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Logic Pro Alchemy source master controls Source components are shown only in advanced view. Alternatives integrated versioning. Resonance behavior varies among filter types. The Main edit window instguments divided into three areas that interact with each other.
Logic Pro X – Apple Pro Tr – David Nahmani – – Customer Reviews
Posted July 22, Link to comment Share on other sites More sharing options Jordi Torres Posted July 22, I’ll keep adding to it as I find things to add. Feel free to add to it too! Nice feature is that LPX will automatically try to assign a proper icon from sounds dragged from the loop browser, also from 3rd party content, probably by analyzing the naming of a file.
Where can the discount coupons used toward the purchase of LPX be found? Logic Pro X Cool, I’ve added the ones you’ve contributed guys, thanks. Keep ’em coming!
Posted July 23, Another thing that keeps being asked. The default shortcut for the tool menu is “T. Mac OS X David Nahmani Posted July 23, Thanks teed, adding those too.
Great idea, David. Bass Amp Designer and new stompboxes in Pedalboard. Round robin sample support in EXS Alternatives integrated versioning. Autosave in the background. Here’s my tip for the day.. With Drummer, you can have separate outs. Initiate drummer track 2. Jordi Torres Posted July 23, Chris D Posted July 23, Posted July 23, edited. Edited July 23, by Chris D.
Eriksimon Posted July 23, Cool, animated gif! How did you create that? Please say it’s a freeware app Why did the chicken cross the Mobius ring? The ability to create your own color palette is gone in Logic X. Ploki Posted July 23, FoolsGold Posted July 23, Is there a way to do Bounce to iTunes Library? Macbook Pro 2. I sorta liked it, saved having to go folder diving through the finder to find your bounce.
Eric Cardenas Posted July 23, Nola Posted July 23, Sorry but back to the letter T for the tools list – I specifically said “Yes” the first time I started LPX when it asked me if I wanted to use the new set of key commands designed for LPX – but Esc is still what brings up the tool list for me, and T does nothing??
You know that can assign it it manually if you want? Type “tool” in its’ search box optional, it’s just for easier searching 2. Oh yes, I know it’s all customizable. Thanks for the tip. Join the conversation You can post now and register later. Reply to this topic Insert image from URL. Go to topic listing. I can’t change my signature.
See Use the ES1 sub-oscillatoron page The lowest setting is 32feet and the highest is 2feet. The use of the term feet to determine octaves comes from the measurements of organ pipe lengths.
The longer and wider the pipe, the deeper the tone. Modulate ES1 pulse width mm Rotate the Wave knob to a position between the square wave and pulse wave symbols. The pulse width can also be automatically modulated in the modulation section see Use the ES1 routeron page Modulating the pulse width with a slowly cycling LFO, for example, allows periodically mutating, fat bass sounds.
ES1 oscillator waveforms The table outlines the basic tones of the oscillator waveformshow they affect your synthesizer sound. Its pitch matches the frequency of the main oscillator. You can choose from the following sub-oscillator waveform options:. Variations of these waveforms, with different mixes and phase relationships, resulting in various sounds. White noise, which is useful for creating percussion sounds as well as wind, surf, and rain sounds.
EXT, which allows you to run an external channel strip signal through the ES1 synthesizer engine, by using a side chain. The global parameters affect the overall sound, or behavior, of the ES1 and are found primarily in the strip that spans the lower edge of the ES1 interface. The Glide slider is displayed above the left end of the strip.
Global parameters Glide slider: Drag to set the amount of time it takes to slide between the pitches of each triggered note. The Glide trigger behavior depends on the value set in the Voices field see below.
Analog field: Drag to slightly, and randomly, change the pitch of each note and the cutoff frequency. This emulates the oscillator detuning and filter fluctuations of polyphonic analog synthesizers, due to heat and age. This can be useful for percussive sounds, when you want to achieve a sharper attack characteristic. Use higher values if you want a warm, analog type of soundwhere subtle sonic variations occur for each triggered voice.
Neg Bender Range slider Extended Parameters area : Drag to set the negative downward pitch bend range in semitone steps. The default value is Pos PB positive pitch bend , which essentially means that there is no downward pitch bend available. Click the disclosure triangle at the lower left of the ES1 interface to access the Extended Parameters area. Voices field: Drag to set the maximum number of notes that can be played simultaneouslyup to 16 voices.
When Voices is set to Legato, the ES1 behaves like a monophonic synthesizerwith single trigger and fingered portamento engaged. This means that if you play legato, a portamento glide from one note to the nextwill happen. If you release each key before pressing a new one, the envelope is not triggered by the new note, and there is no portamento. Use this feature to create pitch bend effects, without touching your keyboards pitch bender, by choosing a high Glide parameter value when using the Legato setting.
Chorus field: Click to choose a classic stereo chorus effect, an ensemble effect, or to disable the effects processor. Filter parameters Cutoff slider: Drag to set the cutoff frequency of the ES1s lowpass filter. Resonance slider: Drag to cut or boost the portions of the signal that surround the frequency defined by the Cutoff parameter. Boost can be set so intensively that the filter begins to oscillate by itself see Drive the ES1 filter to self-oscillateon page Tip:You can simultaneously adjust the cutoff frequency and resonance parameters by dragging vertically cutoff or horizontally resonance on the word Filter, found in the center of the black circle.
Slope buttons: The lowpass filter offers four different slopes of band rejection above the cutoff frequency. Click one of the buttons to choose a slope amount of rejection, expressed in decibels dB per octave :. Turning up the resonance results in a reduction of the low end of the signal. This resembles the behavior of an Oberheim filter. Drive slider: Drag to change the behavior of the Resonance parameter, which eventually distorts the sound of the waveform. Drive is actually an input level control, which allows you to overdrive the filter.
Key slider: Drag to set the effect that keyboard pitch the note number has on filter cutoff frequency modulation. If Key is set to zero, the cutoff frequency does not change, no matter which key you strike.
This makes the lower notes sound comparatively brighter than higher notes. If Key is set to maximum, the filter follows the pitch, resulting in a constant relationship between cutoff frequency and pitch.
This mirrors the properties of many acoustic instruments, where higher notes sound both brighter in tone and higher in pitch. ADSR via Vel sliders: Drag to determine how note velocity affects modulation of the filter cutoff frequency with the envelope generator. Filter Boost button Extended Parameters area : Turn on to increase the output of the filter by approximately 10 decibels. The filter input has a corresponding decrease of approximately 10 decibels, maintaining the overall level.
This parameter is particularly useful when applying high Resonance values. See Drive the ES1 filter to self-oscillate. Drive the ES1 filter to self-oscillate If you increase the filter Resonance parameter to higher values, the filter begins to internally feed back and, as a consequence, begins to self-oscillate. This results in a sine oscillationa sine wavethat is actually audible. You can make the ES1 filter output a sine wave by following the steps below.
This lets you play the filter-generated sine wave with the keyboard. Output a sine wave from the filter 1 Switch the Sub knob to Off.
Filter Boost increases the output of the filter by approximately 10 decibels, making the selfoscillation signal much louder. The parameters in the ES1 Amplifier section allow you to fine-tune the behavior of your sounds level. These are separate from the global Out Level parameter, which acts as the ES1s master volume control. Amplifier parameters Level via Vel slider: Drag to determine how note velocity affects the synthesizer level. The greater the distance between the arrows indicated by the blue bar , the more the volume is affected by incoming velocity messages.
To simultaneously adjust the modulation range and intensity, drag the blue barbetween the arrowsand move both arrows at once. ES1 envelope parameters ES1 envelope parameters overview ES1 provides an attack, decay, sustain, and release ADSR envelope that can shape filter cutoff and the level of the sound over time.
Envelope Parameters A ttack slider: Drag to set the time it takes for the envelope to reach the initial desired level. D ecay slider: Drag to set the time it takes for the envelope to fall to the sustain level, following the initial attack time.
R elease slider: Drag to set the time it takes the envelope to fall from the sustain level to a level of 0. ES1 filter cutoff envelope modulation The envelope generator modulates the filter cutoff frequency over the course of a notes duration.
The modulation intensityand response to velocity informationis set by the arrows on the ADSR via Vel slider in the Filter section. The blue bar between the arrows shows the dynamic range of this modulation. You can simultaneously adjust the modulation range and intensity by dragging the blue bar. Tip:If youre unfamiliar with these parameters, set the Cutoff parameter to a low value, Resonance to a high value, and move both ADSR via Vel arrows upward.
Constantly strike a note on the keyboard while changing the arrows to learn how these parameters work. All ADSR parameters remain active for the filter. The letters A, D, S, and R refer to the attack, decay, sustain, and release phases of the envelope see ES1 envelope parameters overviewon page Gate refers to a control signal used in analog synthesizers that is sent to an envelope generator when a key is pressed. As long as an analog synthesizer key is pressed, the gate signal maintains a constant voltage.
When Gate is used as a modulation source in the voltage-controlled amplifier instead of the envelope , it creates an organ-type envelope without any attack, decay, or release phasein other words, an even, sustained sound.
In-between these phases, the Gate control signal is used to maintain a constant level while a note is held. As soon as you release the key, the release phase begins. GateR: The Gate control signal is used to maintain a constant level while a note is held.
ES1 modulation ES1 modulation parameters overview ES1 offers a number of simple yet flexible modulation routing options. You use modulation to add animation to your sound over time, making it more interesting, lively, or realistic. A good example of this type of sonic animation is the vibrato used by orchestral string players. Router: Enables you to choose the ES1 parameters that are modulated. See Use the ES1 routeron page Modulation Envelope: A dedicated modulation control source that can be used to control various ES1 parameters, or it can be used to control the LFO level.
See Use the ES1 modulation envelopeon page The buttons in the left column set the target for LFO modulation. The buttons in the right column set the target for the modulation envelope. Router parameters Pitch buttons: Click to modulate the pitchthe frequencyof the oscillators.
Pulse Width buttons: Click to modulate the pulse width of the pulse wave. Mix buttons: Click to modulate the mix between the primary oscillator and the sub-oscillator. Cutoff buttons: Click to modulate the cutoff frequency of the filter. Resonance buttons: Click to modulate the resonance of the filter. Volume buttons: Click to modulate the main volume.
Filter FMbutton modulation envelope only : Click to use the triangle wave of the oscillator to modulate filter cutoff frequency. This modulation can result in a pseudo-distortion of the sound, or it can create metallic, FM-style sounds.
