; This .csd is displayed best with tab set to 56 pt or 8 spaces sr = 44100 ksmps = 1 nchnls = 16 ; set to 2 channels for listening, setting it to 9/16 generates ambisonic encoded files only to be decoded later. ; 9 chnls = 2nd order encode, 16 chnls= 3rd order encode - for nchnls 9/16 this .csd will probably not run in real-time zakinit 56, 60 ; isizea, isizek ; SOUNDFILE READ-IN SETTINGS #define INFILE #Filename-# ; insert filename for input here (Example: "File-" if the filename is "File-1.aif") #define HEADER_INFILE #aif# ; insert filetype for input file here (Example: "aif" if the filename is "Test-1.aif") giwsize = 4 ; iwsize of diskin2 in instr. 50 -- interpolation window size, in samples. ; Can be one of the following: ; 1: round to nearest sample (no interpolation, for kpitch=1) ; 2: linear interpolation ; 4: cubic interpolation ; >= 8: iwsize point sinc interpolation with anti-aliasing (slow) ; SOUNDFILE OUTPUT SETTINGS #define FILE #Filename-# ; insert filename for name of output file here (note that the extension is just for the ; generation of the filename, the header of the file is specified below. #define HEADER #aif# ; insert filetype for output file name here giformat = 8 ; 1 = 16 bit integers without header (binary PCM multichannel file) for "fout" ; 2 = 16-bit integers with a header. ; The header type depends on the render format. ; The default header type is the IRCAM format. ; If the user chooses the AIFF format (using the -A flag), ; the header format will be a AIFF type. If the user chooses the WAV ; format (using the -W flag), the header format will be a WAV type. ; 8 = 24 bit integer with header like specified in options ;DEFINITION OF THE SIZE [UNITS] OF THE VIRTUAL PERFORMANCE ROOM FOR EARLY REFLECTIONS AND DOPPLER EFFECT gifrontwall = 1.6 ; position of frontwall at the x-axis [units] gisidewall = 1.3 ; position of sidewall at the y-axis [units] giceiling = 1.1 ; height of the ceiling measured from the zero-point of the coordinate-system [units] giunit = 4 ; radius of the "unit- sphere" (distance of the speaker from the center- point [m]) gidamp = 0.1 ; damping factor of the reflecting surface (gain of reverb) gihighdamp = 15000 ; absorbtion characteristics of the wall in respect to high frequencies: ; frequencies above ihighdamp [Hz] get filtered out gilfboost = 1 ; low frequency boost for sources near a wall ; GLOBAL SCALING OF THE AMPLITUDE girescale = 1 ; no amplitude scaling for test decoding ; girescale = 2.26 ; amplitude scaling for 2chnl decoding ; girescale = 1.4223 ; amplitude scaling for 16-chnl encoding ; ******************************************************************************************************************************* ; Version 2.0 , Dez 2009 coded 1999-2009 by Jan Jacob Hofmann ; jjh@sonicarchitecture.de +++ http://www.sonicarchitecture.de ; Comments and suggestions welcome. ; For non-comercial, artistic, scientific and educational use only - all rights reserved ; ******************************************************************************************************************************* ; This is a sequence of instruments for periphonic spatialisation of any sound input using the 3rd order Ambisonics ; and the Furse- Malham set of equations. It was conceived and coded 1999-2009 by Jan Jacob Hofmann. ; The Instrument is able to spatialize up to 20 soundsources simultaneously (this can be enhanced). To each ; source a trajectory is assigned, so independent movement of the sounds is possible. ; Besides assigning a position in space via Ambisonics, several distance clues are coded separately to enhance ; the perception of distance and depth. These are: lowpassfiltering according to distance, attenuation ; according to distance, ratio of global and local reverb, ratio of reverberated and direct sound, first 6 specular ; early reflections with position and energy of reflections changing according to the position of the sound in space, ; first 6 diffuse early reflections with position and energy of reflections changing according to the position of the sound in space. ; The early reflections also do increase the transparency of the perceived image. Also Doppler Frequency Shift is ; applied to the source. ; Distribution of the sound is for multi- speaker arrays. ; The speakers have to be same type, in-phase, and at the exact geometric location for which the decoding ; was made for. ; SSETTING UP A QUICK TEST ; To get a quick overview please use the soundfile "test1" (e.g. Dr. Boulangers Speech) ; and put them into the "Sound Sample Directory". Now make 7 copies of the file "Test1" and ; name them "test2" through "test8". They represent your audio tracks to be fed into the "Soundfile Reader". Then set the number ; of decoding- channels ("nchnls ") to "2" for stereo listening . ; make sure that you specify for every track a continouus trajectory in Instr. 42/46 ; Now render. If you have chosen more speakers than two, it is important to pass the generated encoded files to the decoder (see "Ambidec.orc") ; and decode according to your listening conditions (stereo? eight speakers? sixteen?). it shoud generate files that ; can be read by your sound editor. Import them into your editor and assign the channels to the speakers according to the position ; of the speaker. See "Ambidec.orc" for correct assignment. Now play the soundfiles simultaneously, each feeding one speaker. ; HOW TO USE THIS ORCHESTRA ; This orchestra works in several steps: ; - First a trajectory is generated (xyz is converted to angle, elevation and distance automatically by instr. 42) ; - Optional: trajectory may be modulated (instr 47 +48) ; - the soundfile is read, Doppler shift is applied (instr 50) ; - Local reverb is generated (instr 60) ; - The sound is spatially encoded, local reverb is added to the sound (instr 72) ; - Global reverb is added ; - Early reflections are added (instr 74+75) ; - Sound is collected and written to disk (instr 80) ; You may read up to 20 tracks into the orchestra simultaneously. The name of the track has to be specified in instr 50. ; To each track you have to generate a trajectory by using either Cartesian coordinates (instr. 40) or polar co-ordinates (instr 46). ; You may divide up the total time of the soundfile for several trajectories and positions in space, but these must sum up at the end to ; an equivalent to the total duration of the track processed. There must be neither overlaps nor gaps in the sequence of trajectories generated, ; one trajectory should follow the other. you may modulate any trajectory by any amount using the "Modulator" (instr. 47) or the ; Randomiser (instr. 48) This modulation may be used at any time. Also multiple instances of this modulations are possible to generate ; complex movements in space automatically by the sum of the modulating waveforms. Instr. 50 is the soundfile reader, you can specify the ; duration and track -number of the soundfile to be processed. This Instrument also triggers the local reverb and the first reflections (Instr. 74 and 75) ; adjust the parameters to taste. ; Instr. 73 is the Global Spatial Reverb. Reverberation occurs from 12 different positions in space, equaly distributed at the unit-sphere. It also ; triggers the output (Instr. 80), so make sure it is running all the time. ; Instr 74 and 75 do add specular and diffuse early reflections, also spatially distributed to each track independently. To save computing ; time these instruments should be activated only for the final rendering. ; Note: The output soundfiles may distort. You can scale them by dividing 32768 by the highest peak of them and put the result into ; "girescale", close to the header. ; ************************************************************************************** ; Please send comments and bug reports to jjh@sonicarchitecture.de on http://www.sonicarchitecture.de you will also find a small Ambisonic-tutorial. ; More info: ; To find out more about Ambisonics, see ; http://www.ambisonic.net/ ; http://www.york.ac.uk/inst/mustech/3dkpeak_audio/ambison.htm (University of York) ; http://www.york.ac.uk/inst/mustech/3dkpeak_audio/ambison.htm (Richard Furse's pages) ; Credits: ; Thanks to Richard Furse and Dave Malham for the wonderful 2nd order equations. Also special thanks to Nora Bluhme, ; Ingo Herwig and Christoph Schuller for helping me with the math. Last not least thanks to all csounders who made this ; possible by their contribution of work and thoughts!!! ; ************************************************************************************************ ; ************************************************************************************************ ; ************************************************************************************************ #define LocalGlobalREVERB ; define LOCAL ( TRACK APPLIED) and GLOBAL REVERB # ; 8 delay line FDN reverb, with feedback matrix based upon ; physical modeling scattering junction of 8 lossless waveguides ; of equal characteristic impedance. Based on Julius O. Smith III, ; "A New Approach to Digital Reverberation using Closed Waveguide ; Networks," Proceedings of the International Computer Music ; Conference 1985, p. 47-53 (also available as