Logic Studio Effects
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Contents Preface 7 8 11 11 An Introduction to the Logic Studio Effects About the Logic Studio Effects About the Logic Studio Documentation Additional Resources Chapter 1 13 13 30 31 37 Amps and Pedals Amp Designer Bass Amp Guitar Amp Pro Pedalboard Chapter 2 53 53 74 74 75 77 Delay Effects Delay Designer Echo Sample Delay Stereo Delay Tape Delay Chapter 3 79 80 81 82 83 83 84 Distortion Effects Bitcrusher Clip Distortion Distortion Effect Distortion II Overdrive Phase Distortion Chapter 4 87 8
101 102 105 108 109 109 Limiter Multipressor Noise Gate Silver Compressor Silver Gate Surround Compressor Chapter 5 113 114 117 118 119 123 129 131 Equalizers Channel EQ DJ EQ Fat EQ Linear Phase EQ Match EQ Single-Band EQs Silver EQ Chapter 6 133 133 139 143 155 159 Filter Effects AutoFilter EVOC 20 Filterbank EVOC 20 TrackOscillator Fuzz-Wah Spectral Gate Chapter 7 163 163 164 167 Imaging Processors Binaural Post-Processing Direction Mixer Stereo Spread Chapter 8 169 169 170 170 171 176 182
199 201 203 204 Rotor Cabinet Effect Scanner Vibrato Effect Spreader Tremolo Effect Chapter 10 205 205 209 210 Pitch Effects Pitch Correction Effect Pitch Shifter II Vocal Transformer Chapter 11 215 216 216 217 220 223 227 Reverb Effects Plates, Digital Reverb Effects, and Convolution Reverb AVerb EnVerb GoldVerb PlatinumVerb SilverVerb Chapter 12 229 230 231 235 241 243 249 Space Designer Convolution Reverb Getting to Know the Space Designer Interface Working with Space Designer’s Impulse Respons
Preface An Introduction to the Logic Studio Effects Logic Studio has an extensive range of digital signal processing (DSP) effects and processors that are used to color or tonally shape existing audio recordings, software instruments, and external audio sources—in real time. These will cover almost every audio processing and manipulation need you will encounter in your day-to-day work. The most common processing options include EQs, dynamic processors, modulations, distortions, reverbs, and delays.
About the Logic Studio Effects The Logic Studio applications are designed for particular types of audio or music work. Similarly, the Logic Studio effects are also designed for specific uses. Given these unique properties and uses, each application provides a custom collection of suitable effects and utilities. Some points to note: • Surround versions of effects, and surround-related parameters that are documented in this manual, are only available in Logic Pro.
Effect category EQ Filter Imaging Metering Modulation Preface Included effects Logic Pro MainStage WaveBurner Expander Yes Yes Yes Limiter Yes Yes Yes Multipressor Yes Yes Yes Noise Gate Yes Yes Yes Silver Compressor Yes Yes Yes Silver Gate Yes Yes Yes Surround Compressor Yes Channel EQ Yes Yes Yes DJ EQ Yes Yes Yes Fat EQ Yes Yes Yes Linear Phase EQ Yes Yes Yes Match EQ Yes Yes Yes Single-Band EQs Yes Yes Yes Silver EQ Yes Yes Yes AutoFilter
Effect category Pitch Reverb Specialized Utility 10 Preface Included effects Logic Pro MainStage WaveBurner Flanger Effect Yes Yes Yes Microphaser Yes Yes Yes Modulation Delay Yes Yes Yes Phaser Effect Yes Yes Yes Ringshifter Yes Yes Yes Rotor Cabinet Effect Yes Yes Yes Scanner Vibrato Effect Yes Yes Yes Spreader Yes Yes Yes Tremolo Effect Yes Yes Yes Pitch Correction Effect Yes Yes Yes Pitch Shifter II Yes Yes Yes Vocal Transformer Yes Yes Yes AVer
About the Logic Studio Documentation Logic Studio comes with various documents that will help you get started as well as provide detailed information about the included applications. • Logic Pro User Manual: This onscreen manual provides comprehensive instructions for using Logic Pro to set up a recording system, compose music, edit audio and MIDI files, and output audio for CD productions.
