Where is operator in ableton live 9

Operator is a versatile, easy-to-use and great-sounding software instrument combining classic analog sounds and frequency modulation synthesis, perfectly integrated in Live's award-winning interface. Operator is a powerful instrument that brings an eclectic spectrum of sonic possibilities and musical inspiration to Ableton Live users. True to the Ableton philosophy, Operator invites creativity through a fusion of depth and usability, allowing even the most complex sounds to be created quickly.

Whether you long for evolving synthetic textures, rich expressive leads, gritty percussion, rhythmic atmospheres or anything in between, Operator has the cure. Operator delivers a wide range of sonic possibilities through an extensive array of synthesis tools. Powerful analog-style subtractive synthesis and detailed FM effects are all possible due to Operator's hybrid approach to synthesis.

Operator's dynamic collection of musical presets was created by an expert team of sound designers to be versatile, inspiring and immediately playable. Inharmonics can also be modulated by velocity via the slider below the knob. Opening, which is only available for the Pipe resonator, scales between an open and closed pipe.

This parameter can also be modulated by velocity via the slider below the knob. The Listening L and R controls adjust the location on the left and right resonator where the vibrations are measured. Higher values move the listening point closer to the edge. These parameters are not used with the Pipe or Tube resonators, which are always measured in the middle of their permanently open end.

The Hit knob adjusts the location on the resonator at which the object is struck or otherwise activated. Higher values move the activation point closer to the edge. The Hit position can also be randomized by increasing the value of the Rd. Random slider below the knob. Each resonator has its own Volume and Pan controls. Pan can also be modulated by note pitch via the K Key slider below the knob.

The Bleed control mixes a portion of the original oscillator signal with the resonated signal. At higher values, more of the original signal is applied. This is useful for restoring high frequencies, which can often be damped when the tuning or quality are set to low values. Additionally, they can modulate each other. The choices are sine, square, triangle, sawtooth up, sawtooth down and two types of noise.

Depth sets the overall intensity of the LFO, while Rate adjusts its speed. The sliders below these parameters allow for additional modulations; Depth can be modulated by velocity while Rate can be modulated by note pitch. With Retrig.

The Offset slider adjusts the phase. Each LFO can modulate two targets, which are set via the Destination choosers. The intensity of the modulations is adjusted with the Amount sliders.

Note that pitch bend is hardwired to pitch modulation, but can still be routed to an additional target. The global section contains the parameters that relate to the overall behavior and performance of Collision.

Collision contains a built-in limiter that automatically activates when the audio level is too high. When in series, Mallet and Noise output to Resonator 1. Note that Resonator 1 must be turned on when using serial mode:.

In parallel mode, the output of Mallet and Noise is mixed and then sent directly to both resonators, which then output to their own mixers. The Voices chooser sets the available polyphony. This can be useful for keeping CPU down when working with long decay times. Although Collision has been designed to model the behavior of objects that exist in the physical world, it is important to remember that these models allow for much more flexibility than their physical counterparts.

To program realistic instrument simulations, it helps to think about the chain of events that produces a sound on a mallet instrument a marimba, for example , and then visualize those events as sections within Collision:. Consequently, it is very easy to program resonances that are much more sensitive to input than any physical resonator could be. Certain combinations of parameters can cause dramatic changes in volume.

Make sure to keep output levels low when experimenting with new sounds. Electric is a software electric piano based on the classic instruments of the seventies, and developed in collaboration with Applied Acoustics Systems. Each component of these instruments has been modeled using cutting edge physical modeling technology to provide realistic and lively sounds. Physical modeling uses the laws of physics to reproduce the behavior of an object.

In other words, Electric solves, in real time, mathematical equations describing how its different components function. No sampling or wavetables are used in Electric; the sound is simply calculated in real time by the CPU according to the values of each parameter. Electric is more than a simple recreation of vintage instruments; its parameters can be tweaked to values not possible with the real instruments to get some truly amazing new sounds that still retain a warm acoustic quality.

The mechanism of the electric piano is actually quite simple. A note played on the keyboard activates a mallet that hits a fork. The sound of that fork is then amplified by a magnetic coil pickup and sent to the output, very much like an electric guitar. The fork is made of two parts, called the tine bar and tone bar.

