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The TimbreModule is intended to be used as an amplitude depending timbre coloring of the incoming signal. The louder (or softer) the amplitude of the incoming, the resonance intensity increases. Idea is to add some viability to the Instrument. Choices are to add some Parametric EQ and/or a Treble cutoff. --- version 1, May 2023 - initial version

2Mod4 1/2 consists of 2 Low Frequency Modulators, where Modulator 2 works on Modulator 1. Modulator 2 can be set to OFF. Several wave types can be selected. The 7th wave type allows to quickly set the this Modulator to a steady signal- some kind of "pause" if you will. All can be changed in realtime. The 2 Modulators can be mixed and so more complex wave patterns occur. 2Mod4 1/2 is the first part of 2; here, the 4 outputs can be switched on, while 2Mod4 2/2 splits these 4 outputs into one in phase, and another that is out of phase. As the phase and out of phase modulators do follow the same modulation values, the out of phase one can get its own offset. This Instrument has 4 possible outputs. 1. ADD - both modulators added together, or Modulator1 only 2. MAX - both modulators added together but only the maximal values are given 3. MIN - both modulators added together but only the minimal values are given 4. MULT - both modulators multiplied. For this to happen, both Modulators needs to be activated. Some extra coding inside the target Instrument is needed. 2Mod4 2/2 produces k-rate modulations, (or when resolution is set, at a-rate) but if i-rate is needed, a conversion inside the target instrument must be implemented. Scaling must be done in the target Instrument. ALL SCALING AND ADDING AN OFFSET TO THE PRODUCED VALUES MUST BE DONE IN THE TARGET INSTRUMENT ITSELF. 2Mod4 only produces values between -1....1. Any offset or scaling back to values between 0...1 must be done in the target Instr. To save CPU, only select the outputs you need! --- version 1, April 2023 - initial version

DESCRIPTION: 2Mod4 2/2 Module is the second half of the Modulator system, the other one is called 2Mod4 1/2. This Module gets up to 4 modulations at k-rate from 2Mod4 1/2. - kADDITION, kADDPHASE and iADDPHASE is derived from the ADD channel - kMAXIMUM, kMAXPHASE and iMAXPHASE is derived from the MAX channel - kMINIMUM, kMINPHASE and iMINPHASE is derived from the MIN channel - kMULTIPLY, kMULTIPHASE and iMULTPHASE is derived from the MULT channel They are at k-rate, k-rate but inverted phase, and i-rate respectively. These modulated signals can be used to manipulate frequencies or amplitudes of the opcodes, or effects like filters in the instrument, and everything that you want to modulate. --- version 1, April 2023 - initial version

This Instrument creates 2 Low Frequency Oscilators: 1 normal and 1 phaase inverted. 1 or 2 modules can be activated. They are then mixed, so more comples modulations are possible. Example: the normal output modulates the amplitude of 1 instrument, and the phase inverted one the amplitude of a 2nd instrument. Result is that one gets louder, the other gets softer in the same speed. 2OutputModulator uses the Global variable system to send the LFO signals to different instruments. In the instruments to be modulated, code must be added. Following the example above, for amplitude for the first instrument: blueMixerOut asigL * gkModulate, asigR * gkModulate for the second instrument: blueMixerOut asigL * gkModulate_inverse, asigR * gkModulate_inverse --- version 1.2, December 2022 - initial version

DESCRIPTION: This instrument controls the Mixer faders of the DAW Reaper. The code for this AlwaysOn instrument can be found in the Always-On Instrument Text. As an AlwaysOn instrument, it is always on :-) So wherever you are on the Timeline, this instrument is active and the faders can be used to make a mix or change 12 Tracks of Reaper. If Blue and Reaper are running on the same computer, you can use one of these IP addresses: localhost or 127.0.0.1 Be sure to choose another Port than 8000 as this Port is used by default by Blue to receive OSC messages. There is a complementary SoundObject in BlueShare called REAPER OSC command, that allows sending static OSC commands to Reaper, in order to Play or Stop the playback head of Reaper. Several SoundObjects must be placed on the TimeLine, providing Reaper with the rigth static commands. --- version 1.0, November 2021 - initial version version 1.1, January 2022 - added Playrate

