MainStage 3 Effects For OS X
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Contents 10 10 11 11 13 19 21 23 24 26 27 27 28 29 30 32 33 34 37 38 38 39 40 41 42 44 45 47 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 58 60 61 67 68 69 70 71 72 73 Chapter 2: Delay effects 75 75 76 77 78 79 79 80 Chapter 3: Distortion effects 81 81 82 83 83 85 87 88 90 91 92 92 92 93 94 95 96 96 97 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 Delay Desi
98 98 98 98 99 100 101 101 101 102 103 104 104 104 105 106 108 109 Chapter 5: Equalizers 110 110 110 110 111 112 113 114 115 115 116 116 117 118 119 120 120 120 121 122 123 124 125 126 128 129 130 131 131 131 133 133 Chapter 6: Filter effects Equalizers overview Channel EQ Channel EQ overview Channel EQ parameters Channel EQ use tips Channel EQ Analyzer Linear Phase EQ Linear Phase EQ overview Linear Phase EQ parameters Linear Phase EQ use tips Linear Phase EQ Analyzer Match EQ Match EQ overview Match
134 Spectral Gate 134 Spectral Gate overview 135 Use Spectral Gate 136 Chapter 7: Imaging processors 136 Imaging processors overview 136 Direction Mixer 136 Direction Mixer overview 137 Stereo miking techniques 139 Stereo Spread 140 140 140 141 141 142 142 143 144 145 145 146 147 Chapter 8: Metering tools 148 148 149 149 150 151 156 159 160 161 162 163 163 164 166 167 167 167 169 171 172 Chapter 9: MIDI plug-ins Metering tools overview BPM Counter Correlation Meter Level Meter plug-in MultiMeter Mult
173 Scripter plug-in 173 Use the Scripter plug-in 173 Use the Script Editor 175 Scripter API overview 175 MIDI processing functions 177 JavaScript objects 181 Create Scripter controls 183 Transposer MIDI plug-in 184 Velocity Processor MIDI plug-in 184 Velocity Processor overview 185 Velocity Processor Compress/Expand mode 186 Velocity Processor Value/Range mode 186 Velocity Processor Add/Scale mode 187 187 188 189 190 190 191 193 194 194 194 195 196 197 197 199 200 200 201 202 203 204 205 206 Chapter 10:
207 207 207 207 208 209 210 211 212 212 213 213 213 214 215 Chapter 11: Pitch effects 217 217 218 218 219 220 221 221 222 223 224 225 Chapter 12: Reverb effects 226 226 227 228 232 232 232 233 234 235 236 237 237 238 239 239 240 242 Chapter 13: Space Designer convolution reverb Pitch effects overview Pitch Correction effect Pitch Correction effect overview Pitch Correction effect parameters Pitch Correction effect quantization grid Exclude notes from pitch correction Use Pitch Correction effect ref
244 244 244 244 245 246 247 248 249 249 249 250 Chapter 14: Specialized effects and utilities 251 251 251 252 253 Chapter 15: Utilities and tools 254 254 254 255 256 256 257 258 259 260 260 261 262 263 263 264 265 266 266 267 268 269 270 Appendix: Legacy effects Specialized effects overview Denoiser Denoiser overview Denoiser smoothing parameters Exciter Grooveshifter Speech Enhancer SubBass SubBass overview SubBass parameters SubBass use tips Utilities and tools overview Gain plug-in Use I/O utili
Amps and pedals 1 Amps and pedals overview MainStage 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.
Amp Designer Amp Designer overview Amp Designer emulates the sound of more than 20 famous guitar amplifiers and the speaker cabinets used with them. Each preconfigured model combines an amp, a cabinet, and EQ that recreates a well-known guitar amplifier sound. You can process guitar signals directly, reproducing the sound of your guitar played through these amplification systems. You can also use Amp Designer for experimental sound design and processing.
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.
Classic American Combos The Blackface, Brownface, and Silverface models are inspired by American combos of the mid 1960s. These tend to be loud and clean with a tight low-end and restrained distortion. They are useful for clean-toned rock, vintage R & B, surf music, twangy country, jazz, or any other style where strong note definition is essential. Model Description Large Blackface Combo A 4 x 10" combo with a sweet, well-balanced tone favored by rock, surf, and R & B players.
British Combos The British Combos capture the brash, treble-rich sound associated with 1960s British rock and pop. The sonic signature of these amps is characterized by their high-end response, yet they are rarely harsh-sounding due to a mellow distortion and smooth compression. Model Description British Blues Combo This 2 x 12" combo has a loud, aggressive tone that is cleaner than the British heads, yet delivers rich distorted tones at high gain settings.
