Logic Pro X Effects For OS X
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Contents 10 10 10 10 13 18 21 23 24 26 27 27 28 29 30 32 33 34 37 38 38 39 40 41 42 44 45 46 50 51 51 52 Chapter 1: Amps and pedals Amps and pedals overview Amp Designer Amp Designer overview Amp Designer models Build a custom Amp Designer combo Amp Designer equalizer Amp Designer amplifier controls Amp Designer effects Amp Designer microphone parameters Bass Amp Designer Bass Amp Designer overview Bass Amp Designer models Build a custom Bass Amp Designer combo Bass Amp Designer signal flow Use the D.I.
53 53 54 54 55 56 59 61 63 68 69 70 71 72 72 73 74 Chapter 2: Delay effects 76 76 77 78 79 79 80 81 Chapter 3: Distortion effects 82 82 83 84 84 86 87 89 90 92 93 94 94 94 95 96 97 98 98 99 Chapter 4: Dynamics processors Delay effects overview Delay Designer Delay Designer overview Delay Designer main display Use the Delay Designer Tap display Create taps in Delay Designer Select, move, and delete taps Edit parameters in the Tap display Delay Designer Tap parameter bar Delay Designer sync mode Dela
100 Surround Compressor 100 Surround Compressor overview 101 Surround Compressor Link parameters 102 Surround Compressor Main parameters 103 Surround Compressor LFE parameters 104 104 104 104 105 106 107 107 107 108 109 110 110 110 111 113 115 116 Chapter 5: Equalizers 117 117 117 117 118 118 119 120 121 121 122 122 123 124 125 126 126 126 127 128 128 129 131 131 133 Chapter 6: Filter effects Equalizers overview Channel EQ Channel EQ overview Channel EQ parameters Channel EQ use tips Channel EQ Analyz
134 EVOC 20 TrackOscillator modulation 135 EVOC 20 TrackOscillator output parameters 136 Fuzz-Wah 136 Fuzz-Wah overview 136 Auto Wah parameters 138 Fuzz-Wah Compressor parameters 138 Fuzz parameters 139 Spectral Gate 139 Spectral Gate overview 140 Use Spectral Gate 141 141 141 142 142 143 145 Chapter 7: Imaging processors 146 146 146 147 147 148 148 149 150 151 151 152 153 153 153 154 155 156 157 158 Chapter 8: Metering tools Imaging processors overview Binaural Post-Processing Direction Mixer Directi
159 159 160 160 161 162 167 170 171 172 173 174 174 175 178 179 179 179 181 183 184 185 185 186 187 187 190 193 195 196 196 197 198 198 Chapter 9: MIDI plug-ins 199 199 200 201 202 202 203 205 206 206 206 207 208 209 209 211 Chapter 10: Modulation effects Use MIDI plug-ins Arpeggiator MIDI plug-in Arpeggiator overview Arpeggiator control parameters Arpeggiator note order parameters Arpeggiator pattern parameters Arpeggiator options parameters Arpeggiator keyboard parameters Use Arpeggiator keyboard pa
212 212 213 214 215 216 217 218 Rotor Cabinet effect Rotor Cabinet effect overview Rotor Cabinet effect motor parameters Rotor Cabinet effect microphone types Rotor Cabinet effect mic processing controls Scanner Vibrato effect Spreader Tremolo effect 219 219 219 219 220 221 222 223 224 224 225 226 226 226 227 Chapter 11: Pitch effects 229 229 230 230 231 232 233 233 234 235 236 237 Chapter 12: Reverb effects 238 238 239 240 243 243 244 245 246 247 248 250 250 251 Chapter 13: Space Designer convolu
252 Space Designer global parameters 252 Space Designer global parameters overview 253 Use Space Designer global parameters 256 Use Space Designer output parameters 259 259 259 259 260 261 262 263 264 264 264 265 Chapter 14: Specialized effects and utilities 266 266 266 267 268 269 270 Chapter 15: Utilities and tools 271 271 271 272 273 273 274 275 276 277 277 278 279 280 280 281 281 282 282 283 284 285 286 Appendix: Legacy effects Specialized effects overview Denoiser Denoiser overview Denoiser sm
Amps and pedals 1 Amps and pedals overview Logic Pro X features an extensive collection of guitar and bass amplifiers and classic pedal effects. You can play live—or process recorded audio and software instrument parts—through these amps and effects. The amplifier models recreate vintage and modern tube and solid-state amps. Built-in effect units, such as reverb, tremolo, or vibrato, are also reproduced. The modeled amplifiers can be paired with a number of emulated speaker cabinets.
The Amp Designer interface is divided into four main parameter sections. Effects parameters Amp parameters Amp parameters Output slider Model parameters Microphone parameters •• Model parameters: The Model pop-up menu in the black bar at the bottom is used to choose a preconfigured model, consisting of an amplifier, a cabinet, an EQ type, and a microphone type. The other pop-up menus in the black bar enable you to independently choose the type of amplifier, cabinet, and microphone.
Switch between the full and small versions of the interface mm Click the disclosure triangle between the Cabinet and Mic pop‑up menus in the full interface to switch to the smaller version. In the small interface you can access all parameters except microphone selection and positioning. mm To switch back to the full interface, click the disclosure triangle beside the Output field in the small interface. Click here in full interface. Click here in small interface.
Amp Designer models Tweed Combos The Tweed models are based on American combos from the 1950s and early 1960s that helped define the sounds of blues, rock, and country music. They have warm, complex, clean sounds that progress smoothly through gentle distortion to raucous overdrive as you increase the gain. Even after half a century, Tweeds can still sound contemporary. Many modern boutique amplifiers are based on Tweed-style circuitry.
Tip: Although these amps tend toward a clean and tight sound, you can use a Pedalboard distortion stompbox to attain hard-edged crunch sounds with sharp treble and extended lowend definition. See Pedalboard distortion pedals on page 45. British Stacks The British Stack models are based on the 50- and 100-watt amplifier heads that have largely defined the sound of heavy rock, especially when paired with 4 x 12" cabinets. At medium gain settings, these amps are suitable for thick chords and riffs.
British Alternatives The late 1960s amplifier heads and combos that inspired the Sunshine models are loud and aggressive, with full mid frequencies. These amps are useful for single note solos, power chords, and big, open chords—making them popular with the “Brit-pop” bands of the 1990s. The Stadium amps are famed for their ability to play at extremely high levels without dissolving into an indistinct distortion. They retain crisp treble and superb note definition, even at maximum gain settings.
Additional Combos The combos and utility models in this category are versatile amps that you can use for a wide variety of musical styles. Model Description Studio Combo A 1 x 12" combo based on boutique combos of the 1980s and 1990s. These models use multiple gain stages to generate smooth, sustain-heavy distortion and bold, bright, clean sounds. Can deliver a heavier sound when paired with a 4 x 12" cabinet.
Cabinet Description Vintage British 4 x 12 This late 1960s closed-back cabinet is synonymous with classic rock. The tone is big and thick yet also bright and lively, due to the complex phase cancelations between the four 30-watt speakers. Modern British 4 x 12 A closed-back 4 x 12" cabinet that is brighter and has a better low end than the Vintage British 4 x 12, with less midrange emphasis. Brown 4 x 12 A closed-back 4 x 12" cabinet with a good low end and complex midrange.
Build a custom Amp Designer combo You can use one of the default models or you can create your own hybrid of different amplifiers, cabinets, and so on. You create your own by using the Amp, Cabinet, and Mic pop-up menus, located in the black bar at the bottom of the interface, as well as the EQ pop-up menu, which you open by clicking the word EQ or Custom EQ above the knobs in the left part of the knobs section.
Choose an Amp Designer cabinet Cabinets have a huge impact on the character of a guitar sound (see Amp Designer cabinets on page 16). Whereas certain amplifier and cabinet pairings have been popular for decades, departing from them can be an effective way to create fresh-sounding tones. For example, most players automatically associate British heads with 4 x 12" cabinets. Amp Designer lets you drive a small speaker with a powerful head, or pair a tiny amp with a 4 x 12" cabinet.
Choose a microphone type and placement 1 Choose a microphone model from the Mic pop-up menu. •• Condenser models: Emulate the sound of high-end studio condenser microphones. The sound of condenser microphones is fine, transparent, and well-balanced. Choose Condenser 87 or Condenser 414. •• Dynamic models: Emulate the sound of popular dynamic cardioid microphones. Dynamic microphones sound brighter and more cutting than Condenser models.
Amp Designer equalizer Amp Designer equalizer overview Hardware amplifier tone controls vary among models and manufacturers. For example, the treble knobs on two different models may target different frequencies or provide different levels of cut or boost. Some equalizer (EQ) sections amplify the guitar signal more than others, thus affecting the way the amp distorts. Amp Designer provides multiple EQ types to mirror these variations in hardware amplifiers.
Amp Designer EQ types This table describes the properties of each Amp Designer EQ type. EQ type Description British Bright Inspired by the EQ of British combo amps of the 1960s, it is loud and aggressive, with stronger highs than the Vintage EQ. This EQ is useful if you want more treble definition without an overly clean sound. Vintage Emulates the EQ response of American Tweed-style amps and the vintage British stack amps that used a similar circuit. It is loud and subject to distortion.
Amp Designer amplifier controls The amp parameters include controls for the input gain, presence, and master output. The Gain knob is located to the left in the knobs section, the Presence and Master knobs are to the right, and the Output parameter is at the lower-right edge of the interface. Gain Presence Master Amplifier parameters •• Gain knob: Rotate to set the amount of pre-amplification applied to the input signal. This control affects specific amp models in different ways.
Amp Designer effects Amp Designer effects overview The effects parameters include reverb, tremolo, and vibrato, which emulate the processors found on many amplifiers. These controls are found in the center of the knobs section. Reverb, which is controlled by an On/Off switch in the middle, can be added to either tremolo or vibrato, or it can be used independently. See Amp Designer reverb effect on page 24.
Amp Designer reverb types This table indicates the properties of each Amp Designer reverb type. Reverb type Description Vintage Spring This bright, splashy sound has largely defined combo amp reverb since the early 1960s. Simple Spring A darker, subtler spring sound. Mellow Spring An even darker, low-fidelity spring sound. Bright Spring Has some of the brilliance of Vintage Spring, but with less surf-style splash. Dark Spring A moody-sounding spring. More restrained than Mellow Spring.
Amp Designer microphone parameters Amp Designer provides seven virtual microphone types. As with other components in the tone chain, different selections can yield very different results. After choosing a cabinet, you can set the type of microphone to emulate and can place the microphone, relative to the cabinet. The Mic pop-up menu is near the right end of the black bar. The speaker-adjustment graphic appears when you move your pointer in the area above the Mic pop-up menu.
Bass Amp Designer Bass Amp Designer overview Bass Amp Designer emulates the sound of three famous bass guitar amplifiers and the speaker cabinets used with them. Each preconfigured model combines an amp and cabinet that recreates a well-known bass guitar amplifier sound. The amp and cabinet can be combined with integrated compression and EQ units to alter the tone. You can process signals directly, reproducing the sound of your bass played through these amplification systems.
•• Output slider: The Output slider is found at the lower-right corner of the interface. It serves as the final level control for Bass Amp Designer’s output that is fed to the ensuing Insert slots in the channel strip, or directly to the channel strip output. Note: This parameter is different from the Master control, which serves the dual purpose of sound design as well as controlling the level of the Amp section.
Bass cabinet models The table below outlines the properties of each cabinet model available in Bass Amp Designer. Cabinet Description Modern Cabinet 15" 1 x 15 inch speaker, closed-back design. Very deep and full tone. Modern Cabinet 10" 1 x 10 inch speaker, closed-back design. A punchy tone. Modern Cabinet 6" 1 x 6 inch speaker, closed-back design. Classic Cabinet 8 X 10" 8 x 10 inch speakers, closed-back design. Flip Top Cabinet 1 X 15" 1 x 15 inch speaker, closed-back design.
•• Single speakers or multiple speakers: The number of speakers is less important than it may appear. Phase cancelations occur between the speakers, adding texture and interest to the tone. Choose a microphone type and placement 1 Click the Mic pop-up menu to choose a microphone model. •• Condenser 87: Emulates the sound of a high-end German studio condenser microphone. The sound of condenser microphones is fine, transparent, and well-balanced.
Pre-amp signal flow The pre-amp section is very flexible, and can be used in several ways when you use different combinations of On/Off and Pre/Post switches. The signal flow indicated in the Mode column is in series when multiple processors are used—that is, the output of one processor signal is fed into the next processor.
Use the D.I. box The D.I. box is modeled on a highly regarded American D.I. unit. D.I. box parameters •• Boost knob: Rotate to set the input gain of the D.I. box. •• HF Cut button: Click to turn on a highpass filter. This is used to reduce noise. •• Tone knob: Rotate to set the tonal color of the D.I. box. Choose from the following preset EQ curves: •• •• 1: An EQ curve with a -6 dB scoop from 100 Hz to 10 kHz, most pronounced around 800 Hz.
Bass Amp Designer amplifier controls The amp parameters include controls for channel selection, input filter and gain, and master output. The Gain knob is located to the left in the knobs section and the Master knob and Output slider are located at the far right. Bright switch Master knob Gain knob Output slider Channel I/II switch Amplifier parameters •• Channel I/II switch: Click to switch between channel I and channel II. •• •• Channel I is active, with a gain of 0 dB.
