Adaptive Filter | ADPCM | ADSP | ADSP-2181 | Aliasing | AMR | Anti-Aliasing | ARMA | Autocorrelation | AutoCovariance | Beamforming | Bessel | Blackfin | Butterworth | C6713 | CCS | Chebyshev | CIC Filter | Circular Convolution | Code Composer Studio | Comb Filter | Compression | Convolution | Cross Correlation | DCT | Decimation | Deconvolution | Demodulation | DM642 | DSP Boards | DSP/BIOS | DTMF | Echo Cancellation | Equalization | Equalizer | ETSI | EZLITE (Ez-kit Lite) | FFT | FFTW | FIR Filter | Fixed Point | FSK | G.711 | G.723 | G.729 | Gaussian Noise | Goertzel | GPIO | Hilbert Transform | IFFT | IIR Filter | Interpolation | Invariance | JTAG | Kalman | Laplace Transform | Levinson | LPC | McBSP | MIPS | Modulation | MPEG | Multirate | Notch Filter | Nyquist | OFDM | Oversampling | Pink Noise | Pitch | PLL | Polyphase | QAM | QDMA | Quantization | Quantizer | Radar | Random Noise | Reed Solomon | Remez | Resampling | RTDX | Sampling | Sharc | TI C6711 | Undersampling | Viterbi | Wavelets | White Noise | Wiener Filter | Windowing | XDS510PP | Z Transform

Levinson-Durbin Vs Generalezed Levinson-Durbin algorithm

Good morning experties, I'm using the generalized Levinson-Durbin algrithm to solve the system R*hinv=q. Where q(l)=h(N-l), N is the length of the filter to be inverted and hinv is its inverse. R is a symmetric Toeplitz matrix. I used the MATLAB code from Bob Cain in http://www.musicdsp.org/sho...

faster algorithm than levinson-durbin recursion

Hello I am looking for a _stable_ algorithm that performs better in terms of speed than the levinson-durbin recursive algorithm. Any suggestions? Thank you. ...

Levinson algo. wtih complex coef. required

Hi, I am desperatly looking for a generalized Levinson algorithm that can cope with complex autocorrelation values and giving complex filter coefficients... I would like to implement it in C, and i have only found the classical LPC analysis but it can not deal with complex coefficients.....

calculating gain with levinson-durbin.

Hi, I'm using the levinson-durbin algorithm to calculate an all-pole filter. This all-pole filter is used to generate background noise to overcome silence when the telephone line is muted. I believe a description of the all-pole filter is : H(z) = G / (a0+a1*z-1 + a2*z-2....). Calcula...

Re: Difference between LPC & LPC vocoder

> I'm doing a speech recognition project. Can LPC vocoder be use for > speech rocognition? > Can anyone help me with the differences between LPC and LPC vocoder? > I'm looking at the code which i found in the net and it seems that LPC > method were used in LPC vocoder and it is dealing with pitc...

Re: Need help about stability of Levinson-Durbin algorithm in Matlab

kenshen wrote: > Hi, expertises, > > I am programming levinson-durbin algorithm in Matlab and desperately find > that it comes out with unstable results, especially for the multivariate > case. LD recursion is an algorithm to find solutions to linear equations with special structure....

Levinson-Durbin algorithm

Good morning experties, I'm using Levinson-durbin algorithm to invert room impulse responses. I used the MAtLAB code writen by Bob Cain in the following page http://www.musicdsp.org/showone.php?id=188 The code works well, but I can't understand why it works! The following equation should be ...

Help in understanding Levinson Durbin

Hello friends, I am an undergraduate student from India and have taken up Advanced DSP as my course. I am referring to the book by Proakis and Manolakis. I have thoroughly read upto filters. However, I find the Linear Prediction chapter a bit heavy to understand. I find the derivations dif...

autocorrelation with 0 lag for LPC in speex library

Hi, I saw in the sources for the speex library (http://www.speex.org/) that they add some value to the first autocorrelation result. The results are used to produce a IIR with a levinson durbin recursion. float d; int i; while (lag--) { for (i = lag, d = 0; i < n...

Re: Inverse FIR computation

Number 6 wrote: > Why are you using FFTs - surely a simple recursion will perform the inverse. Which one might that be? The one that finds the LMS inverse (arbitrary length, no time domain aliasing) in O(N^2) involves solving a Toeplitz matrix within the framework of the Levinson-Dur...

Input parameters of Levinson-Durbin algorithm

Hello experties, I'm using Durbin-Levinson algorithm to calculate the inverse impulse response of a given impulse response h in the time domaine. The algorithm requires three input parameters: - Impulse response h - The delay time of the inverse impulse response of h - The length of the inverse...

