effects.txt (25370B)
1 Effects 2 ------- 3 4 Effects are, generally, black boxes that transform audio signals in a 5 specified way. More exactly, the only input data for an effect in ZynAddSubFX 6 is: 7 8 * an array of samples, which is read *on line* 9 * the current system time (used for LFOs) 10 11 The output is the transformed array of samples. 12 13 NOTE: As described, effects have no information about anything else. For 14 example, key presses are not recognized. Therefore, pressing a key does not 15 initiate the LFO. Phase knobs will always be relative to a *global* LFO, which is 16 only dependent on the system time. 17 18 ZynAddSubFX has 3 types of effects: 19 20 * System Effects 21 * Insertion Effects 22 * Instrument Effects 23 24 TODO: Describe these 3 types (their differences). 25 26 [[effects::general_topics, General Topics]] 27 General topics 28 ~~~~~~~~~~~~~~ 29 30 * *Wetness* determines the mix of the results of the effect and its input. This 31 mix is made the effects output. If an effect is wet, it means that nothing of 32 the input signal is bypassing the effect. If it is dry, then the effect has no 33 effect. TODO: Difference between Volume and D/W? 34 * *Pan* lets you apply panning, which means that the sound source can move to 35 the right or left. Set it to 0.0 to only hear output on the right side, or to 36 the maximum value to only hear output on the left side. 37 * *LRc.* or *L/R* let you apply crossover. 38 * *Filter stages* are the number of times that this filter is applied in series. 39 So, if this number is 1, you simply have this one filter. If it is two, the 40 sound first passes the filter, and the results then pass the same filter again. 41 In ZynAddSubFX, the wetness is applied after all stages were passed. 42 * *LFOs* are, as the name says, oscillators with, compared to the frequency of 43 the sound, low frequency. They often appear in order to control the effect. 44 They can have some of the following controls: 45 ** *LFO Type* determines the shape of the LFO. If not present, the LFO is a 46 sine wave. 47 ** *Freq* determines the LFO's frequency. 48 ** *Dpth* is a multiplier to the LFO. Thus, it determines the LFOs amplitude 49 and its influence. 50 ** *Rnd* is the LFO amplitude's randomness 51 ** *St.df* lets you determine how much left and right LFO are phase shifted. 52 64.0 means stereo, higher values increase the right LFO relatively to the left 53 one. 54 ******************************************************************** 55 Hint: Keep in mind that Effects that can be controlled by LFO can also be 56 controlled arbitrary: Set the LFO depth to zero and manipulate the phase knob 57 (e.g. with NRPNs or maybe via OSC in the future). 58 ******************************************************************** 59 60 Equalizer 61 ~~~~~~~~~ 62 63 Introduction 64 ^^^^^^^^^^^^ 65 66 An https://en.wikipedia.org/wiki/Filter_%28signal_processing%29[equalizer] is a 67 filter effect that applies different volume to different frequencies of the 68 input signal. This can, for example, be used to "filter out" unwanted 69 frequencies. 70 ZynAddSubFX's implementations follow the 71 https://www.musicdsp.org/en/latest/Filters/197-rbj-audio-eq-cookbook.html["Cookbook formulae for 72 audio EQ"] by Robert Bristow-Johnson. 73 74 Filter Types 75 ^^^^^^^^^^^^ 76 77 This topic is completely discussed in <<filters, the Filters section>>. 78 79 Usage 80 ^^^^^ 81 82 We describe all parts of the GUI here. The term passband (or often just "band") 83 refers to the amount of frequencies which are not significantly attenuated by 84 the filter. 85 86 * *Gain* (on the left) defines an offset which is added to the complete filter. 87 * *B.* lets you choose the passband number. Multiple passbands define one 88 filter. This is important if you want multiple filters to be called after each 89 other. Note that filters are commutative. 90 * *T.* lets you choose the current filter's type, as described above. 91 * *Freq* describes the frequencies where the filter has its poles. For some 92 filters, this is called the "cutoff" frequency. Note, however, that a bandpass 93 filter has two cutoff frequencies. 94 * *Gain* is only active for some filters and sets the amount of a special peak 95 these filters have. Note that for those filters, using the predefined gain makes 96 them effectless. 97 * *Resonance* lets you describe a peak at the given frequency for filters with 98 2 poles. This can be compared to real physical objects that have more gain at 99 their resonance frequency. 100 * *St.