Release Notes and New Features Documents Each application offers detailed documentation that covers new or changed features and functions. This documentation can be accessed in the following way: • Open the application Help menu and choose Release Notes or New Features. Logic Studio Website For general information and updates, as well as the latest news on Logic Studio, go to: • http://www.apple.
Amps and Pedals 1 Logic Studio 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 re-create vintage and modern tube and solid-state amps. Built-in effect units, such as reverb, tremolo, or vibrato, are also reproduced.
Amp Designer also emulates classic guitar amplifier effects, including spring reverb, vibrato, and tremolo. The Amp Designer interface can be broken down into four general sections in terms of different kinds of parameters. Amp parameters Effects parameters Amp parameters Microphone parameters • Model parameters: The Model pop-up menu is found at the left of the black bar at the bottom. It is used to choose a preconfigured model, consisting of an amplifier, a cabinet, an EQ type, and a microphone type.
µ To switch between full and smaller versions of the interface Click the disclosure triangle between the Cabinet and Mic pop-up menus in the full interface to switch to the smaller version. To switch back to the full interface, click the disclosure triangle beside the Output field in the small interface. You can access all the parameters, with the exception of microphone selection and positioning, in the small interface. Click here in full interface. Click here in small interface.
Tweed Combos The Tweed models are based on American combos from the 1950s and early 1960s that helped define the sounds of blues, rock, and country music. They have warm, complex, clean sounds that progress smoothly through gentle distortion to raucous overdrive as you increase the gain. Even after half a century, Tweeds can still sound contemporary. Many modern boutique amplifiers are based on Tweed-style circuitry.
Tip: While these amps tend toward a clean and tight sound, you can use a Pedalboard distortion stompbox to attain hard-edged crunch sounds with a biting treble and extended low-end definition. See Distortion Pedals and Pedalboard. British Stacks The British Stack models are based on the 50- and 100-watt amplifier heads that have largely defined the sound of heavy rock, especially when paired with their signature 4 x 12" cabinets. At medium gain settings, these amps are great for chunky chords and riffs.
Model Description Small British Combo A 1 x 12" combo with half the power of the British Combo, this amp offers a slightly darker, less open tone. Boutique British Combo A 2 x 12" combo that is a modern take on the original 1960s sound. The tone is thicker, with stronger lows and milder highs than the other British Combos. Tip: Using high Treble and Presence knob settings that might become strident on other amp types can sound great with the British Combos.
Model Description Modern American Stack A powerful, ultra-high gain amp that is ideal for heavy rock and metal. Use the Mids knob to set an ideal amount of scoop or boost. High Octane Stack Although a powerful, high-gain amp, this model offers a smooth transition between gain settings and excellent natural compression. It is a great choice for fast soloing and for two- and three-note chords. Turbo Stack An aggressive-sounding amp with spiky highs and noisy harmonics, especially at high gain settings.
Building a Customized 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, using the Amp, Cabinet, and Mic pop-up menus, located on the black bar at the bottom of the interface. The EQ pop-up menu is accessed by clicking the word EQ or Custom EQ toward the left of the knobs section.
• Additional Combos Choosing an Amp Designer Cabinet Cabinets have a huge impact on the character of a guitar sound (see Amp Designer Cabinet Reference Table). While certain amplifier and cabinet pairings have been popular for decades, departing from them is an effective way to create fresh-sounding tones. For example, most players automatically associate British heads with 4 x 12" cabinets. Amp Designer allows you to drive a small speaker with a powerful head, or to pair a tiny amp with a 4 x 12" cabinet.
Amp Designer Cabinet Reference Table You can choose a cabinet model from the Cabinet pop-up menu on the black bar at the bottom of Amp Designer’s interface. The table below covers the properties of each cabinet model available in Amp Designer. 22 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 that was originally conceived for bassists, but guitarists love its sparkling presence.
