/******** Csnd.app v2.0 README ********/ Stephen David Beck School of Music Louisiana State University Baton Rouge, LA 70803 USA voice: 1-(504)-388-3261 fax: 1-(504)-388-2562 email: sdb@comp.music.lsu.edu This is a beta release of Csnd.v2.0.app, a NEXTSTEP GUI to MIT/Bath Csound. This distribution contains the following: 1) A fat binary (NeXT/Intel only) Csnd.v2.0.app application 2) A fat csound binary (NeXT/Intel only) compiled using the MIT/Bath sources (v3.41) 3) The file README.csound.newopcodes 4) This README file To install, login as root, place Csnd.v2.0.app.tar.gz in your /LocalApps directory and unzip the file with the following commands: cd /LocalApps gunzip -cd Csnd.v2.0.app.tar.gz | tar -xf - Place the Csnd.v2.0.app in your /LocalApps directory. Be sure your system has the Csound binary located in a public path (i.e., /usr/local/bin). This is strictly a beta release. No warranties, no guarantees, no nothin'! I don't have access to an Intel NEXTSTEP machine, so I'm only about 80% confident that the MAB files work. (i.e. I hope it works...) Our NeXT cube is on its last legs, and so is the development of this program. It has not been tested by my students, and I haven't done much work on it since Fall (1995). But I've had a few days to finish up some stuff, and to make sure everything works. On the other hand, Csnd.v2.0.app has some really cool improvements, like a file inspector (sound files, text files, lpc analysis files), a compiler module (will automatically set flags for midi, extract and scot files), and a much improved GUI for soundfile analysis. Please send your comments, your thanks, and any suggestions. If GNUstep/OPENSTEP takes off, or someone donates an Intel box to the school (wink, wink, nudge, nudge), I might work on it further. Changes since v1.6 - file inspector built-in (sound, text and lpc analysis files) - separate compiler module (will automatically set flags for midi, extract and scot files if present) - improved interface for soundfile analysis - fixed bug of /tmp/score.srt problems (important if multiple users on one machine) - added kill buttons for compiler Changes since 1.5 - MAB csound binary - Preliminary version of info panel for lpc anal files (shows graphs of Pitch, RMS (original and residual) and Error, and displays header data) - More bug fixes Changes since 1.1 - Support of AIFF files for analysis and playback (read-only) - Improved Preferences Manager - MAB for Intel and NeXT hardware (Csnd.app only; csound binary is machine specific) - Messages selected editor for MIDI files - Many bug fixes (I'm sure there are still more) Changes since 1.0 - Compile sounds to floating point - Use Midi files & extract files in compile - Import Midi files into project - Upgrade to March 1994 csound binary - Bug fixes in subprocess manager Csnd.app has several distinct differences from Snd.app: - It can access files from any available directory; - Csnd.app creates a separate process to compile files, returning control to the user as soon as the compiling begins; - It has services to send the following messages from Edit to Csnd.app: - recompile last orc & sco file import file to project - search MIT Csound Quick Reference for selection - open standard NeXT help with MIT Csound Manual - Double-click on browser calls appropriate edit program for file Csnd.app also takes advantage of the analysis programs built-in to MIT Csound, and provides an intuitive interface to them. Some of the analysis features are: - Soundfile display and playback - Analysing all or only selected portion of soundfile - Analysis generated by Csound are automatically imported to projects upon completion. For more information, e-mail: sdb@comp.music.lsu.edu Stephen David Beck School of Music Louisiana State University
------------------------------------------------------------------------
Appendix 7 : Newest Csound opcodes
by
Paris Smaragdis
Berklee College of Music
This appendix describes recent additions to Csound. These
additions include a granular synthesis synthesizer, a new set of
filters, a new variable delay, a multitap delay, a new reverb, an
envelope follower, various noise generators, a power function
generator and two gen routines, GEN20 and GEN21.
1) Granular synthesizer.
ar grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, iwfn, imgdur
Generates granular synthesis textures.
INITIALIZATION
igfn, igdur - igfn is the ftable number of the grain waveform. This
can be just a sine wave or a sampled sound of any length. Each grain
will start from a random table position and sustain for igdur seconds.
iwfn - Ftable number of the amplitude envelope used for the grains
(see also GEN20).
imgdur - Maximum grain duration in seconds. This the biggest value to
be assigned on kgdur.