DESCRIPTION: The Evolver instruments are created because i was looking for string/pad sounds and textures that could evolve in time, with (small or radical) variations in timbre. The instruments are of a modular design, built with ftmorf as a basis module; so you can stream fluidly from one timbre to another within a single note. Other standard modules in these instruments are: Phaser, Swell, Output. The Evolver Type A uses the adsynt opcode. The A in the name represents 'adsynt'. scheme: FTMORF ---------ADSYNT --------(FILTER)--------(PHASER)--------(SWELL)------OUTPUT note example: i2 0 0.8 60 1.0 ; P4 = midi value, P5 = amplitude i2 0 0.2 261.625565 1.0 ; P4 = frequency, P5 = amplitude CONTROLLERS: see the help balloon when hovering over the controllers in the instrument. Help can be turned off by pressing the 'I' button next to the play bar. --- version 2.2, November 2020 - added filter - added ADSR graphic envelope - added P4 frequency/Midi pitch option version 2.1, November 2019 - added 3 ADSRs version 2.0, octobre 2019 - new GUI (more compact) - renaming BSBs - # of voices and Harmonic Ratio can be changed in between Objects version 1, december 2017 - choose from a list of timbre presets - evolve from one timbre to another up to 16 sounds, forwards or backwards - scale fuction to scale wavesequencer sounds over the note - determine where to start the scaling in the note - morphing can also be done manually by drawing an Automation Line (wavedraw) - mono or stereo with setting different seed values with XY controller for Left and Right channel - special effects section (copied and adapted from the Csound manual (adsynt example)) - phaser effect implemented with on/off switch and mixing, lots of options to slowly evolve the timbre even more - Swell for amplitude to give an amp accent in the note

