| I want to start updating my libraries so I can install some current packages
(i.e. Snd). Starting with glibc, I find several possible downloads on the
rpm repository.
glibc-2.1-1a
glibc-debug-2.1-1a
glibc-devel-2.1-1a
glibc-profile-2.1-1a
Which one (or more) do I need?
I know that at startup, my system does get environment variables, PATH, and
export from the /etc/profile file. Is this a different "profile"? or does
this mean I NEED glibc-profile-2.1-1a?
As a MkLinux usr, I have a feeling I will find most of my packages in .src
rather than .bin.
David
Dspondike@aol.com
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To: colug@bopper.wcbe.org
From: "H.Beyer"
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Cc: csound@maths.ex.ac.uk, mseta@yahoo.com
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I want to start updating my libraries so I can install some current packages
(i.e. Snd). Starting with glibc, I find several possible downloads on the
rpm repository.
glibc-2.1-1a
glibc-debug-2.1-1a
glibc-devel-2.1-1a
glibc-profile-2.1-1a
Which one (or more) do I need?
I know that at startup, my system does get environment variables, PATH, and
export from the /etc/profile file. Is this a different "profile"? or does
this mean I NEED glibc-profile-2.1-1a?
As a MkLinux usr, I have a feeling I will find most of my packages in .src
rather than .bin.
David
Dspondike@aol.com
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Date: Wed, 21 Jul 1999 09:08:45 -0500
To: csound@maths.ex.ac.uk
From: Daniel Nass
Subject: table and phasor
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Hello CSounders...
I have a question about GEN 01. I think I finally have a good handle on
how table and phasor works (I think) so I am using this simple orc and sco
just to read through a sound file. I want to make sure I have this much
down before I start trying to get fancy with other things.
Here's my problem. I would expect that the following orc and sco would
simply play back the file called in by GEN01 at normal speed, and play it
only once. Instead, it plays it back faster than normal and it also puts a
huge clip at the end of the soundfile. Is there something I'm not
understanding here?
Thanks for you help...
Dan
;orchestra
instr 1
kenv linen 20000, .01, p3, .01
aph phasor 1/p3
a1 table aph, 1, 1
a1 = a1 * kenv
out a1
endin
;score
f1 0 131072 1 6 0 4 1
;i# st dur(sf duration is 2.136)
i1 0 2.136
*************************************************************
Daniel Nass http://cctr.umkc.edu/~dnass/
"Wherever we are, what we hear is mostly noise.
When we ignore it, it disturbs us. When we listen
to it, we find it fascinating."
John Cage (1961)
*************************************************************
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Have you tried creating 8bit sound files? The garbage effect is often
a byte-order thing, and it is confused by Linux on the wrong byte
order? Just a thought
==John ff
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Have you tried creating 8bit sound files? The garbage effect is often
a byte-order thing, and it is confused by Linux on the wrong byte
order? Just a thought
==John ff
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From: Rob Howiler
To: csound@maths.ex.ac.uk
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Subject: lpc/pvoc files
Date: Wed, 21 Jul 1999 11:00:04 -0400
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Hi,
Can anyone recommend a way to "get inside" an lpc or pvoc file? I've been
working a bit with the vocoder and linear predictive ugens, but I would
really love to see what is actually going on inside the analysis files. The
Dodge/Jerse book has some info about this, but I can't get the files open
except as (I assume) ASCII characters. This may be more of an ASCII
translation question, but does anyone know a "translator" program or any
such thing (PC platform/Win98)? Help please?
thanks
Rob Howiler
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From: J P Fitch
To: Rob Howiler
cc: csound@maths.ex.ac.uk
Subject: Re: lpc/pvoc files
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Try the lpc_export and pv_export programs which change them to readable text.
Or use -v when creating, of -g of course to see what is going on.
....andother thing on teh listr of things to do is allow textual forms
of these files
==John ff
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To: csound@maths.ex.ac.uk
From: Jens Groh
Subject: Re: pluck wierdness
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craig routt wrote:
>i would like to fm a complex sample (like a steam whistle or a
>horn/siren) to test the time domain shift of a pluck.
