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I'm preforming a system identification measurement, and would like to deconvolve the measurement system's response from the measured response. The set up of the system is as follows; the signal is generated on a PC, the signal is then sent to the sound card, which in turn forwards the signal to the power amplifier, thi...
I'm writing a program that averages the user's weight across different days. I'm planning to use a 5-point moving-average (current day, two before and two after). Sometimes, a data point is missing for 1-2 days. How are these cases usually handled? (if there's a better low-pass filter I could use, I'd love suggestions)...
In my slides about signal processing, there is one that mentions the same thing as the beginning of this and this answer, namely that the Fourier transform of a signal, squared is the power spectrum density of the signal. In this talk, it is mentioned that coherence is calculated by dividing the squared cross-spectrum ...
So I have groups of (x,y) coordinates that I'm using to draw lines. The lines aren't perfectly straight but they are connected. In a Cartesian plane the middle would be called the origin, but I'm working with a digital image processing program, so the coordinate (0,0) is at the top left of the image. What I'm trying t...
In my never-ending quest to identify snores, I've found that "spectral flatness" seems to be a fair measure of signal "quality". I'm computing spectral flatness as the geometric mean of the FFT power $(R*2 + I*2)$ data points divided by the arithmetic mean of the same points. I then (a little twist here) am computing...
I have a warping grid, that has to be applied to an image. When I apply the grid, I get the results I expect. Now, I want to apply the same grid to the same image, but the image is modified in the following way: it is rotated 270 degrees counterclockwise. The grid is stored in memory as an array of points. It is [-1, ...
This questions is in conjugation with this one here! Canny edge detector gives ids of pixels in an image. Not the locations of the edges. To be specific, after all it should be an edge between two pixels. Any implementation that actually gives arcs not pixels? One solution i can make is that if i have a pixel, i can si...
Consider the simple first-order low pass filter, described by $\tau \frac{d \mathcal{O}}{dt} + \mathcal{O}(t)=\mathcal{I}(t)$ Considered as a linear system (ie $\mathcal{L} \mathcal{O} = \mathcal{I}$), $\mathcal{L}$ has transfer function $\frac{1}{1 + i \omega \tau}$ (up to maybe some constants, minus signs etc etc dep...
I need help understanding how/what does it mean to "turn" a transformation parameters into a vote. I know how to recover the transformation parameters (in my case: 6 DOF), but how this turns into a vote in 6D (??) space ? Thank You !
I basicaly understood how the Rubberband/Baseline-correction works. The given spectrum is divided into (N) ranges. The lowest points in every range are determined. The initial baseline is built out of those points. Now all the points on the spectrum are drawn down by the difference between the lowest point in the curr...
Again, my snoring detector: When I look at the rhythm of snoring vs the rhythm of talking in terms of peaks/events (which would represent snores) per hour, there's not much difference -- they range from roughly 500 to 900 per hour. But it occurred to me that perhaps there's a significant difference in the regularity o...
I am reading a book for image processing algorithms (Andreas Koschan, Mongi Abidi: Digital Color Image Processing) and for the contrast algorithm it says that I can either go RGB->HSL or RGB->HSI first and than apply a contrast technique for grayscale images, on the lightness component. Then it gives this formula only,...
I have been working on the topic of camera pose estimation for augmented reality and visual tracking applications for a while and I think that although there is a lot of detailed information on the task, there are still a lot of confussions and missunderstandings. I think next questions deserve a detailed step by step...
Can anyone suggest methods for removing the noise (indicated inside the red square) from every where in the following image, while keeping the white lines?
I'm doing some measurements with my sound card, I have one output channel connected to one input channel. I then send the sound card a unit impulse, i.e. a signal 1 value followed by zeros. The recorded response is shown below. I know that the Fourier transform of a Rect function, or LPF, will result in a Sinc function...
Original Question: https://ux.stackexchange.com/q/23040/16006 I've only taken some basic signal analysis courses, so I might be missing some things. Purely theoretical question: What methods exist for representing audio? What methods could be made for representing audio, more specifically musical audio? So far, I'm awa...
I would like to know how to go about calculating the bandwidth of: A constant (real) sinusoidal tone A (real) sinusoidal pulse. The question is as simple as that, but I am having a hard time with the concept of what exactly the bandwidth of a constant tone should be to begin with, and from there, what the bandwidth ...