The latter occurs when the only signal you can hear is the self-oscillation of the resonating filter see Drive the ES1 filter to self-oscillateon page Each waveform has its own shape, providing different types of modulation. You can also choose EXT to assign a side-chain signal as a modulation source.
Choose the side-chain source channel strip from the Side Chain pop-up menu in the upper-right corner of ES1.
When set to 0, the LFO outputs at a constant, full level, which allows you to manually control the LFO speed with your keyboards modulation wheel. This can be useful, for example, if you want to change the pulse width by moving your keyboards modulation wheel.
You would choose the pulse width as the LFO modulation target, using a button in the left router column, and set the modulation intensity range using the Int via Whl slider. Int via Whl slider: The upper arrow defines the intensity of LFO modulation if the modulation wheel is set to maximum.
The lower arrow defines the amount of LFO modulation if the modulation wheel is set to 0. The distance between the arrowsshown as a green bar indicates the range of your keyboards modulation wheel. You can simultaneously adjust the modulation range and intensity by dragging the green bar, thus moving both arrows at once.
Note that as you do so, the arrows retain their relative distance from each other. Use the ES1 modulation envelope The modulation envelope can directly modulate the parameter chosen in the router. It determines the time it takes for the modulation to fade in or fade out. At its center position click Full , modulation intensity is staticno fade-in or fade-out occurs. When set to its full value, modulation intensity is at a constant level.
The modulation envelope allows you to set either a percussive type of decay envelope by choosing low values or an attack type of envelope by choosing high values. Modulation envelope parameters Form slider and field: Drag to set a fade-in attack or fade-out decay time for the modulation.
When set to the full position, the modulation envelope is turned off. The green bar between the arrows displays the impact of velocity sensitivity on the intensity of the modulation envelope. Modulate a parameter with velocity 1 Select a modulation target, such as Pulse Width, from the right column of the router. This results in a velocity-sensitive modulation of the oscillator pulse width. The higher the value, the longer it takes for you to hear the modulation.
The lower the valuecloser to decaythe shorter the fade-out time is. LFO control with envelopes is most often used for delayed vibrato, a technique many instrumentalists and singers employ to intonate longer notes. Set up a delayed vibrato 1 Drag the Form slider to the righttoward attack. ES2 combines subtractive synthesis with elements of FM and wavetable synthesis to help you generate an extraordinary variety of sounds. This makes it the perfect choice for creating powerful pads, evolving textures, rich basses, or synthetic brass.
If youre new to synthesizers, see Synthesizer basics overviewon page, which introduces you to the fundamentals and terminology of different synthesis systems. The three oscillators of the ES2 provide classic analog synthesizer waveforms including noise and single-cycle waveforms, known as Digiwaves. This raw material forms the basis for sounds that range from fat analog to harsh digital sounds, or hybrids of the two.
You can also cross-modulate oscillators, making it easy to create FM-style sounds. Further options include the ability to synchronize and ring-modulate the oscillators or to mix a sine wave directly into the output stage, to thicken the sound. ES2 features a flexible modulation router that offers up to ten simultaneous user-defined modulation routings. Further modulation options include the unique Planar Padwhich provides control of two parameters on a two-dimensional grid.
The Planar Pad itself can be controlled by the sophisticated Vector Envelope. This is a multipoint, loop-capable envelope that makes it easy to create complex, evolving sounds.
If you want to begin experimenting right away, there are a number of settings to try. There are also two tutorials that provide tips and information, and invite you to explore the ES2.
See ES2 sound design from scratch overviewon page94 and ES2 sound design with templateson page Note:You will find tasks that cover the use of parameters as modulation targets or sources throughout these pages. This underlines one of ES2s greatest strengthsnamely, the vast modulation possibilities it offers. Follow the steps in these tasks to create expressive, evolving sounds.
See ES2 modulation overviewon page ES2s graphical interface is divided into the following main areas. Modulation router Global parameters. Random parameters Macro Sound parameters.
Oscillator section: The oscillator parameters are shown in the upper-left area of the ES2interface. The Triangle is used to set the mix relationships between the three oscillators.
See ES2 oscillator parameters overviewon page Global parameters: A number of related global parameters that directly influence the overall output of the ES2, such as Tune, are found to the left of the oscillators, and above the amplifier and filter parameters. See Global parameters overviewon page Filter section: The circular area houses the filter section, including the Drive and Filter FMparameters.
See ES2 filter overviewon page Amplifier parameters: The area at the top right contains the output parameters, where you can set the overall volume of the ES2, and add a sine signal at the output stage. See Use ES2s dynamic stageon page Modulation router or Vector Envelope: The dark strip across the center of the ES2interface is shared by the modulation router and the Vector Envelope.
Use the buttons at the right end of this section to switch between the two. The router links modulation sources, such as the envelopes and other parameters shown in the lower portion of the interface, to modulation targets, such as the oscillators and filters. See Use the modulation routeron page The Vector Envelope is a flexible, powerful envelope generator that provides extensive control over your sound.
See Use the Vector Envelopeon page Modulation controls and parameters: The area immediately below the router is where you can assign and adjust the modulation generator parameters such as LFO and envelope controls. Planar Pad: The square area at the top right is a two-dimensional controller known as the Planar Pad. The Planar Pad facilitates the simultaneous manipulation of two assignable parameters, and can be controlled with the mouse, another controller, or the Vector Envelope.
See Use the Planar Padon page Effect section: The built-in effect-processing options are found to the right of the output parameters. See ES2 integrated effects processoron page The preassigned macro sound parameters are perfect for quick tweaks to the ES2s sound and that of ES2-based GarageBand instruments. You can reassign MIDI control numbers for these parameters. See ES2 macro and controller assignment overviewon page ES2sound sources ES2 oscillator parameters overview ES2 oscillators are used to generate one or more waveforms.
This signal is then sent to other portions of the synthesizer engine for shaping, processing, or manipulation. Oscillators 2 and 3 can be synchronized to, or ring modulated with, oscillator1.
They also have rectangular waves with either user-defined fixed pulse widths or pulse width modulation PWM features. You can use the modulation router to simultaneously change the pulse widths of rectangular waves generated by oscillator1 and the synchronized and ring-modulated rectangular waves of oscillators2 and 3.
Coarse Frequency knob. A green numeric button indicates an active oscillator. A gray numeric button denotes an inactive oscillator. Deactivating an oscillator saves computer processing power. Wave knobs: Rotate to choose the waveform that an oscillator generates. The waveform is responsible for the basic tonal color. See ES2 basic oscillator waveformson page Coarse Frequency knobs: Rotate to set the oscillators pitch, in semitone steps, over a range of 3octaves. Because an octave consists of 12semitones, the 12, 24, and 36 settings represent octaves.
Fine Frequency value fields: Fine-tune the oscillator frequency pitch. For example, an oscillator with the value 12s 30c sounds an octave 12semitones and 30cents higher than an oscillator with the value 0s 0c. Drag vertically to adjust each value. Oscillator Mix Triangle : Move the pointer in the Triangle to cross-fade set the level relationships between the three oscillators.
See Balance ES2 oscillator levelson page ES2 basic oscillator waveforms All ES2oscillators output a number of standard waveformssine, pulse, rectangular, sawtooth, and triangular wavesor, alternately, any of Digiwaves see Use ES2 Digiwaveson page The following table covers the basic waveforms: Waveform.
Useful for basses, clarinets, and oboes. The sine wave of oscillator1 can be frequency modulated by oscillator2. Oscillator synchronization and ring modulation allow for the creation of very complex and flexible harmonic spectra. The principles behind oscillator synchronization are described in Synchronize ES2oscillatorson page Ring modulation principles are described in Use ring modulation in ES2on page Use pulse width modulation in ES2 You can alter the tonal color of rectangular waveforms by scaling the width of waveform pulses to any value.
This is known as pulse width modulation. ES2 pulse width modulation features are extensive. For example, if rectangular waves are chosen for all oscillators, you can simultaneously modulate the pulse width of oscillator1 and the synchronized pulse waves of oscillator2 or the square wave of oscillator2s ring modulator and oscillator3. Set a basic pulse width in oscillator2 or 3 mm Drag the waveform rotary control that surrounds the Wave knob see the highlighted area in the image above.
Only oscillators2 and 3 allow you to define a base default pulse width, prior to any pulse width modulation. Set up a pulse width modulation of oscillator1 in the router 1 Choose a rectangle wave for oscillator 1. Use frequency modulation in ES2 The principle of frequency modulation FM synthesis was developed in the late s and early s by John Chowning. It was popularized by Yamahas range of DXsynthesizers in the s. Although the ES2 cant be compared with the DXseries in the discipline of pure FMsynthesis, it can achieve some of the signature sounds of these instruments.
In pure FM synthesis, the frequency of one signal generator, or oscillator, is altered modulated by another signal generator. Positive values from the second generator increase the frequency of the first generator.
Negative values decrease the frequency. In a synthesizer, this type of modulation takes place in the audible range. Depending on the design of the instrument, you can hear the signals of either the first oscillator alone being modulated by the other oscillator , or both oscillators.
The interaction between the two generators alters the waveform signal of the first oscillator and introduces a number of new harmonics. This harmonic spectrum can then be used as the source signal for further sound processing, such as filtering, envelope control, and so on.
See Frequency modulation FM synthesison page for further information. In ES2, the frequency of oscillator1 with a sine wave chosen11 oclock position for the Wave knob can be modulated by the output signal of oscillator2. The net effect of speeding up or slowing down the frequency of oscillator1 in each waveform cycle is a distortion of the basic wave shape. This waveform distortion also has the side benefit of introducing a number of new, audible harmonics.
Important:The impact of any frequency modulations you perform depends on both the frequency ratio and the modulation intensity of the two oscillators. The pure FMsynthesis method uses a sine wave for both the first and second signal generator both oscillator1 and 2 would be limited to generating a sine wave in ES2 if you stuck with this approach.
ES2, however, provides Digiwaves and countless combinations of modulation intensities and frequency ratios that can be used for either oscillator. This provides a vast pool of harmonic spectra and tonal colors for you to experiment with.
