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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

DESCRIPTION: Stretcher is a sampler for mono and stereo samples. It is capable of stretching the sample in realtime. Or plays them in reverse. Attention: by activating the filter, also the filter envelope drawings become available. In Midi Pitch mode, Midi C4 = normal pitch of sample In Frequency mode, frequency note 8.00 = normal pitch of sample Also, real values can be used. 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 i2 0 0.2 1 1.0 ; P4 = real value, 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, February 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: SFloop loads a Soundfont but does not use any of the transformations and modulations as is embedded in Soundfonts, like looping points, reverb or chorus. You have to set your own looping points here- allowing for new 'sounds' on the existing and well-known samples like piano, banjo, guitar etc. where p4 = frequency or Midi Pitch, p5 = amplitude. For more info: https://github.com/kunstmusik/blue/wiki/Fond-of-Soundfont Reason for this instrument is to have easy acces to those well-known acoustical musical instruments but with different, changeable loop points and interesting transformation options. It makes the well-known and mostly harmonic sounds "only" basic material to start from. Note: Big SoundFonts (> 100MB)take a while to load and play. 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 3.2, December 2020 - added color option, for accessing a sample on a different key version 3.1, November 2020 - added tremolo envelope version 2.1, October 2020 - againbig overhaul: new filters - filters have amplitude or frequency/pitch followers (exponentially) - checkbox for P4 = frequency or MidiPitch - 2 types of filter envelopes, can be mixed version 2.0, July 2020 - big changes: new vibrato, new dynamics, version 1, September 2019 - built with XlooperSolo as a template

DESCRIPTION: SFplayer, based on the splay3 opcode, loads a Soundfont and plays the presets of a SoundFont. It does not use any of the transformations like internal modulations, reverb or chorus as embedde in the preset of the SoundFont, but does however respect the start and end loop points of the Soundfonts' preset. SFplayer is suited for use in microtonal compositions; for instance, the PianoRoll Object can contain an arbitrary tone scale, SFplayer will follow the notes nicely when microtuned. SFplayer can take midi note numbers as entries for p4, but also frequencies. A PianoRoll for example, must be set to Pch Generation --> MIDI or Pch Generation --> Frequency. P5 should be set to values 0 - 1. Reason for this instrument is my wish to have easy acces to (acoustic) musical instruments but with different and interesting transformation options and microtonality. This makes the well-known and mostly harmonic sounds a basic material, to start molding new sounds with an already acoustical property. Note: Big SoundFonts (> 100MB)take a while to load and play. 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 3.1, December 2020 - added color option, for accessing a sample on a different key - added tremolo envelope version 3.0, October 2020 - big overhaul: new filters - filters have amplitude or frequency/pitch followers (exponentially) - switch for P4 = frequency or MidiPitch - 2 types of filter envelopes, can be mixed - optional Score Module version 2.0, July 2020 - big changes in: dynamics, vibrato, version 1.1, August 2019 - added Show Midi Key, so you know what keys are hit and adjust the range of the preset accordingly - added Depth slider for filters version 1, August 2019 - first version- that's why i call it version 1

DESCRIPTION: SFinst, based on the sfinstr3 opcode, loads the instruments of a Soundfont. So it does NOT use the Presets of the Soundfont, but the instruments the Presets are made of. SFinst is suited for use in microtonal compositions; for instance, the PianoRoll Object can contain an arbitrary tone scale, SFinst will follow the notes nicely when microtuned. SFinst can use midi note numbers as entries for p4, but also frequencies. Set the parameter 'scoreP4' accordingly. A PianoRoll for example, must be set to Pch Generation --> Frequency (and not the Midi note pitch), while a PianoRoll using the familiar 12TET set Pch Generation --> MIDI. Or use the AL. Velocity (= p5) is from 0 - 1, but scaled to 0-127 according to the SF expectency. It then follows the normal Soundfont/midi protocol. Reason for this instrument is to have easy acces to those well-known acoustical musical instruments but with different and interesting transformation options and microtonality. That makes the well-known and mostly harmonic sounds "only" basic material to start molding new sounds with an already acoustal property. ATTENTION: SoundFont Presets can be made of stereo samples. According to SF Protocol, these samples are divided over 2 mono Instruments, each with its own number. This means that if you select an Instrument, you load only the Left channel or the Right channel if it is part of a stereo sample. There is a checkbox that makes the mono sound a pseudostereo sound by panning the notes freely over the stereo image. 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 3.2, December 2020 - added color option, for accessing a sample on a different key - changed from mono reading to stereo reading of SF version 3.1, November 2020 - added tremolo envelope version 3.0, October 2020 - big overhaul: new filters - filters have amplitude or frequency/pitch followers (exponentially) - switch for P4 = frequency or MidiPitch - 2 types of filter envelopes, can be mixed - optional Score Module version 2.0, July 2020 - big changes in: dynamics, vibrato, version 1.1, August 2019 - added Show Midi Key, so you know what keys are hit and adjust the range of the preset accordingly - added Depth slider for filters version 1, August 2019 - first version- that's why i call it version 1