a seperate ; publication from CCRMA), as well as some more recent papers by ; Smith and others. ; ; Coded by Sean Costello, October 1999/ modified by Jan Jacob Hofmann August 2005 ; Note: arevsig is the global input to the reverb. afilt1 init 0 afilt2 init 0 afilt3 init 0 afilt4 init 0 afilt5 init 0 afilt6 init 0 afilt7 init 0 afilt8 init 0 afilt9 init 0 afilt10 init 0 afilt11 init 0 afilt12 init 0 ; Delay times chosen to be prime numbers. ; Works with sr=44100/ sr=48000 ONLY. If you wish to ; use a different delay time, find some new ; prime numbers that will give roughly the ; same delay times for the new sampling rate. ; Or adjust to taste. if sr = 48000 goto delaytime48000 ; delaytime chosen for sr=44100 idel1 = (1237.000/sr) idel2 = (1381.000/sr) idel3 = (1609.000/sr) idel4 = (1777.000/sr) idel5 = (1951.000/sr) idel6 = (2063.000/sr) idel7 = (1069.000/sr) idel8 = ( 967.000/sr) idel9 = (1889.000/sr) idel10 = (1693.000/sr) idel11 = (1453.000/sr) idel12 = (1291.000/sr) goto contin delaytime48000: ; delaytime for sr=48000 idel1 = (1361.000/sr) idel2 = (1499.000/sr) idel3 = (1753.000/sr) idel4 = (1933.000/sr) idel5 = (2129.000/sr) idel6 = (2243.000/sr) idel7 = (1163.000/sr) idel8 = (1051.000/sr) idel9 = (2053.000/sr) idel10 = (1693.000/sr) idel11 = (1847.000/sr) idel12 = (1409.000/sr) contin: igain = irvbdecay ; gain of reverb. Adjust empirically ; for desired reverb time. .6 gives ; a good small "live" room sound, .8 ; a small hall, .9 a large hall, ; .99 an enormous stone cavern. ipitchmod = irandpchmod ; amount of random pitch modulation ; for the delay lines. 1 is the "normal" ; amount, but this may be too high for ; held pitches such as piano tones. ; Adjust to taste. itone = icutofffreq ; Cutoff frequency of lowpass filters ; in feedback loops of delay lines, ; in Hz. Lower cutoff frequencies results ; in a sound with more high-frequency ; damping. ; k1-k8 are used to add random pitch modulation to the ; delay lines. Helps eliminate metallic overtones ; in the reverb sound. k1 randi .001, 3.1, .06 k2 randi .0011, 3.5, .9 k3 randi .0017, 1.11, .7 k4 randi .0006, 3.973, .3 k5 randi .001, 2.341, .63 k6 randi .0011, 1.897, .17 k7 randi .0017, 0.891, .19 k8 randi .0006, 3.221, .44 k9 randi .001, 1.891, .88 k10 randi .0011, 2.317, .95 k11 randi .0017, 1.091, .36 k12 randi .0006, 2.821, .41 ; apj is used to calculate "resultant junction pressure" for ; the scattering junction of e.g. 8 lossless waveguides ; of equal characteristic impedance. If you wish to ; add more delay lines, simply add them to the following ; equation, and replace the .25 by 2/N, where N is the ; number of delay lines. apj = .16666666667 * (afilt1 + afilt2 + afilt3 + afilt4 + afilt5 + afilt6 +afilt7 + afilt8 + afilt9 + afilt10 +afilt11 + afilt12) adum1 delayr 1 adel1 deltapi idel1 + k1 * ipitchmod delayw arevsig + apj - afilt1 adum2 delayr 1 adel2 deltapi idel2 + k2 * ipitchmod delayw arevsig + apj - afilt2 adum3 delayr 1 adel3 deltapi idel3 + k3 * ipitchmod delayw arevsig + apj - afilt3 adum4 delayr 1 adel4 deltapi idel4 + k4 * ipitchmod delayw arevsig + apj - afilt4 adum5 delayr 1 adel5 deltapi idel5 + k5 * ipitchmod delayw arevsig + apj - afilt5 adum6 delayr 1 adel6 deltapi idel6 + k6 * ipitchmod delayw arevsig + apj - afilt6 adum7 delayr 1 adel7 deltapi idel7 + k7 * ipitchmod delayw arevsig + apj - afilt7 adum8 delayr 1 adel8 deltapi idel8 + k8 * ipitchmod delayw arevsig + apj - afilt8 adum9 delayr 1 adel9 deltapi idel9 + k9 * ipitchmod delayw arevsig + apj - afilt9 adum10 delayr 1 adel10 deltapi idel10 + k10 * ipitchmod delayw arevsig + apj - afilt10 adum11 delayr 1 adel11 deltapi idel11 + k11 * ipitchmod delayw arevsig + apj - afilt11 adum12 delayr 1 adel12 deltapi idel12 + k12 * ipitchmod delayw arevsig + apj - afilt12 ; 1st order lowpass filters in feedback ; loops of delay lines. afilt1 tone adel1 * igain, itone afilt2 tone adel2 * igain, itone afilt3 tone adel3 * igain, itone afilt4 tone adel4 * igain, itone afilt5 tone adel5 * igain, itone afilt6 tone adel6 * igain, itone afilt7 tone adel7 * igain, itone afilt8 tone adel8 * igain, itone afilt9 tone adel9 * igain, itone afilt10 tone adel10 * igain, itone afilt11 tone adel11 * igain, itone afilt12 tone adel12 * igain, itone ; The outputs of the delay lines are summed ; and sent to the stereo outputs. This could ; easily be modified for a 4 or 8-channel ; sound system. arevsig sum afilt1 , afilt2 , afilt3 , afilt4 , afilt5 , afilt6 , afilt7 , afilt8, afilt9 , afilt10 , afilt11 , afilt12 # ; ************************************************************************************** opcode Encoding_2nd_Order,0,akk asig, kA, kE xin aw, ax, ay, az, ar, as, at, au, av bformenc1 asig, kA, kE ;2nd order Ambisonics encoding zawm aw, 41 ; rewrite and mix zak-location 41 with ambisonics w-channel output zawm ax, 42 ; rewrite and mix zak-location 42 with ambisonics x-channel output zawm ay, 43 ; rewrite and mix zak-location 43 with ambisonics y-channel output zawm az, 44 ; rewrite and mix zak-location 44 with ambisonics z-channel output zawm ar, 45 ; rewrite and mix zak-location 45 with ambisonics r-channel output zawm as, 46 ; rewrite and mix zak-location 46 with ambisonics s-channel output zawm at, 47 ; rewrite and mix zak-location 47 with ambisonics t-channel output zawm au, 48 ; rewrite and mix zak-location 48 with ambisonics u-channel output zawm av, 49 ; rewrite and mix zak-location 49 with ambisonics v-channel output endop ; ************************************************************************************** opcode Encoding_3rd_Order,0,akk asig, kA, kE xin aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc1 asig, kA, kE ;3rd order Ambisonics encoding zawm aw, 41 ; rewrite and mix zak-location 41 with ambisonics w-channel output zawm ax, 42 ; rewrite and mix zak-location 42 with ambisonics x-channel output zawm ay, 43 ; rewrite and mix zak-location 43 with ambisonics y-channel output zawm az, 44 ; rewrite and mix zak-location 44 with ambisonics z-channel output zawm ar, 45 ; rewrite and mix zak-location 45 with ambisonics r-channel output zawm as, 46 ; rewrite and mix zak-location 46 with ambisonics s-channel output zawm at, 47 ; rewrite and mix zak-location 47 with ambisonics t-channel output zawm au, 48 ; rewrite and mix zak-location 48 with ambisonics u-channel output zawm av, 49 ; rewrite and mix zak-location 49 with ambisonics v-channel output zawm ak, 50 ; rewrite and mix zak-location 50 with ambisonics k-channel output zawm al, 51 ; rewrite and mix zak-location 51 with ambisonics l-channel output zawm am, 52 ; rewrite and mix zak-location 52 with ambisonics m-channel output zawm an, 53 ; rewrite and mix zak-location 53 with ambisonics n-channel output zawm ao, 54 ; rewrite and mix zak-location 54 with ambisonics o-channel output zawm ap, 55 ; rewrite and mix zak-location 55 with ambisonics p-channel output zawm aq, 56 ; rewrite and mix zak-location 56 with ambisonics q-channel output endop ; ************************************************************************************************ ; ************************************************************************************************ ; ************************************************************************************************ ; *********************************************************************************************** instr 40 ;x,y,z -READER coded 1999-2001 by Jan Jacob Hofmann. ; *********************************************************************************************** itracknr = p4 ; number of the track the trajectory is applied on ixstart = p5 ; position of x-start ixend = p6 ; position of x-end ixmodfn = abs(p6) ; modulation function input ixmoddir = p6 / ixmodfn ; negative fn causes backwards reading ixmovefn = abs(p7) ; x movement function nr ixmovedir = p7 / ixmovefn ; negative fn causes backwards reading ixmodfn = abs(p8) ; fn for modulation ixmoddir = p8 / ixmodfn ; negative fn causes backwards reading ixmodfncps = p9 ; x movement function cps ixmodfactor = p10 ; x amount of movement iystart = p11 ; position of y-start iyend = p12 ; position of y-end iymovefn = abs(p13) ; y movement function nr iymovedir = p13 / iymovefn ; negative fn causes backwards reading iymodfn = abs(p14) ; fn for modulation iymoddir = p14 / iymodfn ; negative fn causes backwards reading iymodfncps = p15 ; y movement function cps iymodfactor = p16 ; y amount of movement izstart = p17 ; position of z-start izend = p18 ; position of z-end izmovefn = abs(p19) ; z movement function nr izmovedir = p19 / izmovefn ; negative fn causes backwards reading izmodfn = abs(p20) ; fn for modulation izmoddir = p20 / iymodfn ; negative fn causes backwards reading izmodfncps = p21 ; z movement function cps izmodfactor = p22 ; z amount of movement schedule 42, 0, p3, itracknr ; instrument invocation of the x,y,z- converter ; READING UNIT ixdiff = ixend - ixstart ; difference of x-position between start and stop kxpos oscil ixdiff, ixmovedir *(1/p3), ixmovefn ; generation of x-position change trajectory during note kxpos = ixstart + kxpos ; starting the x-component of the trajectory from x-position origin kxmove oscil ixmodfactor, ixmoddir * ixmodfncps, ixmodfn ; reads the function table for modulation of