Tip:The type of modulation that occurs can vary significantly when different waveforms are chosen for oscillator2the modulating oscillatorin particular. Set the frequency ratio and adjust the modulation intensity 1 Adjust the Frequency coarse and fine tune parameter values of one, or both, oscillators. This determines the amount, or intensity, of frequency modulation. Use ring modulation in ES2 Ring modulation is a powerful tool for the creation of inharmonic, metallic, bell-like sounds.
The spectra resulting from its use are inharmonic at almost every frequency ratio. The ring modulator is a device that dates back to the early days of the synthesizer. A ring modulator has two inputs. At the output you hear both the sum and difference frequencies of the input signals. If you ring modulate a sine oscillation of Hz with a sine oscillation of Hz, the output signal of the ring modulator consists of a Hz sum and a Hz difference signal.
Negative frequencies result in a change to the phase polarity of output signals. Tip:Use sawtooth and rectangular pulse width modulated input signals from oscillators 1 and 2, respectively, to create a much more complex output signal. The use of these harmonically rich waveforms results in a number of extra sidebands becoming audible. Create a ring-modulated sound 1 Set the oscillator 2 Wave knob to the Ring setting.
The oscillator2 ring modulator is fed with the output signal of oscillator1 and a square wave, generated by oscillator2 itself. The pulse width of this square wave can be modulated see Use pulse width modulation in ES2on page Use ES2 Digiwaves In addition to the basic synthesizer waveforms, all ES2oscillators provide additional waveforms, called Digiwaves. These are very short samples of the attack transients of various sounds and instruments.
Choose a Digiwave mm Set the Wave knob to Sine 6 oclock position , then do one of the following:. Use the ES2 noise generator The sonic palette of oscillator3 is bolstered by the inclusion of a noise generator, which can be activated by choosing the noise waveform. By default, oscillator3s noise generator generates white noise.
White noise is defined as a signal that consists of all frequencies an infinite number sounding simultaneously, at the same intensity, in a given frequency band. The width of the frequency band is measured in Hertz. Sonically, white noise falls between the sound of the consonant F and breaking waves surf. White noise is useful for synthesizing wind and seashore noises, or electronic snare drum sounds.
You can also modulate the tonal color of the noise signal in real timewithout using the main filters of the ES2by modulating the waveform of oscillator3. Change the noise color 1 Set up a modulation routing as follows:modulation target Osc3Wave, source ModWhl. The modulation amount slider behaves somewhat differently with this routing, essentially acting like a filter. The sound becomes darker red noise as you adjust the mod wheel downwards. When Osc3Wave is modulated positively, the noise becomes brighter blue noise.
ES2 emulation of detuned analog oscillators The Analog parameter randomly alters the pitch of each note and the filter cutoff frequency.
Medium Analog values simulate the tuning instabilities of analog synthesizer circuitry, which can be useful in achieving that much sought-after warmth of analog hardware synthesizers.
High Analog values result in significant pitch instability, which can sound truly out of tune but this may be perfect for your needs. Rotate the Analog knob to randomly alter the pitch of each note, and the filter cutoff frequency.
Much like polyphonic analog synthesizers, all three oscillators maintain their specific frequency deviation from each other, but the pitches of all three oscillators are randomly detuned by the same Analog amount.
In this situation, Analog sets the amount of detuning between the stacked unison voices. For more information about these parameters, see Set the ES2 keyboard modeon page Stretch tuning in ES2 The coarse Frequency knob of each oscillator enables you to tune oscillators 1, 2, and 3 in semitones or octaves.
Precise detuning between oscillators can result in beats, or phasing, between the oscillator frequencies. High notes, therefore, may seem to be somewhat out of tune in comparison with lower notes.
CBD Constant Beat Detuning can be used as a corrective tool to even out the beating between oscillators, or it can be used as a creative tool to emulate stretch tuning. The latter can be particularly important when you use an ES2 sound alongside an acoustic piano recording. This is because acoustic pianos are intentionally tuned out-of-tune from equal temperament. This is known as stretch tuning, and results in the upper and lower keyboard ranges being slightly out of tune with the center octaves but harmonically in-tune with each other.
Choose a CBD value to detune the harmonics of low note frequencies in a ratio proportionate with the fundamental tone of the upper note frequencies.
This value may, however, be too high, because the lower notes might be overly detuned at the point where the phasing of the higher notes feels right. Try lower CBD values in cases where the bass notes are a little too far out of tune with the upper keyboard range. Balance ES2 oscillator levels The position of the pointer in the Triangle is described by two parametersx and y coordinateswhich are used when automating the oscillator mix.
Drag the pointer in the Triangle to cross-fadeset the level relationshipsbetween the three oscillators. This is self-evident in use. If you move the pointer along one of the Triangles sides, it cross-fades between the two closest oscillators, and the third oscillator is muted.
Click or drag in the Triangle to change the level balance between the oscillators. The position of the pointer x and y coordinates in the Triangle can also be controlled with the Vector Envelope. Because the Vector Envelope features a loop function, it can be used as a pseudo-LFO with a programmable waveform. For more information about this feature, see Use the Vector Envelopeon page Modulate triangle coordinates with the modulation wheel 1 Set up a modulation routing as follows:modulation target OscLevelX, source ModWhl.
Adjust the intensity. You can choose other sources for these targets. ES2oscillator start points The oscillators can run freely or can begin at the same phase position of their respective waveform cycles each time a note is struck. Choose free, soft, or hard from the Osc illator Start pop-up menu. Free: The initial oscillator phase start point is random for each played note. This adds life to the sound. The downside is that the output level may differ each time a note is played, making the attack phase sound less punchyeven if the performance is identical each timesuch as when the note is triggered by a MIDI region.
This setting is useful when you are emulating sounds typical of hardware analog synthesizers. Soft: The initial oscillator phase starts at a zero crossing for each played note. This mimics the sonic character and precision typical of digital synthesizers.
Hard:The initial oscillator phase starts at the highest level in the waveform cycle for each played note. The extra punch that this setting can provide is audible only if the ENV3 Attack Time parameter is set to a low valuea very fast attack, in other words.
This setting is highly recommended for electronic percussion and hard basses. Note:Osc Start soft and hard result in a constant output level of the initial oscillator phase every time the sound is played back. This may be of importance when you use the Bounce function of Logic Pro at close to maximum recording levels. Synchronize ES2oscillators Typical oscillator sync sounds tend toward the aggressive, screaming leads that synthesizer manufacturers like to talk about.
The rectangular and sawtooth waveforms of oscillators 2 and 3 feature a Sync option. When this parameter is turned on, the phase of oscillator2 or 3 is synchronized with oscillator1. Every time oscillator1 starts a new oscillation phase, the synchronized oscillator oscillator2 or 3 is also forced to restart its phase from the beginning.
Between the waveform cycles of oscillator1, the waveform cycles of the synchronized oscillators run freely. You can achieve interesting synchronized oscillator sounds by modulating the frequency of the synchronized oscillator with an envelope generator. This constantly changes the number of phases within a section of the synchronization cycle, resulting in corresponding changes to the frequency spectrum. Modulate the synchronized oscillator frequency with an envelope 1 Set the oscillator 2 Wave knob to Sync.
You can find global parameters to the left of the oscillators and above the filter and output sections. Global parameters. Global parameters Keyboard Mode buttons: Switch ES2 between polyphonic, monophonic, and legato behaviors.
See Set the ES2 keyboard modeon page Unison button: Click to turn unison mode on or off. See Use unison and voices in ES2on page Glide knob: Rotate to set the time it takes for the pitch of a played note to slide to the pitch of the following played note.
See Set the ES2 glide timeon page Bend Range fields: Drag to define the upward and downward pitch bend range. See Set the ES2 pitch bend rangeon page Tune field: Drag to set the overall instrument pitch in cents. At a value of 0c zero cents , the central A key is tuned to Hz, or concert pitch. Analog knob: Rotate to randomly alter the pitch of each note and the filter cutoff frequency.
See ES2 emulation of detuned analog oscillatorson page Constant Beat Detuning CBD pop-up menu: Choose a CBD value to detune the harmonics of low note frequencies in a ratio proportionate with the fundamental tone of the upper note frequencies. See Stretch tuning in ES2on page Osc illator Start pop-up menu: Choose free, soft, or hard from the Osc illator Start pop-up menu. See ES2oscillator start pointson page Set the ES2 keyboard mode A polyphonic instrument, such as an organ or a piano, allows several notes to be played simultaneously.
Many older analog synthesizers are monophonic, which means that only one note can be played at a time, much like a brass or reed instrument. This shouldnt be viewed as a disadvantage; instead, it allows playing styles that are not possible with polyphonic instruments. Change the keyboard mode mm Click the Poly, Mono, or Legato button. In Mono mode, staccato playing retriggers the envelope generators every time a new note is played.
If you play in a legato style play a new key while holding another , the envelope generators are triggered only for the first note you play legato, then they continue their curve until you release the last legato played key. Legato mode is also monophonic, but with one difference:the envelope generators are retriggered only if you play staccatoreleasing each key before playing a new key.
If you play in a legato style, envelopes are not retriggered. Note:On several monophonic synthesizers, the behavior in Legato mode is referred to as single trigger, while Mono mode is referred to as multi trigger. Use unison and voices in ES2 One of the great strengths of polyphonic analog synthesizers is unisonor stacked voices mode.
Unison mode in polyphonic analog synthesizers is typically monophonic, with all voices playing simultaneously when a single note is struck. Because the voices of an analog synthesizer are never perfectly in tune, the result is an extremely fat chorus effect with great sonic depth.
Use monophonic unison mode 1 Click the Mono or Legato button, depending on the keyboard mode you want to use. See Set the ES2 keyboard mode. The intensity of the unison effect depends on the number chosen in the Voices parameter field. Increase the Voices value for a fatter sound. See Global parameters overview. The intensity of detuning voice deviation is set with the Analog parameter. See ES2 emulation of detuned analog oscillators.
Use polyphonic unison mode mm Click the Poly and Unison buttons. These two voices are heard when you trigger the note. Set the ES2 glide time The Glide parameter also known as portamento sets the time it takes for the pitch of one played note to travel to the pitch of another played note.
Make portamento active mm Rotate the Glide knob. If the keyboard mode is set to Poly or Mono, and Glide is set to a value other than 0, portamento is active. If Legato is chosen, and Glide is set to a value other than 0, you need to play legato press a new key while holding the old one to activate portamento.