>i tried to do a simple fm of a sample player orc but it would not accept
>an oscil input, you need to put a specific sr as the input to the opcode.
>modulating the vdelay sort of works.
>does any one have an orc to do fm on a sampled input .wav ?
Hi Craig!
You may wish to try my frequency shifter instrument, archived at:
http://media.dr.rhbnc.ac.uk/csound/list/msg01775.html
You will have to replace the variables kfrqshf, ko1frq and ko2frq by a-rate
variables, say afrqshf, ao1frq and ao2frq. Then afrqshf is the modulator
input, awvin the carrier input.
Jens
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Date: Wed, 21 Jul 1999 11:50:56 -0400
From: Peter Lakanen
Organization: Platinum Web Development
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CC: csound@maths.ex.ac.uk
Subject: Re: Karplus-strong bell sounds
References: <199907210926.TAA02695@bohm.anu.edu.au> <37959376.C64D691F@intercom.es>
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> Bell and gong like sounds are bes simulated with bidimensional
> meshes (ie. lots of Karplus-strong waveguides interconnected)
> with lots of dispersion to get that particular inharmonicity
> (putting some allpass filters bettween the waveguides is just ok).
Wow! This sounds really cool. Can anyone point me to a sample
.orc file of one of these instruments?
Thanks!
-peter
--
Peter Lakanen lakanen@platinumweb.com
Platinum Web Development http://www.platinumweb.com
1320 Terrace Street Tallahassee, FL 32303
850.508.4518 FAX: 850.681.6639
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Date: Wed, 21 Jul 1999 09:05:06 -0700 (PDT)
From: "Matt J. Ingalls"
To: Daniel Nass
cc: csound@maths.ex.ac.uk
Subject: Re: table and phasor
In-Reply-To: <4.1.19990721090112.00982520@cctr.umkc.edu>
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> only once. Instead, it plays it back faster than normal and it also puts a
> huge clip at the end of the soundfile. Is there something I'm not
> aph phasor 1/p3
> a1 table aph, 1, 1
> f1 0 131072 1 6 0 4 1
> i1 0 2.136
you didnt specify what your sample rate of your orch or soundfile is, but
i am assuming its the same. if you want to play the entire file your
table size must be greater than the # of sample frames in the sound file
(2.136*sr). what i usually do is store the actual length of the sound
file
in a look-up table, multiply that value with the phasor output, then use
"raw" index mode of the table:
ilenlookup = 50
isndfile = p4
ilen table isndfile-1, ilenlookup
aph phasor 1/p3
a1 table ilen*aph, isndfile, 0
f1 0 131072 1 6 0 4 1
;f2... other sound files here....
;f3
;etc..
f50 0 8 -2 88200 ; other "real" lens here
i1 0 2 1
-matt
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Date: Wed, 21 Jul 1999 14:14:53 -0700 (PDT)
From: Richard Karpen
To: Rob Howiler
cc: csound@maths.ex.ac.uk
Subject: Re: lpc/pvoc files
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On Wed, 21 Jul 1999, Rob Howiler wrote:
> Can anyone recommend a way to "get inside" an lpc or pvoc file?
Below is a little C program I wrote a number of years ago for viewing
pvoc data. I've only compiled it and used it on Unix/Linux systems. It's
very handy for viewing the frequency and amplitude data of the analysis
bins, or even for just reading the header in case you've forgotten the
window size that you used in pvanal. Later this year I'm hoping that we'll
have a java GUI version of this that also allows for graphically editing
the data. The data is stored by frame but this program rearranges the data
and prints it bin-by-bin.