PERFORMANCE
xamp - Total amplitude of the sound.
xpitch - Grain frequency in cps.
xdens - Density of grains measured in grains per second. If this is
constant then the output is synchronous granular synthesis, very
similar to fof. If xdens has a random element (like added noise),
then the result is more like asynchronous granular synthesis.
kampoff - Maximum amplitude deviation from kamp. This means that the
maximum amplitude a grain can have is kamp + kampoff and the minimum
is kamp. If kampoff is set to zero then there is no random amplitude
for each grain.
kpitchoff - Maximum pitch deviation from kpitch in cps. Similar to
kampoff.
kgdur - Grain duration in seconds. The maximum value for this should
be declared in imgdur. If kgdur at any point becomes greater than
imgdur, it will be truncated to imgdur.
2) Butterworth filters.
ar butterhp asig, kfreq
ar butterlp asig, kfreq
ar butterbp asig, kfreq, kband
ar butterbr asig, kfreq, kband
Implementations of second-order hipass, lopass, bandpass and
bandreject Butterworth filters.
PERFORMANCE
These new filters are butterworth second-order IIR filters. They are
slightly slower than the original filters in Csound, but they offer an
almost flat passband and very good precision and stopband attenuation.
asig - Input signal to be filtered.
kfreq - Cuttoff or center frequency for each of the filters.
kband - Bandwidth of the bandpass and bandreject filters.
EXAMPLE
asig rand 10000 ; White noise signal
alpf butterlp asig, 1000 ; cutting frequencies above1K
ahpf butterhp asig, 500 ; passing frequencies above 500Hz
abpf butterbp asig, 2000, 100 ; passing only 1950 to 2050 Hz
abrf butterbr asig, 4500, 200 ; cutting only 4400 to 4600 Hz
3) Vdelay
ar vdelay asig, adel, imaxdel
This is an interpolating variable time delay, it is not very different
from the existing implementation (deltapi), it is only easier to use.
INITIALIZATION
imaxdel - Maximum value of delay in samples. If adel gains a value
greater than imaxdel it is folded around imaxdel. This should not
happen.
PERFORMANCE
With this unit generator it is possible to do Doppler effects or
chorusing and flanging.
asig - Input signal.
adel - Current value of delay in samples. Note that linear functions
have no pitch change effects. Fast changing values of adel will cause
discontinuities in the waveform resulting noise.
Example
f1 0 8192 10 1
ims = 100 ; Maximum delay time in msec
a1 oscil 10000, 1737, 1 ; Make a signal
a2 oscil ims/2, 1/p3, 1 ; Make an LFO
a2 = a2 + ims/2 ; Offset the LFO so that it is positive
a3 vdelay a1, a2, ims ; Use the LFO to control delay time
out a3
Two important points here. First, the delay time must be always
positive. And second, even though the delay time can be controlled in
k-rate, it is not advised to do so, since sudden time changes will
create clicks.
4) Multitap delay
ar multitap asig, itime1, igain1, itime2, igain2 . . .
Multitap delay line implementation.
INITIALIZATION
The arguments itime and igain set the position and gain of each tap.
The delay line is fed by asig.
Example:
a1 oscil 1000, 100, 1
a2 multitap a1, 1.2, .5, 1.4, .2
out a2
This results in two delays, one with length of 1.2 and gain of .5, and
one with length of 1.4 and gain of .2.
5) Reverb2
ar reverb2 asig, ktime, khdif
This is a reverberator consisting of 6 parallel comb-lowpass filters
being fed into a series of 5 allpass filters.
PERFORMANCE
The input signal asig is reverberated for ktime seconds. The
parameter khdif controls the high frequency diffusion amount. The
values of khdif should be from 0 to 1. If khdif is set to 0 the all
the frequencies decay with the same speed. If khdif is 1, high
frequencies decay faster that lower ones.
Example:
a1 oscil 10000, 100, 1
a2 reverb2 a1, 2.5, .3
out a1 + a2 * .2
This results in a 2.5 sec reverb with faster high frequency
attenuation.
6) Envelope follower
kr follow asig, idt
Envelope follower unit generator.
INITIALIZATION
idt - This is the period, in seconds, that the average amplitude of
asig is reported. If the frequency of asig is low then idt must be
large (more than half the period of asig )
PERFORMANCE
asig - This is the signal from which to extract the envelope.
Example
k1 line 0, p3, 30000 ; Make k1 a simple envelope
a1 oscil k1, 1000, 1 ; Make a simple signal using k1
ak1 follow a1, .02 ; ak1 is now like k1
a2 oscil ak1, 1000, 1 ; Make a simple signal using ak1
out a2 ; Both a1 and a2 are the same
To avoid zipper noise, by discontinuities produced from complex
envelope tracking, a lowpass filter could be used, to smooth the
estimated envelope.
