lowpass-filter

audio file sounds bad/noisy after passing through low pass filter

由 时光毁灭记忆、已成空白 提交于 2021-01-29 18:57:18
问题 I am trying to pass my audio through low pass filter, so as to filter out noise from it. However, the output of the wav is very noisy, and I am unable to understand why. Find the original and filtered wav and their resp. spectrograms below link. enter link description here The code I have used is: #https://stackoverflow.com/questions/25191620/creating-lowpass-filter-in-scipy-understanding-methods-and-units import numpy as np from scipy.signal import butter, lfilter, freqz, filtfilt from

How can I implement a low-pass Butterworth filter in Matlab?

由 懵懂的女人 提交于 2020-01-03 07:03:45
问题 From this answer, I know how to create a High-pass Butterworth filter. From this video, I know that, lowpasskernel = 1 - highpasskernel . So, I created the following Low-pass Butterworth Filter, function [out, kernel] = butterworth_lp(I, Dl, n) height = size(I, 1); width = size(I, 2); [u, v] = meshgrid(-floor(width/2):floor(width/2)-1,-floor(height/2):floor(height/2)-1); % lp_kernel = 1 - hp_kernel kernel = 1 - butter_hp_kernel(u, v, Dl, n); % fft the image I_fft_shifted = fftshift(fft2

Accelerometer Low Pass Filter Smoothing

由 天大地大妈咪最大 提交于 2019-12-24 06:38:01
问题 I am using the device accelerometer and try to smooth the Accelerometer Data CMAcceleration . I am doing this with help of this code: -(void)proccessAccelerometerData:(CMAcceleration)accelData { currentAccelX = (kUpdateInterval * accelData.x) + ((1.0 - kUpdateInterval) * currentAccelX); currentAccelY = (kUpdateInterval * accelData.y) + ((1.0 - kUpdateInterval) * currentAccelY); } Which currentAccelX and currentAccelY is the last accelerometer x and y data. Now, I have the smooth x and y

How to write lowpass filter for sampled signal in Python?

由 只谈情不闲聊 提交于 2019-12-23 05:26:18
问题 Sorry for question that may be trivial, but I am new in signal processing, and I didn't find python code with good explanation. So I will be glad to get explanation for dummy :) I have some signal that sampled each 1 nsec (1e-9 sec) and have, let say, 1e4 points. I need to filter high frequencies from this signal. Let say I need to filter frequencies higher than 10MHz. I want that for frequencies lower than cutoff frequency signal will be passed unchanged. It means gain of the filter will be

Python Video Stabilizer Using Low Pass Filter

由 左心房为你撑大大i 提交于 2019-12-23 03:21:51
问题 I have a project which i should implement a video stabilizer using fft and filters (lpf or hpf) Here is some part of the code that i want to modify it: import cv2 import numpy as np # Create a VideoCapture object and read from input file cap = cv2.VideoCapture('Vibrated2.avi') # load data data = np.loadtxt('Vibrated2.txt', delimiter=',') # Define the codec and create VideoWriter object frame_width = int(cap.get(3)) frame_height = int(cap.get(4)) out = cv2.VideoWriter('Stabilized.avi',cv2

Implementing a post-processed low-pass filter using core audio

由 风流意气都作罢 提交于 2019-12-21 21:45:23
问题 I have implemented a rudimentary low-pass filter using a time based value. This is ok, but trying to find the correct time slice is guess work, and gives different results based on different input audio files. Here is what I have now: - (void)processDataWithInBuffer:(const int16_t *)buffer outBuffer:(int16_t *)outBuffer sampleCount:(int)len { BOOL positive; for(int i = 0; i < len; i++) { positive = (buffer[i] >= 0); currentFilteredValueOfSampleAmplitude = LOWPASSFILTERTIMESLICE * (float)abs

Applying low pass filter

由 五迷三道 提交于 2019-12-12 10:03:30
问题 I want to simulate an interpolator in MATLAB using upsampling followed by a low pass filter. First I have up-sampled my signal by introducing 0's. Now I want to apply a low pass filter in order to interpolate. I have designed the following filter: The filter is exactly 1/8 of the normalized frequency because I need to downsample afterward. (it's a specific excersise to upsample interpolate and downsample in this particular order.) However when I apply this filter to my data using the function

IIR Lowpass filter in C# breaks in x86 mode

由 守給你的承諾、 提交于 2019-12-11 11:16:55
问题 I'm trying to use a IIR LP filter in C#. It is a 5th order Butterworth filter. The code works in 64Bit mode but breaks in 32Bit mode. Debugging showed, that the parameters are slightly different and the output raises to infinity/NAN. I am using doubles for the calculation and storing. The correct parameters a[i],b[i] are: -5 -4,9792522401964 10 9,91722403267282 -10 -9,87615728025693 5 4,91765142871949 -1 -0,979465940928259 The 32Bit calculation gets these: -5 -4,97925281524658 10 9

Low-pass filter in carrier modulation and demodulation

由 心不动则不痛 提交于 2019-12-08 03:10:18
问题 I'm designing a project in which an array is passed through quadrature amplitude modulation (QAM) modulator, and then do carrier modulation, make it playable with the sound() command, then demodulate it back for QAM demodulation. Firstly, I have used the standard way of QAM modulation: M = 16; x = randint(5000, 1, M); y = modulate(modem.qammod(M), x); Then, I wrote my own carrier modulation function: function [out] = carriermodulation(x) fs = 16000; T = 1.0 / 4000; fc = 8000; Q = real(x); I =

Applying low pass filter

由 断了今生、忘了曾经 提交于 2019-12-06 13:21:54
I want to simulate an interpolator in MATLAB using upsampling followed by a low pass filter. First I have up-sampled my signal by introducing 0's. Now I want to apply a low pass filter in order to interpolate. I have designed the following filter: The filter is exactly 1/8 of the normalized frequency because I need to downsample afterward. (it's a specific excersise to upsample interpolate and downsample in this particular order.) However when I apply this filter to my data using the function filter(myfilter, data) the following signal is generated: I really don't know what is happening to my
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