casper.cordes@mail.dk

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:

, 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.

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.

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. 

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 

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.


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.

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!

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.

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.

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

Coupla comments from my very surface perspective:

> Here is an example of a (small part of an) LPC analysis:

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)

"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

> samplingPeriod = 2.2675736961451248e-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

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.

> 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" 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.

> 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.

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

I couldn;t find anything obvious on maxobjects.

> 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:

> , 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.

> 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.

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.

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

> I couldn;t find anything obvious on maxobjects.

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

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.

Thanks for the interest, everybody; I was just thinking, that I should describe, excactly what I'm after:

I would like to be able to use formant/spectral information analysed from speech on other sounds. If used on white noise, the effect is like (clear) whispering. When stretched over time, there can be some interesting filtering going on. What's important is, that the calculations have a low cpu cost, and I suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs (frequency/bandwidth) will suffise. But I'm not able to work it out in MAX/MSP. I CAN make it happen, when using PRAAT, (see praat.org), but then its not real time.

there is a nice example of rand~ in the help file,P

> Thanks for the interest, everybody; I was just thinking, that I should describe, excactly what I'm after:

> I would like to be able to use formant/spectral information analysed from speech on other sounds. If used on white noise, the effect is like (clear) whispering. When stretched over time, there can be some interesting filtering going on. What's important is, that the calculations have a low cpu cost, and I suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs (frequency/bandwidth) will suffise. But I'm not able to work it out in MAX/MSP. I CAN make it happen, when using PRAAT, (see praat.org), but then its not real time.

On 06/08/07, Casper Cordes  wrote:
>
> Thanks for the interest, everybody; I was just thinking, that I should describe, excactly what I'm after:
>
> I would like to be able to use formant/spectral information analysed from speech on other sounds. If used on white noise, the effect is like (clear) whispering. When stretched over time, there can be some interesting filtering going on. What's important is, that the calculations have a low cpu cost, and I suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs (frequency/bandwidth) will suffise. But I'm not able to work it out in MAX/MSP. I CAN make it happen, when using PRAAT, (see praat.org), but then its not real time.
>
>
>

Try the rtcmix~ LPCPLAY examples. It does this.

> Thanks for the interest, everybody; I was just thinking, that I

> should describe, excactly what I'm after:

> I would like to be able to use formant/spectral information

> analysed from speech on other sounds. If used on white noise, the

> effect is like (clear) whispering. When stretched over time,

> there can be some interesting filtering going on. What's

> important is, that the calculations have a low cpu cost, and I

> suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs

> (frequency/bandwidth) will suffise. But I'm not able to work it

> out in MAX/MSP. I CAN make it happen, when using PRAAT, (see

> praat.org), but then its not real time.

Try the rtcmix~ LPCPLAY examples.  It does this.

>
> Thanks for the interest, everybody; I was just thinking, that I
> should describe, excactly what I'm after:
>
> I would like to be able to use formant/spectral information
> analysed from speech on other sounds. If used on white noise, the
> effect is like (clear) whispering. When stretched over time,
> there can be some interesting filtering going on. What's
> important is, that the calculations have a low cpu cost, and I
> suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs
> (frequency/bandwidth) will suffise. But I'm not able to work it
> out in MAX/MSP. I CAN make it happen, when using PRAAT, (see
> praat.org), but then its not real time.
>
>
>


There is also an LPC analyser and resynthesis in csound. You can use csound~

> Try the rtcmix~ LPCPLAY examples. It does this.

> > Thanks for the interest, everybody; I was just thinking, that I

> > should describe, excactly what I'm after:

> > I would like to be able to use formant/spectral information

> > analysed from speech on other sounds. If used on white noise, the

> > effect is like (clear) whispering. When stretched over time,

> > there can be some interesting filtering going on. What's

> > important is, that the calculations have a low cpu cost, and I

> > suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs

> > (frequency/bandwidth) will suffise. But I'm not able to work it

> > out in MAX/MSP. I CAN make it happen, when using PRAAT, (see

> > praat.org), but then its not real time.

There is also an LPC analyser and resynthesis in csound. You can use csound~
external to do it real-time  in max.

Peiman
On 06/08/07, garton@columbia.edu  wrote:
>
> Try the rtcmix~ LPCPLAY examples.  It does this.
>
> brad
> http://music.columbia.edu/~brad
>
>
>
> Quoting Casper Cordes :
>
> >
> > Thanks for the interest, everybody; I was just thinking, that I
> > should describe, excactly what I'm after:
> >
> > I would like to be able to use formant/spectral information
> > analysed from speech on other sounds. If used on white noise, the
> > effect is like (clear) whispering. When stretched over time,
> > there can be some interesting filtering going on. What's
> > important is, that the calculations have a low cpu cost, and I
> > suppose, that a maximum of 16 numbers per frame, or maybe 5 pairs
> > (frequency/bandwidth) will suffise. But I'm not able to work it
> > out in MAX/MSP. I CAN make it happen, when using PRAAT, (see
> > praat.org), but then its not real time.
> >
> >
> >
>

1) now I've checked rtcmix~ as suggested by Bradford, and yes: it DOES lpc things, but the problem is how to do an LPC analysis that it can read. It seems, that the object doesnt do analysis, but only reads analyses. I cant get the grasp of how to do an LPC analysis that it will read. The one in the example (a howling dog) comes in an exec file, the content of which, I can't read, and the procedure for making this analysis seems difficult; it's in another program, and in any case it's non-real time.