DESCRIPTION: XlooperSolo is a sampler for mono and stereo samples. It can loop at any arbirary points, the loop points can be changed while playing. A fade time should is a guarantee (most of the times) for clickless loops. Attention: by activating the filter, also the filter envelope drawings become available. In Midi Pitch mode, Midi C4 = normal pitch of sample In Frequency mode, frequency note 8.00 = normal pitch of sample 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 1.2, July 2020 - changed filters section - filter now has Key Follow and Amp Follow option version 1.1, June 2020 - new Vibrato UDO - score format real/midi switch. In midi, accepts 'in-between' note values - better names for ALs - added jitter for amplitude, filter and pitch version 1.0, November 2019 - variable amplitude can create some tremolo effects - ADSR with variable speed - on/off for ADSR

DESCRIPTION: Idea of XlooperSeq is to use a maximum of samples and play them in series or parallel so that it feels like one complicated cluster. Samples can be played simultaneously also, if you set the notes start at the same start time. XlooperSeq, where X stands for crossfade time of the looping and reminds of the XlooperSolo. Turns stereo and mono samples into mono output, but can individually be panned in order to create one dense sonic structure. There is a report function that can be turned on or off to find out if certain samples are loaded all right or invalid. Samples can be muted and unmuted while playing, using their Automation Lines 'mute sample X'. Sometimes a sample module stays silent; it has to do when loop direction is set to 'forward and backwards', due to the in and out fade time the loop gets out of the sample length. CONTROLLERS: Sample x 8: mute - mute this sample start in sample - skip time in sample looprange begin/end - set loop points for begin and end pitch - pitch of sample report - switch to find out if certain samples are loaded all right or invalid direction - loop forward/backward/back-and-forth volume - gain for each sample dirt - subtle random gain variations pan - position this sample in stereo field Vibrato: vibrato depth - amount of gain of vibrato vibrato speed - speed of vibrato Global: ADSR: ADSR ON - switch on/off for envelope generator ADSR attack/decay/sustain/release - set a-d-s-r ADSR speed - retrigger ADSR envelope Xfade: looppoints Xfade - set fade time in order to avoid clicks in loop for all samples looppoints dirt - small variations in loop fades for all samples autopanning - distribute and move sample across stereo field pan speed - speed of moving across stereo field tuning - total tuning of XlooperSeq score format - how to read p4, p4: as real values, midi notes or autodetection L/R - output gain Left/Right channel SCORE EXAMPLE: real value: freq amp sample i1 0 2 100 1 4 midi note: note veloc sample i1 0 2 60 100 4 --- version 1.1, June 2020 - added comments on controllers and description - renamed ALs - revamped sample choice system - added Comments version1, November 2019 - 2 types of ADSR