Bass Amp Designer effects Bass Amp Designer effects overview Bass Amp Designer provides multiple EQ types to sculpt your instrument tones. It provides a basic EQ that mirrors the tonal qualities of the integrated EQ of the amplifier model you choose, if applicable. All amplifier model EQs have identical controls: Bass, Mids, and Treble. See Bass Amp Designer EQ. Bass Amp Designer also offers an additional Graphic or Parametric EQ that you turn on with the EQ switch above the Master knob at the far right.
Bass Amp Designer compressor The internal compression circuit is custom-built for use with Bass Amp Designer. It features an AutoGain function that compensates for volume reductions caused by compression. Compressor parameters •• Compressor on/off switch: Click to turn the Compressor on or off. •• Fast/Easy switch: Click to switch between two compression algorithms: •• Fast: Stronger compression, with good control over levels, which makes it easier to fit the bass into an arrangement.
Bass Amp Designer Parametric EQ Bass Amp Designer offers an additional Graphic or Parametric EQ that you turn on with the EQ switch above the Master knob at the far right. The Parametric EQ provides two EQ bands: •• HiMid: Controls frequencies in the high and high-mid range. •• LoMid: Controls frequencies in the low and low-mid range. Parametric EQ parameters •• Type switch: Click the up position to choose the Graphic EQ. Click the down position to choose the Parametric EQ.
Bass Amp Designer microphone parameters Bass Amp Designer offers three virtual microphone types. As with other components in the tone chain, different selections can yield different results. After choosing a cabinet, you can choose the type of microphone to emulate and you can adjust the position of the microphone, relative to the cabinet. The Mic pop-up menu is near the right end of the black bar. The speaker-adjustment graphic appears when you move your mouse in the area above the Mic pop-up menu.
Pedalboard Pedalboard overview Pedalboard simulates the sound of a number of famous “stompbox” pedal effects. You can process any audio signal with a combination of stompboxes. You can add, remove, and reorder pedals. The signal flow runs from left to right in the Pedal area. The addition of two discrete busses, coupled with splitter and mixer units, enables you to experiment with sound design and precisely control the signal at any point in the signal chain.
Use the Pedal Browser Pedalboard offers dozens of pedal effects and utilities in the Pedal Browser on the right side of the interface. Each effect and utility is grouped into a category, such as distortion, modulation, and so on.
Use Pedalboard’s import mode Pedalboard has a feature you can use to import parameter settings for each type of pedal. In contrast to the plug-in window Settings pop-up menu, which you use to load a setting for the entire Pedalboard plug-in, this feature can be used to load a setting for a specific stompbox type. Turn import mode on or off mm Click the Import Mode button to show all pedals used in the most recent Pedalboard setting.
Replace a pedal setting in the Pedal area with an imported pedal setting 1 Click the pedal you want to replace in the Pedal area. It is highlighted with a blue outline. 2 Click the stompbox in the Pedal Browser to replace the selected pedal (or pedal setting) in the Pedal area. The blue outlines of the selected pedal in the Pedal area and Pedal Browser blink on and off to indicate an imported setting. The setting name area at the bottom of the Pedal Browser displays “Click selected item again to revert.
Replace a pedal in the Pedal area Do one of the following: mm Drag the stompbox from the Pedal Browser directly over the pedal you want to replace in the Pedal area. mm Click to select the stompbox you want to replace in the Pedal area, then double-click the appropriate pedal in the Pedal Browser. Note: You can replace “effect” pedals, but not the Mixer or Splitter utilities. Bus routings, if active, are not changed when an effect pedal is replaced. See Use Pedalboard’s Router on page 42.
Create a second bus routing Do one of the following: mm Move your pointer immediately above the Pedal area to open the Router, and click the name of a stompbox in the Router. Two gray lines appear in the Router—the lower one representing Bus A and the upper one Bus B—and the pedal name moves to the upper line. The chosen stompbox is now routed to Bus B, and a Mixer utility pedal is automatically added to the end of the signal chain.
Change a Splitter utility position in the Pedal area mm Drag the Splitter utility to a new position, either to the left or to the right. If you move the Splitter utility to the left, the split between Bus A and Bus B occurs at the earlier insertion point. Relevant effect pedals are moved to the right and are inserted into Bus A. If you move the Splitter utility to the right, the split between Bus A and Bus B occurs at the later insertion point.
Pedalboard distortion pedals This table below describes the distortion effects pedals. Stompbox Description Candy Fuzz A bright, “nasty” distortion effect. Drive controls the input signal gain. Level sets the effect volume. Double Dragon A deluxe distortion effect. It offers independent level controls for input (Input) and output (Level). Drive controls the amount of saturation applied to the input signal. The Tone knob sets the cutoff frequency.
Stompbox Description Rawk! Distortion A metal/hard rock distortion effect. Crunch sets the amount of saturation applied to the input signal. Output gain is set with Level. Tonal color is set with the Tone knob, making the sound brighter at higher values. Tube Burner A vacuum tube-based distortion that provides a wide palette of sounds, ranging from warm grain to crispy overdrive.
Stompbox Description Phaze 2 A flexible dual-phaser effect. LFO 1 and LFO 2 Rate set the modulation speed and can run freely, or be synchronized with the host application tempo when you enable the Sync button. Ceiling and Floor determine the frequency range that is swept. Order switches between different algorithms, with higher (even) numbers resulting in a heavier phasing effect. Odd order numbers result in more subtle combfiltering effects.
Stompbox Description Roto Phase A phaser effect that adds movement to, and alters the phase of, the signal. Rate sets the modulation speed and can either run freely or be synchronized with the host application tempo when you enable the Sync button. When synchronized, you can specify bar, beat, and note values, including triplets and dotted notes, with the Rate knob. Intensity sets the strength of the effect.
Stompbox Description the Vibe A vibrato/chorus effect based on the Scanner Vibrato unit found in the Hammond B3 organ. You can choose from three vibrato (V1–3) or chorus (C1–3) variations with the Type knob. Rate sets the modulation speed and can either run freely or be synchronized with the host application tempo when you enable the Sync button. When synchronized, you can specify bar, beat, and note values, including triplets and dotted notes. Depth sets the strength of the effect.
Pedalboard delay pedals This table describes the Delay effects pedals. Stompbox Description Blue Echo A delay effect. Time sets the modulation speed and can either run freely or be synchronized with the host application tempo when you enable the Sync button. When synchronized, you can specify bar, beat, and note values, including triplets and dotted notes. The Repeats knob determines the number of delay repeats. Mix sets the balance between the delayed and source signals.
Pedalboard filter pedals This table describes the filter effects pedals. Stompbox Description Auto-Funk An auto-wah (filter) effect. Sensitivity sets a threshold that determines how the filter responds to incoming signal levels. Cutoff sets the center frequency for the filter. The BP/LP switch enables either a bandpass or lowpass filter circuit. Signal frequencies just above and below the cutoff point are filtered when the BP switch position is chosen.
Pedalboard utility pedals This table describes the parameters of the Mixer and Splitter pedals. Stompbox Description Mixer Controls the level relationship between Bus A and Bus B signals. It can be inserted anywhere in the signal chain but is typically used at the end of the chain—at the extreme right of the Pedal area. See Use Pedalboard’s Router on page 42 for more information. The A/Mix/B switch solos the “A” signal, mixes the “A” and “B” signals, or solos the “B” signal.
Delay effects 2 Delay effects overview Delay effects store the input signal—and hold it for a short time—before sending it to the effect input or output. The held, and delayed, signal is repeated after a given time period, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. Most delays also allow you to feed a percentage of the delayed signal back to the input. This can result in a subtle, chorus-like effect or cascading, chaotic audio output.
Delay Designer Delay Designer overview Delay Designer is a multitap delay. Unlike traditional delay units that offer only one or two delays (or taps) that may or may not be fed back into the circuit, Delay Designer provides up to 26 individual taps. These taps are all fed from the source signal and can be edited to create unique delay effects. Delay Designer provides control over the level, pan position, and pitch of each tap. Each tap can also be lowpass or highpass filtered.
Delay Designer main display Delay Designer’s main display is used to view and edit tap parameters. You can choose the parameter to show and quickly zoom or navigate through all taps. Toggle buttons Tap display Identification bar View buttons Autozoom button Overview display Main display parameters •• View buttons: Click to choose the parameter or parameters shown in the Tap display. See Use the Delay Designer Tap display. •• Autozoom button: Zooms the Tap display out, making all taps visible.
Use the Delay Designer Tap display The view buttons determine the parameter shown in Delay Designer’s Tap display. The Toggle bar is shown below the view buttons. You can use it to turn parameters on or off for each tap. You can use Delay Designer’s Overview display to zoom and to navigate the Tap display area. Overview display Tip: If the Overview display is hidden behind a tap, you can move it to the foreground by holding down Shift.
Use Delay Designer’s tap toggle buttons Each tap has its own toggle button in the Toggle bar. These buttons provide a quick way to graphically turn parameters on and off. The parameter being toggled is determined by the current view button selection. 1 Click the view button for the parameter you want to toggle. 2 Click the toggle button of each tap that you want to change: •• Cutoff view: Turn the filter on or off. •• Reso view: Switch the filter slope between 6 dB and 12 dB.
Zoom the Tap display Do one of the following: mm Vertically drag the highlighted section (the bright rectangle) in the Overview display. mm Horizontally drag the highlighted bars—to the left or right of the bright rectangle—in the Overview display. Note: The Autozoom button needs to be turned off when you manually zoom in the Overview display. When you zoom in on a small group of taps, the Overview display continues to show all taps.
Create taps in Delay Designer You can create new delay taps in three different ways: by using the Tap pads, by creating them in the Identification bar, or by copying existing taps. The fastest way to create multiple taps is to use the Tap pads. If you have a specific rhythm in mind, you might find it easier to tap out your rhythm on dedicated hardware controller buttons, instead of using mouse or trackpad clicks. If you have a MIDI controller, you can assign the Tap pads to buttons on your device.
Create a tap in the Identification bar mm Click the position where you want to add a tap. Copy taps in the Identification bar mm Option-drag a selection of one or more taps to the position where you want to add the tap or taps. The delay time of copied taps is set to the drag position.
Select, move, and delete taps There is always at least one selected tap. You can easily distinguish selected taps by color—the Toggle bar icons and the Identification bar letters of selected taps are white. You can move a tap backward or forward in time or completely remove it. Note: When you move a tap, you are actually editing its delay time. Select a tap Do one of the following: mm Click a tap in the Tap display. mm Click the tap letter in the Identification bar.
Select multiple taps Do one of the following: mm To select multiple taps: Drag across the background of the Tap display. mm To select multiple nonadjacent taps: Shift-click specific taps in the Tap display. Move a selected tap in time mm In the Identification bar, drag a tap to the left to go forward in time, or to the right to go backward in time. This method also works when more than one tap is selected.
Edit parameters in the Tap display You can graphically edit any tap parameter that is represented as a vertical line in Delay Designer’s Tap display. The Tap display is ideal if you want to edit the parameters of one tap relative to other taps or when you need to edit or align multiple taps simultaneously. Edit a tap parameter in the Tap display 1 Click the view button of the parameter you want to edit.
Align the values of several taps 1 Command-click in the Tap display, and drag while holding down the Command key. A line trails behind the pointer as you drag. 2 Click the appropriate position to mark the end point of the line. The values of taps that fall between the start and end points are aligned along the line. Reset the value of a tap You can use Delay Designer’s Tap display or Tap parameter bar to reset tap parameters to their default values.
Edit filter cutoff in the Tap display In Cutoff view, each tap actually shows two parameters: highpass and lowpass filter cutoff frequency. mm Drag the cutoff frequency line—the upper line is lowpass and the lower line is highpass—to independently adjust filter cutoff values. Both cutoff frequencies can be adjusted simultaneously by dragging in the area between them. When the highpass filter cutoff frequency value is lower than that of the lowpass cutoff frequency, only one line is shown.
Edit pan in the Tap display The way the Pan parameter is represented in the Pan view is entirely dependent on the input channel configuration—mono to stereo, stereo to stereo, or surround. •• In mono input/stereo output configurations, all taps are initially panned to the center. •• In stereo input/stereo output configurations, the Pan parameter adjusts the stereo balance, not the position of the tap in the stereo field. Note: Pan is not available in mono configurations.
mm To adjust the stereo balance in stereo input/stereo output configurations: Drag the Pan parameter— which appears as a dot on the tap—up or down the tap to adjust the stereo balance. By default, stereo spread is set to 100%. To adjust the spread width, drag either side of the dot. As you do so, the width of the line extending outward from the dot changes. Keep an eye on the Spread parameter in the Tap parameter bar while you are adjusting.
Delay Designer Tap parameter bar The Tap parameter bar provides access to all parameters of the selected tap. It also shows several parameters that are not available in the Tap display, such as Transpose and Flip. Editing the parameters of a single, selected tap is fast and precise because all parameters are visible, with no need to switch display views or estimate values with vertical lines. If you choose multiple taps in the Tap display, the values of all selected taps are changed relative to each other.