DESCRIPTION: This PADsynth Xtended instrument has 30 sinewaves to produce a sample - a nice sound, but the PADsynth algorithm always sounds a bit (too) clean to my ears (https://zynaddsubfx.sourceforge.io/doc/PADsynth/PADsynth.htm). With this instrument, several methods are used to try to obtain an evn more interesting, evolving timbre. Modules like FM distortion and/or hyperbolic tangent distortion spices things up. i have chosen for the distorion to be placed AFTER the filter section, so any resonaces of the filter will add to a dirty, screamy sound. Of course, all off these modules can be swtiched off. This instruments can morph sounds from heaven to hell - and back! From LoFi to HiFi... It takes a little while for the PADsynth GEN to generate the sample; the bigger amount of 'sonic quality in bits' that are to be computed, the longer it takes. But the smoother the sound gets, as more bits mean s bigger samples.... connections of the modules: PADSYNTH --> FREQUENCY MODULATION --> FILTERS --> DISTORTION --> OUTPUT CONTROLLERS: fundamental in Hertz - fundamental frequency for the generated table (cycles per second). when set to -1, it is 261.625565 Hz. But you can set a different frequency. score format - see p4 and p5 as real values or as midi note events sonic quality in bits - the higer, the longer samples harmonics - gain of 1st fundamental frequencies + 29 harmonics partial bandwith in cents - bandwidth of each partial in cents scale bandwith in partials - ratio that bandwidth increases with harmonic number (log of increase/decrease with partial frequency, 0 is no stretch or shrink) harmonic stretch - ratio of stretch of the overtones (1 is harmonic) profile shape - extra parameter for the bandwidth profile (0.5 is default) harmonic shape bandwidth - number specifying the shape of the bandwidth profile: Gaussian, Square or Exponential FM ON/OFF: carrier - frequency of carrier mod - frequency of modulator index - number of (in)harmonics created direct/FM - mix direct and FM signal Filter type: frequency - frequency cutoff of filter bandwith - influences region of frequencies resonance - amount of feedback of filter Distortion ON/OFF: compress - compresses signal before distortion is applied pregain - determines the amount of gain applied to the signal before waveshaping positive - determines the shape of the positive part of the curve negative - determines the shape of the negative part of the curve gain - amount of gain after distortion has been applied direct/distort - mix direct and distorted signal envelopes per note: FM env. (on/off) - envelope for FM distortion Filter freq. (on/off) - envelope for filter cutoff Filter Depth (on/off) - envelope for filter depth Distortion (on/off) - envelope for distortion Pitch Bend (on/off) - envelope for pitch bend scale x - repeat envelope in note several times (also for Amplitude) Output: dirt (on/off) - amount of randomness for amplitude tuning - tuning of total sounding result ADSR: on/off - set retriggering of ADSR on or off speed - frequency of retriggering amplitude attack/decay/sustain/release - ADSR controllers L/R - output gain of resulting sound score example: ; frq amp i1 0 2 66 1 ; real values p4, p5 or ; note velocity i1 0 2 66 100 ; midi values p4, p5 --- version 3.1, June 2020 - added midi note/real value MACRO - added on/off checkbox for Filter - renaming of ALs - added comments on controllers version 3.0, April 2019 - complete overhaul of all envelopes - envelopes for Amplitude, FM, Filter Frequency, Filter Depth, Distortion and Pitch Bend can be set on a per note basis - envelopes have on/off switch - cleaned up menu by removing harmonics and better naming version 2.1, october 2017 - replaced Unique ID number with INSTRUMENT ID --> Unique ID could be removes - some further GUI tweaking version 2.0, march 2017 - latest GUI (Blue 2.7.0 or greater) - produce sample in Global Orchestra - changed ADSR to madsr - when basefreqency = -1, then basefrequnecy = 261.625565 - removed redundant code - renamed AL for better names - When you want to use more than 1 of this instrument, you will have to give NO: unique numbers. version 1.01, november 2016 - improved checkbox code version 1.0, september 2016 - table resolution can be set of PADsynth, controlling the quality of the produced "sample". A setting of 18 produces a sample table of 2^18, and is a compromise between a fast response and sound quality. Lower settings are faster, but can produce some LoFi results. - FM on/off switch and mix for direct/FM - ADSR envelope for index of FM - Low, band, highpass filter (or no filter) is placed BEFORE the distortion unit. This can produce sqeaking sounds for extra "dirt" - distortion module with on/off switch, - compression on/off to control big amplitude differences of distortion - ADSR envelope for distortion - dirt setting for amplitude creates a random fluctuation in amplitude

b64Xtended version 3.0 score example: i5 0 2 100 1 where P4 = frequency in Hertz. Pitch-class units can be used if pitch < 15. and P5 = amplitude This instrument is inspired by the SID chip of the Commodore 64 and the b64 instrument by Steven Yi. The instrument has some significant changes compared to version 2.2. Some presets are added. A nice feature is that for every one of the 3 oscillators a "swell" is implemented. That means that, within a tone of one oscillator, an extra amount - positive or negative - of amplitude ( or swell ) can be given. Thus the 3 oscillators together can be fine modulated by different subtle peaks or falls in amplitude within the note, making it more lively and expressive. Another feature is the stereo/mono switch: in stereo the 3 oscillators are divided over 2 channels. The Base Frequency (as set with p4) is in the middle of the stereo image. The other 2 timbres are placed left and right. In mono mode you can treat the sounds of the 3 oscillators more as one sound with a certain timbre. You can use the Panning knob to set this mono sound in the stereo image. It is possible to switch to and from stereo while the note is playing. version 3.0, December 2018: - the Arpeggio section has been extended to 3 separate subsections, allowing for complex Arpeggios. - revamped GUI to modern layout and colours - replaced Amplitude Envelope with a highly configurable envelope, with a switch for linear or exponential. version 2.2: RE UDO k_MultiOscil - MultiOscilx has 2 extra parameters, 1 for switch and 1 for sinefreq - added schwitch for PWM and a sine LFO for PWM to work with pulse - changed PW to 0.05 and 0.95 (according to manual the right specs) - kcount is a tric to divide a too large amp when waves are added together - atemp is used to be able to add waves together - added UDO k_Zwel, so 3 extra parameters RE UDO k_MultiFilter - low and high pass have a Q, bandpass has Bandwith and no Q - added band reject - uses svfilter for lowpass and highpass with Q, uses butter filter for band pass and band reject - Band pass/reject value up to 1500 Hz ======================================================== THE FOLLOWING UDO'S HAVE ON / OFF SWITCHES: RE UDO k_Zwel - one per voice - udo Zwel (= for each note give an extra accent (positive or negative) to achieve fine modulation in the composite sound) using adapted code from The Csound Book page 179 RE UDO Tremolo - one per voice - setksmps 1 to be able to set a shorter port time to filter sharp edges of square and saw RE UDO Vibrato - has standard LFO types, but also can add some jitter. Vibrato is global for all voices. ======================================================== RE Envelope - i have choosen for transeg, because i think it approaches better the feeling of the Commodore 64 envelope - curve of decay and release can be varied between convex and concave RE Ringmodulator - = responsable for mixing the 3 oscilators to the output - added info on which RM is influencing which - added extra variable aringx, for later use in stereo/mono setting RE stereo/mono - conditions to check if ringmodulated or not, and if stereo or not. - added coarse tuning that effects tuning of all oscillators