/*******************************************************************/
/* PVLOOK.C by Richard Karpen 1993 */
/*******************************************************************/
#include
#include
#define PVSHORT 2 /* for .format .. 16 bit linear data */
#define PVFLOAT 4 /* for .format .. 32 bit float data */
#define PVDFLTBYTS 4
#define PVMAGIC 517730 /* look at it upside-down, esp on a 7-seg display */
#define PVDFLTBYTS 4
typedef struct pvstruct
{
long magic; /* magic number to identify */
long headBsize; /* byte offset from start to data */
long dataBsize; /* number of bytes of data */
long dataFormat; /* (int) format specifier */
float samplingRate; /* of original sample */
long channels; /* (int) mono/stereo etc */
long frameSize; /* size of FFT frames (2^n) */
long frameIncr; /* # new samples each frame */
long frameBsize; /* bytes in each file frame */
long frameFormat; /* (int) how words are org'd in frms */
float minFreq; /* freq in Hz of lowest bin (exists) */
float maxFreq; /* freq in Hz of highest (or next) */
long freqFormat; /* (int) flag for log/lin frq */
char info[PVDFLTBYTS]; /* extendable byte area */
} PVSTRUCT;
main( argc, argv ) int argc ; char *argv[] ; {
int i, j, k, fd;
FILE *fp ;
float *pvdataF, *pvdataA;
register PVSTRUCT *phdr;
int numframes, framesize;
int l, FuncSize,num;
long int m;
int firstBin, lastBin, numBins, lastFrame;
int printInts=1;
int firstFrame = 1;
if ( argc == 1 ) {
fprintf( stderr,"pvlook is a program which reads a Csound pvanal's pvoc.n file and outputs frequency and magnitude trajectories for each of the analysis bins. \n");
fprintf(stderr, "usage: pvlook [-bb X] [-eb X] [-bf X] [-ef X] [-i X] file > output\n" ) ;
fprintf( stderr,
" -bb X begin at anaysis bin X. Numbered from 1 [defaults to 1]\n" ) ;
fprintf( stderr,
" -eb X end at anaysis bin X [defaults to highest]\n" ) ;
fprintf( stderr,
" -bf X begin at anaysis frame X. Numbered from 1 [defaults to 1]\n" ) ;
fprintf( stderr,
" -ef X end at anaysis frame X [defaults to last]\n" ) ;
fprintf( stderr,
" -i X prints values as integers [defaults to floating point]\n" ) ;
exit( -1 ) ;
}
if ( ( fd = open( argv[argc-1], O_RDONLY) ) < 0 ) {
fprintf( stderr, "pvlook: Unable to open '%s'\n Does it exist?", argv[argc-1] ) ;
exit( -1 ) ;
}
if ( ( fp = fdopen( fd, "r" ) ) == NULL ) {
fprintf( stderr, "pvr: Unable to fdopen '%s'\n", argv[argc-1] ) ;
exit( -1 ) ;
}
phdr = (PVSTRUCT *)malloc( sizeof(PVSTRUCT)) ;
pvdataF = (float *) malloc(sizeof(float));
pvdataA = (float *) malloc(sizeof(float));
rewind( fp ) ;
fread(phdr, 1, sizeof(PVSTRUCT), fp);
if(phdr->magic != PVMAGIC) {
fprintf( stderr, "'%s' is not a pvoc file\n", argv[argc-1] ) ;
exit( -1 ) ;
}
framesize = phdr->frameSize+2;
numframes =((phdr->dataBsize/4) / framesize);
firstBin = 1;
lastBin = (framesize/2);
lastFrame = numframes;
for ( i = 1 ; i < argc ; i++ ) {
if (!strcmp( argv[i], "-bb") )
firstBin = atoi( argv[++i] ) ;
if (!strcmp( argv[i], "-eb") )
lastBin = atoi( argv[++i] );
if (!strcmp( argv[i], "-bf") )
firstFrame = atoi( argv[++i] ) ;
if (!strcmp( argv[i], "-ef") )
lastFrame = atoi( argv[++i] ) ;
if (!strcmp( argv[i], "-i") )
printInts = 0;
}
numframes = (lastFrame - firstFrame) + 1;
numBins = (lastBin - firstBin) + 1;
for(l = 2; l < numframes; l*=2);
FuncSize = l+1;
printf("; Bins in Analysis: %d\n", framesize/2);
printf("; First Bin Shown: %d\n", firstBin);
printf("; Number of Bins Shown: %d\n", numBins);
printf("; Frames in Analysis: %d\n", ((phdr->dataBsize/4) / framesize));
printf("; First Frame Shown: %d\n", firstFrame);
printf("; Number of Data Frames Shown: %d\n", numframes);
