distinguish between fax and speech

Started by VSP in comp.dsp13 years ago 5 replies
LPC

hi ppl, Please help me with this problem I got stuck at during my DSP project: I am getting a transmission along a channel and I have to tell...

hi ppl, Please help me with this problem I got stuck at during my DSP project: I am getting a transmission along a channel and I have to tell whether it is speech (LPC coded say) or FAX transmission. The sender of the transmission is anonymous so, I have no info from him. Are there any unique or invariant properties in a fax transmission that can be used for this purpose? Thanks in advance. ...


LP residual

Started by Anonymous in comp.dsp13 years ago 4 replies

Hi, I would like to compute LP residual of a speech signal s(n). Firstly, s(n) is divided into frames of 256 samples each. Secondly,...

Hi, I would like to compute LP residual of a speech signal s(n). Firstly, s(n) is divided into frames of 256 samples each. Secondly, linear prediction filter (order 12) is estimated for each frame using matlab's LPC( ). However, having concatenated residual from each frame, I found there are a number of spikes in the residual, the space is about mulitples of 256. Should I overlap the frame...


levinson durbin

Started by 22139 in comp.dsp4 years ago 1 reply

I was trying to implement Linear Predictive Coding in Matlab. The program I wrote for Levinson Algorithm gives same coefficients as Matlab's...

I was trying to implement Linear Predictive Coding in Matlab. The program I wrote for Levinson Algorithm gives same coefficients as Matlab's levinson function. But i found out that Matlab's lpc function gives different set of coefficients. Why is it so? _____________________________ Posted through www.DSPRelated.com


Levinson-Durbin Recursion revisited

Started by panu in comp.dsp11 years ago 4 replies

I know this subject has been discussed in the past on this site in relation to inverse filter design, and this is also the subject of this post....

I know this subject has been discussed in the past on this site in relation to inverse filter design, and this is also the subject of this post. My specific application is acoustics. If you have experience with the Levinson-Durbin algorithm in C/C++ I could use your help. I have been using the code found at http://kbs.cs.tu-berlin.de/~jutta/gsm/lpc.html to try to implement the algorithm in C t...


Verifying my algorithms

Started by muka...@gmail.com in comp.dsp13 years ago 2 replies

hi all, I am working on voice verification using MATLAB and have devloped few algorithms for measuring energy and pitch etc. Now I want to check...

hi all, I am working on voice verification using MATLAB and have devloped few algorithms for measuring energy and pitch etc. Now I want to check the validity of these algos so can anyone pls guide me. Wat I am lookin for specifically is if I can get a voice sample with all its paramets like energy, pitch LPC, formants etc calculated, so that I can use the sample as an input to my system an...


problem when calculating roots based on LPC coefficients

Started by John in comp.dsp13 years ago 3 replies
LPC

Hi If I have a polynomial A(z)=1+a1*z^(-1)+a2*z^(-2)+.....+a10*z^(-10) and I solve A(z)=0 and write the solution as a vector [r1,r2,....,r10]...

Hi If I have a polynomial A(z)=1+a1*z^(-1)+a2*z^(-2)+.....+a10*z^(-10) and I solve A(z)=0 and write the solution as a vector [r1,r2,....,r10] then I can say that the action of finding the roots is a one-to-many transformation from the coefficient vector [a1,a2,.....,a10] to the root vector [r1,r2,...,r10].... Is it possible somehow to ensure that the above transformation of a given ...


autocorrelation with 0 lag for LPC in speex library

Started by Anton in comp.dsp13 years ago

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...

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; i++) d += x[i] * x[i-lag]; ac[lag] = d; } ac[0] += 10; ^^^^^^^^^^^^


fastest way of calculating the roots of a polynomium??

Started by John in comp.dsp13 years ago 13 replies
LPC

Hello I am working on a speech enhancement algorithm. Part of my algorithm has to do with 10th order LPC-analysis that provides me with the...

Hello I am working on a speech enhancement algorithm. Part of my algorithm has to do with 10th order LPC-analysis that provides me with the real coefficients [a1,a2,.....,a10] for a 10th order polynomium A(z)=1+a1*(z^-1)+a2*(z^-2)+......+a10*(z^-10) Is there a way to do fast calculation of all the roots of A(z) such that they are written on the complex form x+j*y ?? Thank you in a...


ofdm coding with Rayleigh channel

Started by jorgesan_garcia in comp.dsp12 years ago 1 reply

Hello all, I am trying to simulate an algorithm for voice + data in order to send throuhg a Rayleigh Fading channel, by using lpc-10 (2.4...

Hello all, I am trying to simulate an algorithm for voice + data in order to send throuhg a Rayleigh Fading channel, by using lpc-10 (2.4 kbps) vocoder and an ofdm channel (with 36 subcarriers) and I realized some troubles related to SNR and BER which are not good (i.e I expect 10e-6 for 14 db or more) and the output is bad. I was wondering if there is a trouble related to synchro. due to of...


I need Some help regarding speech to Text Conversion task

Started by Anonymous in comp.dsp14 years ago 2 replies

Hello all, I am implementing Speech to Text Conversion task using MATLAB. I implemented with LPC and Kohonnen's neural network...