If you dont play in a legato style, portamento wont work. This behavior is also known as fingered portamento. Set the ES2 pitch bend range The Bend range fields determine the range for pitch bend modulation, typically performed with your keyboards pitch bend wheel. Set independent upward and downward bend ranges mm Drag in either field to set a bend range.
Set an identical upward and downward bend range 1 Set the upward Bend range field to Link mode. This locks the upward and downward bend ranges, making them identical. This is mirrored in the upward Bend range field. Note:A downward bend of 4semitones results in a combined bend range of 8semitones9 if you include the standard pitch, or no bend position. Filter parameters Filter button: Turns the entire filter section on or off. Deactivating the filter section makes it easier to hear adjustments to other sound parameters, because the filters always heavily affect the sound.
Disabling the filters also reduces processor load. Filter Configuration button: Switches between a parallel or series filter configuration. See ES2 filter configurationon page Filter Blend slider: Sets the balance between Filter1 and Filter2. See Cross-fade between ES2 filterson page Filter1 Mode buttons: Switch Filter1 between lowpass, highpass, bandpass, band reject, or peak filter types. See ES2 Filter1 modeson page Filter2 Slope buttons: Switch Filter2 between different slopes.
See ES2 Filter2 slopeson page Cutoff and Resonance: Rotate the Cutoff and Resonance knobs to determine the cutoff frequency and resonance behavior of each filter.
See Filter cutoff and resonance overviewon page Filter Drive knob: Rotate to overdrive the filter, which affects each voice independently. See Overdrive ES2 filterson page Filter FMknob: Rotate to set the amount of Filter2 cutoff frequency modulation with the oscillator1 frequency.
ES2 filter configuration The Filter Configuration button lets you switch between a parallel and series filter routing. When either is chosen, the entire circular filter element rotates, and the positions and direction of the filter controls clearly indicate the signal flow. The button name also changes in each mode. In the figure to the left, the filters are cabled in series. This means that the signal of all oscillators combined at the Oscillator Mix Triangle passes through the first filter, then this filtered signal passes through Filter2, if Filter Blend is set to 0, the middle position.
The output signal of Filter2 is then sent to the input of the dynamic stage Amplifier section. In the figure to the right, the filters are cabled in parallel. The output signals of the two filters are then sent to the input of the dynamic stage.
See Cross-fade between ES2 filters. Regardless of whether parallel or series filter configurations are chosen, a Filter Blend setting of 1 results in only Filter1 being audible.
The figures illustrate the signal flow between the Oscillator Mix stage the Triangle and the dynamic stage the Amplifier. The signal flow through the filters and the filter overdrive circuit the Drive parameter are dependent on the Filter Blend setting. When zero or positive Filter Blend values are used, there is only one overdrive circuit for both filters. Use of negative Filter Blend values introduces another overdrive circuit, which distorts the output signal of the oscillator mix stage before it is fed into the first filter.
The filters receive a mono input signal from the output of the overdrive circuit. The outputs of both filters are mixed to mono via Filter Blend. Filter 1 Mix. The Filter Blend parameter is available as a modulation target in the router. You can use manual control sources, such as the modulation wheel, to change the filter blend; but the Filter Blend target can also be used creatively, to rapidly switch or smoothly fade between the two filters.
You can also use velocity, or a combination of the Vector Envelope and Planar Pad as sources. The latter allows for interesting filter control possibilities that evolve independently, or alongside oscillator parameters that are also being controlled with the Vector Envelope.
Cross-fade between filters mm Drag the Filter Blend slider to cross-fade between the two filters when cabled in parallel. You can also cross-fade the filters when they are cabled in series. In this situation, the distortioncontrolled by the Drive parameteralso needs to be considered, as this can be positioned either before or in between the filters, depending on the Filter Blend setting you choose.
ES2 Filter1 modes Filter1 can operate in several modes, allowing specific frequency bands to be filtered cut away or emphasized. Lo lowpass : Allows frequencies that fall below the cutoff frequency to pass. Hi highpass : Allows frequencies above the cutoff frequency to pass. Peak: Filter1 works as a peak filter. This allows the level in a frequency band to be increased. The center of the frequency band is determined by the Cutoff parameter. The width of the band is controlled by the Resonance parameter.
BR band reject : The frequency band directly surrounding the cutoff frequency is rejected, but frequencies outside this band can pass. The Resonance parameter controls the width of the rejected frequency band. BP bandpass : The frequency band directly surrounding the cutoff frequency is allowed to pass. All other frequencies are cut. The Resonance parameter controls the width of the frequency band. ES2 Filter2 slopes Most filters do not completely suppress the portion of the signal that falls outside the frequency range defined by the Cutoff parameter.
The slope, or curve, chosen for Filter2 expresses the amount of rejection below the cutoff frequency in decibels per octave. The steeper the slope, the more severe the effect on signal levels below the cutoff frequency. Fat button: Click the Fat button for 24dB per octave of rejection.
Fat mode has a built-in compensation circuit that retains the sounds bottom end. By comparison, the standard 24dB setting tends to make lower end sounds less rich. ES2 filter cutoff and resonance Filter cutoff and resonance overview In every lowpass filter ES2: Lo mode for Filter1; Filter2 is a lowpass filter , all frequency portions above the cutoff frequency are suppressed, or cut off, hence the name. If youre new to synthesizers and the concepts behind filters, see Synthesizer basics overviewon page Cutoff and resonance parameters Cutoff Frequency knob: Rotate to control the brilliance of the signal.
In a lowpass filter, the higher the cutoff frequency is set, the higher the frequencies of signals that are allowed to pass. In a highpass filter, the cutoff frequency determines the point where lower frequencies are suppressed and only upper frequencies are allowed to pass. Resonance knob: Rotate to emphasize or suppress portions of the signal above or below the defined cutoff frequency.
Control two filter parameters simultaneously The ability to change the Cutoff and Resonance controls at the same time is essential for creating expressive synthesizer sounds. Click here to simultaneously adjust the cutoff and resonance of Filter 1. The chain between Cut and Res of Filter1 controls both the resonance drag horizontally and cutoff frequency drag vertically simultaneously. The chain between Cut and Res of Filter2 controls both the resonance drag horizontally and cutoff frequency drag vertically simultaneously.
The chain between Filter1 Cut and Filter2 Cut controls the cutoff frequency of Filter1 drag vertically and Filter2 drag horizontally simultaneously. Force ES2 filters to self-oscillate If you increase the filter Resonance parameter to higher values, the filter begins to internally feed back and, as a consequence, begins to self-oscillate.
To start this type of oscillation, the filter requires a trigger. In an analog synthesizer, this trigger can be the noise floor or the oscillator output. In the digital domain of the ES2, noise is all but eliminated. Therefore, when the oscillators are muted there is no input signal routed to the filter. Filter Reset provides a trigger signal that can be used to drive the filter to self-oscillate. Compensate for high resonance values with the Fat ness parameter mm Click to turn on the Fat ness buttonbelow the other filter slope buttons.
An increase of the resonance value results in a rejection of basslow frequency energywhen using lowpass filters. Use the Fatness button to compensate for this side effect and to obtain a richer sound.
Overdrive ES2 filters The filters are equipped with discrete overdrive modules. You can set the overdrive intensity by rotating the Drive parameter. Drive affects each voice independently. When every voice is overdriven individuallylike having six fuzz boxes for a guitar, one for each stringyou can play extremely complex harmonies over the entire keyboard range.
Each voice sounds clean, without unwanted intermodulation effects spoiling the overall sound. Certain Drive settings can lead to a different tonal character for the following reason:the way analog filters behave when overdriven forms an essential part of a synthesizers sonic character. Each synthesizer model is unique in the way its filters behave when overdriven. ES2 is very flexible in this area, allowing tonal colors that range from the most subtle fuzz to the hardest of distortions.
If the filters are connected in series, the position of the overdrive circuits is dependent on the Filter Blend parameter. Tip:Because Filter2 can cut away the overtones introduced by the distortion, Drive can be used as another tool for deforming oscillator waveforms. Polyphonic distortions in the real world ES2 provides a dedicated distortion effect in the Effects section.
Given this inclusion, you may wonder what benefit the filters Drive function offers. The Distortion circuit in the Effects section affects the entire polyphonic output of the ES2. Every rock guitarist knows that more complex chordsother than major chords, parallel fifths, and octavessound rough when using distortion.
Therefore, distorted guitar playing generally involves few voices or parallel fifths and octaves. Because the filter Drive parameter affects each voice individually, you can play complex chords without introducing the unpleasant intermodulations that the Distortion effect can add to your sound. Modulate ES2s Filter2 Frequency Filter2 cutoff frequency can be modulated by the sine wave of oscillator1, which is always generated, even when the oscillator is switched off.
The level of this sine signal can be mixed in at the output stage with the Sine Level parameter see Sine Level enhanced ES2 soundson page The effect of such filter modulations in the audio spectrum is unpredictable, but the results tend to remain harmonic if you avoid high modulation intensity values.
The FMparameter is used to define the intensity of this filter frequency modulation. Note:Dont confuse this filter frequency modulation with the oscillator FMfeature oscillator1 is modulated by oscillator2.
If oscillator1 is frequency-modulated by oscillator2, it does not influence the sine wave signal used to modulate the cutoff frequencies. See Use frequency modulation in ES2. Filter2 can also be driven to self-oscillation. If you set a very high resonance value, it produces a sine wave. This self-oscillating sine wave distorts at the maximum resonance value.
If you mute all oscillators, youll only hear this sine oscillation. By modulating the cutoff frequency, you can produce effects similar to those produced by modulating the frequency of oscillator1 with oscillator2. A sine wave, at the frequency of oscillator1, is always used as the modulation source. Given this default assignment and the direct relationship between the filter FMintensity and oscillator1s frequency, you can set up a second routing to modulate Oscillator 1s pitch.
ES2 amplifier parameters Use ES2s dynamic stage The dynamic stage of a synthesizer defines the level, or perceived volume, of a played note. The change in level over time is controlled by an envelope generator. For more information about envelope generators, see Synthesizer basics overviewon page ENV3 is hard wired to the dynamic stage of the ES2it is always used to control the level of each note. See ES2 envelopes overview.