Does anyone have any suggestions on good resources or sites for learning about computer vision?
How wavelet analysis works as a dimension reduction techniques? The approximations coefficients at higher level of decomposition are the fewer retained coefficients from the originial ?Is it correct?
I'm trying to conceptually understand what is happening when the forward and inverse Short Time Fourier Transforms (STFT) are applied to a discrete time-domain signal. I've found the classic paper by Allen and Rabiner (1977), as well as a Wikipedia article (link). I believe that there is also another good article to ...
I want to design a digital filter for pulse shaping. Pulses are of 100us Fall time. and the sampling rate is 100MegaSamples/sec. and the Shaping time is 5us. What should my coefficients be??? And how to obtain them using matlab or any other related software.
An image is corrupted by Gaussian noise,then salt & pepper noise. How can you eliminate it and why?
While reading some DSP literature, I've noticed integrals of the following form. An example integral (Equation 7.17) is given on pg. 124 of a book that I've found very challenging to read (Seismic Inverse Q filtering): $u(t) = h(t)\int\limits_{ - \infty }^\infty {u(\tau ,t)d} \tau$ This is exactly how the author has...
I am wondering about the correlation between input size and number of coefficients given by a discrete wavelet transform. I am using Daubechies wavelets to describe a 1D function and I'm using PyWavelets to implement it (which is analogous to the MATLAB toolbox). I started by implementing it using Haar wavelets, which ...
I'm able to read wav files and its values. I need to find peaks and pits positions and their values. First time, I tried to smooth it by $\frac{(i-1 + i + i + 1)}{3} $ formula, then searching on array as array[i-1] > array[i] & direction == 'up' --> pits style solution. But because of noise and other reasons of future...
I have two waves, and I would like to quantify the difference between them. They are exponential sine waves where the frequency decreases with time. I'm interested in the difference in decay constant between the signals. I thought maybe I could use some kind of FFT method to get the envelope, but I don't know what to ...
I don't know whether this is the right place to post this, but I suppose it is. I know that frequency multiplication = circular convolution in time space for discrete signals (vectors). I also know that "the convolution theorem yields the desired linear convolution result only if $x(n)$ and $h(n)$ are padded with zeros...
This came from a homework question, which I've solved. Just need some clarification. I am asked to determine the bandwidth in a given amount of spectrum at a given wavelength. To find this, I know that; $$\delta_f = \frac{c\cdot \delta_\lambda}{\lambda^2}$$ where $\delta_f$ is the bandwidth, $\delta_\lambda$ is the amo...
I have implemented a simple V.23-like FSK modem in C here. The peculiarity of the chosen modulation is such that 0's and 1's are sent as tones of two different frequencies (2100 Hz and 1300 Hz respectively) and the duration of each symbol is 1/1200th of a second, which is between one and two full periods of the symbol ...
I am trying to understand how to interpolate discrete sinusoidal data that I (for now) generate myself, and I get problems that I don't understand. I'm a newbie in this field so do not hesitate to point out even the most basic errors. I have tried two approaches. The first is to take (Python's/numpy's) fft(), get ampli...
If I understood correctly (and this page should confirm: http://www.cs.ioc.ee/~khoros2/linear/convolution-teo/front-page.html) if I convolve linearly (the usual point-to-point multiplication and addition without tricks for each element of the image matrix) an image with a kernel I obtain RESULT1 if I calculate the DF...
I have a 1-by-10000 matrix of double`s stored in a file named "fecg.mat". The matrix represents the magnitude of a recorded FECG signal. I've plotted it against time (from 0 to 9999): For removing the baseline I wonder if I can use a high-pass filter. How do I design a proper filter? P.S Signal processing isn't my f...
I did a bit of searching on this site but surprisingly, I didn't find much relevant information, and my knowledge of DSP is very very limited. My goal is quite simple : I want to program an algorithmic reverb in C++ which sounds really good. Or more exactly, the best option would be to let the end user choose the trad...
I am trying to do my own blob detection. My program will receive a real time video, and try to detect a white paper sheet, even if something is written on the paper. I need to detect the paper and its corner, because what I really want is to draw a opengl polygon over the paper, such that each corner of the paper wil...
I want to fill the black areas with the value of neighboring pixels. Kind of interpolation. Can any one suggest me how Can I do in matlab.
What are the internal workings of the OpenCV function findChessboardCorners?