2) I've also started on looking at Csound. It seems more capable of doing what I'm after (for newcomers: real time LPC analysis and resynthesis), but apparently I have to study the thing for 5 years before getting close to anything interesting. I need A KEY to enter the fabulous world of Csound + max/msp!

3) Finally, what would really be interesting is if someone was capable of implementing the PRAAT algorithms in max/msp. I might be wrong, and there might exist initiatives that make this demarche unnessesary, but I think, that the coupling of these two things would be beneficial for both composers and linguists, and, -not least - ME!

Well if anyone can see the potential in this, and are able to work it out, that would really be great!

I wouldn't say you need 5 years to make an interesting sound in csound (if

that was the case I would need to trash half of my music!). You are not

going to use it as a score generator but to do a clear task (in this case

LPC), for that all you need to do is read the tutorial for that particular

opcode (LPC synthesis) or even just copy and paste the tutorial example.

Csound it not like max/msp in that you are not faced with the basic building

blocks but already designed opcodes that do particular synthesis tasks. This

means that to do LPC for instance you just need to give the right parameters

to the right opcodes which with csound~ you can control in real-time and

> 1) now I've checked rtcmix~ as suggested by Bradford, and yes: it DOES lpc

> things, but the problem is how to do an LPC analysis that it can read. It

> seems, that the object doesnt do analysis, but only reads analyses. I cant

> get the grasp of how to do an LPC analysis that it will read. The one in the

> example (a howling dog) comes in an exec file, the content of which, I can't

> read, and the procedure for making this analysis seems difficult; it's in

> another program, and in any case it's non-real time.

> 2) I've also started on looking at Csound. It seems more capable of doing

> what I'm after (for newcomers: real time LPC analysis and resynthesis), but

> apparently I have to study the thing for 5 years before getting close to

> anything interesting. I need A KEY to enter the fabulous world of Csound +

> 3) Finally, what would really be interesting is if someone was capable of

> implementing the PRAAT algorithms in max/msp. I might be wrong, and there

> might exist initiatives that make this demarche unnessesary, but I think,

> that the coupling of these two things would be beneficial for both composers

> Well if anyone can see the potential in this, and are able to work it out,

I wouldn't say you need 5 years to make an interesting sound in csound (if
that was the case I would need to trash half of my music!). You are not
going to use it as a score generator but to do a clear task (in this case
LPC), for that all you need to do is read the tutorial for that particular
opcode (LPC synthesis) or even just copy and paste the tutorial example.
Csound it not like max/msp in that you are not faced with the basic building
blocks but already designed opcodes that do particular synthesis tasks. This
means that to do LPC for instance you just need to give the right parameters
to the right opcodes which with csound~ you can control in real-time and
graphically in max.

On 08/08/2007, Casper Cordes  wrote:
>
>
> OK:
>
> 1) now I've checked rtcmix~ as suggested by Bradford, and yes: it DOES lpc
> things, but the problem is how to do an LPC analysis that it can read. It
> seems, that the object doesnt do analysis, but only reads analyses. I cant
> get the grasp of how to do an LPC analysis that it will read. The one in the
> example (a howling dog) comes in an exec file, the content of which, I can't
> read, and the procedure for making this analysis seems difficult; it's in
> another program, and in any case it's non-real time.
>
> 2) I've also started on looking at Csound. It seems more capable of doing
> what I'm after (for newcomers: real time LPC analysis and resynthesis), but
> apparently I have to study the thing for 5 years before getting close to
> anything interesting. I need A KEY to enter the fabulous world of Csound +
> max/msp!
>
> 3) Finally, what would really be interesting is if someone was capable of
> implementing the PRAAT algorithms in max/msp. I might be wrong, and there
> might exist initiatives that make this demarche unnessesary, but I think,
> that the coupling of these two things would be beneficial for both composers
> and linguists, and, -not least - ME!
>
> Well if anyone can see the potential in this, and are able to work it out,
> that would really be great!
>
>
>

> going to use it as a score generator but to do a clear task

I'm not sure what this means 'clear task', but I have an idea: something to do with not preparing files outside MAX/MSP

> LPC), for that all you need to do is read the tutorial for that particular

> opcode (LPC synthesis) or even just copy and paste the tutorial example.

...now how to find this famous tutorial?!!?