* lets you define multiple filter stages. This is equivalent to having 101 multiple copies of the same filter in sequence. 102 103 Chorus 104 ~~~~~~ 105 106 Introduction 107 ^^^^^^^^^^^^ 108 109 In a chorus, many people sing together. Even if each of them sings at exactly 110 the same frequency, all their voices usually sound different. We say they have a 111 different timbre. Timbre is the way we perceive sound and makes us differ 112 between different music instruments. This is, physically, achieved by varying 113 both the amplitude envelope and the frequency spectrum. Multiple sounds with 114 slightly different timbres make a sound more shimmering, or powerful. This is 115 called the chorus effect. 116 117 Function 118 ^^^^^^^^ 119 120 The chorus effect can be achieved by multiple people singing together. In 121 a concert, there are many instruments, resulting in the same effect. When making 122 electronic music, we only have an input wave and need to generate these 123 different timbres by ourselves. ZynAddSubFX therefore simply plays the sound, 124 pitch modulated by an LFO, and adds this to the original sound. This explains 125 the diagram below: The multiple pitches are generated by a delayed version of 126 the input. This version is being pitched by an LFO. More detailed, this pitch 127 is generated by varying the reading speed of the delayed sound; the variation 128 amount is controlled by an LFO. 129 130 image:./gen/chorus.png[width=700, 131 title="The chorus effect. z^(-n.m) describes the delay."] 132 133 TODO: Add LFO pointing to delay? 134 135 Related effects to Chorus are Flangers. Flangers can be described as Chorus 136 with very short LFO delay and little LFO depth. You can imagine a flanger as two 137 copies of a sound playing at almost the same time. This leads to interference, 138 which can be clearly heard. It is popular to apply flangers to guitars, giving 139 them more "character". 140 While in standard CHORUS and FLANGER mode there is only one additional voice that 141 can be blended with the original one using the amount knob, the DUAL and TRIPLE 142 mode adds two/three voices with their LFOs each being phase shifted by 180 / 120 degrees. 143 Those have been used by famous roland string synthesizers. 144 145 Usage 146 ^^^^^ 147 148 * First, crossover is applied. 149 * The following 5 knobs (*Freq*, *Rnd*, *LFO Type*, *St.df*, *Depth*) control 150 the LFO for the pitch. If the depth is set to zero, the pitch will not be 151 changed at all. 152 * *Delay* is the time that the delayed sound is delayed "on average". Note that 153 the delay also depends on the current pitch. 154 * After the correct element of the sound buffer is found using the LFO, the 155 *Fb* knob lets you set how loud it shall be played. This is mostly redundant to 156 the *D/W* knob, but we have not applied panning and subtraction yet. 157 * Next, the signal can be negated. If the *Subtract* checkbox is activated, 158 the amplitude is multiplied by -1. 159 * Finally, *Pan* lets you apply panning. 160 * *Mode* selector lets you choose between Chorus, Flanger, Dual- and Triple Chorus. 161 162 Distortion 163 ~~~~~~~~~~ 164 165 Introduction 166 ^^^^^^^^^^^^ 167 168 Distortion means, in general, altering a signal. Natural instruments 169 usually produce sine like waves. A wave is transformed in an unnatural way when 170 distortion is used. The most distorted waves are usually pulse waves. It is 171 typical for distortion to add overtones to a sound. Distortion often increases 172 the power and the https://en.wikipedia.org/wiki/Loudness[loudness] of a signal, 173 while the db level is not increased. This is an important topic in the 174 https://en.wikipedia.org/wiki/Loudness_war[Loudness War]. 175 176 NOTE: As distortion increases loudness, distorted music can cause ear damage 177 at lower volume levels. Thus, you might want to use it a bit careful. 178 179 Distortion can happen in many situations when working with audio. Often, this is 180 not wanted. In classical music, for example, distortion does not occur 181 naturally. However, distortion can also be a wanted effect. It is typical for 182 Rock guitars, but also present in electronic music, mostly in Dubstep and 183 Drum & Bass. 184 185 The basic components of distortion are mainly 186 187 * a preamplifier 188 * the waveshaping function 189 * filters 190 191 Preamplification changes the volume before the wave is shaped, and is indeed the 192 amount of distortion. For example, if you clip a signal, the louder the input 193 gets, the more distortion you will get. This can have different meanings for 194 different types of distortions, as described below. 