rewind( fp ) ;
if(printInts!=0) {
for(k=0; k < numBins*2; k+=2) {
j=((firstBin*2)-1) + k;
printf("\nBin %d Freqs.", firstBin+k/2);
for(i = firstFrame-1; i < lastFrame; i++) {
m = 56 + ((j + (i * framesize) ) * 4);
fseek(fp, m, 0);
num+=fread( pvdataF, sizeof(float), 1, fp );
printf("%.3f ", *pvdataF);
}
printf("\n");
j=((firstBin*2)-2) + k;
printf("\nBin %d Amps. ", firstBin+k/2);
for(i = firstFrame-1; i < lastFrame; i++) {
m = 56 + ((j + (i * framesize) ) * 4);
fseek(fp, m, 0);
num+=fread( pvdataA, sizeof(float), 1, fp );
printf("%.3f ", *pvdataA);
}
printf("\n");
}
}
else
{
for(k=0; k < numBins*2; k+=2) {
j=((firstBin*2)-1) + k;
printf("\nBin %d Freqs.", firstBin+k/2);
for(i = firstFrame-1; i < lastFrame; i++) {
m = 56 + ((j + (i * framesize) ) * 4);
fseek(fp, m, 0);
num+=fread( pvdataF, sizeof(float), 1, fp );
printf("%d ", (short)*pvdataF);
}
printf("\n");
j=((firstBin*2)-2) + k;
printf("\nBin %d Amps. ", firstBin+k/2);
for(i = firstFrame-1; i < lastFrame; i++) {
m = 56 + ((j + (i * framesize) ) * 4);
fseek(fp, m, 0);
num+=fread( pvdataA, sizeof(float), 1, fp );
printf("%d ", (short)*pvdataA);
}
printf("\n");
}
}
fclose( fp ) ;
free(pvdataF);
free(pvdataA);
free(phdr);
}
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To: csound@maths.ex.ac.uk
Date: Wed, 21 Jul 1999 16:24:24 -0700
Subject: problem with soundin on winsound 3.52
Message-ID: <19990721.162428.14326.1.kurtnelson2@juno.com>
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From: Kurt S Nelson
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To all Csounders:
I am running winsound 3.52 on my PC.
The soundin opcode will not open .wav files
It seems to work about half the time on the soundin.aiff file included
with toot8, but not with any of the .wav files on my drive regardless of
the directory they are in.
I would try it with other .aiff files, but I don't have any. Nor do I
have a utility for writing .aiff files other than winsound itself, and
soundin won't open those either.
Any suggestions or bugs I should be aware of?
Kurt
___________________________________________________________________
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To: csound@maths.ex.ac.uk
Date: Wed, 21 Jul 1999 16:15:18 -0700
Subject: new Csounder
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From: Kurt S Nelson
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Dear Csounders:
I am a composer/experimental musician from Portland, Oregon, USA. I have
19 years experience playing the cello in orchestras and chamber
ensembles, and a 1994 B.S. in biology from Portland State University. I
have worked three years now as a part-time janitor in a church near the
university, but I spend most of my time thinking about music, and finding
ways to realize my ideas. I dallied with rock bands for a couple of
years, and have done five or six studio sessions for rock, pop, and
gothic local bands, but now I do just solo performances, about one every
six months; I find my ideas have been too exacting and labor intensive
for any effective collaborations.
In spite of my poverty, and lack of equipment, I have finally finished a
solo album, called "Zendrome", to be released soon on a very young local
independant label. Because I have no decent microphones or pick-ups,
there is no cello at all on this album; I produced it entirely with the
Software Audio Workshop, and Cakewalk Pro Audio on a Windows 95 PC with a
SoundBlaster32 card, then mixed it down on a twelve year old Yamaha
4-track with two Boss guitar pedals. It's really quite primitive.