I want to implement the Fast Cosine Transform. I read on wikipedia, that there is a fast version of the DCT which is similarly computed to the FFT. I tried to read the cited Makhoul* paper, for the FTPACK and FFTW implementations that are also used in Scipy, but I were not able to extract the actually algorithm. This i...
I have a file that is composed of n data records with a sample rate s. Each of these records has a duration d. Now due to postprocessing I have to change the duration to d=d/3, thus afterwards I'll have 3*n data records stored in the file. I'm now wondering if there is some signal processing rule that says that I also...
Say I have 5 measurements of a signal as a column vector: x. I can smooth the signal by multiplying it with a smoothing matrix: $\ \mathbf{xs=S*x} $ For instance I might use [1 2 1] as a smoothing kernel in which case I get: S = 2 1 0 0 0 1 2 1 0 0 0 1 2 1 ...
I performed 512 point FFT on a signal. I got another set of 512 Numbers. I understand that those numbers represent amplitude of the various sine and cosine waves having different frequencies. If my understanding is correct can somebody tell me how to know the frequencies of those sine and cosine waves from the knowle...
I have a signal that I sample at 500khz. I am trying to detect a rise, fall and the peak in the incoming data. The base of the peak could be  for 250 usec or 2.5msec, amplitude could be 6db or 15db above the noise floor. I don't have good snr unfortunately.  The dc level of the signal is not constant but move much more...
Sanger DNA sequencing produces a chromatogram trace which can be visualized with a number of programs, including FinchTV or ChromasLite. The raw data consists of co-ordinates for each of the four DNA bases (A, C, G and T). However, plots are displayed as smoothed peaks, as shown in the image previously. Many programs a...
If I have an image and I find the DCT and then apply hard thresholding on the coefficients and then IDCT then I have attenuated the noise. Can someone please explain in detail or point me to the answer as to why this works? I understand why a filter blocking high frequency will work in denoising (because we assume tha...
Can we take advantage of the fact that high frequency components in the FFT of an image generally correspond to edges, to implement an edge detection algorithm in the fourier domain? I did try multiplying a high pass filter with the FFT of an image. Although the resultant image sort of corresponded to edges, it wasn't...
I am a novice user in image processing and working on a project to determine number of yarns in a fabric image i.e. warps and wefts (Perpendicular and parallel yarns to x-axis). I am trying to detect the yarns by taking the sum of columns and rows for warps and weft respectively but it doesn't seem working because the...
In the article J. Matas, O. Chum, M. Urban, T. Pajdla: Robust Wide Baseline Stereo from Maximally Stable Extremal Regions it is stated that: The set (of extremal regions) is closed under continuous (and thus perspective) transformation of image coordinates, ... I know what perspective transformations (and consequentl...
So the exercise is basically a signal $f(t)$ that is going to modulate the carrier $A\cos(\omega_ct)$ using a modulation index of $m=1$. I have to find $A$ and the power of the modulated signal: $$ f(t)=\cos(\omega_mt)+2\cos(2\omega_mt) $$ The minimum amplitude of $f(t)$ is $-2$. Then $A = 2$. The power of the signal i...
With regard to DSSS and CDMA(DS), what does it mean to spread a signal over a spectrum of frequencies in terms of individual frequencies? Is it about sending the same signal redundantly on multiple frequencies, or is the data stream actually divided and sent as multiple discreet signals over various frequencies? I've r...
The Gabor transform (Short-Time Fourier Transform with Gaussian window) is represented as $U(t,\omega)$, where $t$ is the time, and $\omega$ is the frequency. The author of a monograph (Seismic Inverse Q filtering, pg. 190) is asking for the Gabor spectrum to be transformed from 2D to 1D in the following fashion. Def...
I have a three-dimensional binary image of a collection of discrete, individual voxels ("seeds") contained in a connected 3-dimensional surface ("skin"). (Like a small fruit, with a surface delineated by a one-pixel boundary, that contains seeds.) The binary matrix was derived from a three-dimensional intensity image o...
I am not very familiar with DSP techniques, and I am currently working on my research project to develop the firmware of a DSP processor, using C/Linux. It involves the modulation and demodulation of a dataset using Binary FSK and frequency hopping Tx and Rx algorithms. I am able to modulate the signals and have verifi...
