| I've been playing with creating dense sine granule textures aligned on a
harmonic lattice. I've found the easiest way to create the effect I want is
to ignore the granule ugens, as they don't offer enough fine control, and
write some simple code to create a monster score file with tens of
thousands of events.
The problem is, the latest version of Csound barfs around 1.47 seconds into
a 30 second render, and tries to access some memory that doesn't exist.
Is there a practical limit on the number of events in a score file? Or is
this some other problem?
(I won't post the score file here as it's 10,000 lines long!)
R.
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From: Toby
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> The problem is, the latest version of Csound barfs around 1.47 seconds into
> a 30 second render, and tries to access some memory that doesn't exist.
>
> Is there a practical limit on the number of events in a score file? Or is
> this some other problem?
I have a piece which uses your method
to combine short clips of a cello sound
at various tunings. This is about 7700
lines long, and runs fine on csound3.44
under Linux with 32Meg.
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Date: Sun, 16 Feb 1997 11:50:01 -0500
From: Jean Piche
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Yes: 46 No: 0. I'll take that as a yes...
How we proceed:
1- Committee is formed of the following people:
Thomas Trenka, Paulo Mouat, Jean Piche, Mathieu Bezkorowajny
Alexandre Burton, John Fitch and Tomy Hudson. I'll immodestly
appoint myself as chairman of committee.
2- Committee is open to any member of the list who wishes to join and
contribute thoughts, work and expertise. Email me.
3- Mandate of the committee:
Explore technologies that permit:
Central document control,
Automatic generation of variants.
Formulate and implement DTD (Document Type Definition) for Csound
manual.
Formulate and distribute protocol for manual updates.
Code current manual pursuant to the DTD.
Generate variants.
Distribute variants through ftp site.
4- Committee will attempt to support variants in the following formats:
HTML, ASCII and RTF. However, at this time, no formal engagements are
taken to support all or any of these variants.
5- Committee will report to this mailing list as work progresses but its
discussions will be kept off the list.
6- To make lives easier, public communications to and from the committee
should be posted to the list with the subject prefixed by [MANUAL}.
7- This project is supported by the community of Csound users at large
and is strictly volunteer-based. No guarantees whatsoever are offered as
to its date of completion.
Thanks!
________________________________________________________
Jean Piche
Universite de Montreal
http://mistral.ere.umontreal.ca/~pichej
http://www.musique.umontreal.ca/Org/CompoElectro/CEC/
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From: Michael Gogins
To: Csound list ,
John Fitch
Subject: Csound build
Date: Sun, 16 Feb 1997 13:21:33 -0500
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I appreciate the lack of C++ on many platforms and do not suggest it.
However, there are some things that could be done that would improve the
Csound code greatly.
I have harped sufficiently on plugin opcodes, so won't do more than mention
it.
Csound should be rebuilt with type double everywhere instead of type float.
The machines are big enough now that this should not bulk up the code, and
most FPUs by default take and return doubles, so this is really the native
type for computer math. It should actually be a bit faster on many
machines. I bet we would distinctly hear a favorable difference. I think
audio quality should be our number one object, especially now that that
professional audio is moving to higher quality, e.g. 24 bits.
Similarly the score and orchestra parsers should be able to read number
strings with 15 digits.
Although Csound should not be rewritten in C++, it could be rewritten with
classes. What I mean is structures that collect data used for some purpose
and work with functions of the form function(class *ptr). There are
already, implicitly, a number of classes in Csound. They should be made
more regular. This would help when it comes to making Csound re-entrant.
This I fear is a dismaying task and certainly requires either mucho time or
someone intimately familiar with what is really going on in there.
The class reorganization should make it possible to have a cleaner
separation of layers between the orchestra compiler, the score compiler,
the DSP stuff, and the input and output stuff, which in turn would make
maintaining the code base for so many platforms easier.
I do not particularly care one way or the other about changing the syntax
of the orchestra as some have suggested. If it is changed of course it is
vital that all the old syntax be kept for backward compatibility.
Well, that's my 2 cents again.
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From: Michael Gogins
To: Csound list , John Fitch
Subject: Csound build
Date: Sun, 16 Feb 1997 13:21:33 -0500
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I appreciate the lack of C++ on many platforms and do not suggest it.
However, there are some things that could be done that would improve the
Csound code greatly.
I have harped sufficiently on plugin opcodes, so won't do more than mention
it.
