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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. Changing this parameter will affect notes that are already playing. It is therefore listed in the section on envelopes see Glide G — With Glide on, notes will slide from the pitch of the last played note to their played pitch.
Note that all envelopes are not retriggered in this case if notes are being played legato. Glide Time Time — This is the time it takes for a note to slide from the pitch of the last played note to its final pitch when Glide is activated.
This setting has no effect if Glide is not activated. Pitch Envelope to Osc Destination A-D — The pitch envelope affects the frequency of the respective oscillator if this is turned on. Pitch Envelope Destination B — This sets the second modulation destination for the pitch envelope. Filter On — This turns the filter on and off. Turning it off when it is unused saves CPU power. Filter Type — This chooser selects from lowpass, highpass, bandpass, notch, and Morph filters.
Circuit Type — This chooser selects from a variety of circuit types that emulate the character of classic analog synthesizers. Filter Frequency Freq — This defines the center or cutoff frequency of the filter.
Note that the resulting frequency may also be modulated by note velocity and by the filter envelope. Filter Resonance Res — This defines the resonance around the filter frequency of the lowpass and highpass filters, and the width of the bandpass and notch filters.
The center point for this function is C3. It is therefore listed in the section on envelopes. Filter Drive Flt. Drive — Applies additional input gain to the signal before it enters the filter. Shaper Drive Shp. Drive — This boosts or attenuates the signal level being sent to the waveshaper. Turning it off when it is unused saves some CPU power. All waveforms are band-limited to avoid unwanted clicks.
Due to the possible high frequencies, the LFO can also function as a fifth oscillator. Retrigger R — When enabled, the LFO restarts at the same position in its phase each time a note is triggered. Note that the actual effect also depends on the LFO envelope. Osc Coarse Frequency Coarse — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters.
Coarse sets the ratio in whole numbers, creating a harmonic relationship. Osc Fine Frequency Fine — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters. Fine sets the ratio in fractions of whole numbers, creating an inharmonic relationship. This frequency is constant, regardless of note pitch. Osc Fixed Multiplier Multi — This is used to adjust the range of the fixed frequency. Osc Output Level Level — This sets the output level of the oscillator.
If this oscillator is modulating another, its level has significant influence on the resulting timbre. Osc Waveform Wave — Choose from a collection of carefully selected waveforms. You can then edit them via the harmonics editor. Osc Feedback Feedback — An oscillator can modulate itself if it is not modulated by another oscillator.
The modulation is dependent not only on the setting of the feedback control but also on the oscillator level and the envelope. Higher feedback creates a more complex resulting waveform. Osc Phase Phase — This sets the initial phase of the oscillator.
The range represents one whole cycle. Retrigger R — When enabled, the oscillator restarts at the same position in its phase each time a note is triggered. Repeat — Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser. If activated, the sonic result is the same as manually changing the Coarse parameter for each note. Applying this to modulating oscillators creates velocity-dependent timbres.
Envelope Attack Time Attack — This sets the time it takes for a note to reach the peak level, starting from the initial level. For the oscillator envelopes, the shape of this segment of the envelope is linear. For the filter and pitch envelopes, the shape of the segment can be adjusted. Envelope Decay Time Decay — This sets the time it takes for a note to reach the sustain level from the peak level. For the oscillator envelopes, the shape of this segment of the envelope is exponential.
Envelope Release Time Release — This is the time it takes for a note to reach the end level after a note-off message is received. For the oscillator envelopes, this level is always -inf dB and the shape of the segment is exponential. For the filter and pitch envelopes, the end level is determined by the End Level parameter and the shape of the segment can be adjusted.
This envelope segment will begin at the value of the envelope at the moment the note-off message occurs, regardless of which segment is currently active. Envelope Sustain Level Sustain — This is the sustain level at the end of the note decay. The envelope will stay at this level until note release unless it is in Loop, Sync or Beat Mode. Envelope Loop Mode Loop — If this is set to Loop, the envelope will start again after the end of the decay segment. If set to Beat or Sync, it will start again after a given beat-time.
In Sync Mode, this behavior will be quantized to song time. In Trigger mode, the envelope ignores note off. When retriggered, the envelope will move at the given attack rate from the current level to the peak level. The time it takes to move from the sustain level to the initial value is defined by this parameter. This is especially interesting if the envelopes are looping.
Note that this modulation does not influence the beat-time in Beat or Sync Modes, but the envelope segments themselves. The filter and pitch envelopes also provide parameters that adjust the slope of their envelope segments. These include:. There is also a command to export the waveform as an. When enabled, E3 is the center.
