signal-processing

Closed. This question is off-topic . It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 4 years ago . As part of a university project I have to do some signal processing and would like to output the results using the PC sound card. The software has to be written in C and needs to work with Windows (preferably 7 and XP). I have found code examples for outputting .wav and similar files, but I am interested in continuously outputting data rather than outputting from files. It is likely that the data for output will

Sorry to ask a similar question to the one i asked before ( FFT Problem (Returns random results) ), but i've looked up pitch detection and autocorrelation and have found some code for pitch detection using autocorrelation. Im trying to do pitch detection of a users singing. Problem is, it keeps returning random results. I've got some code from http://code.google.com/p/yaalp/ which i've converted to C++ and modified (below). My sample rate is 2048, and data size is 1024. I'm detecting pitch of both a sine wave and mic input. The frequency of the sine wave is 726.0, and its detecting it to be

Description of the problem FFT can be used to compute cross-correlation between two signals or images. To determine the delay or lag between two signals A and B , it suffices to locate the peak of: IFFT(FFT(A)*conjugate(FFT(B))) However, the amplitude of the peak is related to the amplitude of the frequency spectra of the individual signals. Thus to determine the Pearson correlation (rho) , the amplitude of this peak must be scaled by the total energy in the two signals. One way to do this is to normalize by the geometric mean of the individual autocorrelations . This gives a reasonable

I want to find the peaks of the raw ecg signal so that I can calculate the beats per minute(bpm). I Have written a code in matlab which I have attached below.In the code below I am unable to find threshold point correctly which will help me in finding the peaks and hence the bpm. %input the signal into matlab [x,fs]=wavread('heartbeat.wav'); subplot(2,1,1) plot(x(1:10000),'r-') grid on %lowpass filter the input signal with cutoff at 100hz h=fir1(30,0.3126); %normalized cutoff freq=0.3126 y=filter(h,1,x); subplot(2,1,2) plot(y(1:10000),'b-') grid on % peaks are seen as pulses(heart beats) beat

I've currently got this bounty running on how to resample audio data with the intention of increasing the pitch . Many solutions have been made and I have to admit I'm feeling a bit overwhelmed by the choices and information. I was directed to this solution and found this chunk of code: public static float InterpolateCubic(float x0, float x1, float x2, float x3, float t) { float a0, a1, a2, a3; a0 = x3 - x2 - x0 + x1; a1 = x0 - x1 - a0; a2 = x2 - x0; a3 = x1; return (a0 * (t * t * t)) + (a1 * (t * t)) + (a2 * t) + (a3); } public static float InterpolateHermite4pt3oX(float x0, float x1, float