The dynamic stage can be modulated by any router modulation source. A tremolo effect is created, with the level changing periodically, based on the current LFO1 Rate value. Sine Level enhanced ES2 sounds The Sine Level knob mixes a sine wave at the frequency of oscillator1 directly into the dynamic stage, independent of the filters.
Even if you have filtered away the basic partial tone of oscillator1 with a highpass filter, you can reconstitute it with this parameter. In cases where oscillator1 is frequency-modulated by oscillator2 if you have turned up FM with the waveform selector , only the pure sine wave is mixed into the dynamic section, not the distorted FMwaveform. Any modulation of oscillator1s pitch, set in the router, affects the frequency of the sine wave mixed in at this stage.
Note:The Sine Level knob is useful for adding warmth and a fat bass quality to the sound. Extra body can be added to thin sounds with this parameter, given that oscillator1 actually plays the basic pitch. ES2 modulation ES2 modulation overview ES2 is equipped with a huge number of modulation sources and targets, making it a synthesizer that can generate extraordinary sounds that constantly evolve, sound like audio loops, or are just plain expressive to play.
Modulation router. Modulation router: The modulation routeror router, for shortlinks modulation sources, such as the envelope, to modulation targets, such as the oscillators and filters. The router features ten modulation routings, arranged into columns. Modulation sources: The modulation sources include the LFOs and envelopes.
Vector Envelope: The Vector Envelope is an extremely sophisticated, loop-capable, multipoint envelope that can control the Planar Pad and Triangle oscillator mix parameter. The Vector Envelope shares the space occupied by the modulation router and can be viewed by clicking the Vector Envelope button to the right of the router. Planar Pad: The Planar Pad is a two-dimensional controller that facilitates the simultaneous manipulation of two, freely assignable, parameters.
It can be controlled with the Vector Envelope. ES2 modulation router Use the modulation router The modulation routeror routerspans the center of the ES2interface. Click the Router button to view it if the Vector Envelope is displayed these components share the same section of the interface. If you are new to synthesizer modulation routings, see Modulation overviewon page Via sources are shown in the middle of each modulation routing.
Modulation targets are shown at the top of each modulation routing. The modulation intensity slider divides into two halves when a via source is active. The modulation intensity slider is not divided when there is no active via source. Any modulation source can be connected to any modulation target, much like an old-fashioned telephone exchange or a studio patch bay.
The modulation intensityhow strongly the target is influenced by the sourceis set with the vertical slider to the right of the modulation routing. The intensity of the modulation can itself be modulated:the via parameter defines a further modulation source, which is used to control the modulation intensity. When via is active, you can specify upper and lower limits for the modulation intensity. Ten such modulation routings of source, via, and target can take place simultaneously.
It doesnt matter which of the ten modulation routings you use. You can even select the same target in several parallel modulation routings. You can also use the same sources and the same via controllers in multiple modulation routings. Create a basic modulation routing 1 Choose the parameter you want to modulate from the Target pop-up menu.
When via is active, this slider sets the minimum modulation intensity. Control ES2 modulation intensity with via sources In a basic modulation routing comprised of a target and source, you can set a fixed modulation intensity by vertically dragging the Intensity slider to the right of the routing.
The slider value always defines a constant modulation intensity. You can choose a further modulation source from the via pop-up menu, which controls modulation intensity. Choosing a value other than off for via divides the Intensity slider into two halves. Each half has its own arrowhead. The upper half of the slider defines the maximum modulation intensity when the via controller is set to its maximum value.
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Note that the chosen filter type can change both the name and function of available controls, notably the Cutoff, Resonance, and Drive parameters. The three filters are independent and can have unique settings. The LED at the top of each button shows on lit or off unlit status. You can step through the available filter types with the Previous and Next buttons the arrows.
Series runs from filter 1 into filter 2. Parallel runs the two filters side-by-side. Resonance behavior varies among filter types. This can lead to intense distortions and aliasing, depending on filter type. Logic Pro Alchemy source filter use tips Alchemy provides multiple filter types at different positions in the signal path.
You can use filters at the source level, and the main filters, and you can also insert filters in the effects section. The positioning can have a significant bearing on the sound produced and can also impact on the processing resources required.
Depending on currently available resources, you may need to pay close attention to envelope settings, the number of voices, and other parameters to achieve the sonic result you are chasing. The downside is that source-level filtering requires more processing resources. Processing is per voice. Filtering at this stage of the signal path is often used to refine the overall sound or to provide a performance control variation. Use an FM filter as a sound source The FM filter produces a sine wave that is modulated by your source signal.
Because the FM process adds harmonics to the sound, the more complex the sound you feed into the filter and the louder it is the more quickly the sound is distorted. FM in Alchemy is great for aggressive sounds, but is also useful for basses and other sounds. Unlike dedicated FM synths, Alchemy does not have preconfigured algorithms or a modulation matrix set up for FM synthesis. This means following the approach outlined in the steps below is not always the best option in Alchemy if you want to recreate classic digital FM sounds.
Such sounds are often more easily achieved by other means, such as with additive synthesis or resynthesis. FM in Alchemy is more like working with FM on analog synths where you modulate oscillator frequency rather than phase.
In Logic Pro X, from the Name bar, click the File button and choose Initialize Preset from the pop-up menu to reset all Alchemy parameters to default settings. Switch to advanced view, then click the A button to show source A parameters.
Click the source A filter On button to enable the filter, then choose FM from the Filter type pop-up menu. The filter is assigned to track keyboard pitch. By default, the centered knob at Hz provides a medium pitched sound that works well. If you want to change the octave, multiply or divide by two, and round to the nearest whole number that sounds best. For example, set the Frequency knob at Hz for one octave lower.
Adjust the modulation Depth on the filter to increase the impact the source signal has on the sine wave generated by the FM filter, and listen to the results. For more grit, try adjusting the Feedback control, which allows the filter output to apply modulation to itself. FM in Alchemy requires experimentation to develop more complex sounds. Here are a few things to try:. When doing so, resist high initial modulation and feedback depths so you can get a feel for the degree of control you have in shaping the overall sound.
Important: Due to technical requirements, FM is often best done at the source filter level. As you progress through the signal path, gain increases and therefore FM become increasingly heavily modulated and distorted. You will find that it is easier to work with FM at the source level than as a master filter or effect. Use a comb filter as a sound source When making comb filters your primary sound source, you may find that longer master envelope release settings are required for a natural feel when playing.
This results in more overlapping notes, higher polyphony, and therefore more CPU overhead. Because of this, you may need to carefully adjust envelope release times and reduce the maximum number of voices in the Master section. By default, the centered knob at Hz will provide a medium pitched sound that works well. Set up an envelope for your impulse signal to excite the comb filter.
You can choose any impulse type, from VA noise to resynthesized samples. The aggressive nature of FM also makes this a good choice of impulse for comb filters. The impulse requires its own envelope, separate from the master or any envelope you have controlling the comb.
The best settings for the envelope depend on the needs of the sound, but a good rule of thumb is to start with zero attack, zero hold, a very short decay, zero sustain, and zero release.
This provides a quick spike that starts comb movement and allows the remainder of sound generation be handled by the comb itself. Tip: The chosen impulse can have a large impact on the tone so it is worthwhile experimenting with different sound sources.
One approach is to import a sample with a strong initial attack using the Additive import method, then use the Additive Harmonic effect knobs to adjust the tone. These controls plus comb filtering can provide numerous fast and easy adjustments, letting you dramatically alter the perceived hardness, material, and tone of your modeled sound. You can also import a drum loop and set it to sustain with Continuous Loop mode. Because drum loops typically contain short bursts of sound that vary in tone, they work well with comb filters.
Use the Damp control to reduce ringing or other artifacts in the sound, if required. Logic Pro Alchemy source elements overview Source components are shown only in advanced view. Each source can make use of multiple synthesis elements that operate on different functional principles.
You can use a synthesis method independently, or you can combine multiple synthesis types by turning on all required elements. For example, you can combine granular synthesis with virtual analog synthesis, or additive synthesis with spectral synthesis. Each synthesis method has inherent strengths and weaknesses, making them more suitable for certain sound types than other synthesis engines. If you are new to synthesis or are unfamiliar with different synthesis approaches, see Synthesizer basics overview.
Not all of these buttons are available for use at any one time. The combination of active buttons updates to reflect the audio data specified in the source select field and the import method used, if applicable.
See Logic Pro Alchemy Import browser. Logic Pro Alchemy additive element controls Source components are shown only in advanced view. The parameters in this section are shown when the Additive button is active in a source subpage. The additive element controls also include a number of additive effects. See Logic Pro Alchemy additive element effects. Use Complex to choose a resynthesis waveform from the Shape pop-up menu. Sine mode results in the most accurate resynthesis of the original sample and makes it easy to work with the additive effects and formant controls.
In Complex mode, choosing any non-sine waveform can have a dramatic and often unusual effect on the overall timbre of the sound. Note: The additive effects and formant controls are named on the assumption that each partial is a sine wave. To simplify working with these controls, it is recommended that the Sine option is chosen in most cases. When multiple elements are used in a source, use this control to set the relative level of the additive component. The number of oscillators required depends on the sound.
For example, a flute has a limited number of harmonics and requires fewer partials than a cello or a violin. The playable register can also affect the number of oscillators required: high notes will accommodate only a small number of higher harmonics before reaching the limits of audibility, while low bass notes may have hundreds of harmonics without reaching the limit.
Alchemy automatically sets a suitable Num Partials value when re-synthesizing additive data from imported audio files. You can reduce this value in some cases, but removing higher partials can cause unwanted changes to certain sounds. Note: The additive engine processes partials in groups of four. Set the Num Partials parameter to a multiple of four to achieve the best compromise between CPU load and sound quality. Always set Num Partials to the lowest number of partials that are required by the sound because this helps to reduce CPU load.
The sonic impact of this parameter is highly dependent on the audio material: sounds with strong inharmonic content such as bells are dramatically changed by reducing pitch variations. If all partials are tuned to the harmonic series, however, the knob has no influence on the sound. The pitch variation knob is most useful when dealing with resynthesized audio.
For example, if you analyze a vocal sample recorded with vibrato, this knob lets you reduce the vibrato depth, or remove it entirely with a setting of zero. Removing all pitch variations from a vocal can result in a synthetic, artificial character. The audible effect is similar when the knob is turned in either direction. Symmetry alters waveforms until they are no longer pure sine waves in shape, resulting in each partial developing independent harmonics and making the sound brighter.