Order selection of AR process

Hi Can anyone explain how one choses the order of the AR process when trying to estimate it's coefficient (Via levinson durbin algorithm). Is this a try and error? Thanks ...

Re: Fitting AR model to stohastic signal

anglomont@yahoo.com (TI) wrote in message news: ... > Hi > I have a sample of a Stohastic signal with very slowly variable mean, > and would like to find AR model for it perhaps using Burg parameter > estimation model (explained in book by Kay). Kay has written several books. I assume yo...

Re: LPC Coefficients

Hi, I am using an autocorrelatoin method using the Levinson recursion. The algorithm I am using is adapted to java from some non-java code I found in a book. Do you know anywhere that I might find another basic LPC algorithm to test my own. Thanks, Tom wrote in message news:1107...

inversion of a matrix

Hi I need to inverse a rather large symmetric matrix. Using 'inv' takes much time and I need to speed this up. What I actually have is the the follwing: A'*inv(R)*A Where A is convmtx(a,n) where a is a vector of length less than 'n' and R is A*A' .(or actually the pre-whitened, adde...

Levinson Durbin / Reflection coefficients

Hi ! I want to learn more about different representations of the LP coefficients (ie reflection coefficients, spectral frequency line, and so on). How to transform between those representations and what they actually mean. I'm searching for an easy representation which is robust against ...

Vector Quantization: Problem with designing codebook

Hi I am trying to do vector quantization for noise-free speech segments. The algorithm I have programmed works like this: 1) Update 240-point frame F with pre-emphasized block x of 10 speech samples. 2) Update estimate of power spectrum P based on updated frame 3) Calculate vector B=...

Re: Computational complexity of FLAC?

On Oct 24, 5:26 pm, Chris Eilbeck wrote: > How difficult do you guys think it would be to recode FLAC to run on a > microcontroller like an AVR? I'm tinkering with a small audio recorder and > figured I could get more recording time without having a re-design for a > second flash device....

Re: Article

Randy Yates wrote: > Sorry - you might actually have to get up from > your chair and drive to a university library. Assuming one realistically has that option, then what? Living in Palo Alto I used to be able to browse Stanford's stacks (mid '70s) but I doubt that I could now get in the b...

Re: Inverse impulse response

> stef wrote: > > > Hello experties, > > > > Any body have an idea how to calculate the inverse impulse response in the > > time domaine of an impuse response h using MATLAB ? > > I'll be happy if you help me. > > Thank you in advance > > You probably want something like this: > ...

Levinson durban recursion

Hi, I am trying to get forward/backward predictor coefficients from the ladder relection coefficients using the following recursion. (From haykins book) a_m(n) = [a_{m-1}(n) 0] + k_{f, m} [0 c_{m-1}(n-1)] --------- (1) c_m(n) = [0 c_{m-1}(n-1) + k_{b, m} [a_{m-1}(n) 0] --------- (2) wher...

Demixing two audio Signals in the time domaine

Hello experties, I have the following System in the time domaine that I want to solve with the unknown x1 and x2: y1 = h11*x1 + h21*x2 (1) 0 = h12*x1 + h22*x2 (2) with: h11, h21, h12, h22 are the room impulse responses. I found the following solution: x1 = [(h11-h21(h22^-1)h12)^-1]*y1 x2...

Re: Inverse matrix

> > "stef" wrote in message > news:6q-dnSbKc7TRSbbZRVn-qw@giganews.com... > > Hello, i'm new to this group > > I'm trying to demix two audio signal. > > My audio signals are y1 and y2 (Y=[y1;y2]) > > The impulse resonses: H11, H12, H21, H22. > > The matrix form of the system is: y=H...

Re: Regarding the Linear prediction

On Sep 25, 7:44 am, "neelufar.2005" wrote: > Hi, > > I am currently trying to implement the linear prediction analysis in c to > decorrelate the speech signal. I have implemented the livenson durbin code > in C. Is there any method to cross check that the linear filter coefficient > a...

spectral factorization using cepstral deconvolution

Hi there, I want to find the minimum phase spectral factor of a real autocorrelation sequence using cepstral deconvolution. However I run into problems when the spectrum has a null. Is there a way to get around this problem, or is the cepstral deconvolution doomed to fail in case of a null? Could...

Re: Levinson-Durbin Adaptive Filter....How?

OK, I'm still being a bit thick! What would anyone recommend as a simple test signal that I could use to put through the L-D block in simulink and what should I expect as the output? If I have only one signal can I put that through the L-D? I know that the LD solves for a number of equatio...