Delay Designer sync mode Delay Designer can either synchronize to the project tempo or can run independently. When you are in synchronized mode (sync mode), taps snap to a grid of musically relevant positions, based on note durations. You can also set a Swing value in sync mode, which varies the precise timing of the grid, resulting in a laid-back, less robotic feel for each tap. When you are not in sync mode, taps don’t snap to a grid, nor can you apply the Swing value.
•• Swing field: Drag to determine how close to the absolute grid position every second grid increment will be. •• A setting of 50% means that every grid increment has the same value. •• Settings below 50% result in every second increment being shorter in time. •• Settings above 50% result in every second grid increment being longer in time. Tip: Use subtle grid position variations of every second increment (values between 45% and 55%) to create a less rigid rhythmic feel.
Use Delay Designer in surround Delay Designer is optimized for use in surround configurations. With 26 taps that can be positioned in the surround field, you can create interesting rhythmic and spatial effects. Note: Delay Designer generates separate automation data for stereo pan and surround pan operations. This means that when you use it in surround channels, it does not react to existing stereo pan automation data, and vice versa. Delay Designer always processes each input channel independently.
Echo This simple echo effect always synchronizes the delay time to the project tempo, enabling you to quickly create echo effects that run in time with your composition. Echo parameters •• Time pop-up menu: Choose the grid resolution of the delay time in musical note durations, based on the project tempo. •• “T” values represent triplets. •• “.” values represent dotted notes. •• Repeat slider and field: Drag to determine how often the delay effect is repeated.
Stereo Delay Stereo Delay lets you set the Delay, Feedback, and Mix parameters separately for the left and right channels. The Crossfeed knob (for each stereo side) sets the feedback intensity level of each signal being routed to the opposite stereo side. You can use Stereo Delay on mono tracks or busses when you want to create independent delays for the two stereo sides.
Common parameters •• Beat Sync button: Turn on to synchronize delay repeats to the project tempo. •• Output Mix (Left and Right) sliders and fields: Drag to independently control the level of the left and right channel signals. •• Low Cut and High Cut sliders and fields: Drag to cut frequencies below the Low Cut value and above the High Cut value from the source signal. Tape Delay Tape Delay simulates the sound of vintage tape echo machines.
•• Low Cut and High Cut sliders and fields: Drag to cut frequencies below the Low Cut value and above the High Cut value from the source signal. You can shape the sound of taps (delay repeats) with the highpass and lowpass filters. The filters are located in the feedback circuit, which means that the filtering effect increases in intensity with each delay repeat. If you want an increasingly muddy and confused tone, move the High Cut slider toward the left.
Distortion effects 3 Distortion effects overview Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology. They are still used in musical instrument amplifiers today. When overdriven, tubes produce a musically pleasing distortion that has become a familiar part of the sound of rock and pop music. Analog tube distortion adds a distinctive warmth and bite to the signal.
Bitcrusher Bitcrusher is a low-resolution digital distortion effect. You can use it to emulate the sound of early digital audio devices, to create artificial aliasing by dividing the sample rate, or to distort signals until they are unrecognizable. Bitcrusher parameters •• Drive slider and field: Drag to set the amount of gain applied to the input signal (in decibels). Note: Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well.
Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra. It can simulate warm, overdriven tube sounds and can also generate heavy distortions. Clip Distortion has an unusual combination of serially connected filters. The incoming signal is amplified by the Drive value, passes through a highpass filter, then is subjected to nonlinear distortion. Following the distortion, the signal passes through a lowpass filter.
Distortion effect The Distortion effect simulates the low fidelity distortion generated by a bipolar transistor. You can use it to simulate playing a musical instrument through a highly overdriven amplifier or to create unique distorted sounds. Distortion parameters •• Drive slider and field: Drag to set the amount of saturation applied to the signal. •• Display: Shows the impact of parameters on the signal. •• Tone knob and field: Rotate to set the frequency for the high cut filter.
Overdrive Overdrive emulates the distortion produced by a field effect transistor (FET), commonly used in solid-state musical instrument amplifiers and hardware effects devices. When saturated, FETs generate a warmer-sounding distortion than bipolar transistors, such as those emulated by the Distortion effect. Overdrive parameters •• Drive slider and field: Drag to set the saturation amount for the simulated transistor. •• Display: Shows the impact of parameters on the signal.
Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (see Modulation effects overview on page 199). Unlike these effects, however, the delay time is not modulated by a low frequency oscillator (LFO) but rather by a lowpass-filtered version of the input signal itself, using an internal sidechain. This means that the incoming signal modulates its own phase position.
Dynamics processors 4 Dynamics processors overview Dynamics processors control the perceived loudness of your audio, add focus and punch to tracks and projects, and optimize the sound for playback in different situations. The dynamic range of an audio signal is the range between the softest and loudest parts of the signal—technically, between the lowest and highest amplitudes. Dynamics processors enable you to adjust the dynamic range of individual audio files, tracks, or an overall project.
•• Noise gates: Noise gates alter the signal in a way that is opposite to that used by compressors or limiters. Whereas a compressor lowers the level when the signal is louder than the threshold, a noise gate lowers the signal level whenever it falls below the threshold. Louder sounds pass through unchanged, but softer sounds, such as ambient noise or the decay of a sustained instrument, are cut off. Noise gates are often used to eliminate low-level noise or hum from an audio signal.
•• Output meters: Show output levels, enabling you to see the results of the limiting process. The Margin field shows the highest output level. You can reset the Margin field by clicking it. •• Mode buttons (Extended Parameters area): Click to choose the type of peak smoothing: •• OptFit: Limiting follows a linear curve, which allows signal peaks above 0 dB. •• NoOver: Avoids distortion artifacts from the output hardware by ensuring that the signal does not exceed 0 dB.
Compressor parameters •• Circuit Type pop-up menu: Choose the type of circuit emulated by Compressor. The choices are Platinum, Studio or Vintage VCA or FET, and Vintage Opto. •• Side Chain Detection pop-up menu: Choose the signal type to exceed or fall below the threshold. Max uses the maximum level of each side-chained signal. Sum uses the summed level of all side-chained signals. •• If either of the stereo channels exceeds or falls below the threshold, both channels are compressed.
Use Compressor The following explains how to effectively use the main Compressor parameters. Compressor Threshold and Ratio The most important Compressor parameters are Threshold and Ratio. The Threshold parameter sets the floor level in decibels. Signals that exceed this level are reduced by the amount set as the Ratio. The Ratio parameter is a percentage of the overall level; the more the signal exceeds the threshold, the more it is reduced.
Other Compressor parameters As Compressor reduces levels, the overall volume at its output is typically lower than the input signal. You can adjust the output level with the Gain slider. You can also use the Auto Gain parameter to compensate for the level reduction caused by compression (choose either −12 dB or 0 dB). When you use the Platinum circuit type, Compressor can analyze the signal using one of two methods: Peak or root mean square (RMS).
DeEsser does not use a frequency-dividing network—a crossover utilizing lowpass and highpass filters. Rather, it isolates and subtracts the frequency band, resulting in no alteration of the phase curve. The Detector parameters are on the left side of DeEsser’s interface, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider.
Use Ducker Ducking is a common technique used in radio and television broadcasting. When the DJ or announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level. Ducker provides a simple means of achieving this result with existing recordings. It does not work in real time. Note: For technical reasons, Ducker can be inserted only in output and aux channel strips.
•• Release slider and field: Drag to control how quickly the volume returns to the original level. Set it to a high value if you want the music mix to slowly fade up after the announcement. Use the Ducker plug-in 1 Insert Ducker into an aux channel strip. 2 Assign all channel strip outputs that are supposed to “duck” (dynamically lower the volume of the mix) to a bus—the aux channel strip chosen in step 1.
Enveloper parameters •• Threshold slider and field: Drag to set the threshold level. Signals that exceed the threshold have their attack and release phase levels altered. In general, you should set the Threshold to the minimum value and leave it there. Only when you significantly raise the release phase level, which also boosts any noise in the original recording, should you raise the Threshold slider slightly. This limits Enveloper to affecting only the useful part of the signal.
Expander Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic range above the threshold level. You can use Expander to add liveliness and freshness to your audio signals. Expander parameters •• Threshold slider and field: Drag to set the threshold level. Signals above this level are expanded. •• Peak/RMS buttons: Click to determine whether the Peak or RMS method is used to analyze the signal.
Limiter Limiter works much like a compressor but with one important difference: where a compressor proportionally reduces the signal when it exceeds the threshold, a limiter reduces any peak above the threshold to the threshold level, effectively limiting the signal to this level. Limiter is used primarily when mastering. Typically, you apply Limiter as the very last process in the mastering signal chain, where it raises the overall volume of the signal so that it reaches, but does not exceed, 0 dB.
Multipressor Multipressor overview Multipressor (an abbreviation for multiband compressor) is a versatile audio mastering tool. It splits the incoming signal into different frequency bands—up to four—and enables you to independently compress each band. After compression is applied, the bands are combined into a single output signal. The advantage of compressing different frequency bands separately is that it allows more compression to be applied to bands that need it, without affecting other bands.
Display parameters •• Graphic display: Shows and allows adjustment of frequency and gain for each frequency band. The amount of gain change from 0 dB is indicated by blue bars. The band number appears in the center of active bands. You can adjust each frequency band in the following ways: •• Drag the horizontal bar up or down to adjust the gain makeup for that band. •• Drag the vertical edges of a band to the left or right to set the crossover frequencies, which adjusts the band’s frequency range.
•• Attack fields: Drag to set the time before compression starts for the selected band, after the signal exceeds the threshold. •• Release fields: Drag to set the time before compression stops on the selected band, after the signal falls below the threshold. •• Band on/off buttons: Turn each band on or off. When enabled, the button is highlighted, and the corresponding band appears in the graphic display area above. •• Byp(ass) buttons: Turn on to bypass the selected frequency band.
Use Multipressor In the graphic display, the blue bars show the gain change—not merely the gain reduction—as with a standard compressor. The gain change display is a composite value consisting of the compression reduction, plus the expander reduction, plus the auto gain compensation, plus the gain make-up. Compression parameters The Compression Threshold and Compression Ratio parameters are the key parameters for controlling compression.
Noise Gate Noise Gate overview Noise Gate is commonly used to suppress unwanted noise that is audible when the audio signal is at a low level. You can use it to remove background noise, crosstalk from other signal sources, and low-level hum, among other uses. Noise Gate works by allowing signals above the threshold level to pass unimpeded, while reducing signals below the threshold level. This effectively removes lower-level parts of the signal, while allowing the desired parts of the audio to pass.
Use Noise Gate In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold value are completely suppressed. Setting Reduction to a higher value attenuates low-level sounds but still allows them to pass. You can also use Reduction to boost the signal by up to 20 dB, which is useful for ducking effects. The Attack, Hold, and Release knobs modify the dynamic response of Noise Gate.
Surround Compressor Surround Compressor overview Surround Compressor, based on the Compressor plug-in, is specifically designed for compression of complete surround mixes. It is commonly inserted in a surround output channel strip or in audio or aux channel strips—busses—that carry multichannel audio. You can adjust the compression ratio, knee, attack, and release for the main, side, surround, and LFE channels, depending on the chosen surround format.
Surround Compressor Link parameters Surround Compressor’s Link section provides the following parameters. Link parameters •• Circuit Type pop-up menu: Choose the type of circuit emulated by Surround Compressor. The choices are Platinum, Classic A_R, Classic A_U, VCA, FET, and Opto (optical). •• Grp. (Group) pop-up menus: Set group membership for each channel (A, B, C, or no group (indicated by -).
Surround Compressor Main parameters Surround Compressor’s Main section provides the following parameters. Main parameters •• Ratio knob and field: Rotate to set the ratio of signal reduction when the threshold is exceeded. •• Knee knob and field: Rotate to set the ratio of compression at levels close to the threshold. •• Attack knob and field: Rotate to set the time it takes to reach full compression, after the signal exceeds the threshold.
Surround Compressor LFE parameters Surround Compressor’s LFE section provides the following parameters. LFE parameters •• Ratio knob and field: Rotate to set the compression ratio for the LFE channel. •• Knee knob and field: Rotate to set the knee for the LFE channel. •• Attack knob and field: Rotate to set the attack time for the LFE channel. •• Release knob and field: Rotate to set the release time for the LFE channel.
Equalizers 5 Equalizers overview An equalizer (commonly abbreviated as EQ) shapes the sound of incoming audio by changing the level of specific frequency bands. Equalization is one of the most-used audio processes, both for music projects and in postproduction work for video. You can use EQ to subtly or significantly shape the sound of an audio file, an instrument, a vocal performance, or a project by adjusting specific frequencies or frequency ranges.
Channel EQ parameters The left side of the Channel EQ window features the Gain and Analyzer controls. The central area of the window includes the graphic display and parameters for shaping each EQ band. Channel EQ parameters •• Master Gain slider and field: Drag to set the overall output level of the signal. Use it after boosting or cutting individual frequency bands. •• Analyzer button: Turns the Analyzer on or off.
Note: The Q parameter of band 1 and band 8 has no effect when the slope is set to 6 dB/Oct. When the Q parameter is set to an extremely high value, such as 100, these filters affect only a very narrow frequency band and can be used as notch filters. •• Link button: Turns on Gain-Q coupling, which automatically adjusts the Q (bandwidth) when you raise or lower the gain on any EQ band, to preserve the perceived bandwidth of the bell curve.