Metal Stacks The Metal Stack models are inspired by the powerful, high gain amplifier heads favored by modern hard rock and metal musicians. All are paired with 4 x 12" cabinets. Their signature tones range from heavy distortion to extremely heavy distortion. These models are ideal if you want powerful lows, harsh highs, and long sustain in your guitar tones. Model Description Modern American Stack A powerful high-gain amp that is ideal for heavy rock and metal.
Amp Designer cabinets This table outlines the properties of each cabinet model available in Amp Designer. Cabinet Description Tweed 1 x 12 A 12" open-back cabinet from the 1950s with a warm and smooth tone. Tweed 4 x 10 A 4 x 10" open-back cabinet from the late 1950s that was originally conceived for bassists but that guitarists use for its sparkling presence. Tweed 1 x 10 A single 10" open-back combo amp cabinet from the 1950s with a smooth sound.
Cabinet Description Stadium 4 x 12 A tight, bright, closed-back British cabinet with bold upper/mid peaks. Stadium 2 x 12 A nicely balanced modern British open-back cabinet. Tonally, it is a compromise between the warmth of the Blackface 4 x 10 and the brilliance of the British 2 x 12. Boutique Retro 2 x 12 A 2 x 12" cabinet based on the British 2 x 12. It has a rich, open midrange and is more powerful in the treble range.
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.
•• Combos or Stacks: Combo amps include both an amplifier and speakers in a single enclosure. These usually have an open back, so the sound resonates in multiple directions. The resulting sound is open—with bright, airy highs and a spacious sound. Amplifier stacks consist of an amplifier head, with the speakers in a separate cabinet. These cabinets generally have a closed back and project the sound forward in a tight, focused beam.
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 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 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.
Pedalboard modulation pedals This table describes the modulation effects pedals. Stompbox Description Dr Octave A classic octaver effect with two independent octave controls plus an integrated overdrive. Flange Factory A deluxe flanging effect that allows precise control of every aspect of your sound. Heavenly Chorus A rich, sweet-sounding chorus effect that thickens the sound.
Stompbox Description Robo Flanger Flexible flanging effect. 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. Feedback determines the amount of effect signal that is routed back into the input.
Stompbox Description Total Tremolo A flexible tremolo effect—modulation of the signal level. 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. Wave and Smooth work in combination to change the LFO waveform shape.
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 with the Tap pad 1 Click the upper pad (Start). Note: Whenever you click the Start pad, it automatically erases all existing taps. Because of this behavior, after you create your initial taps, you will want to create subsequent taps by clicking in the Identification bar. The upper pad label changes to Tap, and a red tap recording bar appears in the strip below the view buttons. 2 To begin recording new taps, click the Tap button. 3 To create new taps, click the Tap button.
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.
Delete a tap Do one of the following: mm Select a tap, then press the Delete key. mm In the Identification bar, drag a tap letter downward, out of the Tap display. This method also works when more than one tap is selected. Delete all selected taps mm Control-click (or right-click) a tap, then choose “Delete tap(s)” from the shortcut menu. 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.
Note: The method outlined above is slightly different for the Filter Cutoff and Pan parameters. See the tasks below. Set the values of multiple taps mm Command-drag horizontally and vertically across several taps in the Tap display. Parameter values change to match the pointer position as you drag across the taps. Commanddragging across several taps lets you draw value curves, much like using a pencil to create a curved line on a piece of paper.
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. mm To reset a parameter to its default setting in the Tap display: Option-click a tap to reset the selected parameter to its default setting. If multiple taps are selected, Option-clicking any tap resets the chosen parameter to its default value for all selected taps.
When the highpass filter cutoff frequency value is lower than that of the lowpass cutoff frequency, only one line is shown. This line represents the frequency band that passes through the filters—in other words, the filters act as a bandpass filter. In this configuration, the two filters operate serially, meaning that the tap passes through one filter first, then the other.
mm To edit the pan position in mono input/stereo output configurations: Drag vertically from the center of the tap in the direction you want to pan the tap or taps. A white line extends outward from the center in the direction you have dragged, reflecting the pan position of the tap or taps. Lines above the center position indicate pans to the left, and lines below the center position denote pans to the right. Left (blue) and right (green) channels are easily identified.
Edit taps with shortcut menu commands mm Control-click (or right-click) a tap in the Tap display, then choose one of the following commands from the shortcut menu: •• Copy sound parameters: Copies all parameters (except the delay time) of the selected tap or taps to the Clipboard. •• Paste sound parameters: Pastes the tap parameters from the Clipboard into the selected tap or taps. If there are more taps in the Clipboard than are selected in the Tap display, the extra taps in the Clipboard are ignored.