Re: solving systems of linear equations

galreuven@gmail.com wrote: > Hi folks, > I'm looking for code which solves systems of linear equations for TI's > C67 processors. > I have the "numerical recipes" code but I rather use an optmized code. What is "optimized"?? One thing that I recall from my Matrix Numerical Methods cour...

Re: Linear predictive coefficient computation

RAMU wrote: > To find the Linear predictive coefficents all are using auto > correlation with levinson durbin equation. I want to know is there any > alternative method availble that still reduces the computational > complexity to find the LPC coefficients. If you are willing to trade f...

QRD-LSL

Hi all, I am having problems regarding weight extraction in QRD-LSL algorithm i.e., I want to know the unknown filter coefficients of the QRD-LSL adaptive filter. I have implemented QRD-LSL algorithm, given on page 666 of symon hakin book on adaptive filtering. On page 686 he mentions weight ex...

Re: Filter design, minimal group delay, maximal variance

crowlogic skrev: > Hi all, > > I want to design a filter (invertable if possible) that has the minimum > amount of group delay I may be wrong now, but I *think* "minimum group delay" means that the filter has to be "minimum phase", which in turn means it is invertible. > and captu...

Noise Subtraction

My boss showed me a feature on some commercial software he wants me to incorporate into our system. Basically, you show the software a patch of noise (samples w/o the signal). Later, the program lets you subtracts that noise from a signal contaminated by that same noise. IOW, subtract the fa...

Re: Professional Exams for Digital Signal Processing!?

On 30 Aug, 14:20, Scott Seidman wrote: > Rune Allnor wrote in news:1188451007.296145.109460 > @r29g2000hsg.googlegroups.com: > > > Whenever I encountered students who knew DSP from > > matlab, I had a hard time de-teaching that stuff out of them > > so I could teach practical DS...

Re: FIR Lattice Filter Reflection Coefficients in Terms of Direct Coefficients?

You can find the relevant information in the following textbooks: (1) "Advanced Digital Signal Processing", J.G.Proakis, C.M.Rader, F.Ling, C.L.Nikias, ISBN 0-02-396841-9, Chapter 4 (Linear Prediction and Optimum Linear Filters), pp 219-221 (2) "Optimum Signal Processing, An Introduction, ...

Re: De-mixing two audio streams using Room Impulse Response (Matlab)

jeremyscerri@gmail.com wrote: > Hi, new to this group...I am using Matlab to SIMULATE the de-mixing of > two audio streams. What I did is as follows: > > 1. found two clean audio sources on the internet S1, S2. (4 seconds at > 11kHz, 16bit) > 2. found four room/hall impulse response...

Re: AR coefficients for complex numbers, etc.

sharathreddy@hotmail.com (Sharath) wrote in message news: ... > Rune- > > Thank you very much for your assistance. Now I have it working for > complex numbers too. The code was indeed helpful. > > I would appreciate it if you/anyone could give me some pointers on > another issue...

Re: Parametric Modelling/Filtering

On 20 Feb, 16:06, "kirLIan" wrote: > Hi There > > I am looking at parametric model structures in DSP and Control systems. and > am interested to know if you guys have experience of using the ones outlined > (apart from the obvious ones) ... > Also are there different names for th...

Re: Block Processing Issue

> What type of signal are you filtering? A noise degraded speech signal x=s+n where s is clean speech and n is colored, gaussian noise. > How important is continuity? I don't know to be honest. Maybe you can tell me why you ask that particular question? > How/why are the coefficie...

Re: Why minimising in the mean-error sense.

Bob Cain wrote in message news: ... > Matt Timmermans wrote: > > > > - Orthogonal transforms that preserve energy, like the Fourier transform, > > also preserve squared error. So if your goal is to minimize squared error, > > you can do it in the time or frequency domain and it w...

Re: Inverse of a simple non-linear-phase FIR filter

Shafik wrote: > Hello everyone, > > Can anyone tell me if there is a known way to find the inverse of a > given FIR filter? Generalized Levinson-Durbin in Matlab: function inv=invimplms(den,n,d) %syntax inv=invimplms(den,n,d) % den - denominator FIR (to be inverted) % n ...

Re: OT: Where did you get your start?

Bob Cain wrote in news:dq9d4m02rmr@enews1.newsguy.com: > > > Chris.Gammell wrote: > > Hey all, > > > > I was just wondering how everybody got their start in DSP. I am > > personally geting mine by working with FPGAs and Simulink tools. > > Also, I am working my way into han...