DESCRIPTION: Idea of Xlooper8 is to use up to 8 samples and play them in together so that it feels like one big, complicated cluster. It is a way to get colors in the clusters by combining more than just 1 sample. Samples can be muted and unmuted while playing, for example by setting their respective Automation Lines 'mute sample X'. Xlooper8, where X stands for crossfade time of the looping. Turns stereo and mono samples into mono output, but can individually be panned in order to create one dense sonic structure. Or you can choose the autopanning. There is a report function to find out if certain samples are not loaded or invalid. Sometimes a sample module stays silent; it has to do when loop direction is set to 'forward and backwards', due to the in and out fade time the loop gets out of the available sample length. In Midi Pitch mode, Midi C4 = normal pitch of sample In Frequency mode, frequency note 8.00 = normal pitch of sample note example: i2 0 0.8 60 1.0 ; P4 = midi value, P5 = amplitude, P6 = sample i2 0 0.2 261.625565 1.0 ; P4 = frequency, P5 = amplitude, P6 = sample 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 1.3, November 2020 - added filter - added envelopes version 1.2, October 2020 - added BSB information comments - changed P4, P5 UDO version 1.1, June 2020 - added comments on controllers and description - renamed ALs - revamped sample choice system - added Comments version1, November 2019 - 2 types of ADSR

DESCRIPTION: Cycloop can be positioned in between the sampler and the granulator, as it has some aspects of both of them. Cycloop can be seen as an effect, as transformation is done on the signal. On the other hand, the direct signal can be mixed in and together they can sound as one instrument. The instrument takes in the sample and loops it. By retriggering the overlaps and other features like pitch and reading forward/backwards, interesting and evolving textures can be created. The number of overlaps gives the sound an echoic quality that reminds of the granulators. Attention: by activating the filter, also the filter envelope drawings become available. 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. SCORE EXAMPLE: option 1, real values: frequency amplitude i1 0 2 261.625565 1.0 option 2, Midi pitch: pitch amplitude i1 0 3 60 1.0 --- version 1.2, October 2020 - big filter changes - added BSB information comments - changed P4 freq/midi pitch version 1.1, September 2020 - changes: filter - added mix direct/instr version 1.0, June 2020 - initial version

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: Partikkulary is a wrapper for the partikkel opcode, and more.... The opcode is designed to stretch granualtion methods to the limit. - there are 4 samples as input to granulate. - these 4 samples can be mixed - the endresult has an amplitude envelope and ADSR note examples: 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 1.1, October 2020 - added filter, that allows key follow and amp follow - allows switching between Midi pitch/frequency mode

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

SFluidly is a wrapper for Fluidsynth: http://www.fluidsynth.org/ Soundfonts have some modulators embedded. They were put there at creation. Modulations depends on if and how much of these embedded modulators are present. All modulators are global. This means that they effect all the notes playing at that point in time. For more info: https://github.com/kunstmusik/blue/wiki/Fond-of-Soundfont version 1.0, September 2019

The ImagePlayer is a troublemaker; this instrument plays images. The image must in the PNG format. But still it's hard to predict what the sounding result will be, even if the 3 colors red, blue and green scanned and thus have a steady pattern as they are not moving or evolving entities in the image... A few things to understand: - No sound will be produced when only black image; black is the silent canvas - 3 basic colours (RGB) on which the image opcodes react: red, blue and green. - The three colours (red, green and blue) have volume faders. - The image is scanned from left to right (X axis) and from top to bottom (Y axis) or reverse and rotated - Top of the image = the lower frequencies, bottom of the image = the higher frequencies or reverse - take a low scanning speed for adsyn, so with low base frequency, there will be no problems with Nyquist distortion (aliasing). The harmonics will be built upon the frequency set by p4 - can take a lot of CPU The lpf18 filter can be used to avoid further aliasing when experimenting at higher base frequencies . . example of score: i1 0 20 244 where p4 = frequency. please have a look at the Wiki page: https://github.com/kunstmusik/blue/wiki/A-Blue-instrument:-ImagePlayer version 3.1, december 2017 - rotates picture if image width is smaller then image height - replaced ADSRs by multicurve envelopes - added more envelopes - added scaling for amp, filter frequency and filter depth and pitch bend - changed AL names to more understandable ones version 3.0, december 2017 - layout converted to JavaFX standard, blue version >=2.7 - renewed layout so that: - XY controller represents the portion of the image to be scanned - added filter ADSR with depth control - added stereo option that puts RED to the left, GREEN in the middle and BLUE to the right - rescaling of base frequency when playing - more than ImagePlayer can be used in a composition - option to show image in external viewer (xviewer <file> for example). Please change according to your needs version 2.3, december 2015 - improved text for labels - added a setting for basefrequency instead of p4 - added anti-aliasing switch & aliasing filter is adjustable version 2.2, nvember 2015 - changed label 'logaritmic' to 'exponential' - XY positions are now automatable - added labels to explain the range of image to be scanned version 2.1 , november 2015 - fixed file input selector - removed fill table option because prone to errors - removed switch for inverse X Y read out (not interesting enough) - XY position for number of overtones - select log or linear ftable to hold frequencies - choose the part of the image width to play - changed filter to LPF18 with distortion - red, green and blue volume now undependent from each other version 2 , october 2015 - stereo option - add limit to voices to save CPU - variable ftable to hold frequencies - changed filter to LPF18 with distortion - switch for inverse X Y read out - ADSR - 5 basic waveforms * version 1.1, april 2013 - repaired initialization bug - show width and heigth of image - transition from raw sound (=low resolution) to clean sound (+high resolution) is fluid now - added panning * initial version 1, march 2013