Channel EQ Analyzer The Analyzer uses a mathematical process called a Fast Fourier Transform (FFT) to provide a realtime curve of all frequency components in the incoming signal. This is superimposed over any EQ curves you have set. The Analyzer curve uses the same scale as the EQ curves, making it easy to recognize important frequencies in the incoming audio. This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies and frequency ranges.
Linear Phase EQ parameters The left side of the Channel EQ window incorporates the Gain and Analyzer controls. The central area of the window includes the graphic display and parameters for shaping each EQ band. Linear Phase EQ parameters •• Master Gain slider and field: Drag to set the overall output level of the signal after boosting or cutting individual frequency bands. •• Analyzer button: Click to turn the Analyzer on or off.
Note: The Q parameter of band 1 and band 8 has no effect when the slope is set to 6 dB/Oct. When the Q parameter is set to an extremely high value (such as 100), these filters affect only a very narrow frequency band and can be used as notch filters. •• Link button: Click to turn on Gain-Q coupling, which automatically adjusts the Q (bandwidth) when you raise or lower the gain on any EQ band, to preserve the perceived bandwidth of the bell curve.
Linear Phase EQ Analyzer The Analyzer uses a mathematical process called a Fast Fourier Transform (FFT) to provide a realtime curve of all frequency components in the incoming signal. This is superimposed over any EQ curves you have set. The Analyzer curve uses the same scale as the EQ curves, making it easy to recognize important frequencies in the incoming audio. This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies or frequency ranges.
Match EQ parameters Match EQ offers the following parameters. Match EQ parameters •• Analyzer button: Turns the Analyzer function on or off. •• Pre/Post button: Click to determine if the Analyzer looks at the signal before (Pre) or after (Post) the filter curve is applied. •• View pop-up menu: Set the information shown in the graphic display. Choices are: •• •• Auto: Displays information for the current function, as set with the active button below the graphic display.
•• Settings between 0 and 100% blend these values with your filter curve changes for each channel. This results in a hybrid curve. Note: The Channel Link parameters are disabled when you use the effect on a mono channel. •• LFE Handling buttons (Extended Parameters area): In surround instances, click to process or bypass the LFE channel. •• Graphic display: Displays the filter curve created by matching the template to the current material.
Use Match EQ These tasks are those commonly used with Match EQ to match the frequency spectrum of a mix with the spectrum of a source audio file. You can adapt some, or all, to your own workflow. Learn or create a Match EQ template You can drag an audio file to the Template Learn or Current Material Learn buttons for use as either the template or the current material. A progress bar appears while Match EQ is analyzing the file.
•• Generate Current Material Spectrum from audio file: Generates a frequency spectrum for an audio file that you have chosen. Refine the Match EQ curve Each time you match two audio signals, either by loading or learning a new spectrum while Match is activated or by turning on Match after a new spectrum has been loaded, any existing changes to the filter curve are discarded and Apply is set to 100%.
Edit the Match EQ filter curve You can edit the filter curve in the graphic display by adjusting the various points shown in each band. As you drag a point, the current value appears in a small box inside the graphic display, allowing precise changes. Adjust Match EQ curve values Do any of the following: mm To shift the peak frequency for the band (over the entire spectrum), drag horizontally. mm To adjust the gain of the band, drag vertically. mm To adjust the Q Factor, Shift-drag vertically.
Single-Band EQ The single-band EQ can operate in several modes. When you choose an EQ from the EQ Mode pop-up menu, the parameters shown below change. You can choose: •• Low Cut or High Cut Filter: Low Cut Filter attenuates the frequency range that falls below the selected frequency. High Cut Filter attenuates the frequency range above the selected frequency. •• High Shelf or Low Shelf EQ: Low Shelving EQ affects only the frequency range that falls below the selected frequency.
6 Filter effects Filter effects overview Filters are used to emphasize or suppress frequencies in an audio signal, resulting in a change in the tonal color of the audio. Logic Pro X contains a variety of advanced filter-based effects that you can use to creatively modify your audio. These effects are most often used to radically alter the frequency spectrum of a sound or mix. Note: Equalizers (EQs) are special types of filters.
The main areas of the AutoFilter window are the Threshold, Envelope, LFO, Filter, Distortion, and Output parameter sections. •• Threshold slider: Sets an input level that—if exceeded—triggers the envelope or LFO that dynamically modulates filter cutoff frequency. See AutoFilter threshold on page 118. •• Envelope parameters: Define how the filter cutoff frequency is modulated over time. See AutoFilter envelope on page 118.
•• Sustain knob and field: Rotate to set the sustain time for the envelope. If the input signal falls below the threshold level before the envelope sustain phase, the release phase is triggered. •• Release knob and field: Rotate to set the release time for the envelope. This is triggered as soon as the input signal falls below the threshold. •• Dynamic knob and field: Rotate to determine the input signal modulation amount. You can modulate the peak value of the envelope section by varying this control.
AutoFilter filter The Filter parameters allow you to precisely tailor the tonal color. Filter parameters •• Cutoff knob and field: Rotate to set the cutoff frequency for the filter. Higher frequencies are attenuated, whereas lower frequencies are allowed to pass through in a lowpass filter. The reverse is true in a highpass filter. When the State Variable Filter is set to bandpass (BP) mode, the filter cutoff determines the center frequency of the frequency band that is allowed to pass.
AutoFilter distortion The Distortion parameters can be used to overdrive the filter input or filter output. The distortion input and output modules are identical, but their different positions in the signal chain—before and after the filter, respectively—result in remarkably dissimilar sounds. Distortion parameters •• Input knob and field: Rotate to set the amount of distortion applied before the filter section processes the signal.
EVOC 20 Filterbank EVOC 20 Filterbank overview EVOC 20 Filterbank consists of two formant filter banks. The input signal passes through the two filter banks in parallel. Each bank features level faders for up to 20 frequency bands, allowing independent level control of each band. Setting a level fader to its minimum value completely suppresses the formants in that band. You can control the position of the filter bands with the Formant Shift parameter. You can also crossfade between the two filter banks.
EVOC 20 Filterbank Formant Filter The parameters in this section provide precise level and frequency control of the filters. Formant Shift knob Lowest button High and Low Frequency parameters Boost A knob Fade AB slider Slope pop-up menu Highest button Boost B knob Resonance knob Frequency band faders Bands value field Formant Filter parameters •• High and Low Frequency parameters: Drag to determine the lowest and highest frequencies allowed to pass by the filter banks.
•• Boost A and Boost B knobs: Rotate to set the amount of boost—or cut—applied to the frequency bands in filter bank A or B. You can use these knobs to compensate for the reduction in volume caused by lowering the level of one or more bands. If you use Boost A and Boost B to set the mix relationship between filter bank levels, you can use Fade AB (see “Fade AB slider” below) to alter the tonal color, but not the levels.
EVOC 20 Filterbank output parameters The output parameters provide control over the level and stereo width. The output section also incorporates an integrated overdrive (distortion) circuit. Output parameters •• Overdrive button: Click to turn the overdrive circuit on or off. Note: To hear the overdrive effect, you might need to boost the level of one or both filter banks. •• Level slider: Drag to set the volume of the output signal. •• Stereo Mode pop-up menu: Choose the input/output mode.
EVOC 20 TrackOscillator EVOC 20 TrackOscillator overview EVOC 20 TrackOscillator is a vocoder with a monophonic pitch tracking oscillator. The tracking oscillator tracks, or follows, the pitch of a monophonic input signal. If the input signal is a sung vocal melody, the individual note pitches are tracked and mirrored, or played, by the synthesis engine. EVOC 20 TrackOscillator features two formant filter banks, an analysis bank, and a synthesis filter bank. Each offers multiple input options.
An envelope follower is coupled to each filter band. The envelope follower of each band tracks, or follows, volume changes in the audio source—or, more specifically, the portion of the audio that has been allowed to pass by the associated bandpass filter. In this way, the envelope follower of each band generates dynamic control signals. These control signals are then sent to the synthesis filter bank—where they control the levels of the corresponding synthesis filter bands.
EVOC 20 TrackOscillator analysis in parameters The parameters in the Analysis In section determine how the input signal is analyzed and used by the EVOC 20 TrackOscillator. Analysis In parameters •• Attack knob: Rotate to determine how quickly each envelope follower—coupled to each analysis filter band—reacts to rising signals. •• Release knob: Rotate to determine how quickly each envelope follower—coupled to each analysis filter band—reacts to falling signals.
Freeze the input signal mm Click the Freeze button to hold, or sustain, the sound spectrum of the analysis input signal. By freezing the input signal you can capture a particular characteristic of the signal, which is then imposed as a complex sustained filter shape on the Synthesis section.
A short introduction to formants A formant is a peak in the frequency spectrum of a sound. In the context of human voices, formants are the key component that enables humans to distinguish between different vowel sounds—based purely on the frequency of the sounds. Formants in human speech and singing are produced by the vocal tract, with most vowel sounds containing four or more formants. U/V detection parameters •• Sensitivity knob: Rotate to determine how responsive U/V detection is.
EVOC 20 TrackOscillator synthesis in parameters The Synthesis In section controls various aspects of the tracking signal for the synthesizer. The tracking signal is used to trigger the internal synthesizer. Synthesis in parameters •• Synthesis In pop-up menu: Choose the tracking signal source: •• Track: Uses the input audio signal of the channel strip that EVOC 20 TrackOscillator is inserted into as the synthesis signal, which drives the internal synthesizer.
•• At a value of 0, the FM tone generator is disabled and a sawtooth wave is generated. •• At values above 0, the FM tone generator is activated. Higher values result in a more complex and brighter sound. •• Coarse Tune field: Drag to set the pitch offset of the oscillator in semitones. •• Fine Tune field: Drag to set the pitch offset in cents. One cent equals 1/100 of a semitone.
EVOC 20 TrackOscillator formant filter EVOC 20 TrackOscillator features two formant filter banks—one for the Analysis In section and one for the Synthesis In section. The entire frequency spectrum of an incoming signal is analyzed by the Analysis section and is divided equally into a number of frequency bands. Each filter bank can control up to 20 of these frequency bands. The Formant Filter display is divided in two by a horizontal line.
When combined, Formant Stretch and Formant Shift alter the formant structure of the resulting vocoder sound, which can lead to interesting timbre changes. For example, using speech signals and tuning Formant Shift up results in “Mickey Mouse” effects. Formant Stretch and Formant Shift are also useful if the frequency spectrum of the synthesis signal does not complement the frequency spectrum of the analysis signal.
EVOC 20 TrackOscillator output parameters The output section provides control over the type, stereo width, and level of signal that is sent from the EVOC 20 TrackOscillator. Output parameters •• Signal pop-up menu: Choose the signal that is sent to the plug-in’s main outputs: •• Voc(oder): Hear the vocoder effect. •• Syn(thesis): Hear only the synthesizer signal. •• Ana(lysis): Hear only the analysis signal. Note: The last two settings are mainly useful for monitoring purposes.
Fuzz-Wah Fuzz-Wah overview The Fuzz-Wah plug-in emulates classic wah effects, combined with compression and fuzz distortion effects. The name wah wah comes from the sound it produces. It has been a popular effect—usually a pedal effect—with electric guitarists since the days of Jimi Hendrix. The pedal controls the cutoff frequency of a bandpass, a lowpass, or—less commonly—a highpass filter. Drag a panel to determine the order of the effects chain.
•• Classic Wah: This setting mimics the sound of a popular wah pedal with a slight peak characteristic. •• Retro Wah: This setting mimics the sound of a popular vintage wah pedal. •• Modern Wah: This setting mimics the sound of a distortion wah pedal with a constant Q(uality) Factor setting. The Q determines the resonant characteristics. Low Q values affect a wider frequency range, resulting in softer resonances. High Q values affect a narrower frequency range, resulting in more pronounced emphasis.
Fuzz-Wah Compressor parameters The Compressor effect is normally used just before the Fuzz (distortion) effect. This allows you to increase or decrease the perceived gain, thus providing a suitable input level to the distortion circuit. You can, however, place the Compressor at any position in the effects chain or can disable it completely. Compressor effect parameters •• On/off button: Turns the Compressor effect on or off. •• Ratio knob: Rotate to adjust the compression slope.
Spectral Gate Spectral Gate overview Spectral Gate is an unusual filter effect that can be used as a tool for creative sound design. It works by dividing the incoming signal into two frequency ranges—above and below a central frequency band that you specify with the Center Freq and Bandwidth parameters. The signal ranges above and below the defined band can be individually processed with the Low Level and High Level parameters and the Super Energy and Sub Energy parameters.
Use Spectral Gate One way you can familiarize yourself with the operation of Spectral Gate is to start with a drum loop. Set Center Freq to its minimum value (20 Hz) and Bandwidth to its maximum value (20,000 Hz) so that the entire frequency range is processed. Turn up the Super Energy and Sub Energy knobs, one at a time, and then try different Threshold settings to get a sense of how different Threshold levels affect the sound of Super Energy and Sub Energy.
Imaging processors 7 Imaging processors overview The imaging processors are tools for manipulating the stereo image. You can use them to make certain sounds, or the overall mix, seem wider and more spacious. You can also alter the phase of individual sounds within a mix to enhance or suppress particular transients. Binaural Post-Processing Each channel strip in Logic Pro allows you to use a special version of the Pan knob, known as the Binaural Panner.