Hello all, I am implementing Speech to Text Conversion task using MATLAB. I implemented with LPC and Kohonnen's neural network to do Speaker Identification task . Now i am able to idenify the speaker. With the same method i am able to identify the Vowel sounds also. Further i am not able to proceed to implement Consonent Conversion. I do not know about Phoneme also. ...


Detect wide band noise

Started by Vladimir Vassilevsky in comp.dsp10 years ago 7 replies

Hello All, I need to detect the bursts of the wide band noise in the sampled signal. The detector should ignore the narrow band pulses or the...

Hello All, I need to detect the bursts of the wide band noise in the sampled signal. The detector should ignore the narrow band pulses or the stationary signals. This is intended for blanking of the signal processing in the instrument. A solution could be based on the FFT, LPC or some sort of adaptive filter. However I am looking for less demanding method, which can be very approxi...


wavelet based analysis for speech processing!

Started by santosh nath in comp.dsp15 years ago 2 replies

Speech is short stationary signals. Can anybody use "wavelet transform" for speech/voice or audio processing? Do they get any extra benefit out...

Speech is short stationary signals. Can anybody use "wavelet transform" for speech/voice or audio processing? Do they get any extra benefit out of it e.g using variable window size - since speech is sparse in nature also? How about complexity of wavelet based analysis compared to conventional LPC analysis - the question just came out of head. Regards, Santosh


whitening filter

Started by Anonymous in comp.dsp14 years ago 3 replies

Hi, I am trying to implement a high frequency regeneration system where I take a narrowband speech signal and reconstruct high frequecies...

Hi, I am trying to implement a high frequency regeneration system where I take a narrowband speech signal and reconstruct high frequecies to obtain a wideband signal. I have done LP analysis on the NB speech and have LPC's and the residue. I now want to pass the residue (after upsampling)through a whitening filter to obtain a residue which has a flat spectrum. I just cant figure out how ...


Looking for an FEC coding scheme with variable coding bits

Started by dudelmann in comp.dsp10 years ago 5 replies
LPC

Background: I am currently writing a lossy LPC source coder for real time transmission over a wireless channel. This coder produces a stream with...

Background: I am currently writing a lossy LPC source coder for real time transmission over a wireless channel. This coder produces a stream with a fixed AVERAGE bit rate (that means it varies around that rate). To keep the MOMENTARY bit rate within certain limits, I reduce quality when the bit rate gets too high and I INSERT USELESS BITS, WHEN IT GETS TOO LOW (hint: this was the important part :...


adaptive complex coefficient filter

Started by elliot in comp.dsp14 years ago 6 replies

Hello, I am working on some fetal heart rate algorithm. To verify it, I decide to do the reverse thing. I use a LPC model(similar to the model...

Hello, I am working on some fetal heart rate algorithm. To verify it, I decide to do the reverse thing. I use a LPC model(similar to the model of voiced speech, using some period impulse to excite an all pole filter) the input of this model are: 1. the original signal 2. fetal heart rate calculated out from this original signal the output is my reconstructed signal, I want to use this ...


LFCC (Linear Frequency Cepstral Coefficients)

Started by Biruntha Gnaneswaran in comp.dsp4 years ago

It is used to extract the features in audio signal. It has steps Signal Normalization, Compute FFT power spectrum, Compute autoregressive model...

It is used to extract the features in audio signal. It has steps Signal Normalization, Compute FFT power spectrum, Compute autoregressive model analysis, Convert LPC to cepstra , Apply DCT and Convert to cepstra, Apply lifter to matrix of cepstra, Calculate the deltas of a output mel sequence . Can anyone briefly explain about LFCC and it's steps? Thankyou


Audio - Noise removal

Started by hurry in comp.dsp12 years ago 4 replies

Hi I am using Kalman filter in building a noise removal tool for speech signal. The questions I am faced with are: a) To find right...

Hi I am using Kalman filter in building a noise removal tool for speech signal. The questions I am faced with are: a) To find right estimate of Signal variance b) To find right estimate of Noise variance I do a rough estimate of signal variance from LPC analysis of the noisy i/p and amd updating noise variance during silence period. But, with this technique a short lived noise durin...


Determining the best filter order for linear predictive coding

Started by bennjo in comp.dsp5 years ago 1 reply

Hi All, I am wondering if there is an established method for choosing the best filter order to use when performing linear predictive coding (such...

Hi All, I am wondering if there is an established method for choosing the best filter order to use when performing linear predictive coding (such as that used in audio file formats like FLAC). My current approach is: 1. Take a chunk of signal 2. Window the signal with a 0.5 tukey window 3. Get the auto-correlation coefficients 4. Calculate the LPC coefficients using the auto-correlation ...


Fixed-point implementation of levinson durbin algorithm

Started by John McDermick in comp.dsp6 years ago 17 replies

Hi, Any links to a fixed point implementation of the Levinson Durbin algorithm ...something like this: void levdur(pLpcCoeff, pAutoCorr,...

Hi, Any links to a fixed point implementation of the Levinson Durbin algorithm ...something like this: void levdur(pLpcCoeff, pAutoCorr, pReflec, nOrder) where pLpcCoeff is a pointer to the lpc coefficients where pAutoCorr is a pointer to the autocorrelation coefficients (range -40 to 40) where pReflec is a pointer to the reflection coefficients where nOrder is 10 The autocorrelat...