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

Midi Control (heavily based on MidiCC Control) sends notes AND midi Continuous Controller data to a software synth or external hardware synth. with Midi Control, you can assign the midi controllers of an external or software synth. CCs for brightness, volume, panning etc. etc., in order to draw Automation Lines in the Object. version 1, octobre 2018 - needs Blue >= 2.7.2

MidiCC Control (was MidiControl) sends midi Continuous Controller data to a software synth or external hardware synth. This instrument is meant to be used together with another instrument that produces the notes. This other midi instrument migth be as simple as: midion p1, p4 , p5 ;channel, number of note, velocity of note Both instruments (the note producer and midiControl) must run in parallel, using the same midi channel. with MidiCC Control, you can assign for example the brightness, volume, panning etc. etc. to one of the faders and draw the Automation Lines. version 1.2, oktober 2018 - small redesign and new name - removed speed transfer rule version 1, november 2017 aka MidiControl - needs Blue >= 2.7.2

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 G uses the grain3 opcode. The G in the name represents 'grain3'. scheme: FTMORF ---------GRAIN3--------(PHASER)--------(SWELL)-------OUTS I l I (LFOs for phasing) I ( LFOs for deviation ) version 1.1, december 2017 - added Arpeggio module (UDO) 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 - optional dirt gives the deviation pitch irregular accents - 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 - exp or linear envelope score in Generic Object may look like: i1 0 2 190 1 where p4 = freq p5 = amp (0-1)

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

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 F uses the foscili opcode. The F in the name "Type F" stands for 'foscili'. scheme: FTMORF ---------FOSCIL --------(PHASER)--------(SWELL)------OUTPUT mono/stereo I l I l I l ( LFOs for INDEX ) (LFOs for phasing) 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 - dirt gives the Index of the FM irragular accents - Amplitude Modulation is optional - 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 score in Generic Object may look like: i1 0 2 190 1 where p4 = freq p5 = amp (0-1)