Click a source select field, then choose Import Audio from the pop-up menu. In the Import browser, click the Additive button to change the analysis mode. Single notes with a strong harmonic character tend to work well. A good source of such files is the Vocals subfolder of the Factory samples. Play the file up and down the keyboard and reduce the value of the Num Partials knob to remove upper harmonics.
Note that when playing higher notes you will need to turn this knob down further before you hear it start to take effect. Reduce the PVar knob value to remove all pitch variations and force all partials to a perfect harmonic series. A PVar value of zero completely removes any vibrato or pitch changes in the source sound.
Adjust the Sym knob value to change the symmetry of source sine waves, and note the extra brightness that is introduced by new harmonics that are generated. Logic Pro Alchemy additive element effects Source components are shown only in advanced view.
Three effects units are available in the lower half of the additive parameters shown in the source subpage. See Logic Pro Alchemy additive element controls. These are not audio effects in the traditional sense; rather they provide ways to control entire groups of partials simultaneously. Each unit provides a different selection of effects. Unit 1 is devoted to amplitude effects, unit 2 to pitch effects, and unit 3 to panning effects.
The initial configuration of the effects units changes when you use the default preset and when you import a sample. A sound with an imported sample loads the Harmonic module into unit 1. Note: Some effects are included only for compatibility with older Alchemy versions.
It is best to turn down your amplifier or mixer levels to avoid signal spikes that may damage your speakers or hearing. Your choice determines the controls that are shown. Choosing None results in an equal volume level for all partials, which can sound extremely bright and harsh.
Set to zero to completely remove the fundamental tone. Higher values tend to make the sound thicker. Set to zero to completely remove harmonics 1, 2, 4, 8, 16, and so on, while boosting the levels of non-octave harmonics.
Set to higher values to boost the levels of octave harmonics while reducing the levels of other harmonics. Low values increase the level of odd-numbered partials 1, 3, 5, 7, and so on , making the sound more hollow. Higher values boost harmonics 1, 3, 9, 27, and so on, with a corresponding reduction in the levels of other harmonics. Low values have the reverse effect and can make the sound more cutting and edgy.
Loaded by default when no sample has been imported into the additive element of the source. Turn toward Pulse to mute even harmonics. Low values increase the level of odd-numbered partials 1, 3, 5, 7, and so on, making the sound more hollow. The group of partials affected is defined by the Partial knob. The range of this knob is determined by the position of the Tuned knob. Set to 2 to limit the detuning affect to only partials 2, 4, 6, 8, and so on; set to 3 to apply detuning to only partials 3, 6, 9, 12, and so on.
Higher values affect fewer partials, which in turn changes the impact of the Amount knob, making it more subtle. This knob behaves like a switch. On: At the zero position, the selected partial is tuned down to the pitch of the second partial below. Off: At the zero position, the selected partial is tuned down to the pitch of the second partial below.
This stretching of the harmonic series is typical of instruments that use vibrating strings or tines. Higher values increase the intervals between partials and shift higher partials upward. Lower values decrease the intervals between partials and shift higher partials downward. This is a quick way to tune upper partials without the need to manually edit partial pitch values in the additive editor. See Logic Pro Alchemy additive edit window overview.
Small increases can result in a subtle sweetening of the sound without altering its basic character. Larger increases can add an inharmonic, metallic, or bell-like quality to upper partials. Modulate this parameter with an envelope to add a plucked string type articulation to the start of a note. In a sound with a fundamental frequency of Hz and a second harmonic an octave higher at Hz, an upward frequency shift of Hz results in partials at Hz and Hz, with the second partial no longer an octave higher than the first.
This effect type can radically alter the harmonic structure, leading to inharmonicities and atonalities, in addition to a perceived change of the fundamental pitch. All subsequent partials are shifted by the same amount in hertz rather than in semitones because this would result in a simple pitch change.
Defining the frequency shift in this way retains a consistent timbre as you play up and down the keyboard. Because the shift does not track the keyboard, the timbre of the resulting sound changes from note to note.
When combined with another harmonic sound in the VA section, for example , small shifts can create beating, chorus-type effects between the source elements.
This can lead to unusual effects and can result in dramatic transformations of the sound. Subtle use of this parameter can turn harmonic pitched sounds into atonal bell-like timbres, for example. All partials are shifted toward the target when the Amount knob is rotated. At low positions, partial pitches shift down. Adjust the knob to balance the shift and to control the brightness of any atonal, inharmonic elements that may be introduced to the sound.
This can lead to chaotic effects. Partial knob: Set the amount that noise affects low or high partials by specifying the minimum partial that is altered by the effect. This can create a subtle widening of the sound, with less obvious left to right movement in the lowest partials. At zero, all partials are in- phase. Higher values lead to a finer, more subtle and complex effect. The range is from 16 Hz to 20 kHz. Crossover frequencies are centered at Hz, Hz, and Hz, with a one octave transition between bands.
Partials are panned left to right in a regular pattern, with every second partial position inverted. Modulating the partial number can create rhythmic melodic effects. Shorter release times and slower modulation of the partial number can create strumming of partials. For example, a value of 10 applies a sine wave shape over partials , , , and so on. Modulate this with a ramp up LFO for smooth sound changes.
Logic Pro Alchemy spectral element controls Source components are shown only in advanced view. The parameters in this section are shown when the Spectral button is active in a source subpage. Two Logic Pro Alchemy spectral element effects units are available in the lower half of the spectral parameters shown in the source subpage. Note: You must first perform a sample import with a spectral analysis or draw in the Spectral edit window before you can use any of the spectral engine parameters.
In Alchemy, the audible spectrum of a signal is split into a large number of spectral bins. Energy distribution across these bins is analyzed and the sound is recreated by filling each spectral bin with the required amount of signal, using either sine waves or filtered noise.
The results are then summed. Use Noise to fill spectral bins with filtered noise. The spectral bins are filled with sine waves, which is generally the best choice to recreate the entire original signal. Noise mode can be useful for transforming normal speech into whispering, for example. The spectral engine is used only to recreate noisier aspects of the sound because this is not a strength of additive resynthesis. In this case, the mode is set to Noise.
When multiple elements are used in a source, use this control to set the relative level of the spectral component. All signals above this frequency are allowed to pass. Signals below the frequency are cut. All signals below this frequency are allowed to pass. Signals above the frequency are cut. The Low Cut and High Cut parameters work in conjunction with each other to act as a bandpass filter, where signals that fall within the two cutoff ranges are allowed to pass.
Alchemy spectral element effects provide a number of creative options in the spectral synthesis engine. Two effects units are available in the lower half of the spectral parameters shown in the source subpage. See Logic Pro Alchemy spectral element controls for information on other spectral element parameters.
See spectral effect descriptions below. This parameter is common to all spectral effect types. The parameter name and function vary with each effect type. Note that this effect requires a small amount of calculation time to collect and release a group of frequencies.
As a result, there may be a gap between playing a note and hearing the effect. Tip: Try single note samples with a strong initial attack, such as a piano, and set Mix to a value that introduces the effect as part of the tail of the sound. When centered 0 , the original frequency balance is used. Blur Blur produces a frequency blurring effect. Tip: Try a melodic loop with pitch variations to best hear the impact of this effect. For example, when used on a loop, higher settings produce a simplified sound with more frequent gaps in the effect output.
Cloud Cloud produces what might best be described as a cloud of frequency grains, resulting in a textured chorus effect. Depending on your settings and source material, this can either produce a choppy sound or a smoother one. Tip: Drum loops are an ideal starting point when learning uses for this effect.
This reduces detail and enhances prominent frequencies in the source. Glide Glide creates adjustable, repeating upward filter sweeps that are based on the source content. Note that this effect requires a small amount of calculation time before the results of your adjustments are heard. Tip: Sources with wide frequency ranges produce a more pronounced filter sweep sound, whereas sources with limited frequencies can result in unique melodic drones as narrow filters sweep across their ranges.
Freq Shift Freq uency Shift moves the spectral bins higher or lower in the spectrum, reducing the level of or entirely dropping some frequencies while emphasizing others. This is a powerful sound design tool that can dramatically alter the sound and can introduce inharmonic overtones. Tip: Try single note samples with a strong initial attack, such as a piano, and blend the mix level so that the effect comes in as part of the tail of the sound.
Start with small adjustments because this parameter has a wide range. Higher frequencies are attenuated. Tip: This effect is highly dependent on the available frequency range in the imported sample. For example, the Alpha and Beta knobs are useful across the entire range with drum loops, whereas the most useful Alpha and Beta ranges are small positive or negative deviations from the center position when used with spoken vocals.
A setting of 1 centered is closest to the source sound. Metallize Metallize produces classic comb filter style effects. Tip: Experiment with drum loops to clearly hear the impact of the controls.
Higher settings emphasize harmonics, creating metallic resonances. Shimmer Shimmer sweeps the frequencies to varying degrees and speeds, imparting either slow frequency shifts or fast shimmering sounds. Tip: Try pure organ samples to clearly see the results of the effect in the real-time spectrogram display, particularly at moderate rate settings. Tip: Import a bell sample, and start with very low settings to see and hear the impact of controls on the sound. Note that bins are numbered and selected sequentially.
Smear Smear averages between blocks of frequencies to create a smoother, more consistent sound. It delivers different results to the Blur effect. Tip: Try melodic loops that have pitch variations to showcase this effect. Higher settings have less sonic variation, so make small changes. This natural variation in the sound provides a more organic cloud-like effect.
Disable to lock the phases of the source, resulting in a tight, metallic sound. Logic Pro Alchemy pitch correction controls Source components are shown only in advanced view. The parameters in this section are shown when the Pitch button is active in an additive or spectral source subpage. Higher values result in stronger correction. This is shown as a percentage. Logic Pro Alchemy formant filter controls Source components are shown only in advanced view.
The parameters in this section are shown when the Formant button is active in an additive or spectral source subpage. When audio is imported into the additive or spectral engines with the Formant option enabled, the signal is analyzed and resonances in the original signal are extracted and converted into a formant filter shape.
The formant filter scales the amplitude of additive partials or spectral bins over time to recreate the characteristic resonances of the instrument, rather than processing the audio signal like a conventional filter. This more detailed analysis attempts to determine the resonant frequencies of the source audio data.