Csound should be rebuilt with type double everywhere instead of type float.
The machines are big enough now that this should not bulk up the code, and
most FPUs by default take and return doubles, so this is really the native
type for computer math. It should actually be a bit faster on many
machines. I bet we would distinctly hear a favorable difference. I think
audio quality should be our number one object, especially now that that
professional audio is moving to higher quality, e.g. 24 bits.
Similarly the score and orchestra parsers should be able to read number
strings with 15 digits.
Although Csound should not be rewritten in C++, it could be rewritten with
classes. What I mean is structures that collect data used for some purpose
and work with functions of the form function(class *ptr). There are
already, implicitly, a number of classes in Csound. They should be made
more regular. This would help when it comes to making Csound re-entrant.
This I fear is a dismaying task and certainly requires either mucho time or
someone intimately familiar with what is really going on in there.
The class reorganization should make it possible to have a cleaner
separation of layers between the orchestra compiler, the score compiler,
the DSP stuff, and the input and output stuff, which in turn would make
maintaining the code base for so many platforms easier.
I do not particularly care one way or the other about changing the syntax
of the orchestra as some have suggested. If it is changed of course it is
vital that all the old syntax be kept for backward compatibility.
Well, that's my 2 cents again.
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To: Michael Gogins ,
Csound list , John Fitch
From: Larry Troxler
Subject: Re: Csound build
At 01:21 PM 2/16/97 -0500, Michael Gogins wrote:
>
>Although Csound should not be rewritten in C++, it could be rewritten with
>classes. What I mean is structures that collect data used for some purpose
>and work with functions of the form function(class *ptr).
Yes. I am on a development team for a very large C project, and we've had
lots of success with things like this:
MYCLASS.H :
typedef struct myclass {
...
...
} Myclass;
void MyclassInit( Myclass *raw, ... );
void MyclassSomeFunction( Myclass *raw, ..);
MYCLASS.C :
#include "myclass.h"
void MyclassInit( Myclass *raw, ... ) { ... }
void MyclassSomeFunction( Myclass *raw, ..) { ... };
The syntax is a bit more wordy than the same thing in C++, but it makes it
very easy to understand the modularization.
Larry
-- Larry Troxler -- lt@westnet.com -- Patterson, NY USA --
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To: Michael Gogins ,
Csound list , John Fitch
From: Larry Troxler
Subject: Re: Csound build
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At 01:21 PM 2/16/97 -0500, Michael Gogins wrote:
>
>Although Csound should not be rewritten in C++, it could be rewritten with
>classes. What I mean is structures that collect data used for some purpose
>and work with functions of the form function(class *ptr).
Yes. I am on a development team for a very large C project, and we've had
lots of success with things like this:
MYCLASS.H :
typedef struct myclass {
...
...
} Myclass;
void MyclassInit( Myclass *raw, ... );
void MyclassSomeFunction( Myclass *raw, ..);
MYCLASS.C :
#include "myclass.h"
void MyclassInit( Myclass *raw, ... ) { ... }
void MyclassSomeFunction( Myclass *raw, ..) { ... };
The syntax is a bit more wordy than the same thing in C++, but it makes it
very easy to understand the modularization.
Larry
-- Larry Troxler -- lt@westnet.com -- Patterson, NY USA --
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Hi All
I've been reading about, and would like to get involved with
Csound. But, before I expend the time and energy, can someone on the
list tell me if Csound for the Mac will run on a 030 machine with a
software FPU. I'll be moving to a PPC in June, but would like to get
started now if it's possible.
Thanks for any help!
John
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From: Tobias Kunze
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In-Reply-To: "Michael Gogins" "Csound build" (Feb 16, 1:21pm)
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To: Csound list ,
Michael Gogins
Subject: Re: Csound build
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| Csound should be rebuilt with type double everywhere
| instead of type float. The machines are big enough now
| that this should not bulk up the code, and most FPUs by
| default take and return doubles, so this is really the
| native type for computer math. It should actually be a bit
| faster on many machines.
Are you sure? Most chip designs these days are driven by graphics
applications and thus focus on float performance, as far as I see.
This is especially true for MIPS (SGI) type-of machines, but the
PPC and Pentium lines of processors seem to go in the same direction.
Also, most machines suffer from bus deficiencies already (ie, they can't
get data fast enough to and from the cpu), and using doubles would
more than double the load.