The tine bar is where the mallet hits the fork while the tone bar is a tuned metal resonator, sized appropriately to produce the correct pitch. Once the fork is activated, it will continue to resonate on its own for a long time.

But releasing the key applies a damper to the fork, which mutes it more quickly. The Electric interface is divided into five main sections, some of which are further divided into related subsections. The first four main sections Mallet, Fork, Damper and Pickup correspond to the sound producing components mentioned above. The Global section contains parameters that affect overall behavior and performance, such as pitch bend and polyphony.

Higher values simulate a harder surface, which results in a brighter sound. Lower values mean a softer surface and a more mellow sound. Low values simulate a soft impact while high values mean a hard impact.

The stiffness and force can also be modified by velocity and note pitch, via the Vel and Key sliders found below the knobs. The Noise subsection simulates the impact noise caused by the mallet striking the fork. The Decay knob adjusts how long it takes for this noise to fade to silence, while the Pitch control sets the center frequency. Level adjusts the overall volume of the noise component. An additional Key scaling control adjusts how much the noise volume is determined by note pitch.

The Fork section is further divided into Tine and Tone subsections. The Tine subsection controls the portion of the fork that is directly struck by the mallet. Low values increase the amount of low harmonics, while higher values result in higher harmonics. The amplitude of the tine is adjusted with the Level knob.

This level can be further modulated by note pitch via the Key scaling control. The Tone subsection controls the secondary resonance of the fork. Decay and Level parameters here work in the same way as their Tine counterparts. The metal forks in an electric piano are designed to sustain for a long time when a key is held.

The mechanism that regulates this sustain is called the damper. When the key is released, the damper is applied to the fork again to stop it from vibrating. But the dampers themselves make a small amount of sound, both when they are applied and when they are released.

The Tone knob adjusts the stiffness of the dampers. Turning this control to the left simulates soft dampers, which produces a mellower sound. Turning it to the right increases the hardness of the dampers, producing a brighter sound. The overall amount of damper noise is adjusted with the Level control.

When turned to the left, damper noise is only present during the attack phase of the note. When turned to the right, the noise is present only during the release phase. In the center, an equal amount of noise will be added during both the attack and release.

The Pickup section simulates the behavior of the magnetic coil pickup that amplifies the sound of the resonating fork. The R-W buttons switch between two different types of pickups. In the R position, Electric simulates electro-dynamic pickups, while W is based on an electro-static model. The Output knob controls the amount of signal output by the pickup section. Different combinations of these two knobs can yield very different results. For example, a low amount of input with a high amount of output will produce a cleaner sound than a high input with a low output.

The output level can be further modulated by note pitch via the Key scaling control. The Symmetry and Distance knobs adjust the physical location of the pickup in relation to the tine. Symmetry simulates the vertical position of the pickup. In the center position, the pickup is directly in front of the tine, which results in a brighter sound. Turning the knob to the left or right moves the pickup below or above the tine, respectively. Distance controls how far the pickup is from the tine.

Turning the knob to the right increases the distance, while turning it to the left moves the pickup closer. Note that the sound becomes more overdriven as the pickup approaches the tine. The Global section contains the parameters that relate to the overall behavior and performance of Electric.

The Semi and Detune controls function as coarse and fine tuners. Semi transposes the entire instrument up or down in semitone increments, while the Detune slider adjusts in increments of one cent up to a maximum of 50 cents up or down. Stretch simulates a technique known as stretch tuning, which is a common modification made to both electric and acoustic pianos and is an intrinsic part of their characteristic sound. Stretch tuning attempts to correct this by sharpening the pitch of upper notes while flattening the pitch of lower ones.

The External Instrument device is not an instrument itself, but rather a routing utility that allows you to easily integrate external hardware synthesizers and multitimbral plug-ins into your projects. It sends MIDI out and returns audio.

The top chooser selects either a physical MIDI port see If another track in your set contains a multitimbral plug-in, you can select this track in the top chooser. In this case, the second chooser allows you to select a specific MIDI channel in the plug-in. The Audio From chooser provides options for returning the audio from the hardware synth or plug-in device. Note that the main outputs will be heard on the track that contains the instrument.