195 196 ******************************************************************** 197 The filters are practical. A reason for using them afterwards is that distortion 198 can lead to waves with undesired high frequency parts. Those can be filtered out 199 using the LPF. A reason for using filters before applying is to achieve 200 multiband distortion. ZynAddSubFX has no "real" multiband distortion by now, 201 however. 202 ******************************************************************** 203 204 Types of Distortion 205 ^^^^^^^^^^^^^^^^^^^ 206 207 This topic is completely discussed in 208 <<adsynth::oscilllator::types_of_waveshaping, the Oscillator Section>>. Note 209 that you can use the 210 Oscillator editor in order to find out what your distortion effect does. Also 211 note that while the Oscillator editor's distortion is limited to some 212 oscillators you can produce in the Oscillator editor, the distortion effect can 213 be used on every wave that you can generate with ZynAddSubFX. 214 215 Function 216 ^^^^^^^^ 217 218 We explain the functionality in a diagram and list the components below. 219 220 image:./gen/distort.png[width=700, 221 title="The components of a distortion function."] 222 223 * Negation is the first thing to happen. If the *Neg* checkbox is activated, the 224 amplitude is multiplied by -1. 225 * Panning is applied. Note, however, that you have to activate the 226 Stereo Checkbox, labeled *St*, before. 227 * Preamplification is done next. The amount can be changed using the 228 *Drive* nob. Indeed, this is the amount of distortion. For example, if you clip 229 a signal, the louder the input gets, the more distortion you will get. This can 230 have different meanings for different types of distortion, as described above. 231 * *HPF* and *LPF* are filters with 2 poles. Whether they are used before or 232 after the waveshape, depends on the checkbox labeled *PF*. 233 * The next step is the wave shape. This defines how the wave is 234 actually modified. The *Type* combo box lets you define how. We will discuss some 235 types below. 236 * After the wave shape, we scale the level again. This is called 237 output amplification. You can change the value using the *Level* knob. 238 * Crossover is the last step. This is controlled by the knob *LR Mix* and 239 means that afterwards, a percentage of the left side is applied to the right 240 side, and, synchronously, the other way round. It is a kind of interpolation 241 between left and right. If you set the LR Mix to 0.0, you will always have a 242 stereo output. 243 244 Dynamic Filter 245 ~~~~~~~~~~~~~ 246 247 Introduction 248 ^^^^^^^^^^^^ 249 250 A dynamic filter is, as the name says, a filter which changes its parameters 251 dynamically, dependent on the input and current time. In ZynAddSubFX, 252 frequency is the only variable parameter. It can be used as an "envelope 253 following filter" (sometimes referenced "Auto Wah" or simply "envelope filter"). 254 255 Function 256 ^^^^^^^^ 257 258 Though this filter might look a bit complicated, it is actually easy. We divide 259 the parameters into two classes: 260 261 * *Filter Parameters* are the ones you get when you click on *Filter*. They 262 give the filter its basic settings. 263 * *Effect Parameters* are the other ones that control how the filter changes. 264 265 The filter basically works like this: The input signal is passed through a 266 filter which dynamically changes its frequency. The frequency is an additive of: 267 268 * the filter's base frequency 269 * an LFO from the effect parameters 270 * the "amplitude" of the input wave 271 272 image:./gen/dynamic.png[width=700, 273 title="The components of a dynamical filter"] 274 275 The amplitude of the input wave is not the current amplitude, but the so called 276 https://en.wikipedia.org/wiki/Root_mean_square["Root Mean Square (RMS)"] value. 277 This means that we build a mean on the current amplitude and the past values. 278 How much the new amplitude takes influence is determined by the *Amplitude 279 Smoothness* (see below). 280 281 ******************************************************************** 282 RMS value plays an important role in the term loudness. A fully distorted 283 signal can sound 20 db louder due to its higher RMS value. This filter takes 284 this into account, depending on the smoothness. 