The Evolver instruments are created because i was looking for string/pad sounds and textures that could evolve in time, with (small or radical) variations in timbre. The instruments are of a modular design, built with ftmorf as a basis module; so you can stream fluidly from one timbre to another within a single note. Other standard modules in these instruments are: Phaser, Swell, Output. The Evolver Type O uses the oscbnk opcode. The O in the name represents 'oscilbnk'. scheme: FTMORF ---------OSCBNK--------(PHASER)--------(SWELL)-------OUTS I l l (LFOs for phasing) I ( LFOs for EQ, FREQ, PHASE, AMP) version 1, december 2017 - choose from a list of timbre presets - evolve from one timbre to another up to 16 sounds, forwards or backwards - scale fuction to scale wavesequencer sounds over the note - determine where to start the scaling in the note - morphing can also be done manually by drawing an Automation Line (wavedraw) - all options for oscbnk are exploited (modulations, eq settings...) - mono or stereo with setting different seed values with XY controller for Left and Right channel - phaser effect implemented with on/off switch and mixing, lots of options to slowly evolve the timbre even more - Swell for amplitude to give an amp accent in the note

This instrument is a port of a Michael Gogins CsoundAC instrument. It is moddled after LaMonte Young and Just Intonation. Young?s drones consists of the mere persistence of long loud tones in a reverberant space. Setting the fundamental frequency to low values is nice, but higher settings produces nice aery sounds as well. This is (the condensed version of) the sore of Michaels CsoundAC instrument. ; p1 p2 p3 p4 p5 p6 p7 ; insno onset duration numerator denominator amplitude pan ; just intonation, 3 chords :C major 7, G major 7, G 7, C major with voice leading. i 1 0 6 1 1 .2 -0.875 i 1 0 18 3 2 . 0.000 i 1 0 6 28 15 . 0.875 i 1 30 6 9 8 . 0.875 i 1 30 3 10 7 . -0.875 i 1 30 3 4 3 . 0.875 i 1 60 6 1 1 . -0.875 i 1 60 6 5 4 . 0.000 i 1 60 6 2 1 . 0.875 version 1, January 2016 - AD envelope for amplitude - fundamental frequency can be set at will