movement kx = kxpos + kxmove ; applies modulation onto the x-component of the trajectory iydiff = iyend - iystart ; difference of y-position between start and stop kypos oscil iydiff, iymovedir * (1/p3), iymovefn ; generation of y-position change trajectory during note kypos = iystart + kypos ; starting the y-component of the trajectory from y-position origin kymove oscil iymodfactor,iymoddir * iymodfncps, iymodfn ; reads the function table for modulation of movement ky = kypos + kymove ; applies modulation onto the y-component of the trajectory izdiff = izend - izstart ; difference of z-position between start and stop kzpos oscil izdiff, izmovedir * (1/p3), izmovefn ; generation of z-position change trajectory during note kzpos = izstart + kzpos ; starting the z-component of the trajectory from x-position origin kzmod oscil izmodfactor, izmoddir * izmodfncps, izmodfn ; reads the function table for modulation of movement kz = kzpos + kzmod ; applies modulation onto the y-component of the trajectory ; GLOBAL OUTPUT UNIT zkw kx, itracknr ; read global x-value zkw ky, itracknr + 20 ; read global y-value zkw kz, itracknr + 40 ; read global z-value endin ; ******************************************************************************************************** instr 42 ;x, y, z -CONVERTER coded 1999-2001 by Jan Jacob Hofmann. Thanks to Christoph ; ******************************************************************************************************** itracknr = p4 ; INPUT UNIT kx zkr itracknr ; read global x-value ky zkr itracknr + 20 ; read global y-value kz zkr itracknr + 40 ; read global z-value kx = (kx < 0.01 && kx > -0.01 ? 0.01 : kx) ; fix for that strange y-axis noise ; Distance calculation kD = sqrt( kx^2 + ky^2 + kz^2) ; calculates the distance kx = kx / kD ; this operation generates a vector equivalent at the unit sphere ky = ky / kD ; this operation generates a vector equivalent at the unit sphere kz = kz / kD ; this operation generates a vector equivalent at the unit sphere kD = (kD < 0.3 ? 0.3 : kD) ; limits kD to a minimum of 0.3 unit to prevent overflow ; Elevation calculation kE = cosinv ( sqrt( 1 - kz^2)) ; calculation of elevation kE = (kz < 0 ? -kE : kE) ; if z is negative make elevation- vector negative too ; Angle calculation kA = sininv (ky / cos (kE)) ; calculation of angle kA = (kx >= 0 ? kA : 3.14159265 - kA) ; distinction of positive and negative y kA = (ky <= 0 && kx >= 0 ? 6.28318531 + kA : kA) ; distinction of positive and negative y and x ; RADIANT - DEGREE CONVERSION kA = kA * 57.295779513 ; convert angle from radiant to degree kE = kE * 57.295779513 ; convert elevation from radiant to degree ; GLOBAL OUTPUT UNIT zkw kA, itracknr ; sends out global A value zkw kE, itracknr + 20 ; sends out global E value zkw kD, itracknr + 40 ; sends out global D value endin ; ***************************************************************************************** instr 46 ;A,E,D - MOVEMENT SCORE READER coded 1999-2001 by Jan Jacob Hofmann ; ***************************************************************************************** itracknr = p4 ; number of the track the trajectory is applied on iAstart = p5 ; starting point of angle trajectory iAend = p6 ; end-point of angle trajectory iAdiff = iAend - iAstart ; difference of A-position between start and stop iAmovefn = abs(p7) ; A movement function nr iAmovedir = p7 / iAmovefn ; negative fn causes backwards reading iAmodfn = abs(p8) ; A modulation function nr iAmoddir = p8 / iAmodfn ; negative fn causes backwards reading iAfncps = p9 ; A modulation function cps iAfactor = p10 ; A amount of modulation in degrees iEstart = p11 ; starting point of elevation trajectory iEend = p12 ; end-point of elevation trajectory iEdiff = iEend - iEstart ; difference of E-position between start and stop iEmovefn = abs(p13) ; E movement function nr iEmovedir = p13 / iEmovefn ; negative fn causes backwards reading iEmodfn = abs(p14) ; E modulation function nr iEmoddir = p14 / iEmodfn ; negative fn causes backwards reading iEmodfncps = p15 ; E modulation function cps iEfactor = p16 ; E amount of modulation in degrees iDstart = p17 ; distance at start position iDend = p18 ; distance at end position iDdiff = iDend - iDstart ; difference of D-position between start and stop iDmovefn = abs(p19) ; D movement function nr iDmovedir = p19 / iDmovefn ; negative fn causes backwards reading iDmodfn = abs(p20) ; D modulation function nr iDmoddir = p20 / iDmodfn ; negative fn causes backwards reading iDfncps = p21 ; D modulation function cps iDfactor = p22 ; D amount of modulation ; TRAJECTORY GENERATION UNIT kA oscil iAdiff, iAmovedir * (1/p3), iAmovefn ; generation of angle-position change trajectory during note kA = iAstart + kA ; starting the angle-component of the trajectory from angle-position origin kAmod oscil iAfactor, iAmoddir * iAfncps, iAmodfn ; reads the function table kA = kA + kAmod ; applies modulation onto the angle-component of the trajectory kE oscil iEdiff, iEmovedir *(1/p3), iEmovefn ; generation of elevation-position change trajectory during note kE = iEstart + kE ; starting the elevation-component of the trajectory from elevation-position origin kEmod oscil iEfactor, iEmoddir * iEmodfncps, iEmodfn ; reads the function table kE = kE + kEmod ; applies modulation onto the elevation-component of the trajectory kD oscil iDdiff, iDmovedir * (1/p3), iDmovefn ; generates the movement of kD kD = iDstart + kD ; starting the distance-component of the trajectory from angle-position origin kDmod oscil iDfactor, iDmoddir * iDfncps, iDmodfn ; reads the function table kD = kD + kDmod ; applies modulation onto the distance-component of the trajectory kD = abs(kD) ; kD may only be positive kD = (kD < 0.3 ? 0.3 : kD) ; limits kD to a minimum of 0.3 unit to prevent overflow ; GLOBAL OUTPUT UNIT zkw kA, itracknr ; sends out global A value zkw kE, itracknr + 20 ; sends out global E value zkw kD, itracknr + 40 ; sends out global D value endin ; ************************************************************************************** instr 47 ;AED- MODULATOR coded 1999-2001 by Jan Jacob Hofmann. ; ************************************************************************************** ; Applies aditional modulation onto a generated trajectory according mode setting: ; mode1 = Angle modulation ; mode2 = Elevation modulation ; mode3 = Distance modulation itracknr = p4 ; number of the track the modulation is applied on imode = p5 ; mode setting imodfn = abs(p6) ; AED-movement function nr imoddir = p6 / imodfn ; negative fn causes backwards reading imodfncps = p7 ; frequency of applied modulation imodamt = p8 ; amount of modulation in degree (mode 1+2) or unit (mode 3) imodfactorfn = abs(p9) ; function describing the progression of modulation depth imodamtdir = p9 / imodfactorfn ; negative fn causes backwards reading itracknr = itracknr + (imode * 20 - 20) ; branch the modulation information according mode ; TRAJECTORY GENERATION UNIT kmodamt oscil imodamt , imodamtdir * (1/p3), imodfactorfn ; change modulationamount during note kmod oscil kmodamt, imoddir * imodfncps, imodfn ; create a modulating source kAED zkmod kmod, itracknr ; apply modulation from k-channel 0 on kAED ; GLOBAL OUTPUT UNIT kAED = (imode = 2 && kAED > 90 ? 90 : kAED) ; limit elevation to a maximum of 90 degree kAED = (imode = 2 && kAED < -90 ? -90 : kAED) ; limit elevation to a maximum of 90 degree kAED = (imode = 3 && kAED < 0.3 ? 0.3 : kAED) ; limit Distance to a minimum of 0.3unit zkw kAED, itracknr ; sends out global A, E or D value endin ; ************************************************************************************** instr 48 ;AED RANDOMIZER coded 1999-2001 by Jan Jacob Hofmann. ; ************************************************************************************** ; mode1 = Angle modulation ; mode2 = Elevation modulation ; mode3 = Distance modulation itracknr = p4 ; number of the track the randomization is applied on imode = p5 ; mode setting imodamt = p6 ; amount of modulation in degree (mode 1+2) or unit (mode 3) imodamtfn = abs(p7) ; AED-movement function nr imodamtdir = p7 / imodamtfn ; negative fn causes backwards reading irandcps = p8 ; AED movement function cps imodcpsfn = abs(p9) ; fn number for the change of random speed imodspedir = p9 / imodcpsfn ; negative fn causes backwards reading iseed = p10 itracknr = itracknr + (imode * 20 - 20) ; branch the randomization information according mode ; TRAJECTORY GENERATION UNIT kmodamt oscil imodamt, imodamtdir * (1/p3), imodamtfn ; change modulationamount during note kmodcps oscil irandcps, imodspedir * (1/p3), imodcpsfn ; change random speed during note krand randi kmodamt, kmodcps, iseed ; create a modulating source krand = krand -0.5 * kmodamt ; shift the range also to the negative kAED zkmod krand, itracknr ; apply modulation from k-channel 0 on kAED kAED = (imode = 2 && kAED > 90 ? 90 : kAED) ; limit elevation to a maximum of 90 degree kAED = (imode = 2 && kAED < -90 ? -90 : kAED) ; limit elevation to a maximum of 90 degree kAED = (imode = 3 && kAED < 0.3 ? 