b=intfilt(r,n,alpha) r = factor of sampling =4 b has a lengh of (n+1)*r for n even and a lengh of (n+1)*r -1 for n odd 1___ ...

b=intfilt(r,n,alpha) r = factor of sampling =4 b has a lengh of (n+1)*r for n even and a lengh of (n+1)*r -1 for n odd 1___ if we have the signal of lengh 1024 n=..... how much in this case???? 2____ in some books alpha =1 but in other books aplpha =0.5 what are the different????????

I seek a program that will take two sound files (A and B) of approximately equal duration and produce a third sound that is a blend of A and B...

I seek a program that will take two sound files (A and B) of approximately equal duration and produce a third sound that is a blend of A and B throughout its duration. i.e. I want to be able to produce the middle sound shown below. AA AA AA AA AA ab ab ab ab ab

Hi all, Do you have a general structure or a documentation how to design and compute the filter coefficients for a Farrow interpolation...

Hi all, Do you have a general structure or a documentation how to design and compute the filter coefficients for a Farrow interpolation resampling filter. It's for a Gardner synchronisation scheme... Thanks

Hi fellows, I'm still on my way with the same project, but now, it seems that I'm stuck and need a hint, any pointing into the right...

Hi fellows, I'm still on my way with the same project, but now, it seems that I'm stuck and need a hint, any pointing into the right direction... seems like a brain deadlock which needs an escape... Background: A signal is sampled at 48kS/s. It carries information which is related to a position vector. Since speed is varying, position and time are not proportional, though they are r...

I'm attempting to interpolate a bandlimited sampled signal s(nT) to some arbitrary point s(nT+u) 0

I'm attempting to interpolate a bandlimited sampled signal s(nT) to some arbitrary point s(nT+u) 0

hi, ladies and gentlemen, can you do me a favor ? I am designing an audio encoder and decoder on ADI blackfin 533 EZ-KIT. now i have...

hi, ladies and gentlemen, can you do me a favor ? I am designing an audio encoder and decoder on ADI blackfin 533 EZ-KIT. now i have one problem: can I set the sample rate of ADC to 48KHz and the interpolation mode of DAC to 4*96KHz ? thanks for your help.

How are filters that have their cutoff and e.g. Q variable (such as in a typical parametric equalizer) done? Does this mean that the filter...

How are filters that have their cutoff and e.g. Q variable (such as in a typical parametric equalizer) done? Does this mean that the filter coefficients are recalculated (i.e. the filter is redesigned) every time the parameters are changed? What kind of other design considerations are involved (such as computational efficiency)? What about "smooth parameter changes", is it interpolation (and...

Hi, Has anyone got a sample code in matlab (or any other lang.) for "polar format algorithm" (range and azimuthal interpolation), for SAR...

Hi, Has anyone got a sample code in matlab (or any other lang.) for "polar format algorithm" (range and azimuthal interpolation), for SAR applications ? Or could direct me to a paper which describes the algorithm? All I can find on the net is power point presentations. Thanks

Hi, I have two signals, one sampled at 1ms (ricker), the other anything greater, 4 or 5 or 10ms. I need to do a convolution, that is...

Hi, I have two signals, one sampled at 1ms (ricker), the other anything greater, 4 or 5 or 10ms. I need to do a convolution, that is a multiplication in freq. domain. My question is, given that the second signal is a series of spikes, do I need to use some sort of interpolation (e.g. sinc) or is it enough to add samples (zeros) between the spikes, depending on the ratio of the signals sampli...

Please can someone point me in the right direction? I'm hoping that this is a solved problem for somebody out there. I need a function that...

Please can someone point me in the right direction? I'm hoping that this is a solved problem for somebody out there. I need a function that will limit Acceleration and Velocity, generating at each sample time a new position that meets the V A constraints. The input position is kept within bounds. A vital design requirement is low signal latency, ideally I want a current answer for each samp...

hi everyone, million thanks in advance...i would like to know about EMD which is part one of hilbert huang transform. the procedure of EMD is...

hi everyone, million thanks in advance...i would like to know about EMD which is part one of hilbert huang transform. the procedure of EMD is : 1)For a signal X(t), let m1 be the mean of its upper and lower envelopes as determined from a cubic-spline interpolation of local maxima and minima. The locality is determined by an arbitrary parameter; the calculation time and the effectiveness of the ...

