signal-processing

问题 I've only used MATLAB as a calculator, so I'm not as well versed in the program. I hope a kind person may be able to guide me on the way since Google currently is not my friend. I have a wav file in the link below, where there is a human voice and some noise in the background. I want the noise removed. Is there anyone who can tell me how to do it in MATLAB? https://www.dropbox.com/s/3vtd5ehjt2zfuj7/Hold.wav 回答1: This is a pretty imperfect solution, especially since some of the noise is

问题 I am developing a system as an aid to musicians performing transcription. The aim is to perform automatic music transcription (it does not have to be perfect, as the user will correct glitches / mistakes later) on a single instrument monophonic recording. Does anyone here have experience in automatic music transcription? Or digital signal processing in general? Help from anyone is greatly appreciated no matter what your background. So far I have investigated the use of the Fast Fourier

问题 I'm trying to do real time pitch detection of a users singing, but I'm running into alot of problems. I've tried lots of methods, including FFT (FFT Problem (Returns random results)) and autocorrelation (Autocorrelation pitch detection returns random results with mic input), but I can't seem to get any methods to give a good result. Can anyone suggest a method for real-time pitch tracking or how to improve on a method I already have? I can't seem to find any good C / C++ methods for real time

问题 I have a 3D matrix (MxNxK) and want to resize it to (M'xN'xK') (like imresize in matlab). I am using image pyramid, but its result is not very accurate and need a better one. Any solution? 回答1: You could use interp3 (since you want to interpolate 3D data): im=rand(2,3,4); %% input image ny=3;nx=3;nz=5; %% desired output dimensions [y x z]=... ndgrid(linspace(1,size(im,1),ny),... linspace(1,size(im,2),nx),... linspace(1,size(im,3),nz)); imOut=interp3(im,x,y,z); 回答2: Here is the resize function

问题 UPDATE: I found a Scipy Recipe based in this question! So, for anyone interested, go straight to: Contents » Signal processing » Butterworth Bandpass I'm having a hard time to achieve what seemed initially a simple task of implementing a Butterworth band-pass filter for 1-D numpy array (time-series). The parameters I have to include are the sample_rate, cutoff frequencies IN HERTZ and possibly order (other parameters, like attenuation, natural frequency, etc. are more obscure to me, so any

问题 As it currently stands, this question is not a good fit for our Q&A format. We expect answers to be supported by facts, references, or expertise, but this question will likely solicit debate, arguments, polling, or extended discussion. If you feel that this question can be improved and possibly reopened, visit the help center for guidance. Closed 7 years ago . I'm looking for libraries to: read and write meta data (for example ID3v2 tags in mp3 and all) convert compressed to to raw audio data

问题 Lets assume we have a dataset which might be given approximately by import numpy as np x = np.linspace(0,2*np.pi,100) y = np.sin(x) + np.random.random(100) * 0.2 Therefore we have a variation of 20% of the dataset. My first idea was to use the UnivariateSpline function of scipy, but the problem is that this does not consider the small noise in a good way. If you consider the frequencies, the background is much smaller than the signal, so a spline only of the cutoff might be an idea, but that

问题 Has anybody used the Apple FFT for an iPhone app yet or know where I might find a sample application as to how to use it? I know that Apple has some sample code posted, but I'm not really sure how to implement it into an actual project. 回答1: I just got the FFT code working for an iPhone project: create a new project delete all the files except for main.m and xxx_info.plist going to project settings and search for pch and stop it from trying to load a .pch (seeing as we have just deleted it)

问题 I have a signal into which I want to introduce several offsets and delays, where offsets range from 0.5 to 5 and delays range from 1 to 7 . I'm providing an example signal here to demonstrate the problem I'm having, but the size of my real data is 1x1666520. How do I introduce these changes to the signal? Example code: t = [ 0 : 1 : 50]; % Time Samples f = 45; % Input Signal Frequency Fs = 440; % Sampling Frequency data = sin(2*pi*f/Fs*t)'; T.InputOffset = 5; T.OutputOffset = 5; addoffset =

问题 I'm implementing a 'filter sweep' effect (I don't know if it's called like that). What I do is basically create a low-pass filter and make it 'move' along a certain frequency range. To calculate the filter cut-off frequency at a given moment I use a user-provided linear function, which yields values between 0 and 1. My first attempt was to directly map the values returned by the linear function to the range of frequencies, as in cf = freqRange * lf(x). Although it worked ok it looked as if