Higher values can make sounds seem brighter or thinner. Lower values can create a darker, thicker character. Set to lower values to reduce key tracking which may make some sounds playable over a wider keyboard range. The Size knob works in conjunction with the Center parameter.
Resonances below the center frequency are shifted upward as the Size knob value is increased. A corresponding downward shift occurs to resonances above the center frequency. High values smooth and slow down formant changes. Low values exaggerate and speed up changes. Formant filter synthesized parameters The synthesized controls work with any additive or spectral material and do not require the formants to be analyzed on import. Use these parameters to impose new resonant characteristics on the original signal.
Size works in conjunction with the Center knob. Resonances below the center frequency are shifted upward as the Size value is increased. The displayed value indicates position. Whole numbers indicate a particular filter unit, and fractional values indicate a position between filters. Assign this type to use one of the four filter units as a bypass. Adjust the Select knob to quickly disable synthesized formant processing.
The Size knob can be used to stretch the pattern of cuts and boosts up or down the frequency spectrum, or both, depending on the setting of the Center knob. The negative filter name is used because it recreates the effect of a phase- inverted delayed signal that boosts only odd harmonics, resulting in a hollow sound.
This filter has a brighter sound than the negative comb filter. Experiment with each comb to determine the best choice for your sound. The parallel filters are multipole designs. Signals above or below the set center frequency are attenuated. The Shift knob sets the cutoff frequency. The Size knob changes the filter slope. The frequency band can be moved up or down the frequency spectrum with the Shift knob.
The Size knob sets the width of the band notch. The Size knob sets the width of the band. Classic vowel sounds are warmer, and are similar to synthesizer vowel sound filtering. Smooth variants are more natural-sounding vowel shapes with a gentler filter slope. Each Bright, Classic, and Smooth vowel filter is more of a unique variation on that general sound, with not only brightness differences, but also overall character differences.
Additionally, any vowel filter can be independently modulated, alone or in conjunction with Select knob morphing between filters even from mismatched sets. Use these facilities to dramatically expand your filtering options. Each variation of this complex filter shape has prominent peaks at different frequencies. It is, generally speaking, an open-sounding filter. This filter shape has gentler midrange and upper midrange peaks with a dominant low-mid resonance.
The result is a rounder sound with less brightness and presence than the vowel types above. This filter shape has gentler midrange and upper midrange peaks with a prominent low-mid resonance. Modify formants in a resynthesized additive guitar sound 1. Select source A, then click the source select field and choose Import Audio from the pop-up menu.
Navigate to the Guitars subfolder in the Factory samples folder, and choose a single guitar sample. When loading is complete, click the Formant button to the right side of the source A window. Note that the upper Analyzed section is turned on. Adjust the Shift knob to move resonances up or down in frequency and to change the timbre.
Small amounts of Shift variation work well for subtle changes: try a few semitones in either direction. Play some very low notes, then some very high notes. Gradually turn down the KTrack knob to reduce key tracking for the formant filter, and note the difference when you replay the high and low notes.
Adjust the Size knob value to change the apparent size of the guitar body. Also adjust the Center knob value, and note the effect it has on the tone of the resulting larger or smaller guitar body. Modify formants in a resynthesized spectral drum loop 1. Navigate to the Loops subfolder in the Factory samples folder, and choose a drum loop.
When loading is complete, click the Formant button to the right of the source A window. Adjust the Size knob value to make the drums seem bigger or smaller. Adjust the Smooth knob value to alter the rate of change for the formant filter. Higher values smear the timbre of one drum into the next. Lower values exaggerate changes and create an unusual distortion near the bottom of the knob range.
Create a talking additive sound with synthesized vowel formants 1. Select source A, then turn off the oscillator in the VA section to the right. Click the Additive button, and turn on the additive section. You will hear an additive sawtooth sound if you play some notes. As an option, increase the Num Partials value. This helps to prevent the sound becoming dull if played in lower registers.
Click the Formant button, and turn on the lower Synthesized section. Increase the Select knob value, and play a few notes. Adjust the Shift knob, the Size knob, and the Center knob, to explore the different timbres available.
Switch the order of vowels in the four pop-up menus, and also load different filter types such as Comb. The parameters in this section are shown when the Granular button is active in a source subpage. The Granular section is available only when you import an audio sample using either granular or sampler mode. Note: The sampler and granular engines are mutually exclusive: you can use one or the other within a single source, but not both together. You can, however, enable further sources if both engines are required simultaneously.
Granular synthesis represents continuous sound as a stream of grains, or tiny pieces of sound. Alchemy generates grains by extracting 2- to millisecond pieces from an audio file. The amplitude of each grain is shaped, along with any pitch and pan modifications, before the grain is sent to the output stream. Grains can be reordered, time stretched, and pitch shifted. This provides an inexhaustible supply of potential raw material to use as the basis of your sounds. Granular element parameters In addition to the following controls, granular playback is affected by loop modes and by the settings and modulations of the Position and Speed knobs in each source subpage.
Modulations of the granular element update with each new grain. For an example of the impact this has, modulating the source Coarse Tune parameter with an LFO causes the stream of grains to rise and fall in pitch, but does not create pitch sweeps within each grain. If a large Size value is used in conjunction with a low Density value, modulations of source parameters such as pitch may sound stepped, rather than smooth. The Size and Density parameters interact with each other.
When the Density value is 1, a single grain is sent to the output stream. As soon as one grain finishes, the next one is sent. A Size value of msec sends a new grain every msec.
Increasing Density to 2 adds a second grain that is sent in between those of the first, resulting in a new grain every 50 msec, assuming a Size value of msec. The first and second grains overlap each other. Higher Density values inject additional new grains into the output stream.
These new grains occur more frequently and overlap more heavily. Setting Size to around msec and Density to around 5 grains is often suitable for smooth pad sounds with no sharp transients. Setting Size between 40 and 80 msec and Density to around 2 grains is useful for drums and other sounds featuring sharp transients. Small Size values tend to produce a buzz that masks the original pitch of the sample. Large Size values tend to break up the sound.
You can counteract both tendencies by increasing the Density. Note: Also important to the Size and Density parameters is the shape chosen in the Grain Shape pop-up menu. This can have a significant or subtle impact on sonic artifacts that may be introduced in the stream of grains. The source Stereo button must be on for RPan to have an effect. Taps retrigger the attack phase of the source. Note: Taps that fall within a looped area are retriggered on each loop cycle. Values are shown as a percentage of the overall sound duration.
Set to zero to trigger taps in quick succession at the sound end point. The source Stereo button must be on for Stereo Offset to have an effect. At a basic level, this applies a small fade-in and fade-out to each grain, but some shapes may have a more significant impact, depending on the current Size and Density values and the source material.
You can also step through the available grain shapes with the Previous and Next buttons the arrows. This function is primarily intended to reduce or remove glitches, clicks, and crackles in the playback of a stream of grains, but it can introduce buzzy gaps between grains and can affect the tonality of grains. There are no fixed rules when it comes to the choice of grain shape, given the infinite variety of source audio material. Therefore, you may want to experiment to achieve the required results.
The parameters in this section are shown when the Sampler button is active in a source subpage. The sampler section is available only when you import an audio sample using either granular or sampler mode. The sampler section allows audio files, known as samples, to be played directly. Samples played at a higher pitch than the original play back at a faster speed. Samples played at a lower pitch than the original play back at a slower speed.
The sample waveform is displayed in the center. A progress bar indicates the current playback position for the most recently triggered note. When multiple elements are used in a source, use this control to set the relative level of the sampled component. The parameters in this section are shown when the VA Virtual Analog button is active in a source subpage. When you click the Name bar File button, and choose Initialize Preset from the pop-up menu to initialize Alchemy to default settings, the VA element is automatically enabled.
Basic saw, sine, square, and triangle and many specialized waveforms are provided. You can also step through the available waveforms with the Previous and Next buttons the arrows. When multiple elements are used in a source, use this control to set the relative level of the oscillator component. When a square wave is active, Symmetry acts as a pulsewidth control.
These have different spectral characteristics that can be further refined with filters. You can step through the available waveforms with the Previous and Next buttons the arrows. When multiple elements are used in a source, use this control to set the relative level of the noise component. All frequencies above this value are allowed to pass.
All frequencies below are attenuated. All frequencies below this value are allowed to pass. All frequencies above are attenuated. The Low Cut and High Cut parameters work in conjunction with each other to act as a bandpass filter, where the noise signal that falls within the two cutoff ranges is allowed to pass.
Logic Pro Alchemy source modulations Source components are shown only in advanced view. Parameters that have a modulation assignment are indicated by an orange arc around the control. Note: Parameters that are morphed and have a modulation assignment show both an orange and green arc around the control. This section focuses on Position, which is a modulation target.
The principles discussed apply equally to other source parameter targets. Position determines the playback position of audio data. When modulated, the playback path through the audio data is controlled by the selected modulation source. In sampler mode, the note-on modulation value determines the initial offset for the play position within the audio data. Beginning at that position, the rest of the sound plays in a normal manner, although looped as if the Loop mode is set to All.
In additive, spectral, or granular mode, Position can be continuously modulated forward or backward at any rate including zero. Create tempo-synced loops by modulating position Synchronized playback of looped audio with the Logic Pro X tempo is easy to achieve by modulating the Position parameter.
This technique is possible with any synthesis method that permits continuous modulation of Position. This example uses the granular engine, but the same technique can be applied to the additive and spectral engines. When Position is modulated and Speed has a value greater than zero, the playback path is determined by a combination of modulation value whenever this value changes and the normal path at a rate determined by Speed whenever the modulation value is static.
In source A, import a rhythmic or melodic sample that loops evenly. You will hear that playback is frozen at the very beginning of the sample. Do one of the following:.
Note the orange arc that appears around the Position knob. This indicates that the parameter has a modulation assignment. Also in the LFO, turn off the Bipolar button. This routing increases Position smoothly so that the entire sample plays back from beginning to end, then jumps immediately back to the beginning and continues to loop. Finally, adjust the Logic Pro X tempo as you play additional notes to confirm that the loop is properly synchronized.
Logic Pro Alchemy morph controls Source components are shown only in advanced view. Click the Advanced button to switch to advanced view, then click the Morph button to view and use the morph controls. The morph controls determine how the four Alchemy sources interact. There are two basic types of interaction. This is equivalent to turning the Amp knobs in each source to attain the desired mix.