The Gain knob adjusts the audio level coming back from the sound source. This level should be set carefully to avoid clipping. Since external devices can introduce latency that Live cannot automatically detect, you can manually compensate for any delays by adjusting the Hardware Latency slider.

The button next to this slider allows you to set your latency compensation amount in either milliseconds or samples. If your external device connects to Live via a digital connection, you will want to adjust your latency settings in samples, which ensures that the number of samples you specify will be retained even when changing the sample rate.

If your external device connects to Live via an analog connection, you will want to adjust your latency settings in milliseconds, which ensures that the amount of time you specify will be retained when changing the sample rate.

In this case, be sure to switch back to milliseconds before changing your sample rate. Any latency introduced by devices within Live will be compensated for automatically, so the slider will be disabled when using the External Instrument Device to route internally.

Note: If the Delay Compensation option see Impulse is a drum sampler with complex modulation capabilities. Alternatively, each sample slot features a Hot-Swap button for hot-swapping samples see 5. Imported samples are automatically mapped onto your MIDI keyboard, providing that it is plugged in and acknowledged by Live. C3 on the keyboard will trigger the leftmost sample, and the other samples will follow suit in the octave from C3 to C4.

Mapping can be transposed from the default by applying a Pitch device see Each of the eight samples has a proprietary set of parameters, located in the area below the sample slots and visible when the sample is clicked.

Adjustments to sample settings are only captured once you hit a new note — they do not affect currently playing notes. Note that this behavior also defines how Impulse reacts to parameter changes from clip envelopes or automation, which are applied once a new note starts.

If you want to achieve continuous changes as a note plays, you may want to use the Simpler see This was designed with a specific situation in mind but can, of course, be used for other purposes : Replicating the way that closed hi-hats will silence open hi-hats. Each slot can be played, soloed, muted or hot-swapped using controls that appear when the mouse hovers over it. The Start control defines where Impulse begins playing a sample, and can be set up to ms later than the actual sample beginning.

The Stretch control has values from to percent. Negative values will shorten the sample, and positive values will stretch it. Two different stretching algorithms are available: Mode A is ideal for low sounds, such as toms or bass, while Mode B is better for high sounds, such as cymbals.

The Filter section offers a broad range of filter types, each of which can impart different sonic characteristics onto the sample by removing certain frequencies. The Frequency control defines where in the harmonic spectrum the filter is applied; the Resonance control boosts frequencies near that point. The Saturator gives the sample a fatter, rounder, more analog sound, and can be switched on and off as desired. The Drive control boosts the signal and adds distortion. Extreme Drive settings on low-pitched sounds will produce the typical, overdriven analog synth drum sounds.

The envelope can be adjusted using the Decay control, which can be set to a maximum of Impulse has two decay modes: Trigger Mode allows the sample to decay with the note; Gate Mode forces the envelope to wait for a note off message before beginning the decay.

This mode is useful in situations where you need variable decay lengths, as is the case with hi-hat cymbal sounds. Each sample has Volume and Pan controls that adjust amplitude and stereo positioning, respectively. Both controls can be modulated: Pan by velocity and a random value, and Volume by velocity only. Volume adjusts the overall level of the instrument, and Transp adjusts the transposition of all samples. The Time control governs the time-stretching and decay of all samples, allowing you to morph between short and stretched drum sounds.

When a new instance of Impulse is dragged into a track, its signal will be mixed with those of the other instruments and effects feeding the audio chain of the track. It can oftentimes make more sense to isolate the instrument or one of its individual drum samples, and send this signal to a separate track. Please see the Routing chapter see Operator includes a filter section, an LFO and global controls, as well as individual envelopes for the oscillators, filter, LFO and pitch.

The interface of Operator consists of two parts: the display surrounded on either side by the shell. The shell offers the most important parameters in a single view and is divided into eight sections.

On the left side, you will find four oscillator sections, and on the right side from top to bottom, the LFO, the filter section, the pitch section and the global parameters. If you change one of the shell parameters, the display in the center will automatically show the details of the relevant section. Operator can be folded with the triangular button at its upper left. This is convenient if you do not need to access the display details.