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.
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.
Sample Delay Sample Delay is more a utility than an effect—you can use it to delay a channel by single sample values. When used in conjunction with the phase inversion capabilities of the Gain effect, Sample Delay is useful for correcting timing problems that may occur with multichannel microphones. It can also be used creatively to emulate stereo microphone channel separation. Every sample at a frequency of 44.1 kHz is equivalent to the time taken for a sound wave to travel 7.76 millimeters.
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.
Distortion II Distortion II emulates the distortion circuit of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect or can use it creatively when designing new sounds. Distortion II parameters •• PreGain knob: Rotate to set the amount of gain applied to the input signal. •• Drive knob: Rotate to set the amount of saturation applied to the signal. •• Tone knob: Rotate to set the frequency of the highpass filter.
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 187). 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 DeEsser is a frequency-specific compressor, designed to compress a particular frequency band within a complex audio signal. It is used to eliminate hiss (also called sibilance) from the signal. The advantage of using DeEsser rather than an EQ to cut high frequencies is that it compresses the signal dynamically, rather than statically. This prevents the sound from becoming darker when no sibilance is present in the signal. DeEsser has extremely fast attack and release times.
DeEsser common parameters •• Detector and Suppressor frequency displays: The upper display shows the Detector frequency range. The lower display shows the Suppressor frequency range (in hertz). •• Smoothing slider: Drag to set the reaction speed for the gain reduction start and end phases. Smoothing controls both the attack and release times, as they are used by compressors.
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.
DraftVersion 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.
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.
•• Template Learn button: Starts/stops the process of learning the frequency spectrum of the source file. •• Current Material Learn button: Starts/stops the process of learning the frequency spectrum of the project you want to match with the source file. •• Current Material Match button: Click to match the frequency spectrum of the current material to that of the template (source) file. •• Phase pop-up menu: Switches the operational principle of the filter curve.
Match the EQ of a project mix to the EQ of a source audio file 1 In the project you want to match to the source audio file, insert Match EQ (typically on Output 1-2). 2 Drag the source audio file to the Template Learn button. 3 Return to the start of your mix, click Current Material Learn, and play your mix (the current material) from start to finish. 4 When you are done, click Current Material Match (this automatically turns off the Current Material Learn button).
Use the matched EQ on a channel strip Match EQ creates a filter curve based on the differences between the spectrum of the template and the current material. This curve automatically compensates for differences in gain between the template and the current material, with the resulting EQ curve referenced to 0 dB. A yellow filter response curve appears in the graphic display, showing the average spectrum of your mix. This curve approximates (mirrors) the average spectrum of your source audio file.
Change Match EQ gain with the scales mm Drag either scale to adjust the overall gain of the filter curve from −30 to +30 dB. The left scale—and the right, if the Analyzer is inactive—shows the dB values for the filter curve. 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.
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. MainStage 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 111. •• Envelope parameters: Define how the filter cutoff frequency is modulated over time. See AutoFilter envelope on page 112.
AutoFilter envelope The envelope is used to shape the filter cutoff over time. When the input signal exceeds the set threshold level, the envelope is triggered. Envelope parameters •• Attack knob and field: Rotate to set the attack time for the envelope. •• Decay knob and field: Rotate to set the decay time for the envelope. •• Sustain knob and field: Rotate to set the sustain time for the envelope.
AutoFilter LFO The LFO is used as a modulation source for filter cutoff. LFO parameters •• Coarse Rate slider and field: Drag to set the speed of LFO modulation. Use to set the LFO frequency in hertz. Note: The labels shown for the Rate knob, slider, and field change when you activate Beat Sync. Only the Rate knob and field are then available. •• Fine Rate knob: Rotate to set the speed of LFO modulation. Use to fine-tune the LFO frequency.
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.
Use EVOC 20 TrackOscillator analysis in You should be precise with the Analysis In parameters in order to attain the best possible speech intelligibility and the most accurate tracking. Follow these tasks and tips to obtain the best results. Set Attack and Release times mm Rotate the Attack and Release knobs to set times that provide the most articulated sound. Longer attack times result in a slower tracking response to transients—level spikes—of the analysis input signal.
EVOC 20 TrackOscillator U/V detection parameters Human speech consists of a series of voiced sounds—tonal sounds or formants—and unvoiced sounds. The main distinction between voiced and unvoiced sounds is that voiced sounds are produced by an oscillation of the vocal cords, whereas unvoiced sounds are produced by blocking and restricting the air flow with lips, tongue, palate, throat, and larynx.
