understanding digital signal processing

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[Matlab] Algorithm Craftsman video 1: Digital signal processing simulation and realization the first signal source generation and filtering 1, 2

); fir1_band_stop_filter = Fir1 ( N,[WC1 WC2], ' stop '); %ftype%figure (8); Freqz (fir1_band_stop_filter);% Implementation filter FIR1_BS_S1 = filter (fir1_band_stop_filter,1,sin_s); fft_bs_ Filter = FFT (FIR1_BS_S1), figure (9), subplot (2,1,1);p lot (time, fir1_bs_s1), title (' Band filter After filter waveform '); subplot (2,1,2); Danbianfudu = 2*abs (Fft_bs_filter (1:POINT_S/2));p lot (F,danbianfudu), title (' frequency amplitude characteristics after band-stop Filtering ');   [Matlab] alg

Digital Signal Processing (I)

frequency at will. Assume that there is a sine signal X [N], its frequency f = 100Hz, its amplitude is A, and its initial phase is 0, the signal can be expressed as follows:X (t) = A * sin (2 * pI * 100 * t)Sample it with the sampling frequency fs = Hz. The resulting digital signal X [N] is:X [N] = A * sin (2 * pI * 1

Digital voice Signal Processing learning notes-homomorphic processing voice signals (1)

5.1 OverviewThe method of processing, which can be two signals by multiplying the signal by multiplication, or by convolution synthesis signal separation.for voice signals. Our aim is to separate the original components from the convolution of the channel impulse corresponding to the excitation component.Each signal co

"Digital Signal Processing" Learning Summary

21st century belongs to the digital information age, it is fortunate to learn some of the basic content of digital signals, although it is not clear the application of these theoretical basis, but his application of the technology has given himself a lot of accumulation also let himself feel the greatness of human wisdom, this article mainly discusses the Gaoxi and Ding Yu-mi authoring "

The common methods in digital signal processing

; % Set Frequency Transform range>> subplot (1,2,2);>> plot (F,ftt_sig (1:256))% plot power spectral density distribution map>> title (' Power Spectral density graph ')fftfilt-based on FFT amount FIR FilterUsageY=fftfilt (b,x)The function uses the superposition method to perform the FIR filter based on FFT. The input vector x is filtered by the given coefficient vector b.Y=fftfilt (B,x,n)The function uses the superposition method to perform the FIR filter based on FFT. The input vector x is filt

Digital signal processing 1-moving Average Filter

as the data_array_section, storing the result For i=1:moving_point_right Data_array_filter (i) =data_array_section (i);EndFor i= (col_array+moving_point_left+1): Col_array % This piece of code is to assign values to the first 4 and the last 4 points of a matrixData_array_filter (i) =data_array_section (i);EndFor i= (moving_point_right+1):(col_array+moving_point_left)For J=moving_point_left:moving_point_right Data_array_filter (i) =data_array_filter (i) +data_array_256 (i+j); %moving Averag

MATLAB Digital Signal Processing

'); > Ftt_sig = FFT (SIG, 512); % perform a 512-point fast Discrete Fourier transformation for the SIG signal to zero> P = ftt_sig. * conj (ftt_sig)/512; % calculate the power spectral density of the signal> F = 1000 * ()/512; % set the frequency conversion range> Subplot (1, 2 );> Plot (F, ftt_sig () % to plot the power spectrum density distribution > Title ('power spectral density fig ') Fftfilt-FFT-bas

Various frequency relations in digital signal processing

the s plane, the digital frequency still changes from 0 to 2 pi on the z-plane unit circle. 。。。。。。 The z-plane repeats like this. We know that the Fourier transform of the discrete signal corresponds to the z-transform on the unit circle , so the above conclusion verifies why the Fourier transform of the discrete signal is cyclical: the root cause is th

Signal Processing of full digital B-ultrasound (1)

Abstract: full digital B-ultrasound is the development direction of ultrasound medical instruments. Its basic technical feature is to use digital hardware circuits to achieve real-time processing of ultrasonic signals with extremely large data volumes. The development direction and main signal

Digital signal Processing--fft and butterfly-shaped algorithm

In digital signal processing, it is often necessary to use discrete Fourier transform (DFT) to obtain the frequency domain characteristics of the Signal. Although the traditional DFT algorithm can obtain the signal frequency domain characteristics, The algorithm is computati

Limitations of digital gain in signal processing

In the field of signal processing, the gain is divided into digital gain and analog gain, the digital gain is used to manipulate discrete digitized sampled values, and analog gain directly operates on a continuous analog signal waveform. For most high-end stereos today, the

FPGA implementation of digital signal processing to create super cool PHP Data pie chart Effect implementation code

cake Draw_sector3d ($img, $ox, $oy, $a, $b, $v, $SD, $ed, $clrLst [$i]); $SD, $ed, $clrLst [$i]); Draw Labels Imagefilledrectangle ($img, 5, $ly, 5+ $fw, $ly + $fh, $clrLst [$i]); Imagerectangle ($img, 5, $ly, 5+ $fw, $ly + $fh, $CLRT); Imagestring ($img, $font, 5+2* $fw, $ly, $labLst [$i]. ":". $datLst [$i]. " (". (Round (10000* ($datLst [$i]/$tot))/100). "%)", $CLRT); $str = Iconv ("GB2312", "UTF-8", $labLst [$i]); Imagettftext ($img, $font, 0, 5+2* $FW, $ly +13, $CLRT, "./simsun.ttf", $str.