Operator offers eleven predefined algorithms that determine how the oscillators are connected. An algorithm is chosen by clicking on one of the structure icons in the global display, which will appear if the bottom right global section of the shell is selected.

Signals will flow from top to bottom between the oscillators shown in an algorithm icon. The algorithm selector can be mapped to a MIDI controller, automated, or modulated in real time, just like any other parameter. Typically, FM synthesis makes use of pure sine waves, creating more complex waveforms via modulation. However, in order to simplify sound design and to create a wider range of possible sounds, we designed Operator to produce a variety of other waveforms, including two types of noise.

You can also draw your own waveforms via a partial editor. The instrument is made complete with an LFO, a pitch envelope and a filter section. Operator will keep you busy if you want to dive deep into sound design! The oscillators come with a built-in collection of basic waveform types — sine, sawtooth, square, triangle and noise — which are selected from the Wave chooser in the individual oscillator displays. The first of these waveforms is a pure, mathematical sine wave, which is usually the first choice for many FM timbres.

The square, triangle and sawtooth waveforms are resynthesized approximations of the ideal shape. The numbers included in the displayed name e. Lower numbers sound mellower and are less likely to create aliasing when used on high pitches.

There are also two built-in noise waveforms. You can also select one of the built-in waveforms and then edit it in the same way. The small display next to the Wave chooser gives a realtime overview of your waveform. When your mouse is over the Oscillator display area, the cursor will change to a pencil. Drawing in the display area then raises or lowers the amplitudes of the harmonics. Holding Shift and dragging will constrain horizontal mouse movement, allowing you to adjust the amplitude of only one harmonic at a time.

You can switch between editing the first 16, 32 or 64 harmonics via the switches to the right of the display. Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser. Low Repeat values result in a brighter sound, while higher values result in more high-end roll-off and a more prominent fundamental. With Repeat off, partials above the 16th, 32nd or 64th harmonic are truncated. The context menu also offers an option to toggle Normalize on or off.

When disabled, additional harmonics add additional level. Note that the volume can become extremely loud if Normalize is off.

You can export your waveform in. Ams files can also be loaded into Simpler or Sampler. The frequency of an oscillator can be adjusted in the shell with its Coarse and Fine controls. This can be done for each individual oscillator by activating the Fixed option. This allows the creation of sounds in which only the timbre will vary when different notes are played, but the tuning will stay the same. Fixed Mode would be useful, for example, in creating live drum sounds. Fixed Mode also allows producing very low frequencies down to 0.

Note that when Fixed Mode is active, the frequency of the oscillator is controlled in the shell with the Frequency Freq and Multiplier Multi controls. This feature can be very useful when working with sequenced sounds in which the velocity of each note can be adjusted carefully.

Part of this functionality is the adjacent Q Quantize button. If this control is activated, the frequency will only move in whole numbers, just as if the Coarse control were being manually adjusted. If quantize is not activated, the frequency will be shifted in an unquantized manner, leading to detuned or inharmonic sounds which very well could be exactly what you want The amplitude of an oscillator depends on the Level setting of the oscillator in the shell and on its envelope, which is shown and edited when the Envelope display is visible.

The phase of each oscillator can be adjusted using the Phase control in its display. With the R Retrigger button enabled, the waveform restarts at the same position in its phase each time a note is triggered. With R disabled, the oscillator is free-running.

When an oscillator is modulating another oscillator, two main properties define the result: the amplitude of the modulating oscillator and the frequency ratio between both oscillators.

Any oscillator that is not modulated by another oscillator can modulate itself, via the Feedback parameter in its display.

Aliasing distortion is a common side effect of all digital synthesis and is the result of the finite sample rate and precision of digital systems. It mostly occurs at high frequencies. FM synthesis is especially likely to produce this kind of effect, since one can easily create sounds with lots of high harmonics. Aliasing is a two-fold beast: A bit of it can be exactly what is needed to create a cool sound, yet a bit too much can make the timbre unplayable, as the perception of pitch is lost when high notes suddenly fold back into arbitrary pitches.

