vocoder from LPC analysis

im no DSP whiz, but im thinking those values and coefficients are used for creating an FIR filter? it looks similar to variable declarations in C, without types (but it seems all types are float or double anyways). i dont want to say for sure and lead you in the wrong direction, but maybe that would be a good place to start looking (FIR filter design). for more info on what to do, maybe you can google to find what kind of filters are used in applying LPC analysis.

well, in that case, the values a [1], a [2], ... a[n], must be in dB, and I must convert them to amplitude values (-1. to 1.) before creating a buffir~ content..

Thanks for leading me there, it seems to be the right path!

cool, hope it works out.

im no max object whiz, but im thinking there must be some sort of conversion object between db and Amplitude? without using expr, that is.

there is, in fact. In the 4th max tutorial, there is an patcher called dBtoA (which IS in fact based on expr, but anyhow)

OK... Now I've succeeded in transformaing the LPC analysis generated in PRAAT into a coll file. The good news is that the values, when translated into amplitude go from negative/positive to all positive. Buffir~ seems to prefer positive numbers. The bad news is, that the numbers exeede 1.0... It seems I have the wrong convertion going on from dB to Amplitude.

I've attached the patch which translates the PRAAT analysis into a coll file. In next message I'll upload the text document with the PRAAT analysis.

Here is the PRAAT analysis.

Note that you need to place this file in a folder readable from MAX in order to open it in the patch called lpc_to_buffir

This would be fun if you can get it running in a patch, but you may
be limited by the signal vector size, etc. I'm certainly not an
LPC or DSP person by any stretch of the imagination, but I've used
LPC resynthesis and cross-synthesis a fair amount. 'cross
synthesis' is probably the 'phase vocoder' effect you are chasing
-- you use a different program source through the LPC filters.
rtcmix~ has a working suite of LPC instruments, based on the old
Ken Stieglitz FORTRAN LPC that Paul Lansky, Charles Dodge, etc.
used back in the olden days. The source is all available if you
want to look at how it is implemented, but the code is pretty
dense.

Coupla comments from my very surface perspective:

Quoting Casper Cordes :

> Here is an example of a (small part of an) LPC analysis:
>
> xmin = 0
> xmax = 3.4449951170417563
> nx = 679
> dx = 0.005

As you have noticed, "nx" seems to be the number of frames in the
analysis, and "dx" looks like how far each jump forward is taken
(679*0.005 ~= 3.44499). I can't really tell how large (in samples)
each frame is, unless perhaps:

> x1 = 0.027491460103360062

"x1" is the number of samps/frame (x1*44100 = 1212.37339, which
seems weird). LPC frame sizes are odd, and often not the
power-of-2 that you see in FFT-based analyses. I think the default
for rtcmix~ LPC is 200 samps/frame. NEVER MIND -- I just checked,
and I think your analysis is about 220 samples/frame. Nor sure
what "x1" is.

> samplingPeriod = 2.2675736961451248e-05

samplingPeriod == 1/SR obviously.

> maxnCoefficients = 16
> frame []:
> frame [1]:
> nCoefficients = 16
> a []:
> a [1] = -3.2827861
> a [2] = 6.0996714
> a [3] = -7.8335428
> a [4] = 6.9054546
> a [5] = -3.3981638
> a [6] = -0.90945804
> a [7] = 3.6682715
> a [8] = -3.6352305
> a [9] = 1.4744695
> a [10] = 1.0017649
> a [11] = -2.2189555
> a [12] = 1.8888326
> a [13] = -0.85666442
> a [14] = 0.065275356
> a [15] = 0.18169832
> a [16] = -0.09004765
> gain = 5.8214144e-05

This is where to be careful, as I'm not sure that LPC uses FIR
fiters. The rtcmix~ LPC uses an allpole, recursive (IIR) filter
with 32 poles (coeffs). I think LPC can indeed be implemented as
FIR filters, but the analysis that generates the coeffs is based in
a recursive procedure that may assume IIR-type filters.

What I was saying earlier about the signal vector size and
implementing this at the patch level -- bear in mind that you will
need to access the coeffs for a new frame every 200 samples
(1/0.005), which means ideally you would need a signal vector size
of 200. I'm not sure how you will get around this. And of course
I'm assuming "in time" resynthesis.

brad
http://music.columbia.edu/~brad

Im working on a translator for this. I need a couple of days to sit
down with it, though it will be the formant tracks it reads first for
my own research, though the lpc should be easy to follow. but keep
nagging a I need to get it done pretty soon.

Something that might help is why is text limited to 255 lines. Is most
interesting reading this long?
I couldn;t find anything obvious on maxobjects.

best Pere

On 04/08/07, Casper Cordes wrote:
>
> Hi
>
> I use a software called PRAAT, which is a speech analysis tool. It gives you the possibility (among other things) to do a filtering on a given sound using a socalled LPC analysis from some speech. The effect is close to a vocoder, and LPC analysis seems to be the preferred one for that, by the way.
>
> I'm trying to apply the LPC analysis in MAX/MSP in order to do a real time processing of a given sound file. The problem is, that I don't know how to 'translate' the analysis into parameters understood by the MAX/MSP objects. Here is an example of a (small part of an) LPC analysis:
>
> xmin = 0
> xmax = 3.4449951170417563
> nx = 679
> dx = 0.005
> x1 = 0.027491460103360062
> samplingPeriod = 2.2675736961451248e-05
> maxnCoefficients = 16
> frame []:
> frame [1]:
> nCoefficients = 16
> a []:
> a [1] = -3.2827861
> a [2] = 6.0996714
> a [3] = -7.8335428
> a [4] = 6.9054546
> a [5] = -3.3981638
> a [6] = -0.90945804
> a [7] = 3.6682715
> a [8] = -3.6352305
> a [9] = 1.4744695
> a [10] = 1.0017649
> a [11] = -2.2189555
> a [12] = 1.8888326
> a [13] = -0.85666442
> a [14] = 0.065275356
> a [15] = 0.18169832
> a [16] = -0.09004765
> gain = 5.8214144e-05
>
> , where xmax is the length of the sound file in seconds; nx and dx have to do with window size (I think). It seems, that each of the 16 values should be multiplied by something in the given sample of the filtered sound, but my rudimentary understanding stops here.
>
> I NEED HELP!
>
> Thanks!
>
> PS: I have had succes in translating pitch information from a speech signal into values, that can control sample rate in MAX/MSP, giving me the opportunity to play sounds in a way that imitates the pitch contour of the speech signal. The advantage in using PRAAT is, that the pitch analysis is much better, when dealing with speech, than any max/msp obejct, I have tried.
>

--
www.centuryofnoise.com
www.perevillez.com

Quote: pvillez@gmail.com wrote on Sun, 05 August 2007 15:36
----------------------------------------------------
> Im working on a translator for this. I need a couple of days to sit
> down with it, though it will be the formant tracks it reads first for
> my own research, though the lpc should be easy to follow. but keep
> nagging a I need to get it done pretty soon.
>
> Something that might help is why is text limited to 255 lines. Is most
> interesting reading this long?
> I couldn;t find anything obvious on maxobjects.
>
> best Pere

well coll can open text files too, maybe you could use
another coll.