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| vividshaper_reference_manual [2024/12/28 15:21] – lars | vividshaper_reference_manual [2026/09/09 17:11] (current) – lars | ||
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| * left-right panning | * left-right panning | ||
| * note (including cent note) | * note (including cent note) | ||
| + | * pitch bend information | ||
| * A wave table array containing 128 samples | * A wave table array containing 128 samples | ||
| Line 28: | Line 29: | ||
| vol[1] = VSADSRE(1, | vol[1] = VSADSRE(1, | ||
| active = vol[1] > 0.00001 | active = vol[1] > 0.00001 | ||
| + | pitchbend = pitchbend*2 | ||
| </ | </ | ||
| - | This example | + | This example |
| This is not the playback frequency of the note (i.e. which note that is being played). The speed of the 128 samples is determined by note[1] for oscillator 1. We don't have to set note[1] explicitly, it is already set by the note you are playing on the keyboard. However, if we do wish to change it, for instance transposing one octave, we can do so by setting: **note[1] = notein+12**. Alternatively, | This is not the playback frequency of the note (i.e. which note that is being played). The speed of the 128 samples is determined by note[1] for oscillator 1. We don't have to set note[1] explicitly, it is already set by the note you are playing on the keyboard. However, if we do wish to change it, for instance transposing one octave, we can do so by setting: **note[1] = notein+12**. Alternatively, | ||
| + | |||
| + | The pitch bend value comes from your keyboard and will be added to the note value for each oscillator after the Lua code has finished. This value is stored in the variable pitchbend, which goes from -2 to +2 (i.e. 2 semi-notes). If note[1] == 40 for oscillator 1 and pitchbend == 0.53, the final note value used to play the note will be 43.53. If you want to turn off pitch bend, you can set pitchbend = 0 anywhere in your code. if you want to pitch bend from -4 to +4, you can just multiply it by 2 (pitchbend = pitchbend *2). This is done in the example above. You can also use the pitch bend value to modulate e.g. something else first, then set it to zero if you don't want it to also bend the notes. | ||
| The volume will then be set to an exponential volume ADSR envelope, with attack=1 sec, decay=1 sec, sustain=0.8, | The volume will then be set to an exponential volume ADSR envelope, with attack=1 sec, decay=1 sec, sustain=0.8, | ||
| Line 48: | Line 52: | ||
| -- Input information | -- Input information | ||
| notein | notein | ||
| + | pitchbend | ||
| tempo -- Current tempo (beats per second, e.g. 120). | tempo -- Current tempo (beats per second, e.g. 120). | ||
| timesignnum | timesignnum | ||
| Line 56: | Line 61: | ||
| gate -- Either 1 or 0 depending on if gate is on or off. | gate -- Either 1 or 0 depending on if gate is on or off. | ||
| gatetimeon | gatetimeon | ||
| + | gon -- Same as gatetimeon(short version name) | ||
| gatetimeoff | gatetimeoff | ||
| + | goff -- Same as gatetimeoff (short version name) | ||
| timeon | timeon | ||
| gtimeon | gtimeon | ||
| Line 68: | Line 75: | ||
| -- Output information | -- Output information | ||
| + | pitchbend | ||
| wave[x] | wave[x] | ||
| panning[x] | panning[x] | ||
| Line 79: | Line 87: | ||
| active | active | ||
| midigenerator | midigenerator | ||
| + | legato | ||
| + | portamento[x] | ||
| + | portamentotype -- 0 for linear, 1 for exponential | ||
| - | -- Delay output information (upcoming feature | + | -- Coming soon in a future version (v1.5 probably) |
| + | sample[x] | ||
| + | loopstart[x] | ||
| + | loopend[x] | ||
| + | sxfade[x] | ||
| + | svol[x] | ||
| + | |||
| + | -- Delay output information | ||
| delay1[x] | delay1[x] | ||
| | | ||
| Line 91: | Line 109: | ||
| d1vol[x] | d1vol[x] | ||
| d2vol[x] | d2vol[x] | ||
| - | ovol[x] -- How much of oscillator x that should be sent to main output (1 = default) | + | |
| - | mvol -- Amplify | + | -- Reverb output information |
| + | reverb1[x] | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | | ||
| + | reverb2[x] | ||
| + | rrvol -- Amplification of reberb2 output | ||
| + | |||
| + | -- Other routes of sound | ||
| + | mvol[x] -- How much of oscillator x that should be sent to main output (1 = default) | ||
| + | mrvol | ||
| + | avol[x] | ||
| + | apanning[x] | ||
| + | arvol -- Amplification on alternative route | ||
| + | |||
| -- Views settings | -- Views settings | ||
| Line 122: | Line 161: | ||