The impetus behind the album was this: For years, I had been searching
for hardware or software which would enable me to use just intonation,
and occasionally microtonal equal temperaments in my compositions. I had
written for the Commodore 64 for years, and had done some things with the
system speaker of my PC, but needless to say, the timbre and polyphony,
not to mention the programming interface, were extremely limited. My
attempts to do it via MIDI in my sequencing software were equally
unsuccessful, given the rigid inflexability of the SoundBlaster card (not
to mention the company's mainstream market orientation). In desperation,
I began to breadboard a pulse code modulation control circuit to vary the
speed of a tapedeck motor. I then created a number of taped drones at
standard pitches (A 440 and Middle C) which were to be varied in speed in
order to implement the tuning systems I desired. Fortunately, I
discovered the MUSICV paradigm before I had finished creating this
monster. A CD on the history of digital computer synthesis from my
public library opened me to the world of Max V. Mathews, John Pierce et
al. From there, I discovered the websites of the Stanford Center for
Computer Research in Music and Acoustics, IRCAM, etc. And finally, I
found Csound! Hurray!
The drones I created, incidentally, were liked well enough by my friends,
that I simply compiled some of them together for them to trip out on.
Voila! My first solo album.
Special areas of interest for me are tunings and temperaments, acoustics,
and the neurophysiology of music perception. Although unaffiliated with
any school or university, I am developing a theory of interval perception
which relies on the hypothesis that the sound processing parts of the
brain treat signal from the cocleae in a manner similar to that in which
the visual cortex treats sigals from the retinae. I am a long way from a
publishable paper, but I am working on a website. Since this is probably
an off-topic issue, I won't bring it up on this list again, but I do
forsee using Csound in my research in this area. After all, Roger
Shepard's landmark pitch perception studies made exstensive use of
MUSICV. BTW if anyone wants to discuss this or related topics off-list
please do e-mail me directly. I would welcome any comments or
suggestions.
In light of the above mentioned hypothesis, I have also concieved of a
number of electronic and acoustic instruments, which as of now exist only
on "paper." I hope to refine these designs, to eventually build these
instruments, and to compose music for these ensembles, towards which ends
Csound will be indispensible. Some of my future Csound compositions, I
imagine, will stand admirably on their own, as well.
Other things which draw me to Csound are:
1) The prospect of a completely digital music production process
(meaning a lack of unintended noise, provided one understands what one is
doing)
2) A depth and breadth of control which is unimaginable on hardware
based platforms (I like to think of it as "sonic nanotechnoloy")
3) A depth and breadth of sonic possibilities which is unattainable on
hardware based platforms
4) A "hands-on" approach to sound production that makes Csound a great
acoustics learning tool (and a good technology according to "The Five
Principles of Good Design")
5) The hope of contact with a community of like minds
6) Dare I say it? A jumpstart for my lagging music career
I look forward to a personal creative renaissance now that I have
discovered Csound.
Sincerely,
Kurt Nelson
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Message-ID: <923DE5722529D31186D500A0C999D9890E29@mail2.c-bridge.com>
From: Grant Covell
To: "Csound (E-mail)"
Subject: RE: timout exasperation
Date: Wed, 21 Jul 1999 21:53:58 -0400
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I've figured out my own timout exasperation. I guess it's not used much by
the group, as I got back no comments!
Here's a simple orc/sco that demonstrates using timout without reinit and
rireturn
;;sample.orc
sr = 44100
kr = 44100
ksmps = 1
nchnls = 1
instr 2
;; locations to start snip
idur = p3 - 4
imid unirand idur
iloc1 unirand imid
iloc2 unirand idur - imid
iloc2 = iloc2 + imid + 2
irem = p3 - iloc2
print imid, iloc1, iloc2, irem
;; the work
timout 0, iloc1, end
timout 0, 2, do
timout 0, iloc2, end
timout 0, 2, do
timout 0, irem, end
do:
asig oscil 5000, 400, 10
out asig
end:
endin
;; sample.sco
f10 0 8192 10 1
i2 0 15
====
> -----Original Message-----
> From: Grant Covell [mailto:GCovell@c-bridge.com]
> Sent: Tuesday, July 13, 1999 10:09 PM
> To: Csound (E-mail)
> Subject: timout exasperation
>
>
> Hello--
>
> I'm trying to use multiple timout calls within the same
> orchestra, and am
> frustrated by the somewhat inconsistent results and cryptic
> documentation.