0.3 : kAED) ; limit Distance to a minimum of 0.3 ; GLOBAL OUTPUT UNIT zkw kAED, itracknr ; sends out global A, E or D value endin ; ************************************************************************************************ instr 50 ;SOUNDFILE READER (with processing unit) coded 1999-2001 by Jan Jacob Hofmann ; ************************************************************************************************ itracknr = p4 ; track (=soundfile) to read in igainfactor = p5 ; amplitude scaling iskiptime = p6 ; skip beginning of the soundfile [sec] ireverbsend = p7 ; amount of direct signal sent to local + global reverb irvbdecay = p8 ; reverbdecay for local reverb irandpchmod = p9 ; randomized pitch modulation for local reverb icutofffreq = p10 ; cutofffreq for local reverb ilocalrev = p11 ; switch on local reverb [1=on/0=off] idiffref = p12 ; switch on diffuse reflections [1=on/0=off] ispecref = p13 ; switch on specular reflections[1=on/0=off] ienvfn = abs(p14) ; function describing the amplitude progression of the soundfile ienvdir = p14 / ienvfn ; negative fn causes backwards reading zacl itracknr, itracknr ; clear a-rate channel of current track zacl itracknr + 20, itracknr + 20 ; clear a-rate channel of current track for local reverb kD zkr itracknr + 40 ; take distance from the global output of the movement score ; reading instrument or other kDrev = 1 / kD kDrev = (kD < 1 ? 1 : kDrev ) ; keeps 1/kD < 1 kpitch init 1.00 ; initialisation of kpitch($T.) at i-time-pass ; INSTRUMENT INVOCATION UNIT schedule 70, iskiptime, p3 - iskiptime, itracknr ; invocation of the sound encoder if ilocalrev == 0 goto diffref ; skip (switch off) the local reverb schedule 60, iskiptime, p3 - iskiptime, itracknr, irvbdecay, irandpchmod, icutofffreq ; invocation of the local reverb diffref: if idiffref == 0 goto specref ; skip (switch off) the diffuse reflections schedule 74, iskiptime, p3 - iskiptime, itracknr ; invocation of diffuse early reflections specref: if ispecref == 0 goto reading ; skip (switch off) the specular reflections schedule 75, iskiptime, p3 - iskiptime, itracknr ; invocation of specular early reflections reading: ;READING UNIT - the mono soundfiles from the editor are imported here #define READING(T) # ; define macro for the reader kpitch limit kpitch, 0.3, 2 ; prevent clicks and pops adry diskin2 "$INFILE$T..$HEADER_INFILE", kpitch, iskiptime, 0, 0, giwsize ; importing monotrack ($T.) ; set iwsize to prefered value at the header section of this .orc goto contin # if itracknr = 1 goto track1 if itracknr = 2 goto track2 if itracknr = 3 goto track3 if itracknr = 4 goto track4 if itracknr = 5 goto track5 if itracknr = 6 goto track6 if itracknr = 7 goto track7 if itracknr = 8 goto track8 if itracknr = 9 goto track9 if itracknr = 10 goto track10 if itracknr = 11 goto track11 if itracknr = 12 goto track12 if itracknr = 13 goto track13 if itracknr = 14 goto track14 if itracknr = 15 goto track15 if itracknr = 16 goto track16 if itracknr = 17 goto track17 if itracknr = 18 goto track18 if itracknr = 19 goto track19 if itracknr = 20 goto track20 track1: $READING(1) ; import the source track2: $READING(2) ; import the source track3: $READING(3) ; import the source track4: $READING(4) ; import the source track5: $READING(5) ; import the source track6: $READING(6) ; import the source track7: $READING(7) ; import the source track8: $READING(8) ; import the source track9: $READING(9) ; import the source track10: $READING(10) ; import the source track11: $READING(11) ; import the source track12: $READING(12) ; import the source track13: $READING(13) ; import the source track14: $READING(14) ; import the source track15: $READING(15) ; import the source track16: $READING(16) ; import the source track17: $READING(17) ; import the source track18: $READING(18) ; import the source track19: $READING(19) ; import the source track20: $READING(20) ; import the source contin: #undef READING(T) ; SCALING AND DELAYING UNIT ; this unit is based upon the example in the Charles Dodge/Thomas Jerse book, Computer Music, page 320 ; attenuation according to distance and ratio of reverberation/direct signal according to distance is applied to the sound material of each track. ; ENVELOPING OF SOUNDFILE kgainfactor oscil igainfactor, ienvdir * (1/p3), ienvfn ; create a enveloping source adry = adry * kgainfactor * girescale ; scaling of dry signal arevsend = adry * sqrt(kDrev) * ireverbsend ; scaling according to distance and gaincontrol irevdel = giunit * gifrontwall * 0.00293255132 ; caculation of time of a reflection (reverb) traveling from the wall to the center [s] arevsend delay arevsend, irevdel ; feeding a delayed signal to global reverbsend to simulate the size of the room ; by delaying the response of the room according to its size aglrevsend = arevsend * kDrev ; ratio between global and local reverb zaw aglrevsend, 0 ; global reverb send gets written to Zak-channel 0 if ilocalrev = 0 goto processing ; no local reverb send if localreverb is switched off arev = arevsend * (1 - kDrev) ; ratio between global and local reverb zaw arev, itracknr + 20 ; local reverb send to Zak- channel 21-40 processing: ; FILTERING UNIT [distance] adry = adry * kDrev ; scaling of dry signal acording to distance afilt butterlp adry, 22000 * sqrt(kDrev) ; filtering according to distance afilt balance afilt, adry ; rescale amplitude after filtering zaw afilt, itracknr ; audio zak-output ; DOPPLER SHIFT CALCULATION ; calculation of velocity ; Note: giunit = distance from a listener in the middle to the unit sphere in meters aD interp kD aDdelayed delay aD, 0.01 ; delay distance - amount for 100 ms kV downsamp aDdelayed - aD ; calculate the difference of distances kV = (kV * giunit) * 100 ; speed in m/s ; calculation of doppler kpitch = 345 / (345 - kV) endin ; ************************************************************************************** instr 60 ;LOCAL ( TRACK APPLIED) REVERB with track nr as argument ; ************************************************************************************** p3 = p3 + 2 ; prolong duration to keep the reverb - tail itracknr = p4 irvbdecay = p5 irandpchmod = p6 icutofffreq = p7 arevsig zar itracknr + 20 $LocalGlobalREVERB. zaw arevsig, itracknr + 20 endin ; ******************************************************************************************************************************* instr 70 ;AMBISONIC ENCODING UNIT, 2nd/3rd ORDER AMBISONICS coded 1999-2009 by Jan Jacob Hofmann, ;using the Furse-Malham-Set (FMH) of encoding equations. Thanks to Dave Malham, Richard Furse and Fons Adriaensen ; ******************************************************************************************************************************* p3 = p3 + 2 ; prolong duration to keep the reverb - tail itracknr = p4 kA zkr itracknr ; take angle from the global output of the movement score reading instrument or other kE zkr itracknr + 20 ; take elevation from the global output of the movement score reading instrument or other asig zar itracknr ; zak audio input arevsig zar itracknr + 20 ; zak localreverb input asig = asig + arevsig ; mixing asig with local reverb ; ENCODING 2nd ORDER + ZAK-WRITING UNIT if nchnls==16 goto third_order Encoding_2nd_Order asig, kA, kE ;encoding the sound in 2nd order ambisonic if nchnls = 9 if nchnls==9 goto end ; ENCODING 3rd ORDER + ZAK-WRITING UNIT third_order: Encoding_3rd_Order asig, kA, kE ;encoding the sound in 3rd order ambisonic if nchnls = 16 end: endin ; ******************************************************************************************* instr 73 ;SPATIAL GLOBAL REVERB conceived and coded 1999-2005 by Jan Jacob Hofmann ; using the Furse-Malham-Set (FMH) of encoding equations. ; ******************************************************************************************* p3 = p3 + 2 ; prolong duration to keep the reverb - tail irvbdecay = p4 ; reverbdecay for global reverb irandpchmod = p5 ; randomized pitch modulation for global reverb icutofffreq = p6 ; cutofffreq for global reverb girandpchmod = irandpchmod ; assigning girandpchmod for re-use in instr. 74 and 75 schedule 80, 0, p3 ; invocation of mixing and output arevsig zar 0 ; reading global reverb send as written to Zak-channel 0 in instr. 