Operator minimizes aliasing by working in a high-quality Antialias mode. This is on by default for new patches, but can be turned off in the global section. The Tone parameter in the global section also allows for controlling aliasing. Its effect is sometimes similar to a lowpass filter, but this depends on the nature of the sound itself and cannot generally be predicted. If you want to familiarize yourself with the sound of aliasing, turn Tone up fully and play a few very high notes.

You will most likely notice that some notes sound completely different from other notes. Now, turn Tone down and the effect will be reduced, but the sound will be less bright. The LFO in Operator can practically be thought of as a fifth oscillator. It runs at audio rates, and it modulates the frequency of the other oscillators.

It is possible to switch LFO modulation on or off for each individual oscillator and the filter using the Dest. A slider. The LFO can also be turned off entirely if it is unused. The Dest. B chooser allows the LFO to modulate an additional parameter. The intensity of this modulation is determined by the Dest.

B slider. Sample and hold uses random numbers chosen at the rate of the LFO, creating the random steps useful for typical retro-futuristic sci-fi sounds. The noise waveform is simply bandpass-filtered noise. Tip: FM synthesis can be used to create fantastic percussion sounds, and using the LFO with the noise waveform is the key to great hi-hats and snares. The frequency of the LFO can follow note pitch, be fixed or be set to something in between. With the R Retrigger button enabled, the LFO restarts at the same position in its phase each time a note is triggered.

With R disabled, the LFO is free-running. This parameter scales both the Dest. Operator has seven envelopes: one for each oscillator, a filter envelope, a pitch envelope and an envelope for the LFO. All envelopes feature some special looping modes.

Additionally, the filter and pitch envelopes have adjustable slopes. A rate is the time it takes to go from one level to the next.

As mentioned above, the filter and pitch envelopes also have adjustable slopes. Clicking on the diamonds between the breakpoints allows you to adjust the slope of the envelope segments. Positive slope values cause the envelope to move quickly at the beginning, then slower. Negative slope values cause the envelope to remain flat for longer, then move faster at the end.

A slope of zero is linear; the envelope will move at the same rate throughout the segment. With FM synthesis, it is possible to create spectacular, endless, permuting sounds; the key to doing this is looping envelopes. Loop Mode can be activated in the lower left corner of the display.

If an envelope in Operator is in Loop Mode and reaches sustain level while the note is still being held, it will be retriggered. The rate for this movement is defined by the Loop Time parameter. Note that envelopes in Loop Mode can loop very quickly and can therefore be used to achieve effects that one would not normally expect from an envelope generator.

While Loop Mode is good for textures and experimental sounds, Operator also includes Beat and Sync Modes, which provide a simple way of creating rhythmical sounds. If set to Beat Mode, an envelope will restart after the beat time selected from the Repeat chooser. In Beat Mode, the repeat time is defined in fractions of song time, but notes are not quantized. If you play a note a bit out of sync, it will repeat perfectly but stay out of sync.

In Sync Mode however, the first repetition is quantized to the nearest 16th note and, as a result, all following repetitions are synced to the song tempo. Note that Sync Mode only works if the song is playing, and otherwise it will behave like Beat Mode.

Note: To avoid the audible clicks caused by restarting from its initial level, a looped envelope will restart from its actual level and move with the set attack rate to peak level.

There is also a mode called Trigger that is ideal for working with percussive sounds. In this mode, note off is ignored. This means that the length of time a key is held has no effect on the length of the sound. The rates of all the envelopes in Operator can be scaled in unison by the Time control in the global section of the shell. Note that beat-time values in Beat and Sync Modes are not influenced by the global Time parameter.

These modulations in conjunction with the loop feature can be used to create very, very complex things A slider and the envelope can be turned off altogether via the switch in the pitch section of the shell. Like the LFO, the pitch envelope can modulate an additional parameter as chosen by the Dest. B chooser. The intensity of this modulation is determined by the Amt. B slider and the main Pitch Env value. The pitch and filter envelopes each have an additional parameter called End, which determines the level the envelope will move to after the key is released.