When disabled, C3 is the center. Note that this option is only available when loading Operator presets that were made in versions of Live prior to Live 9. Sampler users who want to share their presets with all Live users can convert their work to Simpler see It has been designed from the start to handle multi-gigabyte instrument libraries with ease, and it imports most common library formats.
Getting started with Sampler is as easy as choosing a preset from the browser. Presets imported from third-party sample libraries are listed here, too, in the Imports folder. Once you have loaded a Sampler preset into a track, remember to arm the track for recording which also enables you to hear any MIDI notes you might want to play , and then start playing!
This technique is used to accurately capture the complexity of instruments that produce dynamic timbral changes. Rather than rely on the simple transposition of a single recorded sample, multisampling captures an instrument at multiple points within its critical sonic range. This typically means capturing the instrument at different pitches as well as different levels of emphasis played softly, moderately, loudly, etc.
The resulting multisample is a collection of all the individually recorded sample files. The acoustic piano, for example, is a commonly multisampled instrument. Sampler is designed to let you approach multisampling on whatever level you like: you can load and play multisample presets, import multisamples from third-party vendors see Play Video. Get It Now. Slate Digital was created to provide the greatest digital audio tools for musicians, producers, and engineers.
But the fun part starts when you feed it a second signal into the sidechain. The two different sounds will overlap based on the effect settings and the selected waveform. You get tremolo , vibrato , and panner and the ability to blend two signals and get unconventional solutions. All in all, you can safely use it both for mixing guitars and other instruments.
You may get inventive with your instruments with Eventide Spring Reverb. We all bemoaned the harsh assault and metallic sounds of spring-based devices back when they were the only inexpensive way to produce fake reverb, but just like with tape distortion, we found ourselves missing it once it is gone!
It is available as a plug-in that supports the standard Mac OS and Windows plug-in formats as well as an iOS app, and it has a wider range of options than the actual spring reverb. The plugin is available for Windows 8 or higher and macOS It may be used to produce vibrant voices, textured synth treatments, and some unpleasant percussion sounds. While the pre-post option, in conjunction with the independent pitch modulation, makes it simple to create movement or even rhythmic pulsations owing to the tempo lock function, the tremolo makes it simple to obtain those classic surf guitar tones.
Giving color to the mix bus or individual instruments is always a good ide a. You get six different processing modules here , which include various types of effects. In addition, you can either apply all the sections at once or disable individual ones, choosing only what you need. And , of course , there is the section that is responsible for the overall output.
It has modules including Noise , Wobble , Distort , Space , and others. Also , it has a pretty powerful Master section , where you can adjust the tone and set the width. Radiator is a digital model of the classic rack-mounted tube mixer Altec A, turned into a dual drive tube input channel and EQ. This mic preamp is loud , colored , and warm-sounding. It has a two-knob tone control, bass, and treble, which recreates the frequency response from the original hardware. Its input and output amplification stages are independent to add more character.
Supports Windows 7 or higher , and macOS Soundtoys worked hard to make the sound of this digital emulation as realistic , warm , smooth , and you can find some good use for it with drums and percussions, though it works well on anything. The graphic interface is intuitive, to a point where you can sit back and relax and let it do its thing. Regardless, it is fair to say that Radiator is a lot darker-sounding than V76U73 and PreX7 but offers a cleaner sound.
B-3 V2 by Arturia offers a highly creative environment with realistic tonewheel organ sounds and a carefully made rotary speaker add-on. With many modulation options for the sound, the B-3 V2 plugin features all the textures most organs have. With great attention to detail, the B-3 V2 will work on an elite quality physical modeling technology.
With advanced controls to optimize the instrument, the plugin also features a realistic interface with two keyboards and a swell pedal, along with percussion settings. This plugin is available for Windows 7 or higher and macOS The B-3 V2 by Arturia delivers versatility and authenticity with a virtual organ instrument. You can define an immense array of variables to accurately shape the sound and the adjustments of the organ model.
Equipped with an easy-to-use interface, the B-3 V2 is an excellent option for anybody that wants to create and experiment with a high-quality organ synthesizer.
Top 14 Arturia Plugins For Musicians Hence, to summarise the sound design process, you can build up your sounds from basic waveshape like sine, sawtooth, etc. Since you can add harmonics and harmonies to the waveform selected in the first step, hence the name — Harmor.
Unfortunately, Harmor is written in Delphi and is not available on Mac. When it comes to generating pluck sounds, Harmor is a great choice. It has a unique additive synthesis approach with its interesting subtractive synthesis blend. However, the plugin may appear all over the place because of its complex arrangement and design.