Suppose I have 2 signals from function $f_1(x)$ and $f_2(x)$, respectively, and assume the sampling rate is above Nyquist frequency, so we can restore the underlying functions $f_1(x)$ and $f_2(x)$. But my question is, from the 2 signals, how to tell if the underlying functions $f_2(x)$ is scaled from $f_1(x)$, i.e. to...
Assuming a signal is sampled adequately, what is the minimum size of an FFT window that would allow detecting a specific frequency? Is it necessary to have samples for at least one complete period at that frequency?
I have the model $\ s(x,y)=x^2+y^2, 0 \leq x \leq 1, 0 \leq y \leq 1 $. Instead of observing the model directly I am observing the derivatives of the model + some noise (e): $\ p(x,y)=s_x+e, q(x,y)=s_y+e$ From measurements of p(x,y and q(x,y) I want to estimate s(x). Say I know that s(0,0)=0. According to the gradie...
I have two sensors which are placed around 30 to 40 cm apart , both of them measure the same signal say 'lambda signal'. so there is a delay in the measurement of the signal from the second sensor. During the travel there is some noise added to the signal. What are the procedure that I can use to calculate the time ...
I have some signal that sampled each 1 ns (1e-9 sec) and have, let say, 1e4 points. I need to filter high frequencies from this signal. Let say I need to filter frequencies higher than 10 MHz. I want that for frequencies lower than cutoff frequency signal will be passed unchanged. It means gain of the filter will be 1 ...
I have a recording in pcm format and I want to do some simple analysis. I have some questions about what is normalization. As far I understand it is to get all the amplitudes between a range i.e. [-1, 1] The obvious way to do that is: max_amplitude = max(array_of_amplitudes) for amplitude in array_of_amplitudes: amp...
Thanks to the Realtek 2832U I'm just discovering the world of Software Defined Radio. There are a lot of signals I'm able to pick up with this device, but identifying them is not always easy and cannot be done automatically. When scanning for radio transmissions, I can look at a waterfall plot and see whether a broadc...
Given a set of a $ 3 $ by $ 3 $ matrices $ {H}_{i} $. Each matrix is an Homography matrix. They are used to stabilize Video Stream. Yet some of them are outliers which creates "Jumps" in the "Stabilized Video Stream". I would like to apply Low Pass filter on matrices in a manner which "Low Pass" the homography (Meaning...
In the context of visual tracking and video processing I need to estimate variations of the orientation related to the angular velocity. I am trying to calculate the Jacobian matrix of a quaternion $\vec{q}_k$ respect to its (supposed to be constant) angular velocity $\vec{\omega}$. $$\frac{\partial \vec{q}_k}{\partial...
Using an inexpensive SDR (such as the USRP or Per Vices Phi), I want to process two or more communication protocols at the same time (e.g. WiFi and GSM). Is it possible to Tx (or Rx) both at the same time, like FDMA, or would they need to do some fast switching between modulations and frequencies, like TDMA? I'm not su...
I am working on a project in Python to detect and classify some bird song, and I have found myself in a position where I need to convert a wave file into frequency vs. time data. This hasn't been too much of a problem, but to be able to classify the different syllables into groups, I need to write something that will d...
I am looking for a pitch detection algorithm that uses the least number of samples. I don't care about the processing time. I hope to use this on audio signals. I have tried using the one at the end of http://online.physics.uiuc.edu/courses/phys193/NSF_REU_Reports/2005_reu/Real-Time_Time-Domain_Pitch_Tracking_Using_Wa...
I've tried looking around for information on this, but I'm really out of my league here. I'm a guy who likes to fool around with Python, and I wanted to make a program that would filter an audio file. I'm using Python and NumPy, with the scipy.io.wavfile module for importing and exporting Wave files. I've gotten it mes...
I'm interested in learning how telephones work, so I did a little bit of reading about signal processing. When I came up with the word convolution, I quickly realized the importance of this term. To begin with, I want to know how classic analog telephones worked. The apparent simplicity of their design appeals to me. W...
Can you give me some advices on what could be the best algorithm(s) to use for detection of fingertips/nails in image. First thing that crossed my mind was Viola - Jones. After rethinking I concluded that maybe it would be possible to use just Hugh transformation after applying edge detection. But I'd like some more ad...
I am implementing a Flanger using a fractional delay line. I am modulating the length of the delay line using a sin function. The delay line already uses linear interpolation to compute the delay output for the fractional sample. When the delay length is decreasing (and pitch is increasing), the modulated length soun...