Cabinet Description Sunshine 4 x 12 A 4 x 12" closed-back cabinet with a thick, rich mid-range. Sunshine 1 x 12 A single 12" open-back combo amp cabinet with a bright, lively sound that has sweet highs, and transparent mids. 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 fatness of the Blackface 4 x 10 and the brilliance of the British 2 x 12.
Despite these less pleasant-sounding possibilities, you should experiment with different amplifier and EQ combinations because many will sound great together. EQ pop-up menu Bass, Mids, and Treble knobs The EQ parameters include the EQ pop-up menu and the Bass, Mids, and Treble knobs. These parameters are found toward the left-end of the knobs section.
EQ type Description Modern Based on a digital EQ unit popular in the 1980s and 1990s. This EQ is useful for sculpting the hyped highs, booming lows, and scooped mids associated with the era’s rock and metal music styles. Boutique Replicates the tone section of a “retro modern” boutique amp. It excels at precise EQ adjustments, though its tone may be cleaner than desired when used with vintage amplifiers. This EQ is a good choice if you want a cleaner, brighter sound.
Getting to Know Amp Designer’s Effects Parameters The effects parameters include Tremolo, Vibrato, and Reverb, which emulate the processors found on many amplifiers. these controls are found in the center of the knobs section. You can use the switch toward the right to select either Tremolo (TREM), which modulates the amplitude or volume of the sound, or Vibrato (VIB), which modulates the pitch.
• Level knob: Sets the amount of reverb applied to the pre-amplified signal. Amp Designer Reverb Type Reference Table You can choose a reverb type by clicking the Reverb label in the center of the Amp section. The table below covers the properties of each reverb type available in Amp Designer. 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.
• Sync/Free switch: When the switch is set to Sync, the modulation speed is synchronized with the host application tempo. The Speed knob lets you select different bar, beat, and musical note values (1/8, 1/16, and so on, including triplet and dotted-note values). When the switch is set to Free, the modulation speed can be set to any available value with the Speed knob. Setting Amp Designer Microphone Parameters Amp Designer offers a choice between three different virtual microphones.
• Mic pop-up menu: You can choose one of the Microphone models from the pop-up menu: • Condenser: Emulates the sound of a high-end German studio condenser microphone. The sound of condenser microphones is fine, transparent, and well-balanced. • Dynamic: Emulates the sound of popular American dynamic cardioid microphones. This microphone type sounds brighter and more cutting than the Condenser model.
Bass Amp Bass Amp simulates the sound of several famous bass amplifiers. You can route bass guitar and other signals directly through the Bass Amp, reproducing the sound of your musical part played through a number of high-quality bass guitar amplification systems. Bass Amp offers the following parameters. • Model pop-up menu: Includes the following amplifier models: • American Basic: 1970s-era American bass amp, equipped with eight 10" speakers. Well-suited for blues and rock recordings.
• Top Class DI Warm: Famous DI box simulation, well-suited for reggae and pop recordings. Mid frequencies, in the range between 500 and 5000 Hz, are de-emphasized. • Top Class DI Deep: Based on the Top Class DI Warm, this model is well-suited for funk and fusion. The mid frequency range is strongest around 700 Hz. • Top Class DI Mid: Based on the Top Class DI Warm, this model features an almost linear frequency range, with no frequencies emphasized. It is suitable for blues, rock, and jazz recordings.
The Guitar Amp Pro window is organized into sections according to different kinds of parameters. Amp section Microphone Position section Effects section Microphone Type section • Amp section: The model parameters at the top are used to choose the type of amp, EQ model, and speaker. See Building Your Guitar Amp Pro Model. Farther down in the Amp section, the knobs in the V-shaped formation are used to set tone, gain, and level. See Using Guitar Amp Pro’s Gain, Tone, Presence, and Master Controls.
• Choosing a Guitar Amp Pro Equalizer • Setting Guitar Amp Pro Microphone Parameters Choosing a Guitar Amp Pro Amplifier You can choose an amplifier model from the Amp pop-up menu near the top of the interface. • UK Combo 30W: Neutral-sounding amp, well-suited for clean or crunchy rhythm parts. • UK Top 50W: Quite aggressive in the high frequency range, well-suited for classical rock sounds. • US Combo 40W: Clean sounding amp model, well-suited for funk and jazz sounds.