FM_Clang Cabbage original written by Iain McCurdy, 2014 Blue Instrument version translated and extended by Steven Yi, 2017.10.06 ============ DESCRIPTION ============ An FM synthesis based instrument that specialises in creating percussive metallic 'clang'-like sounds. AMPLITUDE - controls pertaining to the control of each notes amplitude Level - Global amplitude control Attack - Attack time. Actually just a small range of control but increasing it can soften the attack subtly Vel.Att. - Degree to which lower note velocities lengthen attack time. Duration - Duration of the overall amplitude decay. Note that depending on other settings notes may actually sound for a shorter time. Sustain - Similar to the effect of a sustain pedal being depressed. MODULATOR - controls pertaining to the modulator in the FM algorithm Ratio - Ratio of the modulator frequency to that of the carrier. Vel.Index - Index of modulation. This is also always dependent upon note velocity. Decay - Rate of decay of the index. This is heard as the rate of loss of spectral energy or damping. CARRIER - controls pertaining to the carrier in the FM algorithm Kyb.Scale - Keyboard scaling of the carrier frequency. Zero means that frequencies follow normal keyboard temperament. Values less than zero mean that frequency intervals from note to note will be compressed. Values greater than zero mean that frequency intervals from note to note will be expanded. DETUNE - Two FM carrier/modulator pairings can be created with a detuning offset between them to create beating effects. Detuning will also be unique (and repeatable) from note to note to give the instrument character. Mix - Amplitude of the second FM pair. This will control the amplitude of beating (amplitude modulation) effects resulting from the detuning. Amount - Amplitude of the detuning intervals STEREO - A stereo broadening effect by applying short delays to the left and right channels. The exact durations of the delays will be unique (and repeatable) from note to note to give the instrument character. Dry/Wet - A dry/wet control of this effect Width - A control of the durations of the delays which is heard as a control of the stereo width of the effect. NOISE - Filtered gaussian noise can be added to the modulator ratio to add discordancy to the tone. Amount - Amplitude of the gaussian noise signal Filter - Narrowness of the bandpass filtering applied to the gaussian noise Follow Modulator (checkbox) - Selects whether the bandpass filter follows the modulator frequency. If 'off' it simply follows the carrier frequency. LFO - A sine shaped LFO added to the modulator ratio of an auxilliary FM pair which is then mixed to an unmodulated pair. This can introduce spectral modulation, as in a bell swinging back and forth as it rings. Mix - Amount of modulated signal mixed into the output. Depth - Depth of the LFO modulation. Rate - Rate of the LFO modulation. Kybd.Follow - Extend to which the rate of LFO modulation follows keyboard notes - keyboard following means that lower notes will modulate at lower frequencies, higher notes will modulate at higher frequencies. FILTER - Filtering of the sound LPF - A lowpass filter that follows the fundemental frequency of the FM pair Filter Fundemental (checkbox) - If active, the fundemental will be filtered off. REVERB - a reverb effect using the freeverb opcode USE MIDI PFIELDS - Allow working with p4=MIDI Key (0-127), p5=MIDI Vel (0-127), or with p4=PCH or Freq, p5 = amdbfs [-96, 0] ========= REQUIRES ========= * Blue 2.7.1 * Csound 6+ ======== HISTORY ======== v1.1 * Reworked ftable generation into instrument code to make instrument encapsulated v1.0 * Translated to Blue from Cabbage, added non-MIDI value pfield operation of instrument

Original Author: Atte Andr�© Jensen (atte.dk), 2008 Updated by Steven Yi, 2017-09-24 Spectral pad, with sounds similar to Roland D-50 and other late 80's, early 90's hardware synths. Changelog: 1.2 * fixed missing/additional commas, adjusted code to work with Automatable widgets 1.1 * Added Brightness control 1.0 * Initial release

Original Author of DistortionGuitar version 1: Atte André Jensen (atte.dk), 2007 Author of DistortionGuitar version 2: Menno Knevel, 2016 Originally: Distorted guitar with flanger, but now extended with some features version 2.3, november 2016 - lin --> exp faders for better mixing of fry and wet signals in the 3 resonance modes - option to set in score: -1. When in score p4 or p5 = -1, then Automation Line makes the value for p4 or p5 (experimental) version 2.2, november 2016 - all checbox code now checks out - improved delay start of vibrato version 2.1, november 2016 - added vibrato - added Zwel for every one of the 3 voices - improved mixing of the 3 resonace filters - option to change pitch while the note plays version 2, november 2016 - detune is variable now - now 0 dbfs = 1 (was = 32000) - filter attack and decay variable now - UDO with pluck has setksmps 1, so is independent of the ksmps of the csd - improved parameter value handling for stretching of pluck (method 4) EFFECT SECTION: - added 3 resonance filters for shaping of the body of the guitar - turn distortion on/off - turn Amplitude MOdulation on/off. several presets are present.

Original Author of DistortionGuitar version 1: Atte André Jensen (atte.dk), 2007 Author of DistortionGuitar version 2: Menno Knevel, 2016 Originally: Distorted guitar with flanger, but now extended with some features version 2.2, november 2016 - all checbox code now checks out - improved delay start of vibrato version 2.1, november 2016 - added vibrato - added Zwel for every one of the 3 voices - improved mixing of the 3 resonace filters - option to change pitch while the note plays version 2, november 2016 - detune is variable now - now 0 dbfs = 1 (was = 32000) - filter attack and decay variable now - UDO with pluck has setksmps 1, so is independent of the ksmps of the csd - improved parameter value handling for stretching of pluck (method 4) EFFECT SECTION: - added 3 resonance filters for shaping of the body of the guitar - turn distortion on/off - turn Amplitude MOdulation on/off. several presets are present.