285 ******************************************************************** 286 287 Usage 288 ^^^^^ 289 290 * The 4 knobs in the middle (*Freq*, *Rnd*, *LFO Type*, *St.df*) control the 291 LFO. 292 * Two knobs let you control the way how the RMS value of the amplitudes is 293 measured: 294 ** *A.M* sets the Amplitude Smoothness (this is described above). The higher 295 you set this value, the more slow will the filter react. 296 ** *A.Inv.*, if being set, negates the (absolute) RMS value. This will lower 297 the filter frequency instead of increasing it. Note that this will not have 298 much effect if the effects input is not very loud. 299 * The following controls define the mix of the LFO and the amplitude. 300 ** *A.S* sets the Amplitude Sensing (i.e. how much influence the amplitude 301 shall have). 302 ** *LfoD* sets the LFO depth. 303 * The filter button lets you choose the filter type. 304 * After the input signal has passed through the filter, *Pan* can apply 305 panning. 306 307 Echo 308 ~~~~ 309 310 Introduction 311 ^^^^^^^^^^^^ 312 313 The echo effect, also known as 314 https://en.wikipedia.org/wiki/Delay_%28audio_effect%29[delay effect], simulates 315 the natural reflection of a sound. The listener can hear the sound multiple 316 times, usually decreasing in volume. Echos can be useful to fill empty parts of 317 your songs with. 318 319 Function 320 ^^^^^^^^ 321 322 In ZynAddSubFX, the echo is basically implemented as the addition of the 323 current sound and a delayed version of it. The delay is implemented as in the 324 picture below. First, we add the delayed signal to the effect input. Then, 325 they pass an LP1. This shall simulate the effect of dampening, which means that 326 low and especially high frequencies get lost earlier over distance than middle 327 frequencies do. Next, the sound is delayed, and then it will be output and added 328 to the input. 329 330 image:./gen/echo.png[width="700", 331 title="The echo includes a fb line, labeled as z^-n, and a delay."] 332 333 ******************************************************************** 334 The exact formula in the source code for the dampening effect is as follows: 335 336 latexmath:[$Y(t) := (1-d) \cdot X(t) + d \cdot Y(t-1)$], 337 338 where latexmath:[t] be the time index for the input 339 buffer, latexmath:[d] be the dampening amount and latexmath:[X,Y] be the input, 340 respective the output of the dampening. This solves to 341 342 latexmath:[$Y(z) = Z(Y(t)) = (1-d) \cdot X(z) + d \cdot Y(z) \cdot z^{-1}$] 343 344 latexmath:[$\Leftrightarrow H(z) := \frac{Y(z)}{X(z)} = \frac{1-d}{1 - 345 d \cdot z^{-1}}$] 346 347 which is used in latexmath:[$Y(z) = H(z) \cdot X(z)$]. So latexmath:[$H(z)$] is 348 indeed a filter, and by looking at it, we see that it is an LP1. Note that 349 infinite looping for d=1 is impossible. 350 ******************************************************************** 351 352 Description 353 ^^^^^^^^^^^ 354 355 * *Pan* lets you apply panning of the input. 356 * *Delay* sets the time for one delay. 357 * *LRdl.* means Left-Right-Delay. If it is set to the middle, then both sides 358 are delayed equally. If not, then the left echo comes earlier and the right 359 echo comes (the same amount) later than the average echo; or the other way 360 round. Set the knob to 0 to hear on the right first. 361 * *LRc.* applies crossover. 362 * Feedback describes how much of the delay is added back to the input. Set 363 *Fb.* to the maximum to hear an infinite echo, or to the minimum to just 364 hear a single repeat. 365 * The *Damp* value lets the LP1 reject higher frequencies earlier if 366 increased. 367 368 Reverb 369 ~~~~~~ 370 371 Introduction 372 ^^^^^^^^^^^^ 373 374 A https://en.wikipedia.org/wiki/Reverberation[Reverberation] actually expresses 375 the effect of many echoes being played at the same time. This can happen in an 376 enclosed room, where the sound can be reflected in different angles. Also, in 377 nature, thunders approximate reverbs, because the sound is reflected in many 378 different ways, arriving at the listener at different times. 379 380 In music, reverbs are popular in many ways. Reverbs with large room size can be 381 used to emulate sounds like in live concerts. This is useful for voices, pads, 382 and hand claps. A small room size can simulate the sound board of string 383 instruments, like guitars or pianos. 384 385 Function 386 ^^^^^^^^ 387 388 As mentioned, a reverb consists of permanent echo. The reverb in ZynAddSubFX is 389 more complex than the echo. After the delaying, comb filters and then allpass 390 filters are being applied. These make the resulting sound more realistic. The 391 parameters for these filters depend on the room size. For details, consider the 392 information about https://ccrma.stanford.edu/~jos/pasp/Freeverb.html[Freeverb]. 