50 $LocalGlobalREVERB. ; kA1-kA12 are the angles pointing to the 12 corners of an ikosaeder ; kE1-kE12 is the elevation pointing to the 12 corners of an ikosaeder iA1 = 0 ; angle reverberant point 1 iA2 = 0.6238 ; angle reverberant point 2 iA3 = 1.2566 ; angle reverberant point 3 iA4 = 1.8849 ; angle reverberant point 4 iA5 = 2.5132 ; angle reverberant point 5 iA6 = 3.1415 ; angle reverberant point 6 iA7 = 3.7699 ; angle reverberant point 7 iA8 = 4.3982 ; angle reverberant point 8 iA9 = 5.0265 ; angle reverberant point 9 iA10 = 5.6548 ; angle reverberant point 10 iA11 = 0 ; angle reverberant point 11 iA12 = 0 ; angle reverberant point 12 iE1 = 0.463646 ; height reverberant point 1 ( 26.565) iE2 = -0.463646 ; height reverberant point 2 ( -26.565) iE3 = 0.463646 ; height reverberant point 3 ( 26.565) iE4 = -0.463646 ; height reverberant point 4 ( -26.565) iE5 = 0.463646 ; height reverberant point 5 ( 26.565) iE6 = -0.463646 ; height reverberant point 6 ( -26.565) iE7 = 0.463646 ; height reverberant point 7 ( 26.565) iE8 = -0.463646 ; height reverberant point 8 ( -26.565) iE9 = 0.463646 ; height reverberant point 9 ( 26.565) iE10 = -0.463646 ; height reverberant point 10 ( -26.565) iE11 = 1.570796 ; height reverberant point 11 ( 90) iE12 = -1.570796 ; height reverberant point 12 ( -90) #define GLOBALREVERBENCODING(W) # kchnlx = cos(iA$W.) * cos(iE$W.) ; 2nd order Ambisonic encoding equation kchnly = sin(iA$W.) * cos(iE$W.) ; 2nd order Ambisonic encoding equation kchnlz = sin(iE$W.) ; 2nd order Ambisonic encoding equation kchnlr = 1.5 * sin(iE$W.) * sin(iE$W.) -0.5 ; 2nd order Ambisonic encoding equation kchnls = cos(iA$W.) * sin(2*iE$W.) ; 2nd order Ambisonic encoding equation kchnlt = sin(iA$W.) * sin(2*iE$W.) ; 2nd order Ambisonic encoding equation kchnlu = cos(2*iA$W.) * cos(iE$W.) * cos(iE$W.) ; 2nd order Ambisonic encoding equation kchnlv = sin(2*iA$W.) * cos(iE$W.) * cos(iE$W.) ; 2nd order Ambisonic encoding equation kchnlw = 1 - 0.293 *(kchnlx^ 2 + kchnly^ 2) ; 2nd order Ambisonic encoding equation aw = afilt$W. * kchnlw ;(A,E) ax = afilt$W. * kchnlx ;(A,E) ay = afilt$W. * kchnly ;(A,E) az = afilt$W. * kchnlz ;(A,E) ar = afilt$W. * kchnlr ;(A,E) as = afilt$W. * kchnls ;(A,E) at = afilt$W. * kchnlt ;(A,E) au = afilt$W. * kchnlu ;(A,E) av = afilt$W. * kchnlv ;(A,E) zawm aw, 41 ; rewrite and mix zak-location 41 with ambisonics w-channel output of Track 1-20 zawm ax, 42 ; rewrite and mix zak-location 42 with ambisonics x-channel output of Track 1-20 zawm ay, 43 ; rewrite and mix zak-location 43 with ambisonics y-channel output of Track 1-20 zawm az, 44 ; rewrite and mix zak-location 44 with ambisonics z-channel output of Track 1-20 zawm ar, 45 ; rewrite and mix zak-location 45 with ambisonics r-channel output of Track 1-20 zawm as, 46 ; rewrite and mix zak-location 46 with ambisonics s-channel output of Track 1-20 zawm at, 47 ; rewrite and mix zak-location 47 with ambisonics t-channel output of Track 1-20 zawm au, 48 ; rewrite and mix zak-location 48 with ambisonics u-channel output of Track 1-20 zawm av, 49 ; rewrite and mix zak-location 49 with ambisonics v-channel output of Track 1-20 # $GLOBALREVERBENCODING(1) $GLOBALREVERBENCODING(2) $GLOBALREVERBENCODING(3) $GLOBALREVERBENCODING(4) $GLOBALREVERBENCODING(5) $GLOBALREVERBENCODING(6) $GLOBALREVERBENCODING(7) $GLOBALREVERBENCODING(8) $GLOBALREVERBENCODING(9) $GLOBALREVERBENCODING(10) $GLOBALREVERBENCODING(11) $GLOBALREVERBENCODING(12) endin #undef GLOBALREVERBENCODING(W) #undef LocalGlobalREVERB ; *********************************************************************************************************************************************************** instr 74 ;DIFFUSE EARLY REFLECTIONS conceived and coded 1999-2009 by Jan Jacob Hofmann, using the Furse-Malham-Set (FMH) of encoding equations. ; Thanks to Dave Malham and Richard Furse. Thanks to Ingo. ; Based on a Paper by David Griesinger published in the Proceedings of the 19th AES International Conference, June 2001. ; *********************************************************************************************************************************************************** p3 = p3 + 0.5 ; prolong duration to keep the reverb - tail itracknr = p4 ; gifrontwall - position of frontwall at the x-axis ; gisidewall - position of sidewall at the y-axis ; giceiling - height of the ceiling measured from the zero-point of the coordinate-system ; giunit - radius of the unit- sphere[m] ; TRAJECTORY AND SIGNAL READ IN kA zkr itracknr ; take angle from the global output of the movement score reading instrument or other kE zkr itracknr + 20 ; take elevation from the global output of the movement score reading instrument or other kD zkr itracknr + 40 ; take distance from the global output of the movement score reading instrument or other asig zar itracknr ; zak audio input ; DEGREE - RADIANT CONVERSION kA = kA * 0.01745329252 ; convert 0-360 degrees to radiant-expression kE = kE * 0.01745329252 ; convert 0-90 degrees to radiant-expression kA = (kA < 0 ? kA + 6.283185 : kA) ; prevent negative angle kA kA = (kA > 6.283185 ? kA - 6.283185 : kA) ; prevent angle kA > 360 ; Polar converter -source as cartesian representation kx = kD * cos(kA) * cos(kE) ; position of sound in kartesian coordinates ky = kD * sin(kA) * cos(kE) ; position of sound in kartesian coordinates kz = kD * sin(kE) ; position of sound in kartesian coordinates kx = (kx < 0.01 && kx > -0.01 ? 0.01 : kx) ; fix for that strange y-axis noise asig tone asig, gihighdamp ; filtering of high frequencies due to absorbtion characteristics of the wall #define DIFFREFLECTION(R)# icnt = $R. ; calculation of points of reflection ; frontwall= 1 ; backwall = 2 ; left sidewall =3 ; right sidewall = 4 ; ceiling = 5 ; floor = 6 krefx = ( icnt = 1 ? gifrontwall : kx) ; asigning a position of the reflection at the x-axis if icnt = 1 kgoto yrefcalc$R. krefx = ( icnt = 2 ? -gifrontwall : kx) ; asigning a position of the reflection at the x-axis yrefcalc$R.: krefy = ( icnt = 3 ? gisidewall : ky) ; asigning a position of the reflection at the y-axis if icnt = 3 kgoto zrefcalc$R. krefy = ( icnt = 4 ? -gisidewall : ky) ; asigning a position of the reflection at the y-axis zrefcalc$R.: krefz = ( icnt = 5 ? giceiling : kz) ; asigning a position of the reflection at the z-axis if icnt = 5 kgoto refout$R. krefz = ( icnt = 6 ? -giceiling : kz) ; asigning a position of the reflection at the z-axis refout$R.: ; A, E, D - calculation of the reflecting point kDref = sqrt(krefx^2 + krefy^2 + krefz^2) ; calculates the distance of the reflection kxref = krefx / kDref ; this operation generates a vector equivalent at the unit sphere kyref = krefy / kDref ; this operation generates a vector equivalent at the unit sphere kzref = krefz / kDref ; this operation generates a vector equivalent at the unit sphere ; Elevation calculation kEref = cosinv ( sqrt( 1- kzref^2)) ; calculation of elevation kEref = (kzref < 0 ? -kEref : kEref) ; if z is negative make Elevation- vector negative too ; Angle calculation kAref = sininv (kyref/ cos (kEref)) ; calculation of angle kAref = (kxref >= 0 ? kAref : 3.1415 - kAref) ; distinction of positive and negative y kAref = (kyref <= 0 && kxref >= 0 ? 6.2831 + kAref : kAref); distinction of positive and negative y and x ; calculation of distance- difference (m/s) and delaytime kDdiff = sqrt((kx - krefx)^2 + (ky-krefy)^2 + (kz - krefz)^2 ) ; calculation of the difference of kreflection and ksound kDist = kDdiff + kDref ; distance sourcesurface + distance surface listener kDmtr = kDist * giunit ; calculation of difference in meters kdel = kDmtr / 0.345 ; calculation of the delaytime [ms] ; reflection behind the source kAdist = abs(kA - kAref) kEdist = abs(kE - kEref) kAdist = (kAdist > 6.2831 ? kAdist - 6.2831 : kAdist) ; reduces the angle kAdist to values between 0 and 6,2831 kmute = kAdist + kEdist kmute = ( kmute < 1.5707 ? sin (kmute) : 1) ; if the reflection gets behind the source, it is muted ; calculation of attenuation (gain of delay) kgain = 1 / kDist * gidamp * kmute ; gain of early reflection according calculation of attenuation ; FILTERING UNIT kDdiff = ( kDdiff < 0.05 ? 0.05 : kDdiff) ; limit the distance source reflection klfboost = (sqrt(1 / kDdiff ) + kD) * gilfboost afilt tone asig, (sqrt(1 / kDist)) * sr ; filtering of high frequencies according to distance afilt pareq afilt, 174, klfboost, 0.7071, 1 ; boosting low frequencies for souces near the wall afilt balance afilt, asig ; balancing the output ; REFLECTING UNIT ; girandpchmod ; amount of random pitch modulation ; for the delay lines. 1 is the "normal" ; amount, but this may be too high for ; held pitches such as piano tones. ; Adjust to taste. a$R. randi 2, 3.973, $R. adel interp kdel adel = adel + a$R. * girandpchmod asigref vdelay3 afilt, adel, 200 ; creating a reflection asigref = asigref * kgain ; assigning a gain ; ENCODING UNIT ; RADIANT - DEGREE CONVERSION kAref = kAref * 57.2958 ; convert angle from radiant to degree kEref = kEref * 57.2958 ; convert elevation from radiant to degree ; ENCODING 2nd ORDER + ZAK-WRITING UNIT if nchnls==16 goto third_order$R. Encoding_2nd_Order asigref, kAref, kEref ;encoding the sound in 2nd order ambisonic if nchnls = 9 if nchnls==9 goto end$R. ; ENCODING 3rd ORDER + ZAK-WRITING UNIT third_order$R.: Encoding_3rd_Order asigref, kAref, kEref ;encoding the sound in 3rd order ambisonic if nchnls = 16 end$R.: # $DIFFREFLECTION(1) $DIFFREFLECTION(2) $DIFFREFLECTION(3) $DIFFREFLECTION(4) $DIFFREFLECTION(5) $DIFFREFLECTION(6) #undef DIFFREFLECTION(R) endin ; ************************************************************************************************************************************* instr 75 ;SPECULAR EARLY REFLECTIONS conceived and coded 1999-2009 by Jan Jacob Hofmann, ; using the Furse-Malham-Set (FMH) of encoding equations. Thanks to Dave Malham and Richard Furse. Thanks to Nora. ; Also thanks to Peter Lennox for inspitation and pushing me in the right direction. ; ; Based on a Paper by David Griesinger published in the Proceedings of the 19th AES International Conference, June 2001. ; ************************************************************************************************************************************* p3 = p3 + 0.5 ; prolong duration to keep the reverb - tail itracknr = p4 ; gifrontwall - position of frontwall at the x-axis ; gisidewall - position of sidewall at the y-axis ; giceiling - height of the ceiling measured from the zero-point of the coordinate-system ; giunit - radius of the unit- sphere[m] ; TRAJECTORY AND SIGNAL READ IN kA zkr itracknr ; take angle from the global output of the movement score reading instrument or other kE zkr itracknr + 20 ; take elevation from the global output of the movement score reading instrument or other kD zkr itracknr + 40 ; take distance from the global output of the movement score reading instrument or other asig zar itracknr ; zak audio input ; DEGREE - RADIANT CONVERSION kA = kA * 0.01745329252 ; convert 0-360 degrees to radiant-expression kE = kE * 0.01745329252 ; convert 0-90 degrees to radiant-expression kA = (kA < 0 ? kA + 6.28318531 : kA) ; prevent negative angle kA kA = (kA > 6.28318531 ? kA - 6.28318531 : kA) ; prevent angle kA > 360 ; POLAR CONVERTER - SOURCE AS CARTHESIAN REPRESENTATION kx = kD * cos(kA) * cos(kE) ; convert polar to carthesian representation ky = kD * sin(kA) * cos(kE) ; convert polar to carthesian representation kz = kD * sin(kE) ; convert polar to carthesian representation kx = (kx < 0.01 && kx > -0.01 ? 