法，通过循环找到最佳的截止频率(The random signal is filtered through the square root cosine filter and enters the AWGN channel. The receiver uses the LMS algorithm to find the best cut-off frequency through the loop.)
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