•• •• Blend: Uses the analysis signal after it has passed through a highpass filter for the unvoiced portions of the sound. The Sensitivity parameter has no effect when this setting is used. Level knob: Rotate to set the volume of the signal used to replace the unvoiced content of the input signal. Important: Be careful with the Level knob, particularly when using a high Sensitivity value, to avoid internally overloading the EVOC 20 TrackOscillator.
EVOC 20 TrackOscillator oscillators Tracking oscillator parameters The tracking oscillator follows the pitch of incoming monophonic audio signals and mirrors these pitches with a synthesized sound. The FM tone generator for the tracking oscillator consists of two oscillators, each of which generates a sine wave. The frequency of Oscillator 1, the carrier, is modulated by Oscillator 2, the modulator, which deforms the sine wave of Oscillator 1. This results in a waveform with rich harmonic content.
Use tracking oscillator pitch parameters The tracking oscillator pitch parameters control the automatic pitch correction feature of the tracking oscillator. They can be used to constrain the pitch of the tracking oscillator to a scale or chord. This allows subtle or strong pitch corrections and can be used creatively on unpitched material with high harmonic content, such as cymbals and high-hats.
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.
Auto Wah parameters •• On/off button: Turns the Auto Wah effect on or off. •• Wah Type pop-up menu: Choose a Wah effect type. •• 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.
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.
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.
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.
All memorized chords are moved with the chord trigger range and are automatically transposed. Transpose chords by octaves mm Choose an octave transposition from the Chord Octave pop-up menu. All memorized chords can be transposed up or down by up to four octaves. Assign a chord to a key using the onscreen keyboard 1 Click the Learn button. The Learn button label changes to “Trigger Key” and the button begins to blink. 2 Click a trigger key—within the chord trigger range—on the upper keyboard.
Clear a chord assignment 1 Click the Clear button. •• In Single Chord mode: The assigned chord is erased. •• In Multi Chord mode: The button label changes to “Trigger Key” and begins to blink. 2 Click the trigger key that you want to clear on the upper keyboard. The chord assigned to the trigger key is erased and the trigger key is dimmed, indicating that no chord is assigned. Clear all chord assignments The following applies only to Multi Chord mode. mm Press Option, then click the Clear button.
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.
See the Scripter API overview for Scripter API documentation and code examples. Run Script button Code Editor Interactive Console Script Editor parameters •• Run Script button: Click to evaluate the script and configure the plug-in and parameters. Output, including errors, is shown in the Interactive Console when you click this button. •• •• Code Editor: Type JavaScript code in this area.
Scripter API overview You can create your own MIDI processing plug-ins using the JavaScript API described in these sections. •• MIDI processing functions overview •• JavaScript objects overview •• Create Scripter controls Tip: View the supplied scripts in the Script Editor to see how they are constructed. You can modify and re-use the code to change functions or to create new processors. See Use the Script Editor.
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.
ParameterChanged function The ParameterChanged() function lets you perform tasks triggered by changes to plug-in parameters. ParameterChanged is called each time one of the plug-in’s parameters is set to a new value. ParameterChanged is also called once for each parameter when you load a plug-in setting. ParameterChanged is called with two arguments, first the parameter starting from 0), then the parameter value (a number).
The Event object is not instantiated directly but is a prototype for the following eventspecific object types. All of the following types inherit the methods described above and the channel property. Event types The event types and their properties are passed to HandleMIDI as follows: •• NoteOn.pitch(integer number): Pitch from 1–127. •• NoteOn.velocity(integer number): Velocity from 0–127. A velocity value of 0 is interpreted as a note off event, not a note on. •• NoteOff.
Use the JavaScript TimingInfo object The TimingInfo object contains timing information that describes the state of the host transport and the current musical tempo and meter. A TimingInfo object can be retrieved by calling GetTimingInfo(). TimingInfo properties •• TimingInfo.playing: Uses Boolean logic where “true” means the host transport is running. •• TimingInfo.blockStartBeat: A floating point number indicates the beat position at the start of the process block. •• TimingInfo.
Use the JavaScript MIDI object The MIDI object contains a number of convenient and easy to use functions that can be used when writing your scripts. Note: The MIDI object is a property of the global object, which means that you do not instantiate it but access its functions much like you would the JavaScript Math object. An example is calling MIDI.allNotesOff() directly.
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.