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

An example of graphical enveloping, using an oscilator version 1.1, november 2015 - linear or exponential envelope Usage: p4 = frequency, for example: i1 0 2 1200

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

BdTR with extra features --------------------------------- Author: Mark Van Peteghem Based on the instrument BdTR by Atte Andr? Jensen (atte.dk), 2007 This instrument adds extra features to the instrument BdTR. It has one extra p-field, p4, that is the amplitude. If it is 0 or less it is in decibels below 0dbfs, otherwise it is multiplied with 0dbfs (so should be 1 or less). Instead of custom waveforms, it lets you choose between a sine and a triangle. For the sine you can choose the amplitude of the second harmonic, for the triangle you can choose the pulse width. Furthermore you can choose how the sine or triangle should be distorted, allowing a big variety of waveforms. The envelope type can be exponential or linear. With a linear envelope the perceived duration is closer to the one specified. An optional lowpass filter lets you adjust the tonality. The RM knob lets you optionally generate a second similar signal with a higher frequency, that is multiplied with the original signal. The volume adjust knob lets you adjust the volume so that when p4 is 1 it will actually be 0dbfs. Changing the condition 1==0 to 1==1 prints out what value should be there to achieve this (set p4 to 0 or 1 to do this).

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)

This is a basic instrument using the pluck opcode. 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. It does not allow you to choose a custom table. You can specify a release time, which is necessary if the note ends before the signal dies out by itself. I added an experimental feature: you can silence the begin of the instrument, which takes away the transients at the very beginning, producing a more monotonous but often still beautiful sound. Of course you have to take into account in your score that if you use this feature, the signal will start a little later. 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

Palamino v0.62 : This instrument is a work in progress. Note that the Hi-Hat and Chrumpet patch has been reproduced as static ftables. All pfields from p6 and up are optional. Use a value of -1 in the score to access default (GUI) settings. - Dec. 16, 2009 : turned off automation on widgets to fix Csound API incompatibility inst strt dur amp fq bfn cmfn vdel vrel vdepth vrate pan cfn efn mfn rfn

; Eastcross : Adapted from the Crossfade instrument in the EAST (Early Analogue Synthesis Techniques) tutorial by Jacob Joaquin. By Brian Wong, 2007. - Dec. 15, 2009 - v0.3 - Automation turned off for BSB knobs. For p-fields 6, 7 and 8 use a value of -1 to use GUI settings, or enter any other value from 0 to 1 into the score. inst strt dur amp frq rise dec pan

Fossil v0- This synth is a simple single oscillator foscili-based monophonic FM synth, that is controlled entirely via automation. - Brian Wong, October 2009

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 Sample player, targeted at playing breakbeats. Automode will guess the tempo of the loop, if: a) The tempo of the beat is between "Maximum BPM" and "Maximum BPM" / 2 b) The beat is in 4/4 and either 1/4, 1/2, 1, 2, 4 (etc) bars long Changelog: 1.9 * Minor code cleanup 1.8 * Stretching works in stereo * UI cleanup * Tuning in stretch-mode (doesn't affect tempo) 1.7 * Stretching is done with syncgrain only * Code cleanup 1.6.1 * Experimented with syncgrain in three bands 1.6 * Experimenting with both sndarp, granule and syncgrain 1.5 * Added iwarpgain to reduce output of sndwarp 1.4 * Added "time stretching" (using sndwarp), however the sound quality is quite poor :-( 1.3 * More efficient code in automode 1.2 * Fixed bug in automode 1.1 * Added auto mode, will try to guess number of bars 1.0 * Initial release

This instrument creates a Shepard tone. See http://en.wikipedia.org/wiki/Shepard_tone for more info on what a Shepard tone is. p-fields: p4 is the pitch or frequency p5 is the amplitude BSB widgets: speed is how fast each component varies envelope shape determines when the amplitude of a component is the highest, more to the beginning or to the end; use 0.5 for regular Shepard tone pulse width is used for the vco2 opcode if the waveform is square or triangle, values are limited to the stable range speed, envelope shape, pulse width and panning have an extra text field which in the code is appended to the value of the knob; use e.g. *p6 to change the value from the score For a regular Shepard tone, use a sine wave. The other waveforms have strange artefacts, which can be nice.

