>>>>> "marian" == marian mounir@gmail com <marian.mounir@gmail.com> writes:
marian> I think that i has to down the sampling frquency to 8000 using resample
marian> function.
I don't see why you have to down sample. LPC doesn't really care.
marian> i am thinking now that will write the following code
marian> while j<= N_frames no of frames
marian> A(j,1:11)=lpc(x(1:180,j),10);
marian> j=j+1;
marian> end;
marian> to pass the sound signal as frames to the LPC function the previous
marian> loop i found on this newsgroup but i wonder do i have to use windowing
marian> before as i read in digital processing of speech signals of Rabiner or
marian> what ?
Typically some kind of windowing is down before computing the LPC
coefficients. This tends to the the LPCs more consistent, but biased.
marian> and what is the method to find the number of frames and choosing the
marian> order of the LPC
The frame length is the important value. This depends on your
sampling rate and on how fast you expect your LPC coefficients might
vary over time. The order also depends on what you think the real
LPC order is. There are ways of estimating the order, but I don't
know how well they work in practice.
marian> What i had understand was that the number of coeff. = the order that i
marian> use is that write or not ?
Yes.
marian> i don't understand what is the AR filter, what does AR stands for i
marian> know that AR stands for automatic recognition.
AR here means Auto Regressive. Basically if you estimate the LPC
coefficients a[k], you get an AR model of the signal and the AR filter
has the transform, H(z):
1/H(z) = sum a[k]*z^(-k)
marian> There is somthing also i didn't understand in the text book of Rabineer
marian> i found that he feed the LPC with voiced and unvoiced signals but he
marian> said then that he use impulse signals as an input does he mean that he
marian> pass the sampled sound signal.
marian> does the unvoiced signal is the Gaussian noise.
I think Rabiner explains this. The speech is modeled as two sources.
One is an impulse train that passes through a filter to produce
speech. This is voiced. The other is Gaussian noise through a
filter; this is unvoiced. These are approximations to the speech
signal. The impulse train gives the periodic nature that some voice
samples have.
Ray
Reply by mari...@gmail.com●March 19, 20052005-03-19
Dear Sir,
I am trying to make speech processing task and i am trying to extract
the coeff. of sound signal in a .wav file.
I have read the wave file into a matrix
[x fs]=wavread('s000_001.wav');
x is 5005 * 1 matrix
fs is the sampling frequency.
fs = 12000
I think that i has to down the sampling frquency to 8000 using resample
function.
then i don't know what to do to get the coeff.
i am thinking now that will write the following code
while j<= N_frames no of frames
A(j,1:11)=lpc(x(1:180,j),10);
j=j+1;
end;
to pass the sound signal as frames to the LPC function the previous
loop i found on this newsgroup but i wonder do i have to use windowing
before as i read in digital processing of speech signals of Rabiner or
what ?
and what is the method to find the number of frames and choosing the
order of the LPC
What i had understand was that the number of coeff. = the order that i
use is that write or not ?
And as for Mr. Raymond Toy, u wrote:
to feed some Gaussian noise through an AR filter, skipping over
> initial part of the output and feeding that data to your LPC
> estimator. The result should be close to the coefficients of your
filter.
i don't understand what is the AR filter, what does AR stands for i
know that AR stands for automatic recognition.
There is somthing also i didn't understand in the text book of Rabineer
i found that he feed the LPC with voiced and unvoiced signals but he
said then that he use impulse signals as an input does he mean that he
pass the sampled sound signal.
does the unvoiced signal is the Gaussian noise.
I am asking u because i found that u have experince in that field.
Thanks Very Much,
Best Regards,
Marian Mounir
Raymond Toy wrote:
>
> FruitfulToon> Dear Readers,
> FruitfulToon> I am currently working on some "speaker
recognition" software which
> FruitfulToon> will be using LPC coefficients for feature
extraction. I have written
> FruitfulToon> an algorithm but have no means by which to test it
because of lack of
> FruitfulToon> availability of existing LPC code.
>
> Why not feed some Gaussian noise through an AR filter, skipping over
> initial part of the output and feeding that data to your LPC
> estimator. The result should be close to the coefficients of your
> filter.
>
> Ray
Reply by FruitfulToon●February 2, 20052005-02-02
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
<kunal.yadwadkar@gmail.com> wrote in message
news:1107347508.963669.291450@l41g2000cwc.googlegroups.com...
>
> hi! I had implemented a speaker recognition software which too used LPC
> for feature extraction. So what kind of an algo are you using (pattern
> comparision?). What is the accuracy that you have got with it?
>
Reply by kuna...@gmail.com●February 2, 20052005-02-02
hi! I had implemented a speaker recognition software which too used LPC
for feature extraction. So what kind of an algo are you using (pattern
comparision?). What is the accuracy that you have got with it?
Reply by Nithin●February 1, 20052005-02-01
Hi
You may download the MATLAB code for LPC10 from the link below
http://www.cysip.com/LPC10-T.html
i am not sure if that meets your requirements. Note that it is suppose
to be for the LPC10 standard. You can atleast get a reference output i
guess.
-Nithin
FruitfulToon wrote:
> Dear Readers,
>
> I am currently working on some "speaker recognition" software which
> will be using LPC coefficients for feature extraction. I have written
> an algorithm but have no means by which to test it because of lack of
> availability of existing LPC code.
>
> I would greatly appreciate it if someone could send me some input
> data, and output data for LPC (maybe 20 pieces?). I am using the
> covariance method. My email is sorepharynx@hotmail.com.
>
> Many thanks!
FruitfulToon> Dear Readers,
FruitfulToon> I am currently working on some "speaker recognition" software which
FruitfulToon> will be using LPC coefficients for feature extraction. I have written
FruitfulToon> an algorithm but have no means by which to test it because of lack of
FruitfulToon> availability of existing LPC code.
Why not feed some Gaussian noise through an AR filter, skipping over
initial part of the output and feeding that data to your LPC
estimator. The result should be close to the coefficients of your
filter.
Ray
Reply by FruitfulToon●January 30, 20052005-01-30
Dear Readers,
I am currently working on some "speaker recognition" software which
will be using LPC coefficients for feature extraction. I have written
an algorithm but have no means by which to test it because of lack of
availability of existing LPC code.
I would greatly appreciate it if someone could send me some input
data, and output data for LPC (maybe 20 pieces?). I am using the
covariance method. My email is sorepharynx@hotmail.com.
Many thanks!