Hi all, Please take a look at this curve... http://img211.imageshack.us/img211/375/qqqrq3.jpg What is the best interpolation scheme for...

Hi all, Please take a look at this curve... http://img211.imageshack.us/img211/375/qqqrq3.jpg What is the best interpolation scheme for it? The purpose is to sample as few number of points as possible....

I have a 6-second sample of a single piano note from which I need to extract the frequencies of the components that are near, but not quite,...

I have a 6-second sample of a single piano note from which I need to extract the frequencies of the components that are near, but not quite, harmonics of the fundamental. I have been doing this by finding appropriately-spaced local maxima in the FFT power spectrum, and then using quadratic interpolation (using the amplitude of the immediate bin neighbors together with the amplitude of the pe...

Hello, I am trying to implement a micro DSP with only 8 instruction set. Can you help me with what should be the bare minimum set ...

Hello, I am trying to implement a micro DSP with only 8 instruction set. Can you help me with what should be the bare minimum set required which will cater to most of the applications? FIR, IIR, interpolation, decimation. Forget about FFT for time being, as I am implementing a simple MAC. Maybe even adaptive filtering could be skipped to begin with. ...

Hi Could anyone help me on how to use lpc (linear prediction coeff) for interpolation ? I have a vector where some elements(samples) are...

Hi Could anyone help me on how to use lpc (linear prediction coeff) for interpolation ? I have a vector where some elements(samples) are way off what is reasonable and wondered if I could use lpc to estimate replacements for those but I don't know how to incorporate the neighbouring (those before and after)samples to estimate the new ones. Matlab has a function called 'lpc' but I am a...

Recently I run into the problem with a basic task: design of a polyphase set of filters for the interpolation of a signal. The input is the...

Recently I run into the problem with a basic task: design of a polyphase set of filters for the interpolation of a signal. The input is the sampled signal; the output should be the interpolated values spaced at 1/10 of the sample. So, I designed the LPF at 10 x sample rate by the Parks-McClellan algorithm, and then decimated it into 10 subsets of the coefficients. But, although the...

Hi, I'm performing upsampling for audio to avoid aliasing. This is what I do: 1) Take source buffer and using cubic interpolation convert it...

Hi, I'm performing upsampling for audio to avoid aliasing. This is what I do: 1) Take source buffer and using cubic interpolation convert it into a buffer let's say 16x larger. 2) Perform lowpass on the temporary buffer with cutoff at X / 16, where X is just some kind of factor compensating steepness of the filter. 3) Here comes the blackbox - some effect. But in this testing case it simply do...

Hello All Due to a lack of memory space , I'm trying to compute my Look-up table at startup. My target is a 16bits fixed point DSP (TI...

Hello All Due to a lack of memory space , I'm trying to compute my Look-up table at startup. My target is a 16bits fixed point DSP (TI C54x). I need 16 bits sinus table I first try Taylor interpolation, but did not succeed. (May be this is best method, but if so, i should use 32 bits precision) I already implement floating point operation, and will try in this way. What other a...

I have done some experiments with a Gaussian window applied to real signals. The signals were pure sinusoids of a fixed frequency...

I have done some experiments with a Gaussian window applied to real signals. The signals were pure sinusoids of a fixed frequency and phase. I used this window: w(k) = e^( -(1/2) * (k - (N-1)/2)^2 / (a*(N-1)/2)^2 ) with N=32, k = 0, ..., N-1 and a = 0.265391249974779 I multiplied my signal with the window and did the DFT. Then I calculated the magnitudes around the peak. I took the...

Hello, I am trying to interpolate values of a Gaussian window after discrete fourier transform. I?ve been searching around but figure that I am...

Hello, I am trying to interpolate values of a Gaussian window after discrete fourier transform. I?ve been searching around but figure that I am still missing some basic knowledge about gaussian function. Any suggestions would be very appreciated. I?ve found some information on this page: http://www.dsprelated.com/freebooks/sasp/Gaussian_Window_Transform.html There is a formula for in