--
______________________________________________________________________
Tobias Kunze t@kunze.stanford.edu
CCRMA, Stanford University http://www.stanford.edu/~tkunze
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From: Charles Baker
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Date: Sun, 16 Feb 97 14:37:53 -0800
To: csound@maths.ex.ac.uk
Subject: Re: Csound build
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Hmm... I think Tobias has a point. Throughput is the limiting factor in
many Csound instruments (as it is on many video manipulation programs).
Watch what happens to compile times when you add part of the instrument
that writes and reads memory constantly (ie: reverb). To double the size
of these writes and fetches will probably slow csound down a great deal.
As I understand it, SGI's O2 has spent a great deal of time on this
bus speed problem, and the remarkable speed up in real-time that
has been reported here is more due to this improved io, not the
speed of the processor. My friend in scientific visualisation
at the FloridaState SuperComputerResearchInstitute *swears* that
this new buss design is the cat's pyjamas (and boy, he knows real-time
video as well as anyone I've met!)
Well, I'm ready tobe corrected, but I feel that processor speed is only
one of the issues in speed of csound execution...
CharlieB
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From: Lawrence Troxler
To: csound@maths.ex.ac.uk
Subject: #ifdef BLASTER, anyone?
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Am I right in thinking that this is Soundblaster support for DOS? Is it
still being used? I guess it would still be usefull, however I notice that
3.44 wouldn't work with BLASTER defined anyway - for example, there
doesn't seem to be an sbinit() defined anywhere.
Larry (In "looking for things to throw out" mode)
-- Larry Troxler -- lt@westnet.com -- Patterson, NY USA --
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From: Tobias Kunze
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Date: Sun, 16 Feb 1997 15:36:00 -0800
In-Reply-To: "Michael Gogins" "Re: Csound build" (Feb 16, 6:12pm)
References: <199702162307.SAA102068@mule1.mindspring.com>
Reply-To: t@ulysses.stanford.edu
X-Url: http://www.stanford.edu/~tkunze
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To: Michael Gogins
Subject: Re: Csound build
Cc: csound@maths.ex.ac.uk
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On Feb 16, 6:12pm, Michael Gogins wrote:
| Subject: Re: Csound build
|
| I think you are right about the bus, but I know for a fact
| that double math on Pentiums is faster than float math (not
| by much; about 4%).
Were you really using float math routines to test float and
double performance? If your compiler doesn't give you both,
double and float routines for all its entry points, you are
comparing apples and oranges, since your float inputs will be
casted to doubles before execution (which might account for the
4% overhead you are quoting). For almost any serious test, float
routines *have* to be faster than their standard double
counterparts for cache and bandwidth reasons. Check your
documentation for routines like `fsin', `fmodf', `ffloor', etc.
(Some compilers allow the use of a -float or similar flag, but it's up to
the compiler whether and to what extent it sticks to that, so it's
not a very good test either).
| However, my primary point is sound quality. Even if it were
| slower I think it would sound better.
I doubt anyone will gain noticeable increase in sound quality using
doubles unless s/he writes excessively inefficient code. Using
32 bit floats, you already have 12 bits (or almost 3 1/2 orders of
magnitude) to waste before your code-induced noise shows up in a
final 20bit result.
|
End of excerpt from Michael Gogins
--
______________________________________________________________________
Tobias Kunze t@kunze.stanford.edu
CCRMA, Stanford University http://www.stanford.edu/~tkunze
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Date: Sun, 16 Feb 1997 20:23:39 -0500 (EST)
From: Michael Hamel
To: csound@maths.ex.ac.uk
Subject: csound max
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Does anyone know if there are any copies of John Burkhardt's Csound-MAX
patches floating around anywhere?
Thanks,
Michael Hamel
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Date: Mon, 17 Feb 1997 01:34:19 -0600 (CST)
From: Jeremiah Thomas Isaacs
To: John Schur
Cc: Csound mail list
Subject: Re: Soft FPU?
In-Reply-To: <33073BC0.4C50@usaor.net>
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On Sun, 16 Feb 1997, John Schur wrote:
> Hi All
>
> I've been reading about, and would like to get involved with
> Csound. But, before I expend the time and energy, can someone on the
> list tell me if Csound for the Mac will run on a 030 machine with a
> software FPU. I'll be moving to a PPC in June, but would like to get
> started now if it's possible.