0.01 :kx) ; fix for that strange y-axis noise iwall1 = gifrontwall ; assign distance of the wall iwall2 = -gifrontwall ; assign distance of the wall iwall3 = gisidewall ; assign distance of the wall iwall4 = -gisidewall ; assign distance of the wall iwall5 = giceiling ; assign distance of the wall iwall6 = -giceiling ; assign distance of the wall kdiff1 = iwall1 - kx ; distance of reflection to the frontwall kdiff2 = iwall2 - kx ; distance of reflection to the backwall kdiff3 = iwall3 - ky ; distance of reflection to the left sidewall kdiff4 = iwall4 - ky ; distance of reflection to the right sidewall kdiff5 = iwall5 - kz ; distance of reflection to the ceiling kdiff6 = iwall6 - kz ; distance of reflection to the floor asig tone asig, gihighdamp ; filtering of high frequencies due to absorbtion characteristics of the wall #define SPECREFLECTION(R)# ; calculation of points of reflection icnt = $R. ksum$R. = abs(iwall$R. ) + abs(kdiff$R.) ; distance source wall listener (x,y,z-component) krefx = (kx/ ksum$R. ) * abs(iwall$R.) ; x- position of the reflection krefy = (ky/ ksum$R. ) * abs(iwall$R.) ; y- position of the reflection krefz = (kz/ ksum$R. ) * abs(iwall$R.) ; z- position of the reflection ; assignment of x-,y-,z- component at the wall ; frontwall = 1 ; backwall = 2 ; left sidewall =3 ; right sidewall = 4 ; ceiling = 5 ; floor = 6 krefx = ( icnt = 1 ? iwall1 : krefx) ; asigning a position of the reflection at the x-axis if icnt = 1 igoto ycalc$R. krefx = ( icnt = 2 ? iwall2 : krefx) ; asigning a position of the reflection at the x-axis ycalc$R.: krefy = ( icnt = 3 ? iwall3 : krefy) ; asigning a position of the reflection at the y-axis if icnt = 3 igoto zcalc$R. krefy = ( icnt = 4 ? iwall4 : krefy) ; asigning a position of the reflection at the y-axis zcalc$R.: krefz = ( icnt = 5 ? iwall5 : krefz) ; asigning a position of the reflection at the z-axis if icnt = 5 igoto labelout$R. krefz = ( icnt = 6 ? iwall6 : krefz) ; asigning a position of the reflection at the z-axis labelout$R.: krefx = (krefx < 0.01 && krefx > -0.01 ? 0.01 : krefx) ; avoid zero to avoid division by zero error (NaN) later krefy = (krefy < 0.01 && krefy > -0.01 ? 0.01 : krefy) ; avoid zero to avoid division by zero error (NaN) later krefz = (krefz < 0.01 && krefz > -0.01 ? 0.01 : krefz) ; avoid zero to avoid division by zero error (NaN) later ; A, E, D - calculation of the reflecting point kDref = sqrt (krefx^2 + krefy^2 + krefz^2) ; calculates the distance of the reflection kxref = krefx / kDref ; this operation generates a vector equivalent at the unit sphere kyref = krefy / kDref ; this operation generates a vector equivalent at the unit sphere kzref = krefz / kDref ; this operation generates a vector equivalent at the unit sphere ; Elevation calculation kEref = cosinv ( sqrt( 1 - kzref^2)) ; calculation of elevation kEref = (kzref < 0 ? -kEref : kEref) ; if z is negative make elevation- vector negative too ; Angle calculation kAref = sininv (kyref / cos (kEref)) ; calculation of angle kAref = (kxref >= 0 ? kAref : 3.1415 - kAref) ; distinction of positive and negative y kAref = (kyref <= 0 && kxref >= 0 ? 6.2831 + kAref : kAref); distinction of positive and negative y and x ; calculation of distance- difference (m/s) and delaytime kDdiff = sqrt((kx - krefx)^2 + (ky - krefy)^2 + (kz - krefz)^2 ) ; calculation of the difference of kreflection and ksound kDist = kDdiff + kDref ; distance sourcesurface + distance surface listener kDmtr = kDist * giunit ; calculation of difference in meters kdel = kDmtr / 0.345 ; calculation of the delaytime [ms] ; calculation of attenuation (gain of delay) kgain = 1 / kDist * gidamp ; gain of early reflection according calculation of attenuation ; reflection behind the source kAdist = abs(kA - kAref) kEdist = abs(kE - kEref) kAdist = (kAdist > 6.2831 ? kAdist - 6.2831 : kAdist) ; reduces the angle kAdist to values between 0 and 6,2831 kmute = kAdist + kEdist kmute = (kmute < 1.5707 ? sin (kmute) : 1) ; if the reflection gets behind the source, it is muted ; FILTERING UNIT kDdiff = ( kDdiff < 0.05 ? 0.05 : kDdiff) ; limit the distance source reflection klfboost = sqrt(1 / kDdiff) * gilfboost afilt tone asig, (sqrt(1 / kDist)) * sr ; filtering of high frequencies due to absorbtion characteristics of the air ; according to distance afilt pareq afilt, 174, klfboost, 0.7071, 1 ; boosting low frequencies for souces near the wall afilt balance afilt, asig ; balancing the output ; REFLECTING UNIT ; girandpchmod ; amount of random pitch modulation ; for the delay lines. 1 is the "normal" ; amount, but this may be too high for ; held pitches such as piano tones. ; Adjust to taste. a$R. randi 2, 3.973, $R. adel interp kdel adel = adel + a$R. * girandpchmod asigref vdelay3 afilt, adel, 200 ; creation of a reflection asigref = asigref * kgain * kmute ; assigning a gain ; ENCODING UNIT ; RADIANT - DEGREE CONVERSION kAref = kAref * 57.2958 ; convert angle from radiant to degree kEref = kEref * 57.2958 ; convert elevation from radiant to degree ; ENCODING 2nd ORDER + ZAK-WRITING UNIT if nchnls==16 goto third_order$R. Encoding_2nd_Order asigref, kAref, kEref ;encoding the sound in 2nd order ambisonic if nchnls = 9 if nchnls==9 goto end$R. ; ENCODING 3rd ORDER + ZAK-WRITING UNIT third_order$R.: Encoding_3rd_Order asigref, kAref, kEref ;encoding the sound in 3rd order ambisonic if nchnls = 16 end$R.: # $SPECREFLECTION(1) $SPECREFLECTION(2) $SPECREFLECTION(3) $SPECREFLECTION(4) $SPECREFLECTION(5) $SPECREFLECTION(6) #undef SPECREFLECTION(R) endin ; ****************************************************************************************************************************************** instr 80 ;MIXING, OUTPUT & 16-CHNL UNIT FOR SOUNDFILE OUTPUT, 3RD ORDER AMBISONICS coded 1999-2009 by Jan Jacob Hofmann. ; FURSE - MALHAM - SET OF EQUATIONS Thanks to Richard Furse, Dave Malham and Fons Adriaensen ; More sets of speaker- layouts may be discovered at ; http://www.muse.demon.co.uk/ref/speakers.html ; ****************************************************************************************************************************************** ; First Order aw zar 41 ; zak ambisonics w-channel input ax zar 42 ; zak ambisonics x-channel input ay zar 43 ; zak ambisonics y-channel input az zar 44 ; zak ambisonics z-channel input ; Second Order ar zar 45 ; zak ambisonics s-channel input as zar 46 ; zak ambisonics t-channel input at zar 47 ; zak ambisonics u-channel input au zar 48 ; zak ambisonics v-channel input av zar 49 ; zak ambisonics w-channel input ; Third Order ak zar 50 ; zak ambisonics k-channel input al zar 51 ; zak ambisonics l-channel input am zar 52 ; zak ambisonics m-channel input an zar 53 ; zak ambisonics n-channel input ao zar 54 ; zak ambisonics o-channel input ap zar 55 ; zak ambisonics p-channel input aq zar 56 ; zak ambisonics q-channel input zacl 0, 56 ; clear all a-rate channels zkcl 0, 60 ; clear all k-rate channels if nchnls == 2 goto stereo ; goto 2chnl decode ; UNIT FOR WRITING 9 2ND ORRDER AMBISONIC ENCODED SOUNDFILES kpeak_1 peak aw ; collect peak of encoded channel kpeak_2 peak ax ; collect peak of encoded channel kpeak_3 peak ay ; collect peak of encoded channel kpeak_4 peak az ; collect peak of encoded channel kpeak_5 peak ar ; collect peak of encoded channel kpeak_6 peak as ; collect peak of encoded channel kpeak_7 peak at ; collect peak of encoded channel kpeak_8 peak au ; collect peak of encoded channel kpeak_9 peak av ; collect peak of encoded channel printk p3, kpeak_1, 0 ; write peak to output window printk p3, kpeak_2, 3 ; write peak to output window printk p3, kpeak_3, 6 ; write peak to output window printk p3, kpeak_4, 9 ; write peak to output window printk p3, kpeak_5, 12 ; write peak to output window printk p3, kpeak_6, 15 ; write peak to output window printk p3, kpeak_7, 18 ; write peak to output window printk p3, kpeak_8, 21 ; write peak to output window printk p3, kpeak_9, 24 ; write peak to output window fout "$FILE.w.$HEADER.", giformat, aw ; output of a soundfile with header as specified in the options fout "$FILE.x.