•• Volume Mod(ulation) slider and field: Drag to determine the impact of LFO modulation on the amplitude of the effect signal. •• Output Mix slider and field: Drag to determine the balance between dry and wet signals. •• All Pass button (Extended Parameters area): Turn on to introduce an additional allpass filter into the signal path. An allpass filter shifts the phase angle of a signal, influencing its stereo image.
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.
Ringshifter Ringshifter overview Ringshifter combines a ring modulator with a frequency shifter effect. Both effects were popular during the 1970s and are currently experiencing a renaissance. The ring modulator modulates the amplitude of the input signal using either the internal oscillator or a side-chain signal. The frequency spectrum of the resulting effect signal equals the sum of, and the difference between, the frequency content in the two original signals.
Set the Ringshifter mode The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator. Ringshifter mode parameters •• Single Freq(uency) Shift(er) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted to a positive value on the right side of the Frequency knob or to a negative value on the left side.
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 197). 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 211. •• 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 209. •• 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 parameters Vocal Transformer includes the following parameters. Vocal Transformer parameters •• Pitch knob and field: Rotate to determine the amount of transposition applied to the input signal. See Use the Vocal Transformer Pitch and Formant parameters. •• Robotize button: Turns Robotize mode on or off. This mode is used to augment, diminish, or mirror the melody. See Use Vocal Transformer’s Robotize mode. •• Pitch Base slider and field: Available only in Robotize mode.
Use Vocal Transformer You can change the pitch of performances, inclusive of, or independent from, formants. Robotize mode enables you to augment or diminish the melody. Use the Vocal Transformer Pitch and Formant parameters mm To transpose the pitch of the signal upward or downward: Rotate the Pitch knob. Adjustments are made in semitone steps. Incoming pitches are indicated by a vertical line below the Pitch Base field.
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 226. 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 228.
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 envelopes and EQ Space Designer envelopes and EQ overview Space Designer’s main interface area is used to show and edit envelope and equalizer (EQ) parameters. It consists of the button bar at the top, the main display, and the parameter bar. •• The button bar is used to choose the current view/edit mode. •• The main display shows, and allows you to graphically edit, either the envelope or the EQ curve.
•• Reverse button: Click to reverse the impulse response and envelopes. When the impulse response is reversed, you are effectively using the tail rather than the front end of the sample. You may need to change the Pre-Dly and other parameter values when reversing. 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.
Change Space Designer’s envelope curve shape graphically 1 Drag the envelope curve in the main display. 2 Drag the small nodes attached to a line for fine adjustments to envelope curves. These nodes are tied to the envelope curve itself, so you can view them as envelope handles. Move the nodes vertically or horizontally to change the shape of the envelope curve. 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.
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 233. 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.
4 Drag vertically to increase or decrease the Gain of the band. 5 Vertically drag the highlighted pivot point of a parametric EQ band to raise or lower the Q value. 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.
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 233. 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 configurations. (The slider does not appear in mono or mono to stereo instances of the effect.) mm In stereo instances: Drag the Input slider to determine how a stereo signal is processed.
mm Drag either of the Definition fields vertically to set the crossover point—where the switch to the reduced impulse response resolution occurs. The first Definition field is shown as a percentage, where 100% is equal to the length of the full resolution impulse response. The second field is shown in milliseconds, which indicate the exact crossover point position. (These two fields are linked, so making a change in one automatically changes the other.
mm Rotate the IR Start knob to shift the playback start point of the impulse response. This effectively cuts off the beginning of the impulse response, which can be useful for eliminating level peaks at the start of the impulse response sample. 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.
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 MainStage 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.
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.
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. 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.
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 MainStage version. You can use these plug-ins or you can replace them with other effect plug-ins available in MainStage. You cannot directly insert these plug-ins in MainStage 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.
•• Pre Gain slider: Sets the pre-amplification level of the input signal. •• Bass, Mid, and Treble sliders: Adjusts the bass, mid, and treble levels. •• 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.
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.
Guitar Amp Pro cabinet models The speaker cabinet can have a huge bearing on the type of tones you can extract from your chosen amplifier. The speaker parameters are found near the top of the interface. Speaker cabinet parameters •• Speaker pop-up menu: You can choose one of the 15 speaker models: •• •• UK 1 x 12 open back: Classic open enclosure with one 12" speaker, neutral, well-balanced, multifunctional.
Guitar Amp Pro EQ The EQ pop-up menu and the Amp-EQ Link button are near the top of the interface. EQ parameters •• EQ pop-up menu: Contains the following EQ models: British1, British2, American, and Modern. Each EQ model has unique tonal qualities that affect the way the Bass, Mids, and Treble knobs in the Amp section respond.
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 85. 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 97. 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.