7) Noise generators
All of the following opcodes operate in i-, k- and a-rate. The output
rate depends on the first letter of the opcode, a for a-rate, k for k-
rate and i for i-rate.
xlinrand krange - Linear distribution random number generator. krange
is the range of the random numbers [0 - krange). Outputs only
positive numbers.
xtrirand krange - Same as above only ouputs both negative and
positive numbers.
xexprand krange - Exponential distribution random number generator.
krange is the range of the random numbers [0 - krange). Outputs only
positive numbers.
xbexprnd krange - Same as above, only extends to negative numbers too
with an exponential distribution.
xcauchy kalpha - Cauchy distribution random number generator. kalpha
controls the spread from zero (big kalpha => big spread). Outputs
both positive and negative numbers.
xpcauchy kalpha - Same as above, ouputs positive numbers only.
xpoisson klambda - Poisson distribution random number generator.
klambda is the mean of the distribution. Outputs only positive
numbers.
xgauss krange - Gaussian distribution random number generator. krange
is the range of the random numbers (-krange - 0 - krange). Outputs
both positive and negative numbers.
xweibull ksigma, ktau - Weibull distribution random number generator.
ksigma scales the spread of the distribution and ktau, if greater
than one numbers near ksigma are favored, if smaller than one small
values are favored and if t equals 1 the distribution is exponential.
Outputs only positive numbers.
xbeta krange, kalpha, kbeta - Beta distribution random number
generator. krange is the range of the random numbers [0 - krange). If
kalpha is smaller than one, smaller values favor values near 0. If
kbeta is smaller than one, smaller values favor values near krange.
If both kalpha and kbeta equal one we have uniform distribution. If
both kalpha and kbeta are greater than one we have a sort of gaussian
distribution. Outputs only positive numbers.
For more detailed explanation of these distributions, see:
1. C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books.
pp.265 - 286
2. D. Lorrain. "A panoply of stochastic cannons". In C. Roads, ed.
1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351
- 379.
Examples:
a1 atrirand 32000 ; Audio noise with triangle distribution
k1 kcauchy 10000 ; Control noise with Cauchy dist.
i1 ibetarand 30000, .5, .5 ; i-time random value, beta dist.
8) Power functions
ir ipow iarg, kpow
kr kpow karg, kpow, [inorm]
ar apow aarg, kpow, [inorm]
Computes xarg to the power of kpow and scales the result by inorm.
INITIALIZATION
inorm - The number to divide the result (default to 1). This is
especially useful if you are doing powers of a- or k- signals where
samples out of range are extremely common!
iarg - If you are using ipow this is the base.
PERFORMANCE
karg - If you are using kpow this is the base.
aarg - If you are using apow this is the base.
EXAMPLES:
1. i2t2 ipow 2,2 ; Computes 2^2.
2. kline line 0, 1, 4
kexp kpow kline, 2, 4
This feeds a linear function to kpow and scales that to the line's
peak value. The output will be an exponential curve with the same
range as the input line.
3. iamp ipow 10, 2
a1 oscil iamp, 100, 1
a2 apow a1, 2, iamp
out a2
This will output a sine with its negative part folded over the amp
axis. The peak value will be iamp = 10^2 = 100.
9) GEN20
This subroutine generates functions of different windows. These
windows are usually used for spectrum analysis or for grain envelopes.
f # time size 20 window max opt
size - number of points in the table. Must be a power of 2 ( + 1).
window - Type of window to generate.
1 - Hamming
2 - Hanning
3 - Bartlett ( triangle)
4 - Blackman ( 3 - term)
5 - Blackman - Harris ( 4 - term)
6 - Gaussian
7 - Kaiser
8 - Rectangle
9 - Sinc
max - For negative p4 this will be the absolute value at window peak
point. If p4 is positive or p4 is negative and p6 is missing the
table will be post-rescaled to a maximum value of 1.
opt - Optional argument required by the Kaiser window.
Examples:
f 1 0 1024 20 5
This creates a function which contains a 4 - term Blackman - Harris
window with maximum value of 1.
f 1 0 1024 -20 2 456
This creates a function that contains a Hanning window with a maximum
value of 456.
f 1 0 1024 -20 1
This creates a function that contains a Hamming window with a maximum
value of 1.
f 1 0 1024 20 7 1 2
This creates a function that contains a Kaiser window with a maximum
value of 1. The extra argument specifies how `open' the window is, for
example a value of 0 results in a rectangular window and a value of 10 in
a Hamming like window.