Why do we need to add a window in digital signal processing?

When digital processing of analog signals, the first step is to sample the analog signals. The sampling frequency is determined by the nequest sampling theorem. Perform dtft on the sampled digital signal to obtain its spectrum. According to the calculation formula of dtft, all sampling points of signals are required fo

Digital Signal Processing (2)

,-]; ayplus = Conv (A, axplus) byplus = Conv (B, bxplus) + Conv (xic, axplus) [R, P, C] = residuez (byplus, ayplus) MP = ABS (P), AP = angle (P)/pin = [0: 50]; [x, FS] = audioread ('e: test1.wma '); % self-input WMA file x = x (:, 1); % Single Channel ? Y = filter (B, A, X, xic); figuresubplot (211), plot (x), title ('input signal curve '); subplot (212 ), plot (Y), title ('signal curve after differential e

Principles of digital signal processing

For an explanation of the Fourier transform, the following link: http://blog.jobbole.com/70549/. It's very detailed:Note the point:1, signal processing based on such a concept, the signal to be processed (? ) can be decomposed into sine wave, different amplitude, different phase, different frequency, such as: F=cos (W1*T+Π/4) +100cos (W2*T+Π/6)2, by decomposition

Digital voice Signal Processing learning notes--short time-frequency domain analysis of speech signals (2)

/southeast "style=" font-size:18px "> Total sample Rate (SR) equalswatermark/2/text/ahr0cdovl2jsb2cuy3nkbi5uzxqvam9qb3poyw5nanu=/font/5a6l5l2t/fontsize/400/fill/i0jbqkfcma==/ Dissolve/70/gravity/southeast ">In most real windows, B can be represented as a multiple of fs/n. Thatwatermark/2/text/ahr0cdovl2jsb2cuy3nkbi5uzxqvam9qb3poyw5nanu=/font/5a6l5l2t/fontsize/400/fill/i0jbqkfcma==/ Dissolve/70/gravity/southeast ">c is a proportional constant, which is:In the formula. Sr/fs is the "over-rate samp

Digital signal Processing C language (3)------FFT

]); } printf ("\ n"); } Q1=x[n-1]; Q2=y[n-1]; for(i=0; i) {W=6.28348530718/n*i; W1=cos (w); W2=-sin (w); C1=1.0-a1*w1+a2*W2; C2=a1*w2+a2*W1; C=c1*c1+c2*C2; D1=1.0-a1*q1+a2*Q2; D2=a1*q2+a2*Q1; A[i]= (C1*D1+C2*D2)/C; B[i]= (C2*D1-C1*D2)/C; } printf ("\ Theoretical dft\n"); for(i=0; i2; i++) { for(j=0;j2; j + +) {printf ("%10.7f+j%10.7f", a[2*i+j],b[2*i+J]); } printf ("\ n"); } FFT (X,y,n,1); printf ("\ndft\n"); for(i=0; i2; i++) { for(j=0;j2; j + +) {printf ("%10.7f+j%10.7

Digital signal Processing C language------uniform distribution and Gaussian distribution random number

=gauss (mean,sigma,s); printf ("%13.7f", x); } printf ("\ n"); } returna.exec ();}Gauss.c#include DoubleGaussDoubleMeanDoubleSigmaLong int*s) { inti; Doublex, y; DoubleUniformDouble,Double,Long int*); for(x=0, i=0;i A; i++) x+=uniform (0.0,1.0, s); X=x-6.0; Y=mean+x*Sigma; return(y);}Uniform.cdouble uniform (double A,double B,longint *seed) {double t; *seed=2045* (*seed) +1; *seed=*seed-(*seed/1048576) *1048576; t= (*seed)/1048576.0; t =a+ (b-a) *t; return (t);}

"MATLAB" 002 "digital signal processing theory, algorithm and implementation" 001_rand

%-----------------------------------------------------------------% Exa011001_rand.m:for Example 1.10.1% to test rand.m and to generate the White noise signal u (n)% with uniform distribution% produces a uniformly distributed random white noise signal and observes the histogram of the data distribution%-----------------------------------------------------------------Clear % clear variables that may be saved

Linux signal Signal Processing mechanism

signal is a very important part of Linux programming, this article will detail the basic concept of signal mechanism, the approximate realization method of Linux to signal mechanism, how to use signal, and several system calls about signal.

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