>
> The sco/orc pair below is very simplified from what I am
> hoping to do: given
> a simple one line call in the score, trigger the playback of
> a given sound
> twice at random times within the p3. I've taken out the
> creation of the
> random start points and the reading of the existing file,
> just to focus on
> getting timout to work.
>
> The documentation specifies that timout (timout istart, idur,
> label) will
> occur at istart, last for idur and accomplish the event(s) at
> the label.
> What I have below is the most logical use of timout (without
> reinit, and
> rireturn) I've figured out so far (timout with the reinit and
> rireturn work
> best when the dur in timout is repeatable and not random--if
> someone has a
> solution for a non-periodic use of timout, I'd love to see
> it!). I see how
> timout can be reinitialized at endin, but still think that
> istart restting
> to 0 is somewhat odd. Also, the call to the last event
> appears to continue
> beyond 13 secs. extending itself to the full p3.
>
> Thanks for the help and advice!
>
> Grant.
> gcovell@c-bridge.com
>
>
> ===
> ;;score:
>
> f10 0 8192 10 1
> i1 0 14 1
> e
> ;; end of score
>
>
> ;;orchestra:
>
> sr = 44100
> kr = 441
> ksmps = 100
> nchnls = 2
>
>
> instr 1
>
> timout 0, 3, end ;; ends at 3
> timout 3, 2, do ;; start at 3, end at
> 5, reset to 0
> timout 0, 5, end ;; start at 5 end at 10
> timout 5, 2, do ;; start at 10 end at
> 12, reset to 0
> timout 0, 1, end ;; start at 12 and at 13
>
> do:
> asig oscil 5000, 400, 10
> outs asig, asig
>
> end:
> endin
> ;; end of orchestra
>
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Date: Wed, 21 Jul 1999 22:40:16 -0500
To: "Matt J. Ingalls"
From: Daniel Nass
Subject: Re: table and phasor
Cc: csound@maths.ex.ac.uk
In-Reply-To:
References: <4.1.19990721090112.00982520@cctr.umkc.edu>
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Thanks for you reply, Matt. Unfortunately, I'm still a little thick-headed.
I am making sure the table size is greater than that of the file (there
are 106752 samples in the file, so my table size should be 131072, right?).
The only way I seem to be able to avoid this problem is to add silence so
that the length of the file is EXACTLY 131072 samples in the file, even if
the last 25000 or so are only silence. It also seems that it puts the huge
clip at the exact point the original soundfile ends.
Let me try to explain better. My original file that I'm calling in with
GEN01 is 106752 samples long. The resultant file, after using table and
GEN 01 to call in the file is 131072 samples long with a huge SNAP at
106752. This happens with just about any sample I try and use, unless the
sample's length is close to a power of 2. Sorry if this doesn't make sense
but let me know if you have any other advice.
Thanks again,
Dan
P.S. Here is the orc and sco that I'm using:
;ORCHESTRA
sr=44100
kr=4410
ksmps=10
nchnls=1
instr 1
kenv linen 20000, .01, p3, .01
aph phasor 1/p3
a1 table aph, 1, 1
a1 = a1 * kenv
out a1
endin
;SCORE
f1 0 131072 1 5 0 4 1
i1 0 2.972
*************************************************************
Daniel Nass http://cctr.umkc.edu/~dnass/
"Wherever we are, what we hear is mostly noise.
When we ignore it, it disturbs us. When we listen
to it, we find it fascinating."
John Cage (1961)
*************************************************************
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From: Stan Olejarz
To: "Matt J. Ingalls" , Daniel Nass
Cc: csound@maths.ex.ac.uk
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.omphalos
Subject: Re: table and phasor
Date: Wed, 21 Jul 1999 23:55:19 -0700
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Just a quick question from a rookie Csounder.How do you calculate the file
size?
Ho
-----Original Message-----
From: Daniel Nass
To: Matt J. Ingalls
Cc: csound@maths.ex.ac.uk
Date: Wednesday, July 21, 1999 8:44 PM
Subject: Re: table and phasor
>Thanks for you reply, Matt. Unfortunately, I'm still a little
thick-headed.