393 394 image:./gen/reverb.png[width=700, 395 "The reverb, being similar to the echo."] 396 397 Description 398 ^^^^^^^^^^^ 399 400 * The *Type* combo box lets you select a reverb type: 401 ** *Freeverb* is a preset. It was proposed by Jezar at Dreampoint. 402 ** *Bandwidth* has the same parameters for the comb and allpass filters, but it 403 applies a unison before the LP/HP. The unison's bandwidth can be set using *bw*. 404 ** Random chooses a random layout for comb and allpass each time the type or 405 the room size is being changed. 406 * The room size (*R.S.*) defines parameters only for the comb and allpass 407 filters. 408 * *Time* controls how long the whole reverb shall take, including how slow the 409 volume is decreased. 410 * The initial delay (*I.del*) is the time which the sounds need at least to 411 return to the user. The initial delay feedback (*I.delfb*) says how much of the 412 delayed sound is added to the input. 413 * Low pass filter (*LPF*) and high pass filter (*HPF*) can be applied before 414 the comb filters. 415 * The dampening control (*Damp*) currently only allows to damp low frequencies. 416 Its parameters are being used by the comb and allpass filters. 417 * *Pan* lets you apply panning. This is the last thing to happen. 418 419 420 Phaser 421 ~~~~~~ 422 423 Introduction 424 ^^^^^^^^^^^^ 425 426 The https://en.wikipedia.org/wiki/Phaser_%28effect%29[Phaser] is a special 427 dynamic filter. The result is a sweeping 428 sound, which is 429 often used on instruments with a large frequency band, like guitars or strings. 430 This makes it typical for genres like rock or funk, where it is often modulated 431 with a pedal, but also for giving strings a warm, relaxing character. 432 433 Function 434 ^^^^^^^^ 435 436 The audio signal is split into two paths. One path remains unchanged. The other 437 one is sent to a delay line. The delay time (the so called *phase*) is made 438 dependent on the frequency. Therefore, an all-pass filter is applied to the 439 signal, which *preserves* the amplitude, but determines the delay time. In the 440 end, both paths are added. 441 442 The following picture describes how this works on white noise. Light blue 443 signalizes that the frequency is not present at the current time, and dark blue 444 signalizes the opposite. The dark blue peaks appear if the delay time is very 445 short, because then, the second path almost equals the first one, which results 446 in duplication of the signal. If the delay line is very long, then it is --- in 447 the case of white noise --- totally at random whether the delayed signal 448 currently duplicates the unchanged path, or whether it cancels it out to zero. 449 This random effect results in white noise between the clear blue structures. 450 451 image:./images/phaser-spectrogram.jpg[width="700", 452 title="Spectrogram of an 8-stage phaser 453 modulated by a sine LFO applied to white noise."] 454 455 Phaser Types 456 ^^^^^^^^^^^^ 457 458 ZynAddSubFX offers different types of phasers: 459 460 * Analog and "normal" phasers. Analog phasers are more complicated. They sound 461 punchier, while normal phasers sound more fluently. However, analog filters 462 usually need more filter stages to reach a characteristic sound. 463 * Sine and triangle filters. Note that an analog triangle filter with many poles 464 is a barber pole filter and can be used to generate 465 https://en.wikipedia.org/wiki/Shepard_tone[Shepard Tones], 466 i.e. tones that seem to increase or decrease with time, but do not really. 467 * The LFO function can be squared. This converts the triangle wave into a hyper 468 sine wave. The sine squared is simply a faster sine wave. 469 * TODO: Barber is deactivated, since PLFOtype is only 0 or 1? 470 471 Description 472 ^^^^^^^^^^^ 473 474 For the normal phaser, first, the LFO is generated: 475 476 * There are 4 controls (*Freq*,*Rnd*,*LFO type*,*St.df*) that define the 477 LFO. 478 * *Phase* and *Depth* are applied afterwards in the usual way (TODO: I don't 479 understand the code here for the normal phase...). For the analog phaser, 480 *Phase* is not implemented, yet. 481 ** If *hyp* is being set, then the LFO function is being squared. 482 483 Next, the input is being used. 484 485 * *Analog* decides whether the phaser is analog or "normal". 486 * First, *Pan* applies panning to the original input in every loop. 487 * Next, barber pole phasing is being applied (Analog only). 