•• CTC—Speaker Angle field and slider: Drag to set an angle that matches the physical angle of your stereo speakers, in relation to the listening position. Note: This parameter is available only when the Speaker CTC compensation mode is chosen. Use multiple Binaural Panners on several channels 1 Turn off the integrated conditioning. 2 Route the output of all binaurally panned signals to an aux channel. 3 Insert a Binaural Post-Processing plug-in into the aux channel.
When you are working with MS signals: •• •• Values of 1 or higher increase the level of the side signal, making it louder than the middle signal. •• At a value of 2, you hear only the side signal. Direction knob and field: Rotate to set the pan position for the middle—the center of the stereo base—of the recorded stereo signal. When Direction is set to a value of 0, the midpoint of the stereo base in a stereo recording is perfectly centered within the mix.
XY miking In an XY recording, two directional microphones are symmetrically angled from the center of the stereo field. The right-hand microphone is aimed at a point between the left side and the center of the sound source. The left-hand microphone is aimed at a point between the right side and the center of the sound source. This results in a 45° to 60° off-axis recording on each channel (or 90° to 120° between channels).
Stereo Spread Stereo Spread is generally used when mastering. There are several ways to extend the stereo base (or the perception of space), including using reverbs or other effects and altering the signal’s phase. These options can sound good, but they can also weaken the overall sound of your mix by ruining transient responses, for example. Stereo Spread extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels.
Metering tools 8 Metering tools overview You can use the Metering tools to analyze audio in a variety of ways. These plug-ins offer you different ways to view your audio than the meters shown in channel strips. The Metering plug-ins have no effect on the audio signal and are intended for use as diagnostic aids. Each meter is specifically designed to view different characteristics of an audio signal, making each suitable for particular studio situations.
Correlation Meter Correlation Meter displays the phase relationship of a stereo signal. •• A correlation of +1 (the far right position) means that the left and right channels correlate 100%—they are completely in phase. •• A correlation of 0 (the center position) indicates the widest permissible left/right divergence, often audible as an extremely wide stereo effect.
MultiMeter MultiMeter overview MultiMeter provides a collection of professional gauge and analysis tools in a single window. It includes: •• An Analyzer to view the level of each 1/3-octave frequency band •• A Goniometer for judging phase coherency in a stereo sound field •• A Correlation Meter to spot mono phase compatibility •• An integrated Level Meter to view the signal level for each channel You can view either the Analyzer or Goniometer results in the main display area.
MultiMeter Analyzer parameters In Analyzer mode, MultiMeter’s main display shows the frequency spectrum of the input signal as 31 independent frequency bands. Each frequency band represents one-third of an octave. The Analyzer parameters are used to activate Analyzer mode and to customize the way that the incoming signal is shown in the main display. Analyzer parameters Scale MultiMeter Analyzer parameters •• Analyzer button: Switches the main display to Analyzer mode.
MultiMeter Goniometer parameters A goniometer helps you to judge the coherence of the stereo image and determine phase differences between the left and right channels. Phase problems are easily spotted as trace cancelations along the center line (M—mid/mono). The idea of the goniometer was born with the advent of early two-channel oscilloscopes.
MultiMeter Level Meter The Level Meter displays the current signal level on a logarithmic decibel scale. The signal level for each channel is represented by a blue bar. RMS and peak levels are shown simultaneously, with RMS levels appearing as dark blue bars and Peak levels appearing as light blue bars. When the level exceeds 0 dB, the portion of the bar above the 0 dB mark turns red. Current peak values are displayed numerically (in dB increments) above the Level Meter.
MultiMeter Peak parameters The MultiMeter Peak parameters are used to enable or disable the peak hold function and to reset the peak segments of all meter types. You can also determine a temporary peak hold duration. MultiMeter Peak parameters •• Hold button: Click to turn on peak hold for all metering tools in the MultiMeter, as follows: •• •• Analyzer: A small yellow segment above each 1/3-octave level bar indicates the most recent peak level.
Surround MultiMeter Surround MultiMeter overview The surround version of the MultiMeter is specifically designed for analysis and metering of multichannel surround files. You can view either the Analyzer, Goniometer, or Correlation Meter results in the main display area. Use the controls on the left side of the window to switch the view and set other MultiMeter parameters. The (Peak/RMS) Level Meter is visible on the right.
Surround MultiMeter Analyzer parameters •• Analyzer button: Switches the main display to Analyzer mode. •• Sum and Max buttons: Click to show the summed or maximum level in the Analyzer results in the main display. These buttons are relevant only when multiple channels are selected with the channel buttons. •• Channel buttons: Click to select one or multiple channels for metering. The number and appearance of these buttons vary when different surround output configurations are chosen.
Surround MultiMeter Goniometer parameters •• Goniometer button: Turn on to show the Goniometer results in the main display. •• Auto Gain field: Drag to set the amount of display compensation for low input levels. You can set Auto Gain levels in 10% increments, or you can turn it off. Note: To avoid confusion with the Auto Gain parameter found in other Logic Pro effects and processors (such as the compressors), Auto Gain is only used as a display parameter in the meters.
Surround MultiMeter Balance/Correlation The Surround MultiMeter’s Correlation Meter gauges the balance or sound placement between all incoming signals. Strongly correlated signals are shown as sharp markers and less strongly correlated signals as a blurred area. Click the Balance/Correlation button to view the Correlation Meter in the main display. Depending on the chosen surround format, a number of points that indicate speaker positions are shown (L, R, C, Ls, Rs in a 5.
Surround MultiMeter Peak parameters The Surround MultiMeter offers the following Peak parameters: Surround MultiMeter Peak parameters •• Hold button: Click to turn on peak hold for all metering tools in Surround MultiMeter, as follows: •• Analyzer: A small yellow segment above each level bar indicates the most recent peak level. •• Goniometer: All illuminated pixels are held during a peak hold. •• Level Meter: A small yellow segment above each level bar indicates the most recent peak level.
Use the Tuner utility You can tune instruments connected to your system with the Tuner utility. This ensures that your external instrument recordings are in tune with any software instruments, samples, or existing recordings in your projects. Tune Deviation display Mute button Drag to set pitch. Keynote Tuner parameters •• Graphic Tuning display: Indicates the pitch of the note in cents. At the centered (12 o’clock) position, the note is correctly tuned.
MIDI plug-ins 9 Use MIDI plug-ins MIDI plug-ins are inserted in software instrument channel strips and process or generate MIDI data—played from a MIDI region or a MIDI keyboard—in real time. MIDI plug-ins are connected in series before the audio path of a software instrument channel strip. MIDI plug-ins have a MIDI input, the MIDI processor, and a MIDI output. The output signals sent from MIDI plug-ins are standard MIDI events such as MIDI note or controller messages.
Arpeggiator MIDI plug-in Arpeggiator overview The Arpeggiator MIDI plug-in generates musically interesting arpeggios based on incoming MIDI notes. It provides split and remote features that allow you to control nearly all Arpeggiator functions without taking your hands off the keyboard, making it a powerful live performance tool. An arpeggio is a succession of notes in a chord. Rather than all notes being played at one time, they are played one after the other in a pattern: up, down, random, and so on.
Arpeggiator control parameters The control parameters start and stop the Arpeggiator and determine the latching behavior. You can also capture a live arpeggio as a MIDI region. Control parameters •• Play button: Click to start or stop arpeggiated playback of note input from a MIDI keyboard or a MIDI region. The Play button is lit when in play mode. When the Arpeggiator plug-in is stopped, incoming MIDI notes are passed through, and the settings of the split and remote keyboard parameters are retained.
•• Clear button: Click to remove all notes from the Arpeggiator plug-in latch memory. The arpeggio stops playing and all position identification numbers are reset to zero, enabling you to create a new arpeggio without turning off Latch mode, which can be useful in a live situation when preparing for a chord change. •• Silent Capture checkbox (extended parameter): Click the disclosure triangle at the lower left to display the extended parameters.
•• Lock button: Works in conjunction with the As played button. When you first click the As played button, an open lock symbol is shown. Click the open lock symbol once you have triggered an arpeggio to lock the current note order, indicated by a closed lock. This note order and feel is retained for any newly triggered arpeggios, but with new notes replacing the original notes. Click the lock symbol again to clear the locked note order and to revert to the standard “as played” behavior.
Arpeggiator note order variations The table outlines the Arpeggiator behavior in each note order preset when the Variation switch is set to the four available positions. Note order Variation 1 Variation 2 Variation 3 Variation 4 Up Plays from the lowest to highest note in consecutive order and restarts when all keys are played. Plays the second step first. This variation consists of four steps; all pressed keys are divided into groups of four with the note order applied to all groups.
Note order Variation 1 Variation 2 Variation 3 Variation 4 Outside-in Plays the highest note, then the lowest note, then plays the second highest and the second lowest note, and so on. The arpeggio restarts when all keys are played. Plays the lowest note, then the highest note, then plays the second lowest and the second highest note, and so on. The arpeggio restarts when all keys are played. This is an insideout variation.
Arpeggiator note order inversions The table outlines the Arpeggiator behavior in each note order preset when the Oct Range/ Inversion switch is set to the four positions in Inversions mode (set with the Oct Range/ Inversions button). Inversions change the root note of the chord, resulting in a different start note to arpeggiated patterns. Note order Inversion 1 Inversion 2 Inversion 3 Inversion 4 Up Plays the original chord, then three inversions in consecutive order and restarts.
Arpeggiator pattern parameters Arpeggiator pattern parameters overview Click the Pattern tab to open the Arpeggiator pattern parameters. The Pattern tab includes two distinct functional modes: Live and Grid. The modes are mutually exclusive, so turning on one turns off the other. It also provides a unique Live Capture to Grid facility. When Grid mode is active, it controls the arpeggio’s velocity, cycle length, rests, ties, and chords.
Live mode parameters •• Rest button: Click to insert a rest at the current arpeggiator step position. A position identification number is assigned to the rest, ensuring that its rhythmic position (step number) within the arpeggio is retained, even when different note order presets are chosen. Note: Rests can only be added while building the arpeggio, which means that at least one key must be held if you want to add a rest.
Note: Within an arpeggio, ties are perceived as a rhythmic element rather than a melodic variation. As a consequence, the tied note may change if notes are added after the tie has been entered or when you choose a different note order preset. •• Chord on/off buttons: Click the chord symbol to turn on Chord mode for the respective step. When the Arpeggiator encounters a chord step, it simultaneously plays all notes currently in (latched or held) memory on that step.
Arpeggiator options parameters Click the Options tab to set global Arpeggiator playback parameters, such as note length and velocity. Options parameters •• Note Length knob: Rotate to define the length of the arpeggiated notes. This ranges from 1 to 150%. •• Random knob: Rotate to set the amount of random note length variation. •• Velocity knob: Rotate to determine the maximum range of possible velocity values for arpeggiated notes.
Arpeggiator keyboard parameters Click the Keyboard tab to open the Arpeggiator keyboard parameters. The dots shown on the keyboard represent the output of currently playing notes, including any key and scale adjustments. You can also open the Remote Key editor window from the Keyboard tab. For further details, see Use Arpeggiator keyboard parameters.
Use Arpeggiator keyboard parameters The Arpeggiator keyboard parameters let you split your keyboard into zones that are used for standard note playback, arpeggio note triggering, and remote control of the Arpeggiator plug-in parameters. Resize the keyboard display The default keyboard range spans the 88 notes from C0 to C7. mm Click the keyboard, then drag left or right to reveal additional octaves, in one octave increments.
Remote control the Arpeggiator with a MIDI keyboard Most Arpeggiator parameters can be remote controlled using a MIDI keyboard. By default, only a few Remote commands are available. You can resize the Remote zone to make more commands available. Remote button 1 You must first click the Keyboard Split button to display the Remote (Key editor) button. The type and number of available remote keys is determined by the Remote zone range.
Chord Trigger MIDI plug-in Chord Trigger overview The Chord Trigger MIDI plug-in lets you trigger chords by playing a single MIDI key. The onscreen keyboards have two functions: the display of incoming and outgoing MIDI notes and the assignment of chords to keys. See Use Chord Trigger. Chord Trigger parameters •• Single and Multi buttons: Click either the Single or Multi button to select a mode. •• Single Chord mode: This mode lets you assign a single chord to a trigger key.
Use Chord Trigger Chord Trigger is straightforward to use: choose a mode, set a chord trigger range, select a trigger key, then set up a chord. You can also transpose chords and quickly assign multiple chords— onscreen or with your MIDI keyboard. Define the chord trigger range The shaded chord trigger range is shown on the upper keyboard. Incoming MIDI notes that fall within this range are interpreted as trigger keys that play the chord (Single Chord mode) or the chords (Multi Chord mode) assigned to them.
Transpose chords in the chord trigger range (Multi Chord mode only) You may want to transpose triggered chords in some circumstances. For example, in Multi Chord mode you can move the entire chord trigger range upward by two semitones to change a chord progression in C-Major (starting with the C trigger key) into a progression that plays in D-Major, starting with the D trigger key. Drag left or right to transpose. mm Drag the center of the chord trigger range left or right.