This instrument is inspired by the SID chip of the Commodore 64 and b64 by Steven Yi. A nice feature is that for every one of the 3 oscillators a "swell" is implemented. That means tht within a tone of one oscillator an extra amount - positive or negative - of amplitude ( or swell ) can be given. Thus the 3 oscillators can be fine modulated by different subtle peaks or falls of amplitude within one note, making it more lively. Another nice one is the stereo/mono switch. In stereo the 3 oscillators are divided over 2 channels. In mono mode you can treat the sounds of the 3 oscillators more as one sound with a certain timbre. It is possible to switch to and from stereo while the note is playing RE UDO k_MultiOscil - MultiOscilx has 2 extra parameters, 1 for switch and 1 for sinefreq - added schwitch for PWM and a sine LFO for PWM to work with pulse - changed PW to 0.05 and 0.95 (according to manual the right specs) - kcount is a tric to divide a too large amp when waves are added together - atemp is used to be able to add waves together - added UDO k_Zwel, so 3 extra parameters RE UDO k_MultiFilter - low and high pass have a Q, bandpass has Bandwith and no Q - added band reject - uses svfilter for lowpass and highpass with Q, uses butter filter for band pass and band reject - Band pass/reject value up to 1500 Hz Q: in udo, why all the K instead of k? Because K works at i-time and k-rate, so it works all the time? Q: what is the advantage of having kpch instead of ipch in my instrument? Is there one? Q: why does the setting of PW an influence on the PW sine? Q: i see no other way to put k_Zwel UDO but inside k_MultiOscil. Can you think of a way? Q: is ringmod functional in this sterosetting? ======================================================== THE FOLLOWING UDO'S HAVE ON / OFF SWITCHES: RE UDO k_Zwel - one per voice - udo Zwel (= for each note give an extra accent (positive or negative) to achieve fine modulation in the composite sound) using adapted code from The Csound Book page 179 RE UDO Tremolo - one per voice - setksmps 1 to be able to set a shorter port time to filter sharp edges of square and saw RE UDO Vibrato - has standard LFO types, but also can add some jitter. Vibrato is global for all voices. ======================================================== RE Envelope - i have choosen for transeg, because i think it approaches better the feeling of the Commodore 64 envelope - curve of decay and release can be varied between convex and concave RE Ringmodulator - = responsable for mixing the 3 oscilators to the output - added info on which RM is influencing which - added extra variable aringx, for later use in stereo/mono setting RE stereo/mono - conditions to check if ringmodulated or not, and if stereo or not. - added coarse tuning that effects tuning of all oscillators

Sampler ------------ Created by Mark Van Peteghem This instrument plays audio files, and allows you to alter them. It plays both mono and stereo files (no need to specify this) and takes the sample rate of the sample in account. There are three play modes: play once, loop, and cycle. The first two should be obvious, the third means that the sample will be used as the cycle that is repeated at audio rate, at the frequency given in p5 (pitch or frequency). The first row of UI controls has the basic stuff: distortion of the table (done once to optimize, so not automatable), reversing, selecting begin and end point to be played, begin and end speed up, stutter, optional attack and release, panning and volume adjustment. Stutter length relates to the position in the original sample, so the actual stutter length may be shorter or longer due to the speedup settings. The second row of UI controls has the more advanced stuff: - Tune adjusts the pitch of the sample, either with pvscale or with delays (based on code by Victor Lazzarini). If the pvscale method is used, the preservation of formants can be selected. The amount of pitch adjustment can be selected with the knob in octaves (when 'Octaves' is selected) or by specifying the wanted fequency in p5, if the pitch of the original sample is given in the text box (when 'Pitch in p5' is selected). Note that the perceived pitch is also affected by the speedup knobs. - Shift lets you use pvshift on the sample. - Blur lets you use pvsblur on the sample. - Smooth lets you use pvsmooth on the sample. - Downsample lets you lower the sample frequency. - Auto delay is an experimental feature that uses the sample itself to add a delay. This works best with samples that only have low frequencies, like kick drums. - Chorus adds a chorus, optionally distributing the different voices to the left and right channel. - The filter settings should be obvious. If the play mode is cycle, the speedup, stutter and tune settings are ignored. Some knobs have a text field under them. These can be used to alter the value of the knob, e.g. entering +p6 when the value of the knob is 1, the used value will be 1+p6.