488 * *Fb* applies feedback. The last sound buffer element is (after 489 phasing) multiplied by this value and then added to the current one. For normal 490 filter, the value is added before, for analog after the first phasing stage. 491 * Now, *Stages* phasing stages are being applied. *dist* sets the distortion 492 for when applying the phasing stages. This has only effect for analog phasers. 493 * The feedback is taken now. 494 * In the end, *Subtract* inverts the signal, multiplying it by -1. 495 496 Alienwah 497 ~~~~~~~~ 498 499 Introduction 500 ^^^^^^^^^^^^ 501 502 The AlienWah effect is a special, dynamic 503 https://en.wikipedia.org/wiki/Formant[formant] filter (TODO: is this true?). 504 Paul Nasca named it AlienWah because it sounded "a bit like wahwah, but more 505 strange". The result of the filter is a sound varying between the 506 vocals "Ahhhhh" (or "Uhhhhh") and "Eeeeee". 507 508 Function 509 ^^^^^^^^ 510 511 The way that the filter moves between the two vocals is mainly 512 described by an LFO. A bit simplified, Paul Nasca has stated the formula (for 513 latexmath:[$i^2=-1; R<1$]) as 514 515 latexmath:[$fb=R*(\cos(\alpha)+i*\sin(\alpha))$] 516 517 latexmath:[$y_n=y_{n-delay}*R*(\cos(\alpha)+i*\sin(\alpha))+x_n*(1-R)$]. 518 519 The input latexmath:[$x_n$] has the real part of the samples from the wave file 520 and the imaginary part is zero. The output of this effect is the real part of 521 latexmath:[$y_n$]. latexmath:[$\alpha$] is the phase. 522 523 Description 524 ^^^^^^^^^^^ 525 526 * *Pan* 527 * The following 5 controls (*Freq*,*Rnd*,*LFO type*,*St.df*, *Dpth*) define the 528 LFO. 529 ** *Fb* 530 531 ** *Delay* If this value is low, the sound is turned more into a 532 "wah-wah"-effect. 533 ** *Phase* See latexmath:[$\alpha$] in the above formula. This lets you set 534 where the vocal is between "Ahhhhh" and "Eeeeee". 535 ** *L/R* applies crossover in the end of every stage. This is currently not 536 implemented for the Analog Phaser. 537 538 Sympathetic 539 ~~~~~~~~ 540 541 Introduction 542 ^^^^^^^^^^^^ 543 544 The Sympathetic aims to 'simulate' the sympathetic resonance in a piano. 545 That is the effect, that all other strings on a piano are also oscillating 546 when one string is played. With high Q setting it's more like playing with sustain pedal. 547 548 TIP: map the sustain pedal to the Q parameter 549 550 Function 551 ^^^^^^^^ 552 553 It is made off a bank of combfilters that are by default tuned like the 554 strings of a piano. 555 556 Description 557 ^^^^^^^^^^^ 558 559 ** *Drive* how much input signal is fed to the strings 560 ** *Level* Output Volume 561 ** *Str* Number of strings to use. 12 is a full Octave. On small systems 562 be careful with cpu load 563 ** *Q* How strong is the reflection 564 ** *Uni* Unisono - how many strings should have the 'same' frequency. It's used with the 565 next paremeter. 566 ** *Detune* Amount of detune between the unisono strings 567 ** *Base* midi note of the lowest string 568 569 Reverse 570 ~~~~~~~~ 571 572 Introduction 573 ^^^^^^^^^^^^ 574 575 The reversed effect plays the audio signal backwards. To prevent a time paradoxon 576 the input signal is chopped in segments that are reversed one by one. 577 One classic usage is the reversing of a decaying sound for example from reverb. 578 579 Function 580 ^^^^^^^^ 581 With a melotron it was possible to reverse each key by turning the magentic tape around. 582 While exactly this is not possible with this effect, because there is no prerecorded 583 sound when hitting a key, this effect can create sounds that resemble those of the melotron. 584 585 586 Description 587 ^^^^^^^^^^^ 588 589 ** *Length* Length of the segments being played backwards. 590 ** *Numerator&Denominator* Alternatively the length can be set relative 591 to the global Tempo (bpm) as note value. For example 1/4 at 120bpm is 0.5s 592 ** *Phase* adjust the time when the reversed segments start relative to a 593 global time reference. It's useful only in combination with another 594 timed element like an LFO or delay. 595 ** *Sync* choose how the recording and reverse playing phase is synchronized 596 * *Auto* the effect switches automatically between recording and reverse playing and vice versa after a segment (length) 597 * *MIDI* like auto but speed is taken from MIDI timecode 598 * *Host* like auto but speed is taken from plugin host (when running as plugin mode) 599 * *NoteOn* recording starts at noteOn. Reverse playing starts after a segment (length) 600 * *NoteOnOff* recording starts at noteon. Reverse playing starts when the decaying note reaches silence