>
Soft FPU has always been more trouble than it is worth for me. (Another
extension to conflict with) It in no way speeds things up, it simply
allows you to run programs you normally would not. I have an 68lc040 (no
FPU) and am just experimenting and learning til I have something more
substantial to work with. You will surely appreciate the PPC when you get
it.
jeremiah
----------------------------------------------
she could talk to squirrels
walking back from the convalescent home
staring at sports cars
crying
jti0001@jove.acs.unt.edu
http://people.unt.edu/~jti0001 -beck
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Date: Sun, 16 Feb 1997 12:30:00 +0100
From: Gabriel Maldonado
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Subject: gabsoundin2
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Hello everybody,
here is the code of a new UG for variable pitch
reading in a soundfile.
the sintax is almost the same as soundin:
a1 soundin2 ifilcod, iskptim, iformat, kfrqratio
a1, a2 soundin2 ifilcod, iskptim, iformat, kfrqratio
the new parameter kfrqratio is a multiplier factor and
must be > 0. This allows transposition and continuous modulation
of pitch. Note that the parameters iskiptim and iformat are
not optional now. Also note that this new unit only supports mono and
stereo 16bit integer audio files.
Known bugs:
-Do not use high transposition ratios (8 or 16) with low krate
(or high ksmps values as 100 or 50) otherwise the sound flow
could be broken.
-When the end of input soundfile is reached, some samples could be
lacking
in the output file.
this is the source:
////////////////// WARNING! in this source I often used the c++
////////////////// line comment sintax ( // comment ).
////////////////// Replace with standard c comment ( /* comment */ )
////////////////// if necessary
////////////////// In the SOUNDIO.H file
////////////////// you must REPLACE the old SOUNDIN structure
////////////////// as follows:
////////////////// (this change will not affect the normal behaviour
////////////////// of the old soundin unit)
typedef struct {
OPDS h;
float *r1, *r2, *r3, *r4, *ifilno, *iskptim, *iformat;
/* #ifdef GAB_RT */
float *kfrqratio; //GAB new parameter
// work variables GAB
double phase_gab; // hidden work variables
double fl_buf;
int base_sample_gab;
int initflag_gab;
/* #endif */
short format, channel, nchnls, sampframsiz, filetyp, analonly,
endfile;
long sr, audrem, framesrem, getframes; /* bytes, frames, frames */
AIFFDAT *aiffdata;
void (*bytrev)();
FDCH fdch;
char *inbufp, *bufend;
char inbuf[SNDINBUFSIZ*4]; //gab
} SOUNDIN;
/////////////////
///////////////// in the SOUNDIN.C file you must ADD the following
///////////////// functions:
/////////////////
void soundin2(SOUNDIN *p) //see SOUNDIO.H for GAB modification of type
SOUNDIN
{
/* Warning! use low k rate if tranpose factor is high:
for example, don't use ksmps > 5 if transpose factor is 16 */
short nsmps;
float *r1, *r2, *r3, *r4;
int chnsout, n, ntogo;
float ratio;
long base;
double *phase;
double *fl_buf;
double phase_diff;
base = p->base_sample_gab;
phase = &(p->phase_gab);
ratio = *(p->kfrqratio) * (p->sr/esr);
fl_buf = &(p->fl_buf);
r1 = p->r1; r2 = p->r2; r3 = p->r3; r4 = p->r4;
chnsout = p->OUTOCOUNT;
ntogo = ksmps;
if (p->endfile) goto filend;
nsmps = (p->bufend - p->inbufp) / p->sampframsiz;
if (nsmps > ksmps) nsmps = ksmps;
ntogo -= nsmps;
if (p->initflag_gab) {
size_t frame_size;
p->initflag_gab =0;
if (chnsout == 1) frame_size = sizeof(short);
else frame_size = sizeof(long);
lseek(p->fdch.fd, (long) *phase * frame_size , SEEK_SET );
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) == 0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
p->bufend = p->inbuf + n + SNDINBUFSIZ/32 ;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo) nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
switch (p->format) {
case AE_SHORT:
shorts:
switch(chnsout) {
case 1:
{
register short *base_addr;
register short *inbufp = (short *)p->inbufp;
base_addr = inbufp;
*fl_buf -= (long) *fl_buf;
do {
phase_diff = *phase - (long) *phase;
*r1++ = (float) *inbufp + (float) ( *(inbufp+1) -
*inbufp) * phase_diff; //gab
//*r1++ = (float) *inbufp;
*phase += ratio;
*fl_buf += ratio;
inbufp = base_addr + (long) *fl_buf;
} while (--nsmps);
if (inbufp >= ((short *)p->bufend)- SNDINBUFSIZ/32) {
lseek(p->fdch.fd, (long) *phase * sizeof(short), SEEK_SET
);
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) ==
0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
inbufp = (short *) p->inbuf;
p->bufend = p->inbuf + n + SNDINBUFSIZ/32 ;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo)
nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
p->inbufp = (char *) inbufp;
}
break;
case 2:
{
register long *base_addr;
register long *inbufp = (long *)p->inbufp;
base_addr = inbufp;
*fl_buf -= (long) *fl_buf;
do {
register short left,right, left1, right1;
phase_diff = *phase - (long) *phase;
left = (short) (*inbufp >> 16);
right =(short) ( *inbufp & 0xffff);
left1 = (short) (*(inbufp+1)>> 16);
right1 =(short) ( *(inbufp+1) & 0xffff);
*r1++ = (float) left + (float) ( left1 - left) *
phase_diff;
*r2++ = (float) right + (float) ( right1 - right) *
phase_diff;
*phase += ratio;
*fl_buf += ratio;
inbufp = base_addr + (long) *fl_buf;
} while (--nsmps);
if (inbufp >= ((long *)p->bufend)- SNDINBUFSIZ/32 ) {
lseek(p->fdch.fd, (long) *phase * sizeof(long) ,
SEEK_SET );
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) ==
0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
inbufp = (long *) p->inbuf;
p->bufend = p->inbuf + n + SNDINBUFSIZ/32;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo)
nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
p->inbufp = (char *) inbufp;
}
break;
default:
dies("soundin2 of %s not implemented",
getstrformat((int)p->format));
}
break;
default:
dies("soundin2 of %s not implemented",
getstrformat((int)p->format));
}
return;
filend:
nsmps = ntogo;
switch(chnsout) { /* if past end of file, */
case 1:
do *r1++ = fzero; /* move in zeros */
while (--nsmps);
break;
case 2:
do {
*r1++ = fzero;
*r2++ = fzero;
} while (--nsmps);
break;
}
}
//////////////// At the beginning of the ENTRY.C file you must add
//////////////// the following function declarations.....
void sndin2set(void*), soundin2(void*); //GAB
//////////////// ..... and the following line in the OENTRY opcodlst[]
//////////////// structure array declaration.
{ "soundin2",S(SOUNDIN), 5, "mmmm", "Sook", sndin2set,NULL,
soundin2 } //GAB
//////////////// That's all .... and good luck!
Gabriel Maldonado
g.maldonado@agora.stm.it
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Date: Sun, 16 Feb 1997 12:30:00 +0100
From: Gabriel Maldonado
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Organization: Fantalogia
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Subject: gabsoundin2
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Hello everybody,
here is the code of a new UG for variable pitch
reading in a soundfile.
the sintax is almost the same as soundin:
a1 soundin2 ifilcod, iskptim, iformat, kfrqratio
a1, a2 soundin2 ifilcod, iskptim, iformat, kfrqratio
the new parameter kfrqratio is a multiplier factor and
must be > 0. This allows transposition and continuous modulation
of pitch. Note that the parameters iskiptim and iformat are
not optional now. Also note that this new unit only supports mono and
stereo 16bit integer audio files.
Known bugs:
-Do not use high transposition ratios (8 or 16) with low krate
(or high ksmps values as 100 or 50) otherwise the sound flow
could be broken.
-When the end of input soundfile is reached, some samples could be
lacking
in the output file.
this is the source:
////////////////// WARNING! in this source I often used the c++
////////////////// line comment sintax ( // comment ).