If you crossfade from a source with a high Coarse Tune setting to a source with a low Coarse Tune setting, the high source fades out as the low source fades in.
In the middle of the crossfade you hear both sources. If you morph from a source with a high Coarse Tune setting to a source with a low Coarse Tune setting, you hear a single sound during the morph. The sound tuning falls smoothly from the high value to the low one.
Morphing provides more scope than simple crossfades between sources. It also allows cross-synthesis, where you can combine different aspects of different sound elements. For example, you could apply the formants or other characteristics of an additive source to the spectral element of another source.
See the tutorials found in Logic Pro Alchemy elemental morphs overview. Morphed parameters are indicated by a green arc around the control. Parameters that are morphed and have a modulation assignment show both an orange and green arc around the control.
Parameter settings are shared across all morphed sources, which means that changing a parameter in one source results in the corresponding parameter being changed for all morphed sources. Note: Parameters that do not directly participate in the morph, including most buttons and pop-up menus, are indicated with a lock icon displayed at the top left of the control the lock icons are shown only in source subpages.
Where there is a parameter pairing of an On button and a pop-up menu, only the button shows the lock icon. Neither parameter participates in the morph. Regions of each source encompassed by corresponding warp markers are time- aligned in the morph. See Logic Pro Alchemy zone waveform editor. Also controls VA morph position if the VA element is active. Also controls sampler morph position if the sampler element is active. Also controls morphing of the source filter knobs. These controls morph the timing of the sound.
In cases where source A has a short attack and source B has a long attack, for example, the length of the attack varies as you change the X knob.
Turn off for the best morphing quality. As an example, Auto Align corrects the timing of words of four spoken voice samples saying the same phrase in each of the four morphed sources. Auto Align is automatically turned off when you set warp markers manually. Regions of each source encompassed by corresponding warp markers are time-aligned in the morph. See Logic Pro Alchemy elemental morphs overview. Use Elements to view and edit the X values of five parameters.
Morphs affect only the chosen group of sources. This also controls the sampler morphing position if the sampler element is active. Turn off for higher morphing quality. For example, Auto Align corrects the timing of words of four spoken voice samples saying the same phrase in each of the four morphed sources. Drag the point to change the X or Y value, or both. Drag each point to change the corresponding X and Y values. Edit buttons are shown on all source subpages.
Shown at the top right of all edit windows. By default, the Main edit window is shown. Further Additive and Spectral edit windows can be opened by clicking the buttons at the top of the window. These windows provide additional parameters that let you precisely edit and sculpt your sounds.
The Main edit window is divided into three areas that interact with each other. You can edit parameters graphically in the keymap or waveform editor or can use corresponding fields and other parameters in the inspector. This area interacts with the zone parameters in the inspector. See Logic Pro Alchemy keymap editor. The source edit window is opened by clicking the Edit button on any source subpage.
Click the close window icon X to close the source edit window. The source inspector is divided into three main parameter groupings: global and source parameters, group parameters, and zone parameters. Click the X icon at the top right of the active window to close it. You can also click the Previous and Next buttons the arrows to step through available waveform data. All other sources are muted. Logic Pro Alchemy inspector group controls Source components are shown only in advanced view.
Click the close window icon X at the top right to close the window. The new group name is added below existing group names in the Group list shown under the Group pop-up menu. A sequentially assigned number is appended to the new group name.
Control-click a group name to choose the Delete command. Attack triggers group zones when note-on messages are received. Release triggers group zones for note-off messages. This mode is useful for instruments with a distinctive end-of-note sound such as a key click or hammer thump. Such sounds can be included as a separate group with release triggering enabled.
You can also use this feature creatively to add abstract reverb tails or to create a pad sound that changes dramatically during the release stage. Apply a fade in time value ranging from 0 to for note-off events when the release trigger mode is active.
This parameter is primarily intended for use in conjunction with release triggering, to create a crossfade between the main body of the note and the release sample. A common use of this feature is to create a group containing an open and a closed hi-hat sample. If you set group polyphony to one, either the closed or open hi-hat sample can play, but not both at the same time. You can create and combine multiple rules using boolean logic. The normal state for any newly created group is a single rule set to Always, unless that group was defined as a Round Robin group in the Import browser Dropzone.
Click the field to choose a rule. This adds a new rule pop-up menu below the first and displays a Logic pop-up menu to the right. Choose Delete from the pop-up menu to remove a rule. At least two round robin groups must exist for this parameter to have an effect. Each group is assigned a different value.
If two groups are assigned to the same value, both zones are triggered simultaneously. Each note-on sequentially triggers a round robin group from lowest to highest. Once the highest group number is triggered, the sequence starts again from the lowest group number. Drag vertically in the field or use the arrows to set a value.
No Order pop-up menu is shown. You can also assign keyswitching to a MIDI note or range of notes which prevents the group from triggering until one of these notes is played. Two pop-up menus are displayed, where you can choose keyswitch conditions. For example, set the first field to Snap2 and the second field to Snap4, which results in the group being triggered only when the Transform pad is at position 2, 3, or 4.
Assign multiple groups to different ranges to switch between up to eight different groups. See Logic Pro Alchemy Transform pad. This knob is visible only when the Keysw options are chosen in these menus. For example, set the first field to Keysw1 and the second to Keysw5, which results in the group being triggered when the Keyswitch knob is set to a value that falls within this range.
Up to 10 Keyswitch knob positions are available, enabling you to switch between groups with the knob. Because the Keyswitch knob is available as a modulation target, this lets you create complex automated group switches.
Play a note in this range to switch to a group, which remains active until another group is chosen. Three pop-up menus are displayed where you can choose the controller type and set controller values. You can set other groups with the same type but with a different control range to switch between groups with a single controller. Because the Control 1 knob used in the example is available as a modulation target, this lets you create complex automated group switches.
Drag vertically in the fields or use the arrows to set a value. The different logic conditions result in different outcomes. Crossfade to a different group of samples on note-off 1. Select a sample or multiple samples representing the main sustain portion of your sound, and import using any of the available import modes. Alchemy analyzes each sample to determine the root pitch if not defined in the filename , set the root key, key range, and velocity range for each sample zone such that they span the entire keyboard and the entire dynamic range, and add all zones to a group named Group 1.
Select a sample or multiple samples representing the release portion of your sound, and import the import mode is automatically set to match the existing group. Alchemy again analyzes each sample and adds all zones to a group named Group 2. Double-click Group 2, then click the Trig field and change it to Release. Zones in Group 2 will now trigger when you release each key, playing over zones in Group 1 which continue to sound until AHDSR1 reaches the end of the release stage.
Double-click Group 1 in the list, then click the Fade field and change it to a value other than 0. Zones in Group 1 will now fade out when the note is released, allowing Group 2 to be heard during the release stage of the sound. Higher Fade values result in slower fades. As an option, double-click Group 2 in the list, then click the Fade field and change it to a value other than 0. Zones in Group 2 will now fade in when the note is released, creating a crossfade between Groups 1 and 2 at note-off.
Higher values result in slower fades. Fading in the release group may be unnecessary if your release samples already have a natural fade in at the start, however, or undesirable if a percussive transient is required at note-off.
Try to set Fade values for Group 1 and 2 to an identical small value to create a sudden but click-free crossfade at the end of each note. Create random round robin variations 1. Select a sample or multiple samples representing the second of your round robin variations, and import the import mode is automatically set to match the existing group. Alchemy will again analyze each sample and add all zones to a group named Group 2.
Any notes you play will now randomly trigger either group 1 or group 2, but not both together. Note that if you play a chord, each individual note is randomly assigned to one of those groups. Repeat steps 7 to 9 as needed to configure a group for each further variation you require. Assign the source Keysw knob to switch between groups 1. Click the Rule field, and change it from Always to Keyswitch.
Click the first range field, and change it to Keysw1. The second range field also changes to the same value. As an option, you can click the second range field and increase the value to specify a range of values that will trigger this group, instead of just a single value. Select a sample or multiple samples representing the second of your variations, and import the import mode is automatically set to match the existing group.
Click the Rule field below Group 2, and change it from Always to Keyswitch. Click the first range field, and change it to the first unused Keysw value. If the second range field for Group 1 is set to Keysw3, choose Keysw4.
As an option, you can click the second range field and increase the value to specify a range of values that will trigger Group 2, instead of just a single value. Click the X symbol at the top right to close the source edit window. A new Keysw knob is visible in the source pane, to the left of the Keyscale field. Rotate the Keysw knob to switch between the groups you created. Clear editor. Upload or insert images from URL.
Click here! Logic Pro Share More sharing options Followers 0. Reply to this topic Start new topic. Recommended Posts. David Nahmani Posted July 22, Posted July 22, Link to comment Share on other sites More sharing options Jordi Torres Posted July 22, I’ll keep adding to it as I find things to add. Feel free to add to it too! Nice feature is that LPX will automatically try to assign a proper icon from sounds dragged from the loop browser, also from 3rd party content, probably by analyzing the naming of a file.
Where can the discount coupons used toward the purchase of LPX be found? Logic Pro X Cool, I’ve added the ones you’ve contributed guys, thanks. Keep ’em coming! Posted July 23, Another thing that keeps being asked. The default shortcut for the tool menu is “T. Mac OS X David Nahmani Posted July 23, Thanks teed, adding those too. Great idea, David. Bass Amp Designer and new stompboxes in Pedalboard. Round robin sample support in EXS Alternatives integrated versioning.
Autosave in the background. Here’s my tip for the day.. With Drummer, you can have separate outs. Initiate drummer track 2. Jordi Torres Posted July 23, Chris D Posted July 23, Posted July 23, edited. Edited July 23, by Chris D. Eriksimon Posted July 23, Cool, animated gif! How did you create that? Please say it’s a freeware app Why did the chicken cross the Mobius ring?
The ability to create your own color palette is gone in Logic X. Ploki Posted July 23, FoolsGold Posted July 23, Is there a way to do Bounce to iTunes Library? Macbook Pro 2. I sorta liked it, saved having to go folder diving through the finder to find your bounce. Eric Cardenas Posted July 23, Nola Posted July 23, Sorry but back to the letter T for the tools list – I specifically said “Yes” the first time I started LPX when it asked me if I wanted to use the new set of key commands designed for LPX – but Esc is still what brings up the tool list for me, and T does nothing??