The rate of this envelope segment is determined by the release time. And, since the envelope of the LFO itself can loop, it can serve as a third LFO modulating the intensity of the first! And, since the oscillators also provide you with the classic waveforms of analog synthesizers, you can very easily build a subtractive synthesizer with them. Operator offers a variety of filter types including lowpass, highpass, bandpass, notch, and a special Morph filter.

Each filter can be switched between 12 and 24 dB slopes as well as a selection of analog-modeled circuit behaviors developed in conjunction with Cytomic that emulate hardware filters found on some classic analog synthesizers.

This is available for all of the filter types. The OSR circuit option is a state-variable type with resonance limited by a unique hard-clipping diode. This is modeled on the filters used in a somewhat rare British monosynth, and is available for all filter types. The MS2 circuit option uses a Sallen-Key design and soft clipping to limit resonance.

It is modeled on the filters used in a famous semi-modular Japanese monosynth and is available for the lowpass and highpass filters. The SMP circuit is a custom design not based on any particular hardware. It shares characteristics of both the MS2 and PRD circuits and is available for the lowpass and highpass filters. The PRD circuit uses a ladder design and has no explicit resonance limiting.

It is modeled on the filters used in a legacy dual-oscillator monosynth from the United States and is available for the lowpass and highpass filters. The most important filter parameters are the typical synth controls Frequency and Resonance.

Frequency determines where in the harmonic spectrum the filter is applied; Resonance boosts frequencies near that point. When using the lowpass, highpass, or bandpass filter with any circuit type besides Clean, there is an additional Drive control that can be used to add gain or distortion to the signal before it enters the filter.

The Morph filter has an additional Morph control which sweeps the filter type continuously from lowpass to bandpass to highpass to notch and back to lowpass. Filter cutoff frequency and resonance can be adjusted in the shell or by dragging the filter response curve in the display area. Filter frequency can also be modulated by the following:. The Shaper Drive Shp. If you open a Set that was created in a version of Live older than version 9.

These consist of 12 dB or 24 dB lowpass, bandpass and highpass filters, as well as a notch filter, and do not feature a Drive control. Each Operator loaded with the legacy filters shows an Upgrade button in the title bar. Pressing this button will permanently switch the filter selection to the newer models for that instance of Operator. Note that this change may make your Set sound different. Additionally, the global display area provides a comprehensive set of modulation routing controls.

The maximum number of Operator voices notes playing simultaneously can be adjusted with the Voices parameter in the global display. Ideally, one would want to leave this setting high enough so that no voices would be turned off while playing, however a setting between 6 and 12 is usually more realistic when considering CPU power.

Tip: Some sounds should play monophonically by nature, which means that they should only use a single voice. A flute is a good example. In these cases, you can set Voices to 1. If Voices is set to 1, another effect occurs: Overlapping voices will be played legato, which means that the envelopes will not be retriggered from voice to voice, and only pitch will change. The center of the global display allows for a wide variety of internal MIDI mappings.

For more information about the available modulation options, see the complete parameter list see Operator includes a polyphonic glide function. When this function is activated, new notes will start with the pitch of the last note played and then slide gradually to their own played pitch.

Glide can be turned on or off and adjusted with the Glide Time control in the pitch display. Operator also offers a special Spread parameter that creates a rich stereo chorus by using two voices per note and panning one to the left and one to the right. The two voices are detuned, and the amount of detuning can be adjusted with the Spread control in the pitch section of the shell. Tip: Whether or not spread is applied to a particular note depends upon the setting of the Spread parameter during the note-on event.

To achieve special effects, you could, for instance, create a sequence where Spread is 0 most of the time and turned on only for some notes. These notes will then play in stereo, while the others will play mono.

Note: Spread is a CPU-intensive parameter. If you want to save CPU power, turn off features that you do not need or reduce the number of voices. For the sake of saving CPU resources, you will also usually want to reduce the number of voices to something between 6 and 12, and carefully use the Spread feature.

The Interpolation and Antialias modes in the global display can also be turned off to conserve CPU resources. FM synthesis was first explored musically by the composer and computer music pioneer John Chowning in the mids. In , he and Stanford University began a relationship with Yamaha that lead to one of the most successful commercial musical instruments ever, the DX7. John Chowning realized some very amazing and beautiful musical pieces based on a synthesis concept that you can now explore yourself simply by playing with Operator in Live.