>
>I am making sure the table size is greater than that of the file (there
>are 106752 samples in the file, so my table size should be 131072, right?).
> The only way I seem to be able to avoid this problem is to add silence so
>that the length of the file is EXACTLY 131072 samples in the file, even if
>the last 25000 or so are only silence. It also seems that it puts the huge
>clip at the exact point the original soundfile ends.
>
>Let me try to explain better. My original file that I'm calling in with
>GEN01 is 106752 samples long. The resultant file, after using table and
>GEN 01 to call in the file is 131072 samples long with a huge SNAP at
>106752. This happens with just about any sample I try and use, unless the
>sample's length is close to a power of 2. Sorry if this doesn't make sense
>but let me know if you have any other advice.
>
>Thanks again,
>
>Dan
>
>
>P.S. Here is the orc and sco that I'm using:
>
>;ORCHESTRA
>sr=44100
>kr=4410
>ksmps=10
>nchnls=1
>
>instr 1
>kenv linen 20000, .01, p3, .01
>aph phasor 1/p3
>a1 table aph, 1, 1
>a1 = a1 * kenv
>out a1
>
>endin
>
>
>
>;SCORE
>f1 0 131072 1 5 0 4 1
>
>i1 0 2.972
>
>*************************************************************
>
>Daniel Nass http://cctr.umkc.edu/~dnass/
>
>"Wherever we are, what we hear is mostly noise.
>When we ignore it, it disturbs us. When we listen
>to it, we find it fascinating."
>
> John Cage (1961)
>
>*************************************************************
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To: Stan Olejarz
CC: CSOUND list
Subject: Re: table and phasor
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Stan Olejarz wrote:
>
> Just a quick question from a rookie Csounder.How do you calculate the file
> size?
use sndinfo, a great utility that comes with csound.
For instance:
[pw@slink drums]$ sndinfo church_spirit_kick.wav
church_spirit_kick.wav: WAVE, 15082 samples
WAVE soundfile
srate 44100, monaural, 16 bit shorts, 0.34 seconds
headersiz 44, datasiz 30164 (15082 sample frames)
If you insist on doing it by hand... let's see if I can do it right:
file size in bytes = (length in seconds * sampling rate) * (bits / 8) *
(number of channels)
e.g. for a 5-second, 16-bit, 44.1kHz, 2-channel file, you get:
5 * 44100 * (16/8) *2 = 882000 bytes.
But I think you really want to know the number of samples. That's just
this:
length in seconds * sampling rate
e.g. 5 * 44100 = 220500 samples.
---------------- paul winkler ------------------
slinkP arts: music, sound, illustration, design, etc.
zarmzarm@hotmail.com --or-- slinkp AT ulster DOT net
http://www.ulster.net/~abigoo/
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Message-ID: <792269ee.24c837b2@aol.com>
Date: Thu, 22 Jul 1999 05:00:34 EDT
Subject: Re: new Csounder
To: csound@maths.ex.ac.uk
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Hi list, hi Kurt,
<< I find my ideas have been too exacting and labor intensive
for any effective collaborations.>>
Thats a common problem I guess, particulary with non realtime systems. But I
hope it changes with the raising of computing power and realtimism.
<< written for the Commodore 64 for years>>
I love it, my first synth, followed 1986 by a Korg MS20.
<< I began to breadboard a pulse code modulation control circuit to vary the
speed of a tapedeck motor. >>
Sounds great & crazy, you should finish it! It will sound unique,
mellotronique?!??!.
<< A CD on the history of digital computer synthesis from my
public library opened me to the world of Max V. Mathews, John Pierce et
al.>>
The Vergo CD with pioneers of digital music inc. Bicycle build for two? I
heard it first time at IRCAMs archive at my Paris trip past september. I
thought I will never hear it again but then I discoverd it at EMF where I
bought it. Sounds a lot of like done with PC speaker ;) , but it was recorded
a bit different.
<<...publishable paper, but I am working on a website. Since this is
probably
an off-topic issue, I won't bring it up on this list again>>
But give us the URL.
welcome to the list,
Malte
---------------------------
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