I'm learning about sampling and DSP on my own. I have a hard time to understand how the quantization error results in noise. I think I miss a fundamental understanding but can't tell what it is. So how does the quantization error generate noise?
I recently started playing with the Fourier transform (after spending a few weeks learning about the mathematics behind it). I decided to try hacking together a low-pass filter on the following sound bite: In particular, I took the Fourier transform, zeroed out the highest 1/2 of the frequencies, and then took the ...
I'm reading opencv tutorials and I'm diving into histogram equalization. i have looked in wikipedia, there is a nice example that sums up exactly the problem: original: equalized: but for getting this result i would take a different approach: find the minimum and maximum in the original normalize (remap) everything...
I have some time series datasets which are periodic, and I'd like to pick out the fundamental frequency of the data and take the average over a fundamental period. Questions: 1) Would something simple like taking the FFT of the time series data and looking for the strongest peak frequency do the trick, or are there mor...
In my computer-vision problem, a blob can move either forward or backwards (with a slight chance of moving sideways) along its directional axis. I am seeking a bivariate model because I need to determine whether a blob's new position (with respect to its old one) is "feasible," or "likely." However, the blob can not mo...
I want to extract the envelope of music signals with a sample rate of 44.1kHz. I used the MATLAB command: abs(hilbert(mysignals)), but the resulting signal is similar to the original signal, not the envelope. Does taking the Hilbert transform extract the envelope of an audio signal? Or is my calculation incorrect?
I am looking for a proof for sinusoidal fidelity. In DSP we study a lot about linear systems. Linear systems are homogenous and additive. One more condition it satisifies is that if a signal is a sine or cos wave then the output only changes the phase or amplitude. Why? Why cant the output be a totally different output...
I have long been faced with the confusion regarding entropy and would be obliged if the following are answered in less technical jargon. Following the link Different kinds of entropy raises the following questions Entropy- It is desired that the entropy of the system be maximized. Maximizing entropy means no symbol is...
I have a small question that I am confused. N=16; x=[-pi:2*pi/N:pi-2*pi/N]; y=1.5*cos(x) z=fft(y)/N we will get negtive Fourier coefficients, however when we do it in this way: N=16; x=[0:2*pi/N:2*pi-2*pi/N]; y=1.5*cos(x); z=fft(y)/N we get positive ones. How to explain this? Thank you very much!
I have got the following image: There are curvs on the picture. i would like to find center of the circles containing curves. i tried opencv and hough circle transform but had no results.
I'm trying to calibrate an ultrasonic speaker with the aim of emitting predictable signals, but alas I keep running into trouble, probably due to my lack of DSP-fu. A little background I want to be able to playback sounds as close as possible to a calibrated recording I have. As far as I understand the theory, I need t...
I am trying to make a method in my simple application which is like SetSilence(); Now If I have audio (mono) PCM 16 bit Signed - should I fill all data with 0 values? And for PCM 16 Unsigned with 32768 values?
If a time-domain signal has sharp corners, its frequency spectrum will contain high-frequency components. Truncating the spectrum results in Gibbs' phenomenon. So if you're trying to design an FIR, you really want the target frequency response to be nice and smooth so that windowing the impulse response down to a finit...
I am implementing BFSK frequency hopping communication system on a DSP processor. It was suggested by some of the forum members to use Goertzel algorithm for the demodulation of frequency hopping at specific frequncies. I have tried implementing the goertzel alorithm in C. the code is follows : float goertzel(int numS...
Hello fellow investigators I have two question about optical illusions 1) A photo mosaic is something like this: What are the signal processing principles behind our eye merging the many tiny images into a large coherent one? 2) Are there etiological explanations of optical illusions in general - by this I mean can it...
I have a series of measurements of a signal source, which emits a periodic signal at an unknown interval time of p seconds. Detecting the signal is not easy so I am missing quite a few signals in the data. Data is also noisy so there are errors in the data of up to +/- 10% per signal. Example measurements x might look ...
I am working on OpenCV2.4.2 C++ interface for face recognition functionality. The face recognition feature seems to be working fine on standard databases. (AT&T, Face Recognition Data, University of Essex, UK) I am more interested in negative testing. Here is what I did - Identify clear front face images from the Inte...