10) GEN21
This generates tables of different random distributions. (see also noise
generators, above).
f # time size 21 type lvl arg1 arg2
Time and size are the usual Gen function arguments. Type defines the
distribution to be used.
1 - Uniform
2 - Linear
3 - Triangular
4 - Exponential
5 - Biexponential
6 - Gaussian
7 - Cauchy
8 - Positive Cauchy
9 - Beta
10 - Weibull
11 - Poison
Of all these cases only 9 (Beta) and 10 (Weibull) need extra
arguments. Beta needs two arguments and Weibull one.
Examples:
f1 0 1024 21 1 ; Uniform (white noise)
f1 0 1024 21 6 ; Gaussian
f1 0 1024 21 9 1 1 2 ; Beta (note that level precedes arguments)
f1 0 1024 21 10 1 2 ; Weibull
All of the above additions were designed by the author between
May and December 1994, under the supervision of Dr. Richard Boulanger.
This appendix was written on 20 December 1994 by
Paris Smaragdis, Berklee College of Music.
Internet: psmaragdis@aol.com
------------------------------------------------------------------------
AUTHOR: Greg Sullivan, sullivan@aussie.enet.dec.com
(Based on algorithm given in 'Elements Of Computer Music', by
F. Richard Moore.
CVANAL - Impulse Response Fourier Analysis for CONVOLVE operator
csound -U cvanal [flags] infilename outfilename
cvanal converts a soundfile into a single Fourier transform frame. The
output file can be used by the CONVOLVE operator to perform Fast
Convolution between an input signal and the original impulse response.
Analysis is conditioned by the flags below. A space is optional between
the flag and its argument.
-s<rate> sampling rate of the audio input file. This will over-ride
the srate of the soundfile header, which otherwise applies.
If neither is present, the default is 10000.
-c<channel> channel number sought. If omitted, the default is to
process all channels. If a value is given, only the
selected channel will be processed.
-b<begin> beginning time (in seconds) of the audio segment to be
analysed. The default is 0.0
-d<duration> duration (in seconds) of the audio segment to be analysed.
The default of 0.0 means to the end of the file.
EXAMPLE
cvanal asound cvfile
will analyse the soundfile "asound" to produce the file "cvfile" for the
use with CONVOLVE.
HINT: To use data that is not already contained in a soundfile,
a soundfile converter that accepts text files may be used to
create a standard audio file. E.g, the .DAT format for SOX.
This is useful for implementing FIR filters.
FILES
The output file has a special CONVOLVE header, containing details of the
source audio file. The analysis data is stored as 'float', in rectangular
(real/imaginary) form.
***NOTE***: The analysis file is NOT system independent! Ensure that
the original impulse recording/data is retained. If/when required,
the analysis file can be recreated.
AUTHOR: Greg Sullivan, sullivan@aussie.enet.dec.com
(Based on algorithm given in 'Elements Of Computer Music', by
F. Richard Moore.
------------------------------------------------------------------------------
CONVOLVE unit generator:
ar1[,ar2[,ar3[,ar4]]] convolve ain,ifilcod,channel
Output is the convolution of signal ain and the impulse response
contained in ifilcod. Note that it is considerably more efficient to
use one instance of the operator when processing a mono input to
create stereo, or quad, outputs.
INITIALISATION
ifilcod - integer or character-string denoting an impulse response data
file. An integer denotes the suffix of a file convolve.m; a character
string (in double quotes) gives a filename, optionally a full pathname.
If not a fullpath, the file is sought first in the the current directory,
then in the one given by the environment variable SADIR (if defined).
The data file contains the Fourier transform of an impulse response.
Memory usage depends on the size of the data file, which is read and
held entirely in memory during computation, but which is shared
by multiple calls.
channel - integer denoting the channel of the impulse response to be used
for the convolution. 0 (the default) means to use all channels.
For multi-channel output, the number of channels in the impulse
response must match the number of output signals.