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

Author: Atte André Jensen (atte.dk), 2007 Simple pad with resonance. Changelog: 1.1 * Volume now automateable 1.0 * Initial release

Author: Atte André Jensen (atte.dk), 2007 Emulates snare drum sound found in Roland TR808. Changelog: 1.1 * Volume now automateable 1.0 * Initial release

This is a port of Oeyvind Brandtsegg's partikkelFLTK instrument, created to experiment with his opcode partikkel . This BSB takes live midi input from external midi controler devices and is capable to record midi messages as automations and play them back. It also remembers the last midi fader positions before the performance ended, the same positions are initialised on the next performance. In order to be able to run this instrument you first need to create a plain text file in the same directory where your blue project resides and call it "midinit" (no extensions), and copy and paste the lines below in it. ======= TABLE 10001 size: 16 values ====== flen: 16 lenmask: 15 lobits: 20 lomask: 1048575 lodiv: 0.000001 cvtbas: 0.000000 cpscvt: 0.000000 loopmode1: 0 loopmode2: 0 begin1: 0 end1: 0 begin2: 0 end2: 0 soundend: 0 flenfrms: 16 nchnls: 1 fno: 10001 gen01args.gen01: 0.000000 gen01args.ifilno: 0.000000 gen01args.iskptim: 0.000000 gen01args.iformat: 0.000000 gen01args.channel: 0.000000 gen01args.sample_rate: 0.000000 ---------END OF HEADER-------------- 127.000000 60.000000 0.000000 0.118581 0.001016 0.000000 0.000000 24.000000 69.000000 35.000000 0.000000 0.000000 70706400890471310414389826289664.000000 0.000000 0.385827 0.000000 0.000000 ---------END OF TABLE---------------

Author: Atte André Jensen (atte.dk), 2008 Simple wrapper around the shaker opcode Changelog: 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.

b64 version 1.1 Released: 2008.02.04 Author: Steven Yi This instrument is inspired by the SID sound chip commonly used in Commodore 64 computers, using simliar waveforms and options for ring modulation. [History] v1.1 - 2008.02.04 -Fixed Triangle wave not being generated -Added use of pan_gm2 for panning -Added automatable pan knob to interface v1.0 - 2007.12.14 -initial release

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

Author: Atte André Jensen (atte.dk), 2008 Metronome click. Could be played with a repeating soundObject containing (supposed the click instrument has an instance named "click"): i"click" 0 1 1 i"click" + . 2 i"click" + . 2 i"click" + . 2 Changelog: 1.0 * Initial release

A simple wrapper around the prepiano opcode.

Author: Atte André Jensen (atte.dk), 2008 Spectral pad, with sounds similar to Roland D-50 and other late 80's, early 90's hardware synths. Changelog: 1.1 * Added Brightness control 1.0 * Initial release

Author: Atte André Jensen (atte.dk), 2007 Animated, resonant pad/drone/bass instrument with tempo sync. Changelog: 1.1 * Syncs to tempo using miditempo * Added more subdivisions 1.0 * Initial release

Author: Atte André Jensen (atte.dk), 2008 Filtered noise, with resonance and tempo sync'ed LFO on filter. Changelog: 1.1 * Uses miditempo to sync to tempo * Added hipass filter * Added more subdivisions 1.0 * Initial release

Author: Atte André Jensen (atte.dk), 2008 Additive instrument, featuring detune, pan and drawable amplitude LFO for each of the 8 overtones. Changelog: 1.1 * Fixed bug: kvol/ivol not used in code 1.0 * Initial release

Author: Atte André Jensen (atte.dk), 2007 Distorted guitar with flanger.

Author: Atte André Jensen (atte.dk), 2007 Mellow bell-like sound with reverse.

Author: Atte André Jensen (atte.dk), 2007 Emulates bass drum sound found in the Roland TR series, esp. TR909 and TR808, but has some other sonic capabilities as well.

Author: Atte André Jensen (atte.dk), 2007 Percussive instrument with resonant filter and distortion.

Author: Atte André Jensen (atte.dk), 2007 Simple, round bass.