I've been reading some papers in signal proccesing and I'm very confused about the issue in the title of my question. Consider a continuous function of time $t$, $f(t)$, that I sample at uneven times $t_k$, where $k=1,2,...,N$. To me, it makes sense that the sampled function is: $$f_s(t)=\sum_{k=1}^N\delta_{t,t_k}f(t),...
I'm receiving PCM samples through stream which is one channel, 8000 samples per second. I figured out way of finding out loudness by calculating 'force' of signal by adding abs value of every next 1000 samples and then dividing by samples count which is 1000. It works somewhat good but sometimes my calculations are goi...
I've got some data sampled at 32 Hz. The only interesting stuff is happening between about 0-8 Hz and I've got to do real time batch processing and detect events with a delay of no more than a second. So that means I can only perform FFTs with 32 bins or less, to remedy the lack of resolution I'm using a Chirp $\mathca...
In the documentation for the [XAudio2 filter parameters] it mentions Filter radian frequency calculated as (2*sin(pi*(desired filter cutoff frequency)/sampleRate)) Where does this formual come from? What is "Filter radian frequency" and "filter cutoff frequency" and what is the difference. Can someone translate this t...
I am relatively new to time delay estimation. I am working on a project where I am estimating the time-delay from the phase of the cross-correlation function (CCF), since it is more reliable in compensation for the inaccuracy than merely using the absolute value of the CCF in estimating the time delay. I am also lookin...
I'm not quite sure of the mathematical terminology here but... Is there a variant (or post-processing) of a Discrete Fourier Transform that separates the shape of a signal from any phase shift applied to the whole signal? For example this image: and this one: should differ in at most 2 coefficients (one for each dime...
I'm down sampling voice audio by first performing an FFT, then only taking the parts of the result that I need, and then performing an inverse FFT. However, it's only working properly when I'm using frequencies that are both power of two, say down-sampling from 32768 to 8192. I perform an FFT on the 32k data, discard t...
I am trying to figure out the details on how to implement the structure tensor in Matlab and need some advice! For an image $\ I(x,y) $ the structure tensor S is given by: $$ S=\begin{pmatrix} W \ast I_x^2 & W \ast (I_xI_y)\\ W \ast (I_xI_y) & W \ast I_y^2 \end{pmatrix}$$ where W is a smoothing kernel, $\ \ast $ deno...
I'm essentially a math student working on algorithms for a chemistry problem involving rotational spectra, and I keep coming across references to Wang transformations but have been unable to track down what the Wang basis is. You can assume I have the analysis background to understand what a basis of a function space i...
I have an application with two separated GPS receivers giving live positions and I'm deriving a heading/displacement from the vector between them. Each set of position measurements is noisy and has dropouts but I do have a quality estimate. So far it's classic Kalman filter. The relation between the two antennas isn't...
I have convolved a random signal with a a Gaussian and added noise (Poisson noise in this case) to generate a noisy signal. Now I would like to deconvolve this noisy signal to extract the original signal using the same Gaussian. The problem is that I need a code which does the job of deconvolution in 1D. (I have alread...
Can I get help on how to make frequency axis going from negative frequency to positive frequency, (in Hertz), that will be the x-axis in an FFT result, but given either an even length FFT, or odd length FFT. I am having some troubles making it in MATLAB. (Assume you know the sampling frequency f_s).
How can I detect a quadrilateral signboard from an image captured using a mobile phone? How can i detect shapes like rectangle? rounded rectangle (rounded corners instead of shape corners)? I am using opencv.wrapper but i am new to it. Thanks. Here is the sample: alt text http://www.freeimagehosting.net/uploads/b03442...
Paper describes that a 2D signal(say video) can be projected to 1D for time series analysis by using a transformation function fx=Ax+constant and fy=Ay+constant where Ax , Ay represents the (x,y) coordinates of a single trajectory A. Then,these fx vs t can be plotted and various analysis can be performed. Going by thei...
I'm writing a program to emulate a vocoder and it obviously works, but any speech that comes through is very difficult to understand. I'm going to explain exactly how my implementation works and then maybe somebody with experience with vocoders or DSP can help me. 128 samples of the voice signal go through a Fast Fouri...
Given something like 103 data points ($N=103$), a DFT will give back 103 frequency values. Then to do something like filtering the high frequencies involves setting the high frequency values from the DFT to zero, and doing the inverse DFT to get back 103 data points that represent the original signal without the high f...