I'd like to use Dynamic Time Warping to compare two feature vectors for two audio recordings (of course I'm doing all the necessary preprocessing first). My program should output the similarity between the two audio recordings in percent. For example 100% means that the two recordings are completely identical, and the more different are ...
hi currently i have the following code in my matlab values = [0;1;0;0;1;0;1;0]; % can contain only 0s and 1s h = modem.oqpskmod; y = modulate(h, values); g = modem.oqpskdemod(h); z = demodulate(g,y); BER = sum(logical(values(:)-z(:)))/numel(values);% thanks to gnovice! now my question ...
I want to make a simple audio equalizer for the iPhone, but I don't know how to start. The equalizer should be really simple and just change bass, high, distortion and maybe echo and chorus. How can I start? Maybe a sample code would be helpful. ...
Hello all! I'm trying to read a image from a electrocardiography and detect each one of the main waves in it (P wave, QRS complex and T wave). Now I can read the image and get a vector like (4.2; 4.4; 4.9; 4.7; ...) representative of the values in the electrocardiography, what is half of the problem. I need a algorithm that can walk thr...
hi for the simulation of oqpsk i have written the below values = [0;1;0;0;1;0;1;0]; h = modem.oqpskmod; y = modulate(h, values); g = modem.oqpskdemod(h); z = logical(demodulate(g,y)); BER = sum(values(:)-z(:))/numel(values); my questions are as follows: 1) how can i verify that it does what i expect it to d...
I am doing sound latency test. my device will be receiving either a beep signal or a silence signal. How can i differentiate between these signals. Please help me. Thanks in advance.. ...
Hi...How do i get frequency using FFT? What's the right procedure and codes? ...
I've got a collection of short WAV files that I would like to process in Java using various digital signal processing algorithms. I need to get an array of int valued samples for this purpose, encoded at the 11025 Hz frame rate. The source files have several different sample rates, including 11025 Hz and 44100 Hz. Here's the code I'm tr...
I'd like to make a cross-platform iTunes visualizer (Mac + Windows). I've looked at: Quartz Composer, which only runs on Mac OS, and appears to consist of dragging boxes around. I'd like to be able to actually program. VizKit, which seems kind of heavy, and doesn't come with any easy-to-modify sample projects. Can someone point me to...
using MATLAB, we have converted image into audio (.wav format). Also this audio file is converted into spectrogram image. Now we are trying to convert this spectrogram into an original input image ...
Can somebody please tell me how to load the default target for a DSP 56800 project with FreeScale Codewarrior 5.9 ? Back in 4.2 there was a DSP56800 menu with a 'Load Default Target' option at the very top, but this seems to have disappeared in 5.9. I'm trying to load an S-file onto an inhouse board running this processor, using a newly ...
I have read a code for pitch determination using autocorrelation method. Can anybody please tell what would be the input data (passed as argument to DetectPitch()) function here: double DetectPitch(short* data) { int sampleRate = 2048; //Create sine wave double *buffer = malloc(1024*sizeof(short)); double amplitude = 0....
Is there any dolby/5.1/7.1 audio processing Python library? It would be best if it is platform independent. Would be nice if it looks like: import lib f = lib.open("8channels_audiofile") lib.play(from=f.channel3, to="left rear"); ...
Is there any form of short-time Fourier transform with corresponding inverse transform built into SciPy or NumPy or whatever? There's the pyplot specgram function in matplotlib, which calls ax.specgram(), which calls mlab.specgram(), which calls _spectral_helper(): #The checks for if y is x are so that we can use the same function to ...
Hey I'm in the situation where i need to do some math related stuff in c# and for that i need some external libarys. The tool i look for should do the following actions: Process sound(wave/mp3): Normalize the amplitude Normalize the phase Any idea which way to go? And is there a big difference if I should to it on mp3 instead of w...
hi is there any difference between half-sine pulse shaping oqpsk used in IEEE 802.15.4 and MSK implemented here? ...
Hi everybody I am developing an application to process geophysical signals; Right now I have done everything in octave and its digital signal processing toolbox, speed is not bad, however the application specifications say I need to port to the final algorithm to C; I am doing lots of filtering, re-sampling and signal manipulation/chara...
Hi, at the moment im trying to implement a FIR lowpass filter on a wave file. The FIR coefficients where obtained using MATLAB using a 40 order. Now i need to implement the FIR algorithm in C# and im finding it difficult to implement it. Any help? Thanks ...
I am using .NET 3.5. I need to Capture Sound from Mic/Headphone and trigger action at particular (not exact) frequency (Need to perform some action when player hits ball with stick while playing golf). So, 1. How to capture sound from Mic/Headphone using .NET 3.5? 2. Trigger action at particular (not exact) frequency? Any ideas? ...
hi, i wanted to know what is the pulse shape of the modem.oqpskmod? and if it is not half-sine pulse shape, how is it possible to make it half-sine pulse shape as it is stated in ieee 802.15.4(zigbee) standard where it shows it as follows p(t)=sin(pi*t/2*Tc) if 0<=t<=2Tc p(t)=0 if otherwise ? thanks a lot! ...