For example, that includes effects like blur, prism, harmonize, tremolo, etc. Buy Here Support Integraudio. Soundtoys is a beloved audio plugin company that delivers unique and instantly recognizable plugins. Little plate is a plate reverb that was inspired by five unique EMT plate reverbs.
Soundtoys gathered inspiration from them and created Little Plate as an easy-to-use and instantly inspiring plate reverb catered to all music producers and engineers. The decay knob can inspire countless new sounds , and the infinity mode is great for creative effects, atmospheres, and sound design excellence.
One of the latest additions to the Waves family in , the Clarity Vx Pro is here to eliminate the noise with a simple central knob. It is worth mentioning that this software is designed specifically for voice de-noising. The algorithms, usability, and results are optimized for that use. Waves also come with a lighter version of this plugin — Waves Clarity Vx.
Most of them need to process the audio to show the final result or show a real-time preview at a reduced quality. Clarity Vx Pro clears the audio using an advanced deep learning algorithm that delivers good results.
The main central knob, which activates the suppression, allows you to dial the amount of reduction you want. Besides noise removal, this control shares another interesting feature, the ambiance preserving option. This function will remove the voice and keep the background noise at a good level.
This suppression is achieved by the aforementioned deep learning technology, which introduces different functionalities. This plugin version comes with three Neural Networks that focus on different aspects. Three separate networks focus on different aspects of noise reduction.
The Advanced Controls differentiates this plugin from its younger sibling, the Clarity Vx. The top added functionalities include a Reflections knob that restores a certain amount of the natural voice reflections without adding extra reverb. The Analysis button will process the audio as mono or stereo, achieving better results with the latter.
The stereo processing adds a heavier load on the CPU than the Single mode, so bear that in mind. The simplest is to put your drum machine or sampler plug-in on a track, then record MIDI clips on that track. When it comes to mixing, you might prefer each sound to come up on a different audio track, so you can treat each one separately. So you route the sounds to different tracks if the plug-in supports this. The Drum Rack gives you a single plug-in environment with trigger pads, each of which can control a single drum sound.
Each sound has a separate chain signal path , but all the chains are mixed back together at the output of the instrument.
A mixer track containing a Drum Rack has an expand button to the right of its name. Clicking this slides open a new nested mixer view, showing all the chains in the Rack as separate mixer channels. From here you can quickly set levels and pans, and drop audio effects on individual sounds.
Clicking the name of a nested track displays that chain on its own in the device view, so you can quickly see or edit what devices are in the chain. Drum Rack pads are laid out in four columns, with 16 shown at a time. A small overview shows all the pads, with a square indicating which ones are visible. The coolest bit is that when you move the focus square around, your controller will always play the 16 pads that are currently in view.
A MIDI keyboard will play all of the available pads. In the chain list, as well as setting which note pad triggers the chain, you can set which note is sent to the instrument.
Drop an effect on a pad and it gets placed after any sample or instrument already there. From the expanded mixer view, you can drag any of the pad channels into an empty space in the mixer. The chain gets pulled out of the Drum Rack, becoming a self—contained MIDI track with the chain devices inserted on it.
This is fine if you prefer to make your own kits, but otherwise you need to look at getting the Drum Machines or Session Drums packages. This is as good a reason as any to opt for the full Ableton Suite. The two included kits are basically teasers for the two new drum libraries. The Drum Machine package is available as a download, and does what it says on the tin. Hopefully there will be more sounds on the way.
This more expensive and much larger library 28GB and available only as a two—DVD box set is dedicated to studio drums. 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. Via the mixer, inter-track routing can work two ways:. In this scenario, soloing Track B will still allow you to hear the output of the tracks that are feeding it.
Also, you can still solo Track A and hear its output signal. In this case, all other tracks are muted, including those that might also feed into Track B. Approach 2, on the other hand, leaves Track A unaffected except for Track B tapping its output. If a track contains one or more Drum Racks see Each Rack will also be listed in the Input Channel chooser:.
Soloing a track that taps a Chain at any of these points will still allow you to hear the output at that point. It is certainly powerful to have a separate effects chain per track for applying different effects to different takes — after the fact. You might, however, want to run the guitar signal through effects a noise gate or an amp model, for instance before the recording stage, and record the post-effects signal. This is easily accomplished by devoting a special audio track for processing and monitoring the incoming guitar signal.
We do not record directly into the Guitar track; instead we create a couple more tracks to use for recording. Those tracks are all set up to receive their input Post FX from the Guitar track. Note that we could also tap the Guitar track Post Mixer if we wished to record any level or panning from it. This output lends itself more to representation as an audio waveform than a single note in a MIDI clip, particularly when comparing the editing options.
A setup similar to the one described above see
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