PERFORMANCE
CONVOLVE implements Fast Convolution. The output of this
operator is delayed with respect to the input. The following
formulas should be used to calculate the delay:
For (1/kr) <= IRdur:
Delay = ceil(IRdur * kr) / kr
For (1/kr) > IRdur:
Delay = IRdur * ceil(1/(kr*IRdur))
Where:
kr = Csound control rate
IRdur = duration, in seconds, of impulse response
ceil(n) = smallest integer not smaller than n
One should be careful to also take into account the initial delay,
if any, of the impulse response. For example, if an impulse response
is created from a recording, the soundfile may not have the initial
delay included. Thus, one should either ensure that the soundfile
has the correct amount of zero padding at the start, or, preferably,
compensate for this delay in the orchestra. (the latter method
is more efficient). To compensate for the delay in the
orchestra, _subtract_ the initial delay from the result
calculated using the above formula(s), when calculating the required
delay to introduce into the 'dry' audio path.
For typical applications, such as reverb, the delay will be in
the order of 0.5 to 1.5 seconds, or even longer. This renders
the current implementation unsuitable for real time applications.
It could conceivably be used for real time filtering however, if
the number of taps is is small enough.
Example:
- Create frequency domain impulse response file:
c:\> csound -Ucvanal l1_44.wav l1_44.cv
- Determine duration of impulse response. For high accuracy,
determine the number of sample frames in the impulse
response soundfile, and then compute the duration with:
duration = (sample frames)/(sample rate of soundfile)
This is due to the fact that the SNDINFO utility only
reports the duration to the nearest 10ms. If you have a
utility that reports the duration to the required accuracy,
then you can simply use the reported value directly.
c:\> sndinfo l1_44.wav
length = 60822 samples, sample rate = 44100
Duration = 60822/44100 = 1.379s.
- Determine initial delay, if any, of impulse response.
If the impulse response has not had the initial
delay removed, then you can skip this step. If it has
been removed, then the only way you will know the initial
delay is if the information has been provided separately.
For this example, let's assume that the initial delay is
60ms. (0.06s)
- Determine the required delay to apply to the dry signal, to align
it with the convolved signal:
If kr = 441:
1/kr = 0.0023, which is <= IRdur (1.379s), so:
Delay1 = ceil(IRdur * kr) / kr
= ceil(608.14) / 441
= 609/441
= 1.38s
Accounting for the initial delay:
delay2 = 0.06s
Total delay = delay1 - delay2
= 1.38 - 0.06
= 1.32s
- Create .orc file, e.g:
----CUT----
; Simple demonstration of CONVOLVE operator, to apply reverb.
sr = 44100
kr = 441
ksmps = 100
nchnls = 2
instr 1
imix = 0.22 ; Wet/dry mix. Vary as desired.
; NB: 'Small' reverbs often require a much higher
; percentage of wet signal to sound interesting. 'Large'
; reverbs seem require less. Experiment! The wet/dry mix is
; very important - a small change can make a large difference.
ivol = 0.9 ; Overall volume level of reverb. May need to adjust
; when wet/dry mix is changed, to avoid clipping.
idel = 1.32 ; Required delay to align dry audio with output of convolve.
; This can be automatically calculated within the orc file,
; if desired.
adry soundin "anechoic.wav" ; input (dry) audio
awet1,awet2 convolve adry,"l1_44.cv" ; stereo convolved (wet) audio
adrydel delay (1-imix)*adry,idel ; Delay dry signal, to align it with
; convolved signal. Apply level
; adjustment here too.
outs ivol*(adrydel+imix*awet1),ivol*(adrydel+imix*awet2)
; Mix wet & dry signals, and output
endin
---CUT----
The granule unit generator (Allan Lee) is more complex than grain
(above), but does add new possibilities. This is a shorten manual,
without the pictures,
NAME
granule - Granular synthesis unit generator for Csound.
SYNOPSIS
granule xamp ivoice iratio imode ithd ifn ipshift igskip \
igskip_os ilength kgap igap_os kgsize igsize_os iatt idec [iseed] \
[ipitch1] [ipitch2] [ipitch3] [ipitch4] [ifnenv]
DESCRIPTION
granule is a Csound unit generator which employs a wavetable as input
to produce granularly synthesized audio output. Wavetable data may be
generated by any of the gen subroutines such as gen01 which reads an
audio data file into a wavetable. This enable a sampled sound to be
used as the source for the grains. Up to 128 voices are implemented
internally.The maximum number of voices can be increased by redefining
the variable MAXVOICE in the grain4.h file. granule has a build-in
random number generator to handle all the random offset parameters.
Thresholding is also implemented to scan the source function table at
initialization stage. This facilitates features such as skipping
silence passage between sentences.
The characteristics of the synthesis are controlled by 22
parameters. xamp is the amplitude of the output and it can be either
audio rate or control rate variable. All parameters with prefix i must
be valid at Init time, parameters with prefix k can be either control
or Init values.