| -- Wave math operators - arguments can be either arrays or scalar factors | -- Wave math operators - arguments can be either arrays or scalar factors | ||
| + | -- These are obsolete in v2.0, but will remain for compatibility reasons. | ||
| + | -- In v2.0, you can instead do: wave[x] = wave1*wave2 or wave[x]=1.5*wave1+wave2 | ||
| + | -- Version 2.0 supports: == <= >= < > + - * / % where comparators return 0 or 1 | ||
| wave[x] = VSMul(wave1, | wave[x] = VSMul(wave1, | ||
| wave[x] = VSMul(wave1, | wave[x] = VSMul(wave1, | ||
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| wave[x] = VSRect(wave[x]) -- Input is a wave array | wave[x] = VSRect(wave[x]) -- Input is a wave array | ||
| - | -- Helper functions for delays | + | -- Helper functions for delays |
| - | VSPingPong(feedback, | + | VSPingPong(feedback, |
| - | VSStereoDelay(feedback, | + | VSStereoDelay(feedback, |
| + | VSMonoDelay(feedback, | ||
| -- Envelopes | -- Envelopes | ||
| + | -- ADSR: | ||
| -- The level is the initial volume. Normally, it would start at zero, but could also be initiated to | -- The level is the initial volume. Normally, it would start at zero, but could also be initiated to | ||
| - | -- start higher. | + | -- start higher. You can also call the function without level as argument (v1.4). It will then be zero. |
| - | vol[x] = VSADSR(attack, | + | vol[x] = VSADSR(attack, |
| - | vol[x] = VSADSRE(attack, | + | vol[x] = VSADSR(attack, |
| + | vol[x] = VSADSRE(attack, | ||
| + | vol[x] = VSADSR(attack, | ||
| + | -- Attack: | ||
| + | vol[x] = VSAttack(attack, | ||
| + | |||
| + | -- Release: | ||
| + | vol[x] = VSRelease(release, | ||
| -- Filters (x = oscillator) | -- Filters (x = oscillator) | ||
| + | -- There are two different filters in VividShaper. One that you apply on the actual waveform and | ||
| + | -- one that you apply as an insert after the output of the oscillator. Both are biquad filters. | ||
| + | -- Common for both filters is that you need to create a filter array containing the biquad values. | ||
| + | -- You don't need to set the values yourself. Instead, this array is created using three different | ||
| + | -- functions, depending on what kind of filter you want. The notein argument is optional from v2.0, | ||
| + | -- but can be used to let the cutoff frequency follow the note. | ||
| + | |||
| + | biquadcoeff = VSLowpass(cutoffFreq, | ||
| + | biquadcoeff = VSLowpass(cutoffFreq, | ||
| + | biquadcoeff = VSBandpass(cutoffFreq, | ||
| + | biquadcoeff = VSBandpass(cutoffFreq, | ||
| + | biquadcoeff = VSHighpass(cutoffFreq, | ||
| + | biquadcoeff = VSHighpass(cutoffFreq, | ||
| + | |||
| -- A quad filter consists of 5 coefficents: | -- A quad filter consists of 5 coefficents: | ||
| - | -- consisting of these 5 coefficients. The functions VSLowpass, VSBandpass, and VSHighpass will | + | -- consisting of these 5 coefficients |
| + | -- waveform filter. The next five coefficients are for the insert filter. The last value is | ||
| + | -- turning on or off the insert filter. You can manipulate these values, but it is complicated and | ||
| + | -- unnecessary since we have the helper functions above. | ||
| + | |||
| + | -- Waveform filtering (uses the first five biquad coefficents): | ||
| + | wave[x] = VSBiquad(wave[x], | ||
| + | |||
| + | -- Insert filter (uses the last five biquad coefficients: | ||
| + | afilter[x] = biquadcoeff | ||
| + | |||
| + | -- | ||
| + | -- Waveform filtering: | ||
| + | -- The functions | ||
| + | The functions VSLowpass, VSBandpass, and VSHighpass will | ||
| -- set the coefficients given the values. When setting these filter coefficients, | -- set the coefficients given the values. When setting these filter coefficients, | ||
| -- frequency and the note value are considered. This is because the cutoff frequency is typically | -- frequency and the note value are considered. This is because the cutoff frequency is typically | ||
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| wave[x] = VSBiquad(wave[x], | wave[x] = VSBiquad(wave[x], | ||
| - | -- MIDI output | + | -- MIDI output |
| -- VSTick is a help function to determine if the beat position crosses a certain time interval. | -- VSTick is a help function to determine if the beat position crosses a certain time interval. | ||
| -- For instance, if length=1/4 (corresponding to a quarter note), VSTick will return true every time | -- For instance, if length=1/4 (corresponding to a quarter note), VSTick will return true every time | ||