////////////////// Replace with standard c comment ( /* comment */ )
////////////////// if necessary
////////////////// In the SOUNDIO.H file
////////////////// you must REPLACE the old SOUNDIN structure
////////////////// as follows:
////////////////// (this change will not affect the normal behaviour
////////////////// of the old soundin unit)
typedef struct {
OPDS h;
float *r1, *r2, *r3, *r4, *ifilno, *iskptim, *iformat;
/* #ifdef GAB_RT */
float *kfrqratio; //GAB new parameter
// work variables GAB
double phase_gab; // hidden work variables
double fl_buf;
int base_sample_gab;
int initflag_gab;
/* #endif */
short format, channel, nchnls, sampframsiz, filetyp, analonly,
endfile;
long sr, audrem, framesrem, getframes; /* bytes, frames, frames */
AIFFDAT *aiffdata;
void (*bytrev)();
FDCH fdch;
char *inbufp, *bufend;
char inbuf[SNDINBUFSIZ*4]; //gab
} SOUNDIN;
/////////////////
///////////////// in the SOUNDIN.C file you must ADD the following
///////////////// functions:
/////////////////
void soundin2(SOUNDIN *p) //see SOUNDIO.H for GAB modification of type
SOUNDIN
{
/* Warning! use low k rate if tranpose factor is high:
for example, don't use ksmps > 5 if transpose factor is 16 */
short nsmps;
float *r1, *r2, *r3, *r4;
int chnsout, n, ntogo;
float ratio;
long base;
double *phase;
double *fl_buf;
double phase_diff;
base = p->base_sample_gab;
phase = &(p->phase_gab);
ratio = *(p->kfrqratio) * (p->sr/esr);
fl_buf = &(p->fl_buf);
r1 = p->r1; r2 = p->r2; r3 = p->r3; r4 = p->r4;
chnsout = p->OUTOCOUNT;
ntogo = ksmps;
if (p->endfile) goto filend;
nsmps = (p->bufend - p->inbufp) / p->sampframsiz;
if (nsmps > ksmps) nsmps = ksmps;
ntogo -= nsmps;
if (p->initflag_gab) {
size_t frame_size;
p->initflag_gab =0;
if (chnsout == 1) frame_size = sizeof(short);
else frame_size = sizeof(long);
lseek(p->fdch.fd, (long) *phase * frame_size , SEEK_SET );
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) == 0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
p->bufend = p->inbuf + n + SNDINBUFSIZ/32 ;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo) nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
switch (p->format) {
case AE_SHORT:
shorts:
switch(chnsout) {
case 1:
{
register short *base_addr;
register short *inbufp = (short *)p->inbufp;
base_addr = inbufp;
*fl_buf -= (long) *fl_buf;
do {
phase_diff = *phase - (long) *phase;
*r1++ = (float) *inbufp + (float) ( *(inbufp+1) -
*inbufp) * phase_diff; //gab
//*r1++ = (float) *inbufp;
*phase += ratio;
*fl_buf += ratio;
inbufp = base_addr + (long) *fl_buf;
} while (--nsmps);
if (inbufp >= ((short *)p->bufend)- SNDINBUFSIZ/32) {
lseek(p->fdch.fd, (long) *phase * sizeof(short), SEEK_SET
);
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) ==
0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
inbufp = (short *) p->inbuf;
p->bufend = p->inbuf + n + SNDINBUFSIZ/32 ;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo)
nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
p->inbufp = (char *) inbufp;
}
break;
case 2:
{
register long *base_addr;
register long *inbufp = (long *)p->inbufp;
base_addr = inbufp;
*fl_buf -= (long) *fl_buf;
do {
register short left,right, left1, right1;
phase_diff = *phase - (long) *phase;
left = (short) (*inbufp >> 16);
right =(short) ( *inbufp & 0xffff);
left1 = (short) (*(inbufp+1)>> 16);
right1 =(short) ( *(inbufp+1) & 0xffff);
*r1++ = (float) left + (float) ( left1 - left) *
phase_diff;
*r2++ = (float) right + (float) ( right1 - right) *
phase_diff;
*phase += ratio;
*fl_buf += ratio;
inbufp = base_addr + (long) *fl_buf;
} while (--nsmps);
if (inbufp >= ((long *)p->bufend)- SNDINBUFSIZ/32 ) {
lseek(p->fdch.fd, (long) *phase * sizeof(long) ,
SEEK_SET );
if ((n = sreadin(p->fdch.fd,p->inbuf,SNDINBUFSIZ * 4,p)) ==
0) {
p->endfile = 1;
if (ntogo) goto filend;
else return;
}
inbufp = (long *) p->inbuf;
p->bufend = p->inbuf + n + SNDINBUFSIZ/32;
if (ntogo > 0) {
if ((nsmps = n / p->sampframsiz) > ntogo)
nsmps = ntogo;
ntogo -= nsmps;
goto shorts;
}
}
p->inbufp = (char *) inbufp;
}
break;
default:
dies("soundin2 of %s not implemented",
getstrformat((int)p->format));
}
break;
default:
dies("soundin2 of %s not implemented",
getstrformat((int)p->format));
}
return;
filend:
nsmps = ntogo;
switch(chnsout) { /* if past end of file, */
case 1:
do *r1++ = fzero; /* move in zeros */
while (--nsmps);
break;
case 2:
do {
*r1++ = fzero;
*r2++ = fzero;
} while (--nsmps);
break;
}
}
//////////////// At the beginning of the ENTRY.C file you must add
//////////////// the following function declarations.....