This is a simple instrument that uses the phaser1 or phaser2 opcode on a signal generated by the pluck opcode. The pluck opcode is in the mvpPluck UDO, which checks ipar1 and ipar2, so you don't have to worry about restrictions on these. Despite the simplicity this instrument generates very diverse and interesting sounds. Some knobs have a text field under them, these allow you to change the value (e.g. an extra p-field or a global value). Usage: p4: pitch or frequency p5: in decibels below 0dbfs if 0 or negative, otherwise amplitude as fraction of 0dbfs (0-1)

This instrument modulates a sine wave with the signal generated by the pluck opcode. You can optionally modulate a second time. For both you can choose between phase modulation and frequency modulation. Phase modulation usually sounds more harsh, frequency modulation usually sounds warmer. The pluck opcode is in a UDO called mvpPluck, that adjusts the parameters if necessary, so you can twiddle around with the parameters without caring about the restrictions. The frequency of the sine and the pluck signal are the requested frequency multiplied by an integer from 1 to 5. To have the requested frequency as a basic harmonic, the greatest common divisor of these two integers should of course be 1. The envelope is that of the signal created by the pluck opcode, otherwise you are left with a dull sine when the signal of the pluck opcode has died out. Some knobs have a text field under them, these allow you to change the value (e.g. an extra p-field or a global value). Usage: p4: pitch or frequency p5: in decibels below 0dbfs if 0 or negative, otherwise amplitude as fraction of 0dbfs (0-1)

A supersaw is available on some synthesizers as an extra waveform, besides the familiar sine, triangle, square and saw. A supersaw is typically seven saws that are slightly detuned. I found out that you also get a nice sound if you replace a saw with a square, so I made this instrument that lets you choose between them. They are called pulse in the interface because you can select the pulse width (does not apply if saw is selected). I did not add presets because it usually sounds good, unless the lo pass knob is too low, and because it does not have that much diversity (but a nice instrument anyway). It sounds best around pitch 7.00 to 9.06, especially in chords. Usage: p4: pitch or frequency p5: in decibels below 0dbfs if 0 or negative, otherwise amplitude as fraction of 0dbfs (0-1) Example: i1 0 2 8.00 -10 i1 0 . 8.02 -10 i1 0 . 8.04 -10

Wrapper instrument for the sndwarp opcode.

Flacho, by Mark Van Peteghem ********************************** This instrument lets you experiment with flanger and chorusing, and the combination of them, on a variety of signals. You can optionally put a filter and a parallel filter on them. p-fields: ----------- p4: pitch if below 20, otherwise frequency p5: amplitude, in decibels if 0 or negative (calculated with ampdbfs), fraction of 0dbfs if positive (between 0 and 1) Widgets: ------------ On top you see knobs, a checkbox and a dropdownlist for the envelope, the effects envelope and the LFO. See the comment of the UDO mvpLFO for more explanation. Most knobs below that come in groups of three with two checkboxes: - The first knob sets the 'default' value of the parameter - The first checkbox determines whether the effects envelope is used on this parameter. - The second knob scales the effects envelope when adding it to the default value. This happens such that the parameter has the default value in the sustain part. - The second checkbox determines whether the LFO is used on this parameter. - The third knob scales the LFO signal when adding it to the default value. This happens such that the parameter is the default value in the delay, and after that the default value is the average of the minimum and maximum value. Adding the effects envelope and/or LFO allows you to create dynamic effects. Make these bigger for funny sounds; otherwise keep them low or turn them off. LFO strengths of even less than 0.01 times the maximum value are often still audible and more pleasing and less crazy than higher values. On the left you have the two sources. The dropdown boxes let you choose the source types. Between parentheses you see what par A and par B do with the source, e.g. pw for a square wave. Some source types don't use par A or par B, for these there is nothing between parentheses. Other source types only use par A, there you see the effect between parentheses. Other source types use both, there you see the effects between parentheses separated by a comma. If one of the source types is none, the other is used. They should not be both none. If they are both not none, you can choose how to mix them, their frequency ratio, and whether you want a phase difference between them or detuning, and how the phase difference or detuning changes. See the comment of the UDO mvpMix for more explanation. In the middle you first have the chorusing. You can choose between no chorusing, or 2, 3 or 4 voices. You can also specify the frequency deviation. If you use 3 or 4 voices, you can specify the strength of the outer voices with respect to the middle voice(s). - If strength is 0, only the inner voice(s) are present. - If strength is 0.5, all voices have the same strength. - If strength is 1, only the outer voices are present. Below that you see the flanger. You can turn it on or off, and specify the delay (relative to the period of the signal, 0 to 2) and the amount of feedback. Below that you can specify how the effects envelope and the LFO change the pitch (the 'default' pitch is p4, so there are only two knobs here). On the right you see the filter and the parallel filter. The filter filters the signal if it is turned on. The parallel filter adds a filtered version of the signal to what you already have. If x is the signal after chorusing and flanger, xf is the filtered signal of x, and xp is the parallel filtered signal of x, you get the following: Filter off, parallel filter off: x Filter on, parallel filter off: xf Filter off, parallel filter on: x + xp Filter on, parallel filter on: xf + xp For both filters you can choose balancing, but you may also consider using the volume adjust knob to save CPU load. The panning has three modes: - Normal - Split components: divides the two sources between left and right; 0 means same on left and right If one of the source types is none, normal panning is used - Split chorusing: divides the voices of the chorusing between left and right; 0 means same on left and right If no chorusing is used, normal panning is used The knob 'Volume adjust' lets you adjust the volume, so you can make different presets sound equally loud. It simply multiplies the volume by ampdb of its value. You can let the instrument calculate the optimal value for that by making a condition true near the end of the code, using the UDO mvpPrintAmpCorrection2. This knob can also be used in automation to change the volume. The text box labeled 'Suggested pitches' is only informative, it doesn't influence the produced sound. It helps you remind for which pitches the preset sounds good.