SUMMARY OF PARAMETERS
xamp - amplitude.
ivoice - number of voices.
iratio - ratio of the speed of the gskip pointer relative to output
audio sample rate. eg. 0.5 will be half speed.
imode - +1 grain pointer move forward (same direction of the gskip
pointer), -1 backward (oppose direction to the gskip pointer) or 0 for
random.
ithd - threshold, if the sampled signal in the wavetable is smaller
then ithd, it will be skipped.
ifn - function table number of sound source.
ipshift - pitch shift control. If ipshift is 0, pitch will be set
randomly up and down an octave. If ipshift is 1, 2, 3 or 4, up to four
different pitches can be set amount the number of voices definded in
ivoice. The optional parameters ipitch1, ipitch2, ipitch3 and ipitch4
are used to quantify the pitch shifts.
igskip - initial skip from the beginning of the function table in sec.
igskip_os - gskip pointer random offset in sec, 0 will be no offset.
ilength - length of the table to be used starting from igskip in sec.
kgap - gap between grains in sec.
igap_os - gap random offset in % of the gap size, 0 gives no offset.
kgsize - grain size in sec.
igsize_os - grain size random offset in % of grain size, 0 gives no
offset.
iatt - attack of the grain envelope in % of grain size.
idec - decade of the grain envelope in % of grain size.
[iseed] - optional, seed for the random number generator, default is
0.5.
[ipitch1], [ipitch2], [ipitch3], [ipitch4] - optional, pitch shift
parameter, used when ipshift is set to 1, 2, 3 or 4. Time scaling
technique is used in pitch shift with linear interpolation between
data points. Default value is 1, the original pitch.
EXAMPLE
Listing of orchestra file.
sr = 44100
kr = 4410
ksmps = 10
nchnls = 2
instr 1
;
k1 linseg 0,0.5,1,(p3-p2-1),1,0.5,0
a1 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,p16,p17,p18,p19,
p20,p21,p22,p23,p24
a2 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,p16,p17,p18,p19,
p20+0.17,p21,p22,p23,p24
outs a1,a2
endin
Listing of score file.
; f statement read sound file sine.aiff in the SFDIR directory into f-table 1
f 1 0 524288 1 "sine.aiff" 1 0
i1 0 10 2000 64 0.5 0 0 1 4 0 0.005 10 0.01 50 0.02 50 30 30 0.39 1 1.42 0.29 2
e
The above example reads a sound file called sine.aiff into wavetable
number 1 with 524,288 samples. It generates 10 seconds of stereo audio
output using the wavetable. In the orchestra file, all parameters
required to control the synthesis are passed from the score file. A
linseg function generator is used to generate an envelope with 0.5
second of linear attack and decade. Stereo effect is generated by
using different seeds for the two granule function calls. In the
example, 0.17 is added to p20 before passing into the second granule
call to ensure that all of the random offset events are different from
the first one.
In the score file, the parameters are interpreted as:
p5 (ivoice) the number of voices is set to 64
p6 (iratio) is set to 0.5, it scan the wavetable at half of the speed
of the audio output rate
p7 (imode) is set to 0, the grain pointer only move forward
p8 (ithd) is set to 0, skipping the thresholding process
p9 (ifn) is set to 1, function table number 1 is used
p10 (ipshift) is set to 4, four different pitches are going to be
generated
p11 (igskip) is set to 0 and p12 (igskip_os) is set to 0.005, no
skipping into the wavetable and a 5 mSec random offset is used
p13 (ilength) is set to 10, 10 seconds of the wavetable is to be used
p14 (kgap) is set to 0.01 and p15 (igap_os) is set to 50, 10 mSec gap
with 50% random offset is to be used
p16 (kgsize) is set to 0.02 and p17 (igsize_os) is set to 50, 20 mSec
grain with 50% random offset is used
p18 (iatt) and p19 (idec) are set to 30, 30% of linear attack and
decade is applied to the grain
p20 (iseed) seed for the random number generator is set to 0.39
p21 - p 24 are pitches set to 1 which is the original pitch, 1.42
which is a 5th up, 0.29 which is a 7th down and finally 2 which is
an octave up.
Csound is developed by Barry L. Vercoe at the Experimental Music Studio, Media Laboratory, M.I.T., Cambridge, Massachusetts.
These are the contents of the former NiCE NeXT User Group NeXTSTEP/OpenStep software archive, currently hosted by Netfuture.ch.