void sndin2set(void*), soundin2(void*); //GAB
//////////////// ..... and the following line in the OENTRY opcodlst[]
//////////////// structure array declaration.
{ "soundin2",S(SOUNDIN), 5, "mmmm", "Sook", sndin2set,NULL,
soundin2 } //GAB
//////////////// That's all .... and good luck!
Gabriel Maldonado
g.maldonado@agora.stm.it
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Subject: Re: Possible bug
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Message written at 17 Feb 1997 09:00:20 +0000
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In-reply-to: <3.0.32.19970216144817.006eb91c@sdps.demon.co.uk> (message from
Richard Wentk on Sun, 16 Feb 1997 15:53:01 +0000)
I did not think that there was a limit on the number of events in a
score. In general it expands memory to fit (except in the cases I
missed :-) so that that does not soudn like it. It could be a malloc
which is not adequately checked. How much memory does your machine
have? Which system also? You might like to send me (ie not teh list)
your monster score/orc.
==John
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Subject: Re: Manual....and formats
Date: Mon, 17 Feb 97 22:43:23 +1100
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To: "Thomas R. Trenka" ,
csound
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Your suggestion is great. I also prefer PDF. I think one of the reasons
people have voted against it is that they never heard of it. Is that so?
PDF is a great electronic and printable format.
If you are willing to do the translations, so be it. Let's have it in
multiple formats, and use according to taste.
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Date: Mon, 17 Feb 1997 06:34:31 -0600 (CST)
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To: jpff@maths.bath.ac.uk
From: Russell Pinkston
Subject: Re: help about Csound tools&courses
Cc: csound@noether.ex.AC.UK
>Message written at 14 Feb 1997 22:07:21 +0000
>--- Copy of mail to isidoro.perez@cdmc.es ---
>In-reply-to: <33049B1A.11BE@cdmc.es> (message from adolfo nunez on Fri, 14 Feb
> 1997 17:04:26 +0000)
>
>I have not heard of a Csound score to MIDI file translator. It would
>be impossible in the general case as teh pitch resolution is
>different, and apriori one does not know what the P-fields mean.
>
>I suppose if one restricted the thing sufficiently it should not be
>_very_ hard.
>
>==John ff
>
>
We have a simple score to MIDI file translator on our ftp site at
indigo.pac.utexas.edu, written by Keith Lent. It's called writemid and it's
in the pub/midi/ibmpc directory, with a companion routine called readmidi.
Source code is provided.
----------------------------------
Russell F. Pinkston, D.M.A.
Associate Professor of Composition
Director, Electronic Music Studios
School of Music
The University of Texas at Austin
Austin, TX 78712
[512-471-0865]
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To: jpff@maths.bath.ac.uk
From: Russell Pinkston
Subject: Re: help about Csound tools&courses
Cc: csound@maths.ex.ac.uk
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>Message written at 14 Feb 1997 22:07:21 +0000
>--- Copy of mail to isidoro.perez@cdmc.es ---
>In-reply-to: <33049B1A.11BE@cdmc.es> (message from adolfo nunez on Fri, 14 Feb
> 1997 17:04:26 +0000)
>
>I have not heard of a Csound score to MIDI file translator. It would
>be impossible in the general case as teh pitch resolution is
>different, and apriori one does not know what the P-fields mean.
>
>I suppose if one restricted the thing sufficiently it should not be
>_very_ hard.
>
>==John ff
>
>
We have a simple score to MIDI file translator on our ftp site at
indigo.pac.utexas.edu, written by Keith Lent. It's called writemid and it's
in the pub/midi/ibmpc directory, with a companion routine called readmidi.
Source code is provided.
----------------------------------
Russell F. Pinkston, D.M.A.
Associate Professor of Composition
Director, Electronic Music Studios
School of Music
The University of Texas at Austin
Austin, TX 78712
[512-471-0865]
|