Assign a chord to a key using a MIDI keyboard It can be faster to use your MIDI keyboard when assigning chords to trigger keys. The Learn process can be started and stopped by playing an assigned note on your MIDI keyboard. 1 Click the disclosure triangle at the lower left to open the extended parameters. 2 Choose the MIDI note number you want to use as a remote control for the Learn button from the Learn Remote pop-up menu.
Modifier MIDI plug-in The Modifier MIDI plug-in lets you quickly reassign or filter a single continuous controller (CC). You can also scale or add to event values. Modifier parameters •• Input Thru button: Turn on to define whether the input event is sent to the output in addition to the reassignment. •• Input Event pop-up menu: Choose the type of MIDI input event that you want to reassign or filter. •• Re-Assign To pop-up menu: Choose the type of MIDI output event.
Modulator MIDI plug-in Modulator MIDI plug-in overview The Modulator MIDI plug-in can generate continuous controller, aftertouch, and pitch bend messages. It consists of one syncable LFO and one Delay/Attack/Hold/Release envelope. See Modulator MIDI plug-in LFO and Modulator MIDI plug-in envelope. Both the LFO and envelope can be assigned to output any continuous controller, aftertouch, and pitch bend message.
Modulation LFO parameters •• LFO on/off button: Turns the LFO on or off. •• Waveform Shape buttons: Click to select a waveform shape. Choose from: triangle, sine, square, and random. Each is suited for different types of modulations. •• Waveform display: Shows the LFO waveform shape. •• Symmetry slider: Drag to adjust the symmetry of the waveform. This deforms the waveform in the following ways: •• •• Triangle: Shapes the triangle waveform into either an upward-sawtooth or downwardsawtooth waveform.
Modulator MIDI plug-in envelope Modulation Envelope parameters •• Envelope on/off button: Turns the envelope on or off. •• •• Envelope display: Shows the current envelope shape. Drag the handles in the display to set the following parameters: •• Delay: Delays the onset of the envelope. Ranges from 0 to 10 seconds. •• Attack: Sets the time required to reach the sustain level. Ranges from 0 to 10 seconds. •• Hold: Sets the sustain level and duration. Ranges from 0 to 10 seconds.
•• Env to LFO Amp knob: Rotate to set the maximum amount of LFO output modulation. This enables you to fade the LFO in or out with the envelope. •• To pop-up menu: Choose a continuous controller number, aftertouch, or pitch bend as the envelope output target. •• Output Level slider: Move to scale the envelope output level. •• Oscilloscope: The Oscilloscope to the left of the Output Level slider displays the shape of the envelope control signal before it is scaled.
Note Repeater MIDI plug-in This plug-in mimics an audio delay by generating repeating MIDI notes. Note Repeater parameters •• Input Thru button: Turn on to pass incoming MIDI note events to the output in addition to the delayed note events. Turn off to send only the delayed notes to the output. •• Delay Sync button: Turn on to synchronize the plug-in with the host application tempo. Set the delay time with the Delay slider.
Randomizer MIDI plug-in The Randomizer plug-in randomizes incoming MIDI events in real time. Randomizer parameters •• Event Type pop-up menu: Choose the MIDI event type that you want to randomize. •• Input Range sliders: Drag to set the upper and lower limit of the range of values that are affected. Only parameter values that fall within the range are processed. All values outside the range pass through the plug-in.
Scripter plug-in Use the Scripter plug-in The Scripter plug-in lets you load and use factory or user-created scripts to process or generate MIDI data in real time. You do not need any programming knowledge to use the plug-ins created in this environment, but you can view and modify them with the built-in script editor. Once authored and stored as a setting or patch or as part of a concert or project file, you can use the plug-in just like any other. A number of pre-built Scripter processors are included.
Use the Script Editor The Script Editor is used to edit JavaScript code, enabling you to write your own MIDI plug-ins. Plug-in creation is in real time, which means that you can change and test your plug-in functions immediately. You can define interface elements, such as sliders and menus, that are shown in the Scripter plug-in window and can create the underlying logic and functions addressed by these onscreen controls.
4 Click the Run Script button. 5 Test your plug-in to verify it behaves as intended. 6 Assuming no errors are shown in the Interactive Console, save the host document, setting, or patch containing the script. Scripter API overview You can create your own MIDI processing plug-ins using the JavaScript API described in these sections.
Code example 3 Repeat notes up one octave with 100ms delay and pass all other events through. Text following “//” are comments. function HandleMIDI(event) { event.send(); // send original event if (event instanceof Note) { // if it's a note event.pitch += 12; // transpose up one octave event.sendAfterMilliseconds(100); // send after delay } } ProcessMIDI function The ProcessMIDI() function lets you perform periodic (generally timing-related) tasks.
GetParameter function The GetParameter() function retrieves information from parameters defined with var PluginParameters. The GetParameter name argument must match the defined PluginParameters name value. Code use example Text following “//” describes the argument function. Open the Mod Wheel Glissando JavaScript in the Script Editor to see how the GetParameter function is used. note.
JavaScript objects JavaScript objects overview The Scripter plug-in provides JavaScript objects that describe or represent MIDI information and information about the host application, in addition to performing MIDI processing-related functions.
Replace every MIDI event received with a modulation control change message mm Type the following in the Script Editor window. Text following “//” describes the argument function. Tip: You can use the JavaScript “new” keyword to generate a new instance of an Event object of any type. function HandleMIDI() { var cc = new ControlChange; // make a new control change message cc.number = 1; // set it to controller 1 (modulation) cc.value = 100; // set the value cc.send(); // send the event cc.
•• TimingInfo.rightCycleBeat: A floating point number indicates the beat position at the end of the cycle range. Note: The length of a beat is determined by the host application time signature and tempo. Print the beat position while the transport is running mm Type the following in the Script Editor window: var NeedsTimingInfo = true; function ProcessMIDI() { var info = GetTimingInfo(); if (info.playing) Trace(info.
Create Scripter controls The Scripter Script Editor lets you use a simple shorthand to add standard controllers such as sliders and menus for automated or real-time control of your plug-ins. The only mandatory property to define a new parameter is a name, which defaults to a basic slider. In addition, you can add the following properties to change the type and behavior of controls. Optional properties •• type: Type one of the following strings as the value: •• “lin”: Creates a linear fader.
Retrieve plug-in parameter values Call GetParameter() with the parameter name to return a value (number object) with the parameter’s current value. GetParameter() is typically used inside the HandleMIDI function or ProcessMIDI function. This code example converts modulation events into note events and provides a slider to determine note lengths. mm Type the following in the Script Editor window. Text following “//” describes the argument function.
Transposer MIDI plug-in The Transposer MIDI plug-in can transpose incoming MIDI notes in real time and can correct notes to a selected scale. Transposer parameters •• Transpose slider: Drag to transpose incoming MIDI Notes by ± 24 semitones. •• Root pop-up menu: Choose the root note for the scale. •• Scale pop-up menu: Choose one of several preset scales or create your own custom scale (User) with the onscreen keyboard. •• Keyboard: Click notes on the Keyboard to switch them on or off.
Velocity Processor MIDI plug-in Velocity Processor overview The Velocity Processor MIDI plug-in processes incoming MIDI velocity events—note on and note off—in real time. Among other applications, it allows velocity compression and expansion. Velocity Processor global parameters •• Process buttons: Click either button to process MIDI note on velocity or MIDI note off velocity. Click both buttons to process MIDI note on and MIDI note off velocity. •• Mode pop-up menu: Choose a velocity processing mode.
Velocity Processor Compress/Expand mode In Compress/Expand mode, the Velocity Processor MIDI plug-in behaves like an audio compressor. Compress/Expand mode parameters •• Threshold knob: Rotate to set a velocity value. Incoming velocities above the threshold are processed. MIDI notes with velocity values below the threshold pass through unaffected. •• Ratio knob: Rotate to determine the slope of compression/expansion above the threshold. Processing is done using a “soft knee” characteristic.
Velocity Processor Value/Range mode In Value/Range mode, the Velocity Processor MIDI plug-in can behave like an audio limiter. Value/Range mode parameters •• Value/Range switch: Set to Value to limit all incoming MIDI velocity values to the value set with the Value slider. Set to Range to limit all incoming MIDI velocity values to the range set with the Min and Max sliders. •• Value slider: Drag to set a fixed velocity for all processed notes.
Modulation effects 10 Modulation effects overview Modulation effects—such as chorus, flanging, and phasing—are used to add motion and depth to your sound. Modulation effects typically delay the incoming signal by a few milliseconds and use an LFO to modulate the delayed signal. The LFO may also be used to modulate the delay time in some effects.
Chorus effect The Chorus effect delays the original signal, and the delay time is modulated with an LFO. The delayed, modulated signal is then mixed with the original, dry signal. You can use the Chorus effect to enrich the incoming signal and create the impression that multiple instruments or voices are being played in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several musicians or vocalists perform together.
Ensemble effect Ensemble can add richness and movement to sounds, particularly when you use a high number of voices. It is useful for thickening parts, but you can also use it for strong pitch variations between voices, resulting in a detuned quality to processed material. Ensemble combines up to eight chorus effects. Two standard LFOs and one random LFO enable you to create complex modulations. The graphic display visually represents what is happening with processed signals.
Flanger effect The Flanger effect works in much the same way as the Chorus effect, but it uses a significantly shorter delay time. In addition, the effect signal can be fed back into the input of the delay line. Flanging is typically used to add a spacey or underwater quality to input signals. Flanger parameters •• Feedback slider and field: Drag to set the amount of the effect signal that is routed back into the input. This can change the tonal color and make the sweeping effect more pronounced.
Modulation Delay Modulation Delay is based on the same principles as the Flanger and Chorus effects, but you can set the delay time, allowing both chorus and flanging effects to be generated. It can also be used without modulation to create resonator or doubling effects. The modulation section consists of two LFOs with variable frequencies. Although rich, combined flanging and chorus effects are possible, the Modulation Delay is capable of producing some extreme modulation effects.
•• LFO Phase knob and field: Rotate to control the phase relationship between individual channel modulations. Available only in stereo and surround instances. •• At 0°, the extreme values of the modulation are achieved simultaneously for all channels. •• At 180° or −180°, you achieve the greatest possible distance between the modulation phases of the channels. Note: The LFO Phase parameter is available only if the LFO Left Right Link button is active.
Phaser effect The Phaser effect combines the original signal with a copy that is slightly out of phase with the original. This means that the amplitudes of the two signals reach their highest and lowest points at slightly different times. The timing differences between the two signals are modulated by two independent LFOs. In addition, the Phaser includes a filter circuit and a built-in envelope follower that tracks volume changes in the input signal, generating a dynamic control signal.
Note: When you load a setting that uses the “random” option, the saved phase offset value is recalled. If you want to randomize the phase setting again, choose “new random” from the Distribution pop-up menu. •• Output Mix slider and field: Determines the balance of dry and wet signals. Negative values result in a phase-inverted mix of the effect and direct (dry) signal. •• Warmth button: Click to turn on a distortion circuit, which is suitable for warm overdrive effects.
•• Oscillator parameters: Configure the internal sine wave oscillator, which modulates the amplitude of the input signal in both of the frequency shifter modes as well as in the ring modulator OSC mode. See Ringshifter oscillator parameters on page 208. •• Delay parameters: Delay the effect signal. See Ringshifter delay parameters on page 209. •• Envelope follower parameters: Modulate the oscillator frequency and output signal with an envelope follower.
Ringshifter oscillator parameters In both of the frequency shifter modes as well as in the ring modulator OSC mode, the internal sine wave oscillator is used to modulate the amplitude of the input signal. •• In the frequency shifter modes, the Frequency parameter controls the amount of frequency shifting, either up or down, applied to the input signal. •• In the ring modulator OSC mode, the Frequency parameter controls the frequency content, or timbre, of the resulting effect.
Ringshifter delay parameters The effect signal is routed through a delay, following the oscillator. Delay parameters •• Time knob and field: Rotate to set the delay time—in Hz when running freely, or in note values, including triplet and dotted notes, when the Sync button is selected. •• Sync button: Turn on to synchronize the delay to the host application tempo. You can choose musical note values with the Time knob.
Ringshifter LFO modulation The oscillator Frequency and Dry/Wet parameters can be modulated with the LFO—and the envelope follower (see Ringshifter envelope follower modulation on page 209). The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. The LFO produces continuous, cycled control signals. Ringshifter LFO parameters •• Power button: Turns the LFO on or off. When it is turned on, you can access the following parameters.
Ringshifter output parameters The output parameters are used to set the balance between the effect and input signals and also to set the width and feedback level of the Ringshifter. Ringshifter Output parameters •• Dry/Wet knob and field: Rotate to set the mix ratio of the dry input signal and the wet effect signal. •• Feedback knob and field: Rotate to set the amount of signal routed back to the effect input. Feedback adds an edge to the Ringshifter sound and is used for a variety of special effects.
Rotor Cabinet effect Rotor Cabinet effect overview The Rotor Cabinet effect emulates the rotating loudspeaker cabinet of a Hammond organ. Also known as the Leslie effect, it simulates both the rotating speaker cabinet, with and without deflectors, and the microphones that pick up the sound. Click to choose a cabinet type. Deflector switch Rotation switch Basic Rotor Cabinet parameters •• Rotation switch: Move to change the rotor speed between Slow, Brake, or Fast.