The function of each Operator parameter is explained in the forthcoming sections. Remember that you can also access explanations of controls in Live including those belonging to Operator directly from the software by placing the mouse over the control and reading the text that appears in the Info View. Parameters in this list are grouped into sections based on where they appear in Operator. Tone — Operator is capable of producing timbres with very high frequencies, which can sometimes lead to aliasing artifacts.

The Tone setting controls the high frequency content of sounds. Higher settings are typically brighter but also more likely to produce aliasing. Algorithm — An oscillator can modulate other oscillators, be modulated by other oscillators, or both. The algorithm defines the connections between the oscillators and therefore has a significant impact on the sound that is created.

Voices — This sets the maximum number of notes that can sound simultaneously. If more notes than available voices are requested, the oldest notes will be cut off.

Retrigger R — When enabled, notes that are enabled will be retriggered, rather than generating an additional voice. Interpolation — This toggles the interpolation algorithm of the oscillators and the LFO.

If turned off, some timbres will sound more rough, especially the noise waveform. Turning this off will also save some CPU power. Disabling this modes reduces the CPU load. Pan — Use this to adjust the panorama of each note. This is especially useful when modulated with clip envelopes.

Typically this is used for piano-like sounds. These modulation targets are available as MIDI routing destinations in the global display, and also as modulation targets for the LFO and pitch envelope. OSC Feedback — Modulates the amount of feedback for all oscillators. Note that feedback is only applied to oscillators that are not modulated by other oscillators.

FM Drive — Modulates the volume of all oscillators which are modulating other oscillators, thus changing the timbre. Filter Drive — Modulates the amount of the Drive not available when the Morph filter is selected. Pitch Envelope On — This turns the pitch envelope on and off. Turning it off if it is unused saves some CPU power.

Spread — If Spread is turned up, the synthesizer uses two detuned voices per note, one each on the left and right stereo channels, to create chorusing sounds. Spread is a very CPU-intensive effect. Transpose — This is the global transposition setting for the instrument. Additive synthesis is useful for digital voice textures, and you can create vowel formants by attenuating or emphasizing specific clusters of harmonics see Figure 4.

Bell Harmonics. For bell and chime tones, start with the sine waveform as the fundamental and add several widely spaced harmonics see Fig. These can even be randomly selected, as long as there are only a few and they cover a wide range of frequencies. ProTip: You can also emphasize a few of these chime harmonics in other waves and create unique hybrid sounds. The two algorithms in Figure 6 are perfect starting points for your experiments, with the first offering a single FM modulator simultaneously affecting three carriers.

The second algorithm contains two independent carrier oscillators while adding an FM carrier-modulator pair as a secondary element. Set all oscillators to sine waves and tune the three carriers to 1, 2, and 4. This will give you an organ-like sound consisting of three octaves. Next, slowly raise the level of oscillator D to around dB.

This approach is the essence of FM synthesis. Since every FM oscillator includes its own envelope, you can now manipulate the intensity of the frequency modulation over time—and thus, change the timbre—by adjusting the shape of the modulator envelope.

Short envelopes with quick decays are great for percussive tones, while long attacks and releases will deliver results that slowly evolve over time. Figure 7 shows my all-time favourite FM algorithm for getting quick results from a 4-operator synth.

It consists of dual, independent carrier-modulator pairs and works really well for quickly creating layered effects. The simplified concept is that the carrier oscillator handles tuning and volume, while the modulator governs the overall timbre and its envelope. Here is a quick way to correlate the parameters of FM and subtractive synthesis when using a single FM oscillator pair. In Figure 8, each modulator can affect a different element in your sound.

For example, the first modulator can generate the timbre for a sustained tone, the second modulator can introduce a subtle decay sweep, while the third modulator applies a sharp attack transient see Figure 9. By using a low coarse tuning value and level between 1 and 5 for the sustaining modulator, you can create more subdued tones, as higher harmonics with extreme modulation levels impart harshness.