> Csound it not like max/msp in that you are not faced with the basic building

> blocks but already designed opcodes that do particular synthesis tasks. This

> means that to do LPC for instance you just need to give the right parameters

> to the right opcodes which with csound~ you can control in real-time and

Thanks, Peiman, but as you see, I'm illiterate on this one, so I need everything explained....

>You are not
> going to use it as a score generator but to do a clear task 

>(in this case
> LPC), for that all you need to do is read the tutorial for that particular
> opcode (LPC synthesis) or even just copy and paste the tutorial example.

> Csound it not like max/msp in that you are not faced with the basic building
> blocks but already designed opcodes that do particular synthesis tasks. This
> means that to do LPC for instance you just need to give the right parameters
> to the right opcodes which with csound~ you can control in real-time and
> graphically in max.

Thanks, Peiman, but as you see, I'm illiterate on this one, so I need everything explained....


haha OK, opcodes are like "instruments" similar to msp objects, each

designed to do a certain task. More often than not each opcode is like a

ready to use synthesizer whose parameters can be controlled by you. So in

order to do a single processing or synthesis "task" like filtering you

usually only need one opcode, particularly in the case of csound~ were you

There is a basic tutorial here that shouldn't take over an hour to go

http://www.csounds.com/manual/html/index.html

http://www.csounds.com/manual/html/SpectralLpcresyn.html

> > going to use it as a score generator but to do a clear task

> I'm not sure what this means 'clear task', but I have an idea: something

> to do with not preparing files outside MAX/MSP

> > LPC), for that all you need to do is read the tutorial for that

> > opcode (LPC synthesis) or even just copy and paste the tutorial example.

> ...now how to find this famous tutorial?!!?

> > Csound it not like max/msp in that you are not faced with the basic

> > blocks but already designed opcodes that do particular synthesis tasks.

> > means that to do LPC for instance you just need to give the right

> > to the right opcodes which with csound~ you can control in real-time and

> Thanks, Peiman, but as you see, I'm illiterate on this one, so I need

haha OK, opcodes are like "instruments" similar to msp objects, each
designed to do a certain task. More often than not each opcode is like a
ready to use synthesizer whose parameters can be controlled by you. So in
order to do a single processing or synthesis "task" like filtering you
usually only need one opcode, particularly in the case of csound~ were you
will do the control in max itself.

There is a basic tutorial here that shouldn't take over an hour to go
through:
http://www.csounds.com/toots/index.html

The manual is here:
http://www.csounds.com/manual/html/index.html

Here is the LPC page of the manual:
http://www.csounds.com/manual/html/SpectralLpcresyn.html

On 08/08/2007, Casper Cordes  wrote:
>
>
> Peiman wrote:
>
> >You are not
> > going to use it as a score generator but to do a clear task
>
> I'm not sure what this means 'clear task', but I have an idea: something
> to do with not preparing files outside MAX/MSP
>
> >(in this case
> > LPC), for that all you need to do is read the tutorial for that
> particular
> > opcode (LPC synthesis) or even just copy and paste the tutorial example.
>
> ...now how to find this famous tutorial?!!?
>
> > Csound it not like max/msp in that you are not faced with the basic
> building
> > blocks but already designed opcodes that do particular synthesis tasks.
> This
> > means that to do LPC for instance you just need to give the right
> parameters
> > to the right opcodes which with csound~ you can control in real-time and
> > graphically in max.
>
> ... 'opcodes' qu'es-ce que c'est?
>
> >
> > Best
> > Peiman
>
> Thanks, Peiman, but as you see, I'm illiterate on this one, so I need
> everything explained....
>
>
>

Thank you Peiman! (does anyone 'pei' you? ;-)

I'm a lot wiser, now. So, what I've done, is to

1) install Csound5 on my mac, I've installed the Csound~ object in max.

2) I've learned, that in order to get csound~ to execute the .sco and .orc files from the tutorial, I need to make a message saying 'csound Toot01.orc Toot01.sco', for example, and max/msp can perform the thing.

3) I've learned, that even the stupidest spelling error can be fatal (no surprise to someone who have worked a tiny bit with scripting).

4) I've also learned that I have to write the .sco and .orc files in a text object in max, and save them from here; in Textedit, Word or any other textediting software, there's trouble (maybe I need to know the right format)

But for what is with the LPC-thing, I still need some keys, to open closed doors with.

I can see, that I need to open the door 'LPANAL'; there is apparently no tutorial for this, and there is no example. I tried to write some syntax and send to the csound~ object, but nothing happened; not even an error message in the max window. I've included a screenshot of the patch (the soundfile sound.aiff was in the right folder and should be reachable for max)

(I hope my mental limitations, now publicized, will be helpful to others out there; I'm considering buying the Csound book, but I hesitate until knowing for certain that this is the right path for what I'm after, - I'm reluctant to spend hours and hours on something that leads me nowhere; now it SEEMS to be the right path, and if it is, I'll certainly invest what's needed...)

vocoder from LPC analysis