Rotor Cabinet effect motor parameters The Rotor Cabinet effect provides the following motor control parameters. Motor parameters •• Acceleration knob: Rotate to set the time it takes to get the rotors up to the speed set with the Max Rate knob, and the length of time it takes for them to slow down. The Leslie motors need to physically accelerate and decelerate the speaker horns in the cabinets, and their power to do so is limited.
Rotor Cabinet effect microphone types The Rotor Cabinet effect provides modeled microphones that pick up the sound of the Leslie cabinet. You can specify the microphone type with these parameters. Also see Rotor Cabinet effect mic processing controls. Mic Position switch Click to choose a microphone type. Click to choose a microphone type. Click the microphone icons to choose a microphone type for the horn and drum speakers when Real Cabinet is chosen in the Type pop-up menu.
Rotor Cabinet effect mic processing controls The Rotor Cabinet effect provides the following microphone processing parameters. Mic processing parameters •• Mic Position switch: Choose either the front or rear position for the virtual microphone. See Rotor Cabinet effect microphone types. •• •• •• When Real Cabinet is chosen in the Type pop-up menu: •• Horn knob: Rotate to define the stereo width of the Horn deflector microphone.
Scanner Vibrato effect Scanner Vibrato simulates the scanner vibrato section of a Hammond organ. Scanner Vibrato is based on an analog delay line consisting of several lowpass filters. The delay line is scanned by a multipole capacitor that has a rotating pickup. It is a unique effect that cannot be simulated with simple LFOs. You can choose between three different vibrato and chorus types. The stereo version of the effect features two additional parameters—Stereo Phase and Rate Right.
Spreader Spreader widens the stereo spectrum of a signal by periodically shifting the frequency range of the original signal, thus changing the perceived width of the signal. You can also use Spreader to specify the delay between channels in samples, which adds to the perceived width and channel separation of a stereo input signal. Spreader parameters •• Intensity slider and field: Drag to determine the modulation amount. •• Speed knob and field: Rotate to set the speed of the built-in LFO.
Tremolo effect Tremolo modulates the amplitude of the incoming signal, resulting in periodic volume changes. Tremolo is commonly used in vintage guitar combo amps, where it is sometimes incorrectly referred to as vibrato. The graphic waveform display shows all the parameters except Rate. Tremolo parameters •• Depth slider and field: Drag to determine the modulation amount. •• Waveform display: Shows the resulting waveform. •• Rate knob and field: Rotate to set the frequency of the LFO.
Pitch effects 11 Pitch effects overview You can use the pitch effects to transpose or correct the pitch of audio signals. These effects can also be used for creating unison or slightly thickened parts, or even for creating harmony voices. You can also define a scale to automatically correct some, but not all, sung notes in a vocal performance, for example. This enables you to effectively perfect an imperfect vocal take.
Pitch Correction effect parameters Pitch Correction parameters •• Use Global Tuning button: Turn on to use project Tuning settings for the pitch correction process. Turn off to set the reference tuning with Ref. Pitch. See Use Pitch Correction effect reference tuning on page 223. •• Normal and low buttons: Click to set the pitch range that is scanned (for notes that need correction). See Pitch Correction effect quantization grid on page 221. •• Ref.
Pitch Correction effect quantization grid Use the Pitch Correction effect’s “normal” and “low” buttons to determine the pitch range that you want to scan for notes that need correction. Normal is the default range and works for most audio material. Low should be used only for audio material that contains extremely low frequencies (below 100 Hz), which may result in inaccurate pitch detection.
Exclude notes from pitch correction You can use the Pitch Correction effect’s onscreen keyboard to exclude notes from the pitch quantization grid. When you first open the effect, all notes of the chromatic scale are selected. This means that every incoming note is altered to fit the next semitone step of the chromatic scale. If the intonation of the singer is poor, this might lead to notes being incorrectly identified and corrected to an unwanted pitch.
Use Pitch Correction effect reference tuning Turn on the Use Global Tuning button to use your host application Tuning settings for the pitch correction process. This ensures that all software instruments and your tuned vocal part will be in tune with each other. If Use Global Tuning is turned off, you can use the Ref. Pitch field to set the reference tuning to the root key or note. As an example of where Ref.
Pitch Shifter Pitch Shifter overview Pitch Shifter provides a simple way to combine a pitch-shifted version of the signal with the original signal. Use pitch shifting to achieve the best results. •• Semi Tones slider and field: Drag to set the pitch shift value in semitones. •• Cents slider and field: Drag to control detuning of the pitch shift value in cents (1/100th of a semitone). •• Drums, Speech, and Vocals buttons: Set one of three optimized algorithms for common types of audio material.
Use Pitch Shifter Pitch Shifter is used most effectively when you take a structured approach. Use pitch shifting 1 To set the amount of transposition, or pitch shift, drag the Semi Tones slider. 2 To set the amount of detuning, drag the Cents slider. 3 To select the algorithm that best matches the material you are working with, click the Drums, Speech, or Vocals button.
Vocal Transformer Vocal Transformer overview Vocal Transformer can be used to transpose the pitch of a vocal line, to augment or diminish the range of the melody, or even to reduce it to a single note that mirrors the pitches of a melody. No matter how you change the pitches of the melody, the constituent parts of the signal (formants) remain the same. You can shift the formants independently, which means that you can turn a vocal track into a Mickey Mouse voice, while maintaining the original pitch.
•• •• Formants pop-up menu (Extended Parameters area): Choose how Vocal Transformer processes formants. •• Process always: All formants are processed. •• Keep unvoiced formants: Only voiced formants are processed. This retains sibilant sounds in a vocal performance and produces a more natural-sounding transformation effect with some signals. Detune slider and field (Extended Parameters area): Drag to detune the input signal by the set value. This parameter is of particular benefit when automated.
Use Vocal Transformer’s Robotize mode 1 Click the Robotize button to turn on Robotize mode. In this mode, Vocal Transformer can augment or diminish the melody. You can control the intensity of this distortion with the Tracking parameter. 2 Click one of the following buttons to immediately set the Tracking slider to one of these most useful values: •• −1 button: Sets the slider to −100%. All intervals are mirrored. •• 0 button: Sets the slider to 0%.
12 Reverb effects Reverb effects overview You can use reverb effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or an open space. Sound waves repeatedly bounce off the surfaces—walls, ceilings, windows, and so on—of any space, or off objects within a space, gradually dying out until they are inaudible. These bouncing sound waves result in a reflection pattern, more commonly known as a reverberation (or reverb).
Convolution reverbs work by convolving (combining) an audio signal with the impulse response recording of a room’s reverb characteristics. See Space Designer overview on page 238. EnVerb EnVerb overview EnVerb is a versatile reverb effect with a unique feature: it allows you to adjust the envelope— the shape—of the diffuse reverb tail.
EnVerb time parameters EnVerb provides the following time parameters. EnVerb time parameters •• Dry Signal Delay slider and field: Drag to determine the delay of the original signal. You can hear the dry signal only when the Mix parameter is set to a value other than 100%. •• Graphic display: Shows changes to the reverb shape when knobs below the display are adjusted.
EnVerb sound parameters EnVerb provides the following sound parameters that change the tonal color of the reverb effect. EnVerb sound parameters •• Density slider and field: Drag to set the reverb density. •• Spread slider and field: Drag to control the width of the reverb’s stereo image. At 0% the effect generates a monaural reverb. At 200% the stereo base is artificially expanded. •• High Cut slider and field: Drag to filter frequencies above the set value out of the reverb tail.
PlatinumVerb PlatinumVerb overview PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easy to precisely emulate real rooms. PlatinumVerb splits the incoming signal into two bands: each is processed and can be edited separately.
PlatinumVerb early reflections parameters PlatinumVerb provides the following early reflections parameters. PlatinumVerb early reflections parameters •• Predelay slider and field: Drag to set the time between the start of the original signal and the arrival of the early reflections. •• Extremely short: Predelay setting can color the sound and make it difficult to pinpoint the position of the signal source.
PlatinumVerb reverb parameters PlatinumVerb provides the following reverb parameters. PlatinumVerb reverb parameters •• Initial Delay slider and field: Drag to set the time between the original signal and the diffuse reverb tail. •• Spread slider and field: Drag to control the width of the reverb’s stereo image. At 0%, the effect generates a monaural reverb. At 200%, the stereo base is artificially expanded.
PlatinumVerb output parameters PlatinumVerb provides the following output parameters. PlatinumVerb output parameters •• Dry slider and field: Drag to control the amount of the original signal. •• Wet slider and field: Drag to control the amount of the effect signal.
SilverVerb SilverVerb provides a low frequency oscillator (LFO) that can modulate the reverberated signal. It also includes a high cut and a low cut filter, allowing you to filter frequencies from the reverb signal. High frequency transients in reverb signals can sound unpleasant, can hamper speech intelligibility, or mask the overtones of the original signal. Long reverb tails with a lot of bass generally result in an indistinct mix.
Space Designer convolution reverb 13 Space Designer overview Space Designer is a convolution reverb effect that you can use to place your audio signals in exceptionally realistic recreations of real-world acoustic environments. Space Designer generates reverb by convolving, or combining, an audio signal with an impulse response reverb sample.
Space Designer interface The Space Designer interface consists of the following main sections: Impulse response parameters Envelope and EQ parameters Main display Button bar Global parameters Global parameters Filter parameters Parameter bar •• Impulse response parameters: Use to load, save, or manipulate recorded or synthesized impulse response files. The chosen impulse response file determines what Space Designer will use to convolve with your audio signal. See Use impulse responses on page 240.
Use impulse responses Space Designer can use either recorded impulse response files or synthesized impulse responses. The circular area to the left of the main display contains the impulse response parameters. These are used to determine the impulse response mode (IR Sample mode or Synthesized IR mode), to load or create impulse responses, and to set the sample rate and length. Impulse response parameters •• IR Sample button and pop-up menu: Click the IR Sample button to switch to IR Sample mode.
Turn on IR Sample mode In IR Sample mode, Space Designer loads and uses an impulse response recording of an acoustic environment. This is convolved with the incoming audio signal to place it in the acoustic space provided by the impulse response. 1 Click the IR Sample button in the circular area to the left of the main display. 2 Select an impulse response file from any folder.
Set the impulse response sample rate and preserve length Changing the sample rate upward increases—or changing it downward decreases—the frequency response (and length) of the impulse response, and to a degree the overall sound quality of the reverb. Upward sample rate changes are of benefit only if the original impulse response sample actually contains higher frequencies. When reducing the sample rate, use your ears to decide if the sonic quality meets your needs.
Set impulse response lengths mm Move the Length parameter to set the length of the impulse response—sampled or synthesized. All envelopes are automatically calculated as a percentage of the overall length, which means that if this parameter is altered, your envelope curves will stretch or shrink to fit. Note: When you are using an impulse response file, the Length parameter value cannot exceed the length of the actual impulse response sample.
Space Designer button bar The button bar is used to switch the main display and parameter bar between envelope and EQ views. It also includes buttons that reset the envelopes and EQ or reverse the impulse response. Button bar parameters •• Reset button: Click to reset the currently displayed envelope or EQ to default values. •• All button: Click to reset all envelopes and the EQ to default values. •• Volume Env button: Click to show the volume envelope in the foreground of the main display.
Edit Space Designer envelope parameters You can edit the volume and filter envelopes of all impulse responses and the density envelope of synthesized impulse responses. All envelopes can be adjusted both graphically in the main display as well as numerically in the parameter bar. Whereas some parameters are envelope-specific, all envelopes consist of the Attack Time and Decay Time parameters.
Space Designer volume envelope The volume envelope is used to set the reverb’s initial level and adjust how the volume will change over time. You can edit all volume envelope parameters numerically, and you can also edit many of them graphically (see Edit Space Designer envelope parameters on page 245). Init Level node Decay Time/End Level node Attack/Decay Time node Volume envelope parameters •• Init Level field: Sets the initial volume level of the impulse response attack phase.
Space Designer density envelope The density envelope allows you to control the density of the synthesized impulse response over time. You can adjust the density envelope numerically in the parameter bar, and you can edit the Init Level, Ramp Time, and End Level parameters using the techniques described in Edit Space Designer envelope parameters on page 245. Note: The density envelope is available only in Synthesized IR mode.
Use Space Designer EQ parameters Space Designer has a four-band EQ consisting of two parametric mid-bands plus two shelving filters (one low shelving filter and one high shelving filter). You can edit the EQ parameters numerically in the parameter bar or graphically in the main display. EQ On/Off button Individual EQ band buttons •• EQ On/Off button: Click to turn the entire EQ section on or off. •• EQ band buttons: Click to turn individual EQ bands on or off.
Graphically edit an EQ curve in Space Designer 1 Enable the EQ and one or more bands with the EQ On/Off and EQ band buttons in the top row of the parameter bar. 2 Move the cursor horizontally over the main display. When the cursor is in the access area of a band, the corresponding curve and parameter area are automatically highlighted and a pivot point is displayed. 3 Drag horizontally to adjust the frequency of the band. 4 Drag vertically to increase or decrease the Gain of the band.
Space Designer filter Space Designer filter parameters Space Designer’s filter provides control over the timbre of the reverb. You can select from several filter types and also have envelope control over the filter cutoff, which is independent of the volume envelope. Changes to filter settings result in a recalculation of the impulse response rather than a straight change to the sound as it plays through the reverb. The main filter parameters are located in the lower-left corner of the interface.