$HEADER.", giformat, ax ; output of a soundfile with header as specified in the options fout "$FILE.y.$HEADER.", giformat, ay ; output of a soundfile with header as specified in the options fout "$FILE.z.$HEADER.", giformat, az ; output of a soundfile with header as specified in the options fout "$FILE.r.$HEADER.", giformat, ar ; output of a soundfile with header as specified in the options fout "$FILE.s.$HEADER.", giformat, as ; output of a soundfile with header as specified in the options fout "$FILE.t.$HEADER.", giformat, at ; output of a soundfile with header as specified in the options fout "$FILE.u.$HEADER.", giformat, au ; output of a soundfile with header as specified in the options fout "$FILE.v.$HEADER.", giformat, av ; output of a soundfile with header as specified in the options if nchnls == 9 goto skipend ; just encode the sound in 2nd order if nchnls = 9 ; UNIT FOR WRITING 3RD ORRDER AMBISONIC ENCODED SOUNDFILES kpeak_10 peak ak ; collect peak of encoded channel kpeak_11 peak al ; collect peak of encoded channel kpeak_12 peak am ; collect peak of encoded channel kpeak_13 peak an ; collect peak of encoded channel kpeak_14 peak ao ; collect peak of encoded channel kpeak_15 peak ap ; collect peak of encoded channel kpeak_16 peak aq ; collect peak of encoded channel printk p3, kpeak_10, 27 ; write peak to output window printk p3, kpeak_11, 30 ; write peak to output window printk p3, kpeak_12, 33 ; write peak to output window printk p3, kpeak_13, 36 ; write peak to output window printk p3, kpeak_14, 39 ; write peak to output window printk p3, kpeak_15, 42 ; write peak to output window printk p3, kpeak_16, 45 ; write peak to output window fout "$FILE.k.$HEADER.", giformat, ak ; output of a soundfile with header as specified in the options fout "$FILE.l.$HEADER.", giformat, al ; output of a soundfile with header as specified in the options fout "$FILE.m.$HEADER.", giformat, am ; output of a soundfile with header as specified in the options fout "$FILE.n.$HEADER.", giformat, an ; output of a soundfile with header as specified in the options fout "$FILE.o.$HEADER.", giformat, ao ; output of a soundfile with header as specified in the options fout "$FILE.p.$HEADER.", giformat, ap ; output of a soundfile with header as specified in the options fout "$FILE.q.$HEADER.", giformat, aq ; output of a soundfile with header as specified in the options if nchnls == 16 goto skipend ; just encode the sound -comment this out if encoded+decoded files are wanted ; ******************************************************************************************************** stereo: ; 2chnl (stereo ) decode, 2nd order Ambisonics (second order controlled opposites) output ; ******************************************************************************************************** ;Rig `Controlled Opposites' Decode Matrix (second Order) ; w x y z r s t u v ; Speaker1 <0.0000,1.0000,0.0000> achnl_1 sum aw * 0.7071, ay * 0.5000 ; Speaker2 <0.0000,-1.0000,0.0000> achnl_2 sum aw * 0.7071, ay * -0.5000 kpeak_1 peak achnl_1 ; collect peak of encoded channel kpeak_2 peak achnl_2 ; collect peak of encoded channel printk p3, kpeak_1, 0 ; write peak to output window printk p3, kpeak_2, 3 ; write peak to output window outc achnl_1, achnl_2 ; output the sound directly in stereo fout "$FILE.-stereo.$HEADER.", giformat, achnl_1, achnl_2 ; output of a soundfile with header as specified in the options skipend: endin ; ******************************************************************************************************** ; Collection of useful f- fables ; ******************************************************************************************************** f1 0 65537 10 1 /* sine */ f7 0 32768 14 1 1 0 1 0 0.6 0 0.4 0 0.1 /* hyperbolid */ f8 0 32768 19 0.5 0.5 270 0.5 /* sigmoid rising */ f9 0 32768 19 0.5 1 180 1 /* paraboloid */ f10 0 32768 20 2 1 /* hanning */ /*f11-f20=straight lines */ f11 0 4 7 0 4 0 /* straight line at 0.0 */ f12 0 4 7 0.1 4 0.1 /* straight line at 0.1 */ f13 0 4 7 0.2 4 0.2 /* straight line at 0.2 */ f14 0 4 7 0.4 4 0.4 /* straight line at 0.4 */ f15 0 4 7 0.5 4 0.5 /* straight line at 0.5 */ f16 0 4 7 0.6 4 0.6 /* straight line at 0.6 */ f17 0 4 7 0.8 4 0.8 /* straight line at 0.8 */ f18 0 4 7 0.9 4 0.9 /* straight line at 0.9 */ f19 0 4 7 1 4 1 /* straight line at 1.0 */ f20 0 4 -7 10 4 10 /* straight line at 10 */ /*f21-f30=slopes and linsegs*/ f21 0 8192 7 0 8192 1 /* slope rising from 0-1 */ f22 0 8192 7 0 8192 -1 /* slope fallng from 0-(-1) */ f23 0 4 7 0 1 0 0 1 2 1 0 0 1 0 /* rectangle unipolar, symmetric */ f24 0 4 7 1 2 1 0 -1 2 -1 /* rectangle bipolar */ f25 0 256 7 0 128 1 128 0 /* triangle unipolar */ f26 0 4 7 0 2 0 0 1 2 1 /* rectangle unipolar */ f27 0 256 7 0 128 1 0 -1 128 0 /* sawtooth bipolar */ f28 0 256 7 0 64 1 128 -1 64 0 /* tiangle bipolar */ f29 0 1024 7 0 100 1 824 1 100 0 /* enveloped fn 19 */ /*f31-f40=exponentials */ f31 0 65537 5 0.001 65537 1 /* exponential */ f32 0 65537 5 1 65537 0.001 /* exponential reversed */ /**********************************************************************************************************************************/ /**********************************************************************************************************************************/ /*instr 40 x,y,z - READER */ /*instr 42 x,y,z - CONVERTER */ /*instr 46 A,E,D - MOVEMENT-SCORE READER */ /*instr 47 AED - MODULATOR */ /*instr 48 AED - RANDOMIZER */ /*instr 50 SOUNDFILE READER */ /*instr 60 LOCAL (TRACK-APPLIED) REVERB */ /*instr 72 16-CHNL ENCODING UNIT FOR SOUND, 3RD ORDER AMBISONICS */ /*instr 73 SPATIAL GLOBAL REVERB */ /*instr 74 DIFFUSE EARLY REFLECTIONS */ /*instr 75 SPECULAR EARLY REFLECTIONS */ /*instr 80 MIXING, OUTPUT & 16-CHNL UNIT FOR SOUNDFILE OUTPUT, 3RD ORDER AMBISONICS */ /**********************************************************************************************************************************/ a 0, 0, 0 /**********************************************************************************************************************************/ /* instr 40 x,y,z -READER */ /**********************************************************************************************************************************/ /* This instrument creates a trajectory in carthesian coordinates (x,y,z) */ /* The positions are measured in [unit] = distance centre unit sphere (where the speakers reside) */ /* Positions beyond the unit sphere are possible */ /* The trajectory between start and end point is described by the chosen function. This function should have values between 0 and 1: 0= start, 1= end */ /* Each component of the trajectory may be modulated by another (periodic) function */ /* Several trajectories may be joined one after the other */ /* Note that there must be no gap in position and time between endpoint of previous trajectory and starting point of the following trajectory */ /* p1= instr */ /* p2= start */ /* p3= dur */ /* p4= track ; track (= soundfile) to create a trajectory for [1-20] */ /* p5= x-start ; start point of the x-component of the trajectory [unit/(-N) - N] */ /* p6= x-end ; end point of the x-component of the trajectory [unit/(-N) - N] */ /* p7= x-movefn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p8= x-modulatorfn ; fn number of of the function to create a modulation up upon the trajectory of the x-component. Values between (-1) and 1 */ /* p9= x-modfncps ; cps of modulation of the trajectory of the x-component */ /* p10= x-modfnfactor ; depth of modulation of the trajectory of the x-component [unit/(-N) - N] */ /* p11= y-start ; start point of the y-component of the trajectory [unit/(-N) - N] */ /* p12= y-end ; end point of the y-component of the trajectory [unit/(-N) - N] */ /* p13= y-move fn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p14= y-modulatorfn ; fn number of of the function to create a modulation up upon the trajectory of the y-component */ /* p15= y-modfncps ; cps of modulation of the trajectory of the y-component */ /* p16= y-mfnfactor ; depth of modulation of the trajectory of the y-component [unit/(-N) - N] */ /* p17= z-start ; start point of the z-component of the trajectory [unit/(-N) - N] */ /* p18= z-end ; end point of the z-component of the trajectory [unit/(-N) - N] */ /* p19= z-movefn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p20= z-modfn ; fn number of of the function to create a modulation up upon the trajectory of the z-component */ /* p21= z-modfncps ; cps of modulation of the trajectory of the z-component */ /* p22= z-modfactor ; depth of modulation of the trajectory of the y-component [unit/(-N) - N] */ /* p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 p15 p16 p17 p18 p19 p20 p21 p22 */ /**********************************************************************************************************************************/ /* instr 46 A,E,D - MOVEMENT SCORE READER */ /**********************************************************************************************************************************/ /* This instrument creates a trajectory in polar coordinates (angle,elevation,distance) */ /* The positions are measured in [degree] */ /* Positions beyond the the range of 0-360 are possible */ /* The trajectory between start and end point is described by the chosen function. This function should have values between 0 and 1: 0= start, 1= end */ /* Each component of the trajectory may be modulated by another (periodic) function */ /* Several trajectories may be joined one after the other */ /* Note that there must be no gap in position and time between endpoint of previous trajectory and starting point of the following trajectory */ /* p1= instr */ /* p2= start */ /* p3= dur */ /* p4= track ; track (= soundfile) to create a trajectory for [1-20] */ /* p5= A-start ; start point of the angle-component of the trajectory [ 0 - 360] - values above and below are possible */ /* p6= A-end ; end point of the angle-component of the trajectory [ 0 - 360] - values above and below are possible */ /* p7= A-movefn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p8= A-modfn ; fn number of of the function to create a modulation up upon the trajectory of the angle-component. */ /* ; The function contains values between (-1) and 1 */ /* p9= A-modfncps ; cps of modulation of the trajectory of the angle-component */ /* p10= A-modfnfactor ; depth of modulation of the trajectory of the angle-component [ 0-360] */ /* p11= E-start ; start point of the elevation-component of the trajectory [ (-90) - 90] - values above and below are possible */ /* p12= E-end ; end point of the elevation-component of the trajectory [ (-90) - 90] - values above and below are possible */ /* p13= E-move fn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p14= E-modfn ; fn number of of the function to create a modulation up upon the trajectory of the elevation-component. */ /* ; The function contains values between (-1) and 1 */ /* p15= E-fncps ; cps of modulation of the trajectory of the elevation-component */ /* p16= E-mfnfactor ; depth of modulation of the trajectory of the angle-component [ 0-360] */ /* p17= D-start ; start point of the distance-component of the trajectory [unit], usually between 0.3 - 10 values above are possible */ /* p18= D-end ; end point of the distance-component of the trajectory [unit], usually between 0.3 - 10 values above are possible */ /* p19= D-movefn ; fn number of of the function describing the trajectory between start and end point - negative fn causes backwards reading */ /* p20= D-modfn ; fn number of of the function to create a modulation up upon the trajectory of the distance-component. */ /* ; The function contains values between (-1) and 1 */ /* p21= D-modfncps ; cps of modulation of the trajectory of the distance-component */ /* p22= D-modfnfactor ; depth of modulation of the trajectory of the distance-component [units] */ /*p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 p15 p16 p17 p18 p19 p20 p21 p22 */ i46 0 8 1 0 360 21 11 0 0 0 0 21 11 0 0 1 1 21 11 0 0 /* perform a circle*/ /****************************************************************************************************************************************/ /*instr 47 AED- MODULATOR */ /****************************************************************************************************************************************/ /* This instrument modulates angle, elevation or distance trajectory of a single track created before */ /* mode1 = Angle modulation */ /* mode2 = Elevation modulation */ /* mode3 = Distance modulation */ /* p1= instr ; [47] */ /* p2= start ; does not necessarily have to correspond with start times of instrument 40,46,or 50 */ /* p3= dur ; does not necessarily have to correspond with duration of instrument 40,46,or 50 */ /* p4= track[1-20] ; modulation of trajectory for track (= soundfile) [1-20]. */ /* p5= mode ; mode (choice of component of the trajectory to be modulated) */ /* p6= AEDmodfn ; [fn](periodic) function to modulate with. The function contains values between (-1) and 1 */ /* p7= AEDmodfncps ; cps of modulation */ /* p8= AEDmodfnfactor ; depth of modulation: for mode 1+2 insert value in [degree], for mode 3 use [unit] */ /* p9= AEDmodfactorfn ; [fn]function describing the progression of modulation depth. The function contains values between 0 and 1 */ /* ; Negative fn causes backwards reading */ /* p1 p2 p3 p4 p5 p6 p7 p8 p9 */ /**********************************************************************************************************************************/ /*instr 48 AED RANDOMIZER */ /**********************************************************************************************************************************/ /* This instrument modulates angle, elevation or distance trajectory of a single track created before */ /* mode1 = Angle modulation */ /* mode2 = Elevation modulation */ /* mode3 = Distance modulation */ /* p1= instr ; [48] */ /* p2= start ; does not necessarily have to correspond with start times of instrument 40,46,or 50 */ /* p3= dur ; does not necessarily have to correspond with duration of instrument 40,46,or 50 */ /* p4= track[1-20] ; randomization of trajectory for track (= soundfile) [1-20]. */ /* p5= mode ; mode (choice of component of the trajectory to be randomized) */ /* p6= rnddevamt ; depth of randomization: for mode 1+2 insert value in [degree], for mode 3 use [unit] */ /* p7= rnddevamtfn ; [fn] function describing the progression of randomization depth. */ /* ; The function contains values between (-1) and 1 */ /* ; Negative fn causes backwards reading */ /* p8= rndspeedcps ; random changes per second */ /* p9= rndspeedfn ; function describing the progression of randomization depth. The function contains values between 0 and 1 */ /* ; Negative fn causes backwards reading */ /* p10= seed ; seed value for random */ /* p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 */ /**********************************************************************************************************************************/ /* instr 50 SOUNDFILE READER */ /**********************************************************************************************************************************/ ; This instrument reads the soundfile, controls the Local + Glaobal Reverb and switches on/off the local reverb and reflections */ /* p1= instr ; [50] */ /* p2= start ; start time of soundfile read in [s](only works with a trajectory created before) */ /* p3= dur ; duration time of soundfile [s] */ /* p4= track [1-20] ; track (= soundfile) to read in [1-20]/ soundfile name has to be specified in the header of the .orc */ /* p5= gainfactor ; amplitude scaling of the soundfile */ /* p6= skiptime ; time to be skiped from the soundfilestart [s] */ /* p7= reverbsend (local+global) ; amount of direct sound to be sent to the reverb (1=100% - an amount of 0.2 is recomended) */ /* p8= rvbdecay (local) ; gain of reverb. Adjust empirically for desired reverb time. 0.6 gives a good small "live" room sound, */ /* ; 0.8 a small hall, 0.9 a large hall, 0.99 an enormous stone cavern. */ /* p9= randpchmod (local) ; amount of random pitch modulation for the delay lines. 1 is the "normal" amount, but this may be */ /* ; too high for held pitches such as piano tones. Adjust to taste. */ /* p10= lp-cutofffreq (local) ; Cutoff frequency of lowpass filters in feedback loops of delay lines, in Hz. */ /* ; Lower cutoff frequencies results in a sound with more high-frequency damping. */ /* p11= localreverb on/off ; switch on local reverb [1=on/0=off] */ /* p12= diffuse reflection on/off ; switch on diffuse reflections [1=on/0=off] */ /* p13= specular reflection on/off ; switch on specular reflections[1=on/0=off] */ /* p14= amplitude progression of soundfile ; [fn]function describing the amplitude progression of the soundfile */ /* p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 */ i50 0 8 1 1 0 0.2 0.4 1 8000 1 1 1 19 /********************************************************************************************************************************************/ /*instr 73 SPATIAL GLOBAL REVERB ( has to run all of the total duration, else instr. 80 (output) is not invocated) */ /********************************************************************************************************************************************/ /* p1= instr */ /* p2= start */ /* p3= dur */ /* p4= rvbdecay (global) ; gain of reverb. Adjust empirically for desired reverb time. 0.6 gives a good small "live" room sound, */ /* ; 0.8 a small hall, 0.9 a large hall, 0.99 an enormous stone cavern. */ /* p5= randpchmod (global) ; amount of random pitch modulation for the delay lines. 1 is the "normal" amount, but this may be */ /* ; too high for held pitches such as piano tones. Adjust to taste. */ /* p6= lp-cutofffreq (global) ; Cutoff frequency of lowpass filters in feedback loops of delay lines, in Hz. */ /* ; Lower cutoff frequencies results in a sound with more high-frequency damping. */ /* /* p1 p2 p3 p4 p5 p6 */ i73 0 8 0.4 1 8000 e Widgets 0 44 400 200 true 255 255 255 10 10 400 500 {d3b9aebc-e13a-4254-a55a-0672b64df7ed} true 0 -3 Courier 8 0 0 0 255 255 255 Version: 3 Render: Real Ask: Yes Functions: ioObject Listing: Window WindowBounds: 72 179 400 200 CurrentView: io IOViewEdit: On Options: ioView nobackground {65535, 65535, 65535} ioListing {10, 10} {400, 500}