I want to develop an iPhone application which should have the ability to count the number of phrases that are received when user sing on the mic.
I\'m currently developing a percussion tutorial program. The program requires that I can determine what drum is being played, to do this I was going to analyse the frequency of the drum recording and
There exists one very good linear interpolation method. It performs linear interpolation requiring at most one multiply per output sample. I found its description in a third edition of Understanding D
I analyzed the sunspots.dat data (below) using fft which is a classic example in this area. I obtained results from fft in real and imaginery parts. Then I tried to use these coefficients (first 20) t
I would like to try to compute y=filter(b,a,x,zi) and dy[i]/dx[j] using FFTs rather than in the time domain for possible speedup in a GPU implementation.
I have a series of rr data (distances between r-r peak in PQRST electrocardiogramm signal) and I want to generate realistic ECG signal in matlab or python. I\'ve found some materials fo开发者_开发知识
I\'m trying to reproduce (part of) the work in this paper: http://www.mit.edu/~kimo/publications/harmonization/
I hope this question is not too vague. I\'m trying to take info from an audio buffer in this Xcode project and use it to do some DSP.
I am trying to resample a signal (sound sample) from one sampling rate, to a higher sampling rate. Unfortunately it needs some kind of filter, as some \'aliasing\' appears to occur, and I\'m not famil
Edit: Actually this is not unexpected behaviour, but I still need a solution. findpeaks compares each element of data to its neighboring values.
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