Author: Atte André Jensen (atte.dk), 2007 For background on the wterrain opcode see http://csounds.com/jmc/Articles/Wts/WTS.html This instrument consists of three stacked lines. Each line defines four points that the center point will move between at frequency "Roundtrip Freq" and with a certain radius. The terrain is build using the selected waveform. The first line is panned center, the second panned left and the third right. The panning is controlled by "width". Lines two and three can be detuned in relation to line 1. The resulting stereo signal is applied a single ADSR envelope. TIPS: * For more depth, run the instrument through a chorus effect. * The higher the "radius" the buzzier the sound. * The closer the points are to 0,0

PHASESHAPER - Orch version 1.1 Released: 2007.12.14 Author: Steven Yi This instrument is modeled after the Casio CZ-101 and its Phase Distortion Synthesis (called phaseshaping here by the author, due to its likeness to waveshaping but on the phasor). The PhaseShaper instrument is not meant to exactly replicate the CZ-101 but instead is inspired by it, intentionally leaving out some features (like noise modulation and amp key follow) and adding in unique ones (like the ability to draw in the phaseshaping tables). It is suggested the user read about the CZ-101 before using this instrument. This instruments oscilators are made up of a phasor indexing into a phaseshape table, which in turns yields the indexes to use to read into a sine wave table. [History] v1.1 - 2007.12.14 -fixed use of envelopes (OSC 2 attack was not being used due to incorrect setting of widget property, some OSC2 values were using OSC1 settings) -removed use of balance opcode -fixed calculation of p3 addition to check release time of OSC 1 or OSC 2 v1.0 - 2007.12.08 -initial release

PHASESHAPER version 1.1 Released: 2007.12.14 Author: Steven Yi This instrument is modeled after the Casio CZ-101 and its Phase Distortion Synthesis (called phaseshaping here by the author, due to its likeness to waveshaping but on the phasor). The PhaseShaper instrument is not meant to exactly replicate the CZ-101 but instead is inspired by it, intentionally leaving out some features (like noise modulation and amp key follow) and adding in unique ones (like the ability to draw in the phaseshaping tables). It is suggested the user read about the CZ-101 before using this instrument. This instruments oscilators are made up of a phasor indexing into a phaseshape table, which in turns yields the indexes to use to read into a sine wave table. [History] v1.1 - 2007.12.14 -fixed use of envelopes (OSC 2 attack was not being used due to incorrect setting of widget property, some OSC2 values were using OSC1 settings) -removed use of balance opcode -fixed calculation of p3 addition to check release time of OSC 1 or OSC 2 v1.0 - 2007.12.08 -initial release

2 osc synth, read the comments

Part of a series of nonlinear oscillators

Part of a series of nonlinear oscillators

Part of a series of nonlinear oscillators

Part of a series of nonlinear oscillators

Part of a series of nonlinear oscillators

Part of a series of nonlinear oscillators

A simple fm synth with custom sine wave table size. Use low values (256 for example) to get some sweet oldschool aliasing. You should probably examine the source to understand how it works as labels by the knobs arent very descriptive.

FOF instrument for soprano voice, using formant values from Csound manual

FOF instrument for countertenor voice, using formant values from Csound manual

FOF instrument for alto voice, using formant values from Csound manual

FOF instrument for tenor voice, using formant values from Csound manual

FOF instrument for bass voice, using formant values from Csound manual

Granular synthesis of evolving saw wave

Emulation of a Waldorf Microwave. Code originally by Matt Gerassimoff, translation to blue by Steven Yi.

6 reson percussion instrument (Modal Synthesis)

6 mode percussion instrument (Modal Synthesis)

Emulation of the VCS3 as originally coded by Steven Cook. Six oscillators, filter, envelope, external audio input, joystick input, noise generator, ring modulation, and equalization. Import may take awhile...

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

tabloid is a wavetable synthesizer. at its core, it provides a simple graphical way of editing linear ftables, then provides two oscillators for said ftable, both of which can be detuned. after that, it gets a little crazy. the second oscillator can also be synced to the note frequency, and said sync can be controlled with an envelope. tabloid also has envelopes for the filter (lpf18) and the amplifier. this is a beta release, some rough edges need to be polished. please give feedback on the blue list.

A simple sample player. It loads in one sample, lets you change the pitch, and plays it.

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