In Matlabs recent versions, the specgram function is being replaced by spectrogram, and the documentation states: Note. To obtain the same results for the removed specgram function, specify a 'Hann' window of length 256. Unfortunately, that doesn't seem to work for me, if I use spectrogram(signal,hann(256)), the result is different...
I'm searching ways to identify scores, when someone is playing i.e. guitar. How can I manage that? I've heard that midi stores music data as musical scores. I wonder if it's a good solution. ...
I used ARX function then RESID function from the System Identification Toolbox, but the resulting residuals are: 0 0 0 5 6 8 7 8 the number of zeros=the number of lags, I need a complete vector of residuals ...
Hello, I am a beginning electrical engineer(Analog) trying to understand the essence of DSP techniques. Things that interest me right now is filter implementations, FFT analysis on real time audio in general. I recently bought a MAC mini(OSX10.4 snow leopard) and inherited a jail broken iphone from my wife. I thought i could leverage the...
I have a number of voice sound streams encoded in PCM (PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian) which I want to send over the network. To save bandwidth I currently encode the streams using JSpeex. However to save even more bandwidth I would like to (if possible) combine these multiple streams into a single strea...
I'm implementing a VOIP application that uses pure Java. There is an echo problem that occurs when users do not use headsets (mostly on laptops with built-in microphones). What currently happens The nuts and bolts of the VOIP application is just the plain datalines of Java's media framework. Essentially, I'd like to perform some digita...
x(n) is given need x(-n+3) so to solve it: first advance the x(n) signal by 3 units(time) then fold it, or make a reflection of it are the above steps correct or is the following correct first fold the x(n) signal then advance the signal by 3 units ? ...
I need to find out how I can manipulate sound files on the Iphone. I am mainly seeking ways to add echo sound effect by manipulating the Delay Time, Amplitude and Decay Time. Any assistance will be appreciated ...
I have information (20,000 frames of data) about an audio track that I have auto-correlated using: [r,lags] = xcorr(XX,XX,'biased'); And it looks like this: Which hopefully is so far so good. Ideally I would like to be able to take the frame number that corresponds to the highest part of the second peak. I've read around and tried ...
Heia... I'm trying to write the code QPSK with zeroforcing when N=2 and i got one error which i dont know how to fix it. plz help me. thanks alot here is the code: Modulation = 'QPSK' Decode_Method = 'ZeroForcing' switch Modulation case {'QPSK'} Symbols = [ 1+j 1-j -1+j -1-j ]'; end Symbols = Symbols.'; nSymbols = length...
I am looking for a simple (pseudo)code that spectrum-inverse a sampled audio signal. Ideally C++ The code should support different sample rates (16/32/48KHz). ...
this code is work with 2x2 QPSK but not working with 3x3 QPSK of Maximum likelihood and dont know how to fix it this is the code of 3x3 for Likelihood function [Received_Symbols] = ML_Decode_3_3(Y,H,Symbols) N=3; nSymbols = length(Symbols); Symbol_Vectors = [ kron( ones(1,nSymbols) , Symbols ) ; kron( Symbols , ones(1,nSymbols) ) ]...
Hi everyone I have some values of intensities from images of yeast colony plates. I need to be able to find the peak values from the intensity values. Below is an example image showing how the values look when graphed. Example of some of the values 5.7 5.3 8.2 16.5 34.2 58.8 **75.4** 75 65.9 62.6 58.6 66.4 71.4 53.5 40.5 26.8 14....
I take blocks of incoming data and pass them through fftw to get some spectral information. Everything seems to be working, however I think I'm getting some aliasing issues. I've been trying to work out how to implement a hann window on my blocks of data. Google has failed me for examples. Any ideas or links I should be looking at? dou...
I have come up with an idea for an audio project and it looks like Go is a useful language for implementing it. However, it requires the ability to apply filters to incoming audio, and Go doesn't appear to have any sort of audio processing package. I can use cgo to call C code, but every signal processing library I find uses C++ classes ...
I am looking to build a a small program that reads a sound file and applies echo effect to it. I am seeking guidance for how to accomplish this. ...
I'm trying to combine three signal waveforms into a single, interleaved waveform. I need to know the best way to do it in C++ STL. Better solutions would use as much C++ STL style as possible, avoid redundant code, etc. Is there some STL "tuple" type class that would do this for me? I need contiguous storage at all times for backward com...
Hey Stackoverflow; I currently have an array full of data which from what I believe is the Amplitude of my wave file. It is currently at a low -32768 and at a high 32767. I also have the SampleRate which was 16,000hz. My understanding of sound isn't very good; does anyone know from this how I can calculate the Frequency? Help greatly...
Hi, Im trying to do a screen-flashing application, that flashes the screen according to the music(which will be frequencies, such as healing frequencies, etc...). I already made the player and know how will I make the screen flash, but I need to make the screen flash super fast according to the music, for example if the music speeds up, ...
Hi, I want to remove vocals from mp3 sound tracks. I searched google, and tried few softwares but none of them are convincing. I am planning to read the mp3 file, get a waveform and remove the waveform that is above a specified limit. do you have any suggestions on how to proceed. -- Update I just want code that can read mp3 file for...