Space Designer filter envelope The filter envelope appears in the main display when you click the Filter Env button. You can use it to control the filter cutoff frequency over time. You can adjust all filter envelope parameters either numerically in the parameter bar or graphically in the main display using the techniques discussed in Edit Space Designer envelope parameters on page 245. Note: Activation of the filter envelope automatically enables the main filter.
Space Designer global parameters Space Designer global parameters overview Space Designer’s global parameters affect the overall output or behavior of the effect. See Use Space Designer global parameters and Use Space Designer output parameters. The global parameters are divided into two sections—those around the main display and those below the main display.
Use Space Designer global parameters Space Designer’s global parameters affect the overall output or behavior of the effect. See Space Designer global parameters overview. The tasks below cover the use of Space Designer’s global parameters. Use the Space Designer Input slider The Input slider behaves differently in stereo or surround configurations. (The slider does not appear in mono or mono to stereo instances of the effect.
mm Click the Latency Compensation button to turn it on, which delays the direct signal in the Output section so that it matches the processing delay of the effect signal. Space Designer’s processing latency is 128 samples at the original sample rate, and it doubles at each lower sample rate division. The processing latency increases to 256 samples if you set Space Designer’s sample rate slider to “/2.” Processing latency does not increase in surround mode or at sample rates above 44.1 kHz.
Use Space Designer’s predelay feature Predelay is the amount of time that elapses between the original signal and the initial early reflections of the reverberation. For a room of any given size and shape, predelay is determined by the distance between the listener and the walls, ceiling, and floor. Space Designer allows you to adjust this parameter over a greater range than what would be considered natural. mm To set a suitable predelay time, rotate the Pre-Dly knob.
Use Space Designer output parameters Space Designer’s global parameters affect the overall output or behavior of the effect. See Space Designer global parameters overview. The tasks below cover the use of Space Designer’s output parameters. Set Space Designer mono/stereo output parameters Use the output parameters to adjust the balance between the direct, or dry, signals and the processed signals. The parameters that are available depend on Space Designer’s input configuration.
Set Space Designer surround output parameters Use the output parameters to adjust the balance between the direct, or dry, signal and the processed signals. The parameters that are available depend on Space Designer’s input configuration. In surround configurations, Space Designer provides four output sliders that together comprise a small surround output mixer.
Use the Space Designer Spread parameters The Spread and Xover (crossover) knobs enhance the perceived width of the signal without losing the directional information of the input signal normally found in the higher frequency range. Low frequencies are spread to the sides, reducing the amount of low frequency content in the center—allowing the reverb to encompass the mix. Note: The Spread and Xover knobs function only in Synthesized IR mode.
Specialized effects and utilities 14 Specialized effects overview Logic Pro includes a bundle of specialized effects and utilities designed to address tasks often encountered during audio production: •• Denoiser eliminates or reduces noise below a threshold level. •• Exciter adds life to your recordings by generating artificial high frequency components. •• Grooveshifter creates rhythmic variations in your recordings.
Denoiser main parameters •• Threshold slider and field: Drag to set the threshold level below which the noise signals are reduced. Locate a section of the audio where only noise is audible, then set the Threshold slider to a dB value that filters only signals at or below this level. •• Reduce slider and field: Drag to set the amount of noise reduction applied to signals that fall below the threshold.
Exciter Exciter generates high frequency components that are not part of the original signal. It does this by utilizing a nonlinear distortion process that resembles the one used to produce overdrive and distortion effects. Unlike this process, however, Exciter’s distortion process involves passing the input signal through a highpass filter before feeding it into the harmonics (distortion) generator.
Grooveshifter Grooveshifter allows you to rhythmically vary audio recordings, imparting a swing feel to the input signal. Imagine a guitar solo played in straight eighth or sixteenth notes. Grooveshifter can make this straightforward solo swing. Grooveshifter automatically follows all changes to the project tempo, which it uses as the reference tempo. Note: Grooveshifter relies on perfect matching of the project tempo with the tempo of the treated recording.
Speech Enhancer You can use Speech Enhancer to improve speech recordings made with your computer’s internal microphone, if applicable. It combines denoising, advanced microphone frequency remodeling, and multiband compression. Speech Enhancer parameters •• Denoise slider and field: Drag to determine the noise floor in the recording—from –60 dB to –20 dB—and thus the amount of noise reduction required.
SubBass SubBass overview SubBass generates frequencies below those of the original signal, resulting in artificial bass content. The simplest use for SubBass is as an octave divider, similar to octaver effect pedals for electric bass guitars. Whereas such pedals can only process a monophonic input sound source of clearly defined pitch, SubBass can be used with complex summed signals as well. SubBass creates two bass signals, derived from two separate portions of the incoming signal.
•• Low Center knob and field: Rotate to set the center frequency of the lower frequency band. •• Low Bandwidth knob and field: Rotate to set the width of the lower frequency band. •• Dry slider and field: Drag to set the amount of dry (non-effect, original) signal. •• Wet slider and field: Drag to set the amount of wet (effect) signal. SubBass use tips Unlike a pitch shifter, SubBass generates a waveform that is not based on the waveform of the input signal; instead it uses a sine wave.
Utilities and tools 15 Utilities and tools overview The tools found in the Utility category can help with routine tasks and situations you may encounter during production. Examples include the Gain plug-ins, which you can use to adjust the level or phase of input signals, and I/O Utility, which you can use to integrate external audio effects into your host application mixer.
Gain plug-in Gain amplifies (or reduces) the signal by a specific decibel amount. It is very useful for quick level adjustments when you work with automated tracks during post-processing—for example, when you have inserted an effect that doesn’t have its own gain control, or when you want to change the level of a track for a remix version. Gain plug-in parameters •• Gain slider and field: Drag to set the amount of gain.
Use I/O utility I/O utility enables you to use external audio effects units, similar to using internal effects. Note: I/O utility is not practical unless you are using an audio interface that provides discrete inputs and outputs, either analog or digital, that are used to send signals to and from the external audio effects unit. I/O utility parameters •• Output Volume field and slider: Drag to adjust the level of the output signal.
5 Adjust the Input Volume and Output Volume sliders as required in the I/O utility window. 6 Click the Latency Detection (Ping) button if you want to detect and compensate for any delay between the selected output and input. When you start playback, the signals of any channel strips routed to the aux channel chosen in step 3 are processed by the external effects unit. Multichannel Gain Multichannel Gain allows you to independently control the gain and phase of each channel in a surround mix.
Test Oscillator Test Oscillator is useful for tuning studio equipment and instruments and can be inserted as both an instrument or effect plug-in. It operates in two modes, generating either a static frequency or a sine sweep. In the first mode, which is the default mode, it starts generating the test signal as soon as it is inserted. You can switch it off by bypassing it.
Appendix Legacy effects Legacy effects overview Legacy effects are included for project compatibility. These plug-ins are inserted when you load a project (that contains these plug-ins) created with an older Logic Pro version. You can use these plug-ins or you can replace them with other effect plug-ins available in Logic Pro. You cannot directly insert these plug-ins in Logic Pro unless you override the effects plug-in menu.
Bass Amp Bass Amp simulates the sound of several famous bass amplifiers. You can route bass guitar and other signals directly through Bass Amp, reproducing the sound of your musical part played through a number of high-quality bass guitar amplification systems. Bass Amp Parameters •• Model pop-up menu: Choose one of the following amplifier models: •• American Basic: 1970s-era American bass amp, equipped with eight 10" speakers. Suitable for blues and rock recordings.
•• Mid Freq slider: Sets the center frequency of the mid band (between 200 Hz and 3000 Hz). •• Output Level slider: Sets the final output level for Bass Amp. EQ DJ EQ DJ EQ combines high and low shelving filters, each with a fixed frequency, and one parametric EQ. You can adjust the Frequency, Gain, and Q-Factor of the latter. The DJ EQ allows the filter gain to be reduced by as much as −30 dB. DJ EQ parameters •• High Shelf slider and field: Drag to set the amount of gain for the high shelving filter.
Fat EQ Fat EQ is a versatile multiband EQ that can be used on individual sources or overall mixes. Fat EQ provides up to five individual frequency bands, graphically displays EQ curves, and includes a set of parameters for each band. Fat EQ parameters •• Band Type buttons: For bands 1, 2, 4, and 5, click one of the paired buttons to select the EQ type. Band 3 is parametric. •• Band 1: Click the highpass or low shelving button. •• Band 2: Click the low shelving or parametric button.
Single-Band EQs The single-band EQs are used for different types of equalization tasks. •• High Cut or Low Cut: High Cut attenuates the frequency range above the selected frequency. Low Cut attenuates the frequency range that falls below the selected frequency. •• High Pass or Low Pass Filter: High Pass Filter affects the frequency range below the set frequency. Higher frequencies pass through the filter. You can use High Pass Filter to eliminate the bass below a selectable frequency.
Silver EQ Silver EQ includes three bands—a high shelving EQ, a parametric EQ, and a low shelving EQ. You can adjust the cutoff frequencies for the high shelving and low shelving EQs. You can adjust the center frequency, gain, and Q factor of the parametric EQ. Silver EQ parameters •• High Shelf slider and field: Drag to set the level of the high shelving EQ. •• High Frequency slider and field: Drag to set the cutoff frequency for the high shelving EQ.
GoldVerb GoldVerb overview GoldVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easy to precisely emulate real rooms. Early Reflections parameters Balance ER/Reverb slider Mix slider and field Reverb parameters GoldVerb is divided into four parameter areas: •• Early reflections parameters: Used to emulate the original signal’s first reflections as they bounce off the walls, ceiling, and floor of a natural room.
GoldVerb early reflections parameters The GoldVerb provides the following Early Reflections parameters. GoldVerb early reflections parameters •• Predelay slider and field: Drag to set the time between the start of the original signal and the arrival of the early reflections. •• Extremely short: Predelay settings can color the sound and make it difficult to pinpoint the position of the signal source.
GoldVerb reverb parameters GoldVerb provides the following reverb parameters. GoldVerb reverb parameters •• Initial Delay slider and field: Drag to set the time between the original signal and the diffuse reverb tail. If you are trying to attain a natural-sounding, harmonic reverb, the transition between the early reflections and the reverb tail should be as smooth and seamless as possible.
Guitar Amp Pro Guitar Amp Pro overview Guitar Amp Pro simulates the sound of popular guitar amplifiers and the speakers used with them. You can process guitar signals directly, which enables you to reproduce the sound of your guitar through a number of high-quality guitar amplification systems. Guitar Amp Pro can also be used for experimental sound design and processing. You can use it with other instruments, applying the sonic character of a guitar amp to a trumpet or vocal part, for example.
Guitar Amp Pro amplifier models You can choose an amplifier model from the Amp pop-up menu near the top of the interface. Amp models •• UK Combo 30W: Neutral-sounding amp, suitable for clean or crunchy rhythm parts. •• UK Top 50W: Quite aggressive in the high frequency range, suitable for classical rock sounds. •• US Combo 40W: Clean sounding amp model, suitable for funk and jazz sounds. •• US Hot Combo 40W: Emphasizes the high mid-frequency range, making this model ideal for solo sounds.
•• •• US broad range: Simulation of a classic electric piano speaker. •• Analog simulation: Internal speaker simulation of a well-known British tube preamplifier. •• UK 1 x 12 (GarageBand): A British Class A tube open back with a single 12" speaker. •• UK 4 x 12 (GarageBand): Classic closed enclosure with four 12" speakers (black series), suitable for rock. •• US 1 x 12 open back (GarageBand): Open enclosure of an American lead combo with a single 12" speaker.
Guitar Amp Pro effects The effects parameters include Tremolo, Vibrato, and Reverb, which emulate the processors found on many amplifiers. You can use the pop-up menu to choose either Tremolo, which modulates the amplitude or volume of the sound, or Vibrato, which modulates the pitch. Reverb can be added to either of these effects, or used independently. To use or adjust an effect, you must first enable it by clicking the corresponding On button to the left. The On button is red when active.
Guitar Amp Pro microphone parameters After choosing a speaker cabinet from the Speaker menu, you can set the type of microphone you want to be emulated, and where the microphone is placed in relation to the speaker. The Microphone Position parameters are available in the yellow area to the left, and the Microphone Type parameters in the yellow area to the right. Microphone position parameters •• Centered button: Places the microphone in the center of the speaker cone, also called on-axis.
Silver Compressor Silver Compressor is a simplified version of the Compressor plug-in. See Use Compressor on page 86. Silver Compressor parameters •• Gain Reduction meter: Shows the amount of compression in real time. •• Threshold slider and field: Drag to set the threshold level. Signals that exceed the threshold are reduced in level. •• Attack knob and field: Rotate to set the time it takes for Silver Compressor to react when the signal exceeds the threshold.
Silver Gate Silver Gate is a simplified version of the Noise Gate plug-in. See Use Noise Gate on page 99. Silver Gate parameters •• Lookahead slider and field: Drag to set how far ahead Silver Gate analyzes the incoming signal, allowing it to respond more quickly to peak levels. •• Threshold slider and field: Drag to set the threshold level. Signals that fall below the threshold are reduced in level.