Author: Atte André Jensen (atte.dk), 2008 Simple sampleplayer, using ftables to store sample. Changelog: 1.3 * Minor code cleanup 1.2 * Works with both stereo and mono samples * Pan now pans both left and right 1.1 * Added 'loop' option 1.0 * Initial release

Abacuss v1.1, by Brian Wong, 2008. Freq0 is the fundamental frequency obtained from p5 in the score, all the other frequencies are calculated by multiplying the value in the corresponding knob by the fundamental frequency. The Resonant Lowpass filter can be turned on and off, and is controlled from the lineobject widget. ;inst strt dur amp freq ;--------------------------------------------------------------------------------------------------- p1 p2 p3 p4 p5 v1.0 - Jan. 28, 2008 v1.1- Feb. 6, 2008 : added automatable panning using the pan_gm2 UDO.

Pentadd v1.02 - by Brian Wong, 2008 Pentadd is a 5-oscillator additive synth with dynamic panning. It uses the 5 pfields shown below, and most of the controls are in the BSB lineobject widget. Each oscillator has a separate amplitude envelope and frequency curve that can be edited from the lineobject widget. The pan control is also implemented in the lineobject. All lines are converted to exponential curves except for the panning. The simple design of the instrument should allow easy expansion to add more oscillators or ftables to suit the needs of the moment. I welcome any discussion or suggestions on the blue list. ;inst strt dur amp freq ;---------------------------------------------------------------------------------------------------- p1 p2 p3 p4 p5 v1.01 - added invdur variable to optimize code, changed startup preset to "1st Five Harmonics". v1.02 - changed colours in labels

Outputs audio from fluidsynth engine using fluidOut opcode, to be used in conjunction with the "SoundFont Instrument"

Generic SoundFont playing Instrument -uses Fluid opcodes (Csound 5+) -Should be used with "FluidEngine Out" Instrument

This is a 4 oscillator synthesizer. Each oscillator uses FM synthesis to produce its output (thereby actually making this an 8 oscillator synthesizer), but the output from the oscillators can be combined via additive synthesis or ring modulation. Each FM signal has its choice of waveform for the carrier and the modulating oscillator. There is also some optional HP/LP filtering available for the final signal.

Addtive Synth Organ

3-Oscilator Subtractive Synth