• UK 4 x 12 closed slanted: when used in combination with off-center miking, you will get an interesting mid frequency range; therefore, this model works well when combined with High Gain amps. • US 1 x 10 open back: Not much resonance in the low frequency range. Suitable for use with blues harmonicas. • US 1 x 12 open back 1: Open enclosure of an American lead combo with a single 12" speaker. • US 1 x 12 open back 2: Open enclosure of an American clean/crunch combo with a single 12" speaker.
Using Guitar Amp Pro’s Gain, Tone, Presence, and Master Controls The Gain, Bass, Mids, Treble, Presence, and Master knobs run from left to right in the V-shaped formation in the upper half of the interface. • Gain knob: Sets the amount of pre-amplification applied to the input signal. This control has different effects, depending on which Amp model is chosen. For example, when you are using the British Clean amp model, the maximum Gain setting produces a powerful crunch sound.
Using Guitar Amp Pro’s Tremolo and Vibrato Effects Tremolo and vibrato are controlled by an On button, the FX pop-up menu, the Depth and Speed knobs, and the Sync button in the Effects section. Tremolo modulates the amplitude or volume of the sound, and vibrato modulates the pitch. • FX pop-up menu: You can choose either Tremolo or Vibrato. • Depth knob: Sets the intensity of the modulation. • Speed knob: Sets the speed of the modulation in Hertz.
• Dynamic button: Emulates the sound of a dynamic cardioid microphone. This microphone type sounds brighter and more cutting than the Condenser model. At the same time, the lower-mid frequency range is less pronounced, making this model more suitable for miking rock guitar tones. Tip: Combining both microphone types can sound quite interesting. Duplicate the guitar track, and insert Guitar Amp Pro as an insert effect on both tracks.
All stompbox knobs, switches, and sliders can be automated. Eight Macro controls enable real time changes to any pedal parameter with a MIDI controller. Routing area Macro Controls area Pedal area Pedal Browser • The Pedal Browser shows all pedal effects and utilities. These can be dragged into the Pedal area as part of the signal chain. See Using Pedalboard’s Pedal Browser. This interface area is also used for the alternative import mode. See Using Pedalboard’s Import Mode.
Using Pedalboard’s 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. For information about these types of stompboxes, see Distortion Pedals, Modulation Pedals, Delay Pedals, Filter Pedals, Dynamics Pedals, and Utility Pedals.
Note: Double-clicking a stompbox in the Pedal Browser when a stompbox is selected in the Pedal area will replace the selected pedal. Using 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 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.
2 Click the Select Setting button and select a setting, then click Open. Dependent on the chosen setting, one or more stompboxes appear in the Pedal Browser. The name of the imported setting is shown at the bottom of the Pedal Browser. To add an imported pedal to the Pedal area Do one of the following: µ Drag the stompbox that you want to add from the Pedal Browser to the appropriate Pedal area position. This can be to the left, to the right, or in-between existing pedals.
To add a pedal to the Pedal area Do one of the following: µ Drag the stompbox that you want to insert from the Pedal Browser to the appropriate Pedal area position. This can be to the left, to the right, or in-between existing pedals. µ Ensure that no pedal is selected in the Pedal area, then double-click a stompbox in the Pedal Browser to add it to the right of all existing effects in the Pedal area. Note: You insert Mixer and Splitter utility pedals in a different way.
µ 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 only replace “effect” pedals, not the Mixer or Splitter utilities. Bus routings, if active, are not changed when an effect pedal is replaced. To remove a pedal from the Pedal area Do one of the following: µ µ Drag the pedal out of the Pedal area. Click the pedal to select it and press the Delete key.
µ µ Remove all stompboxes from the Pedal area. This automatically removes an existing Mixer utility. To remove an effect from the second bus Click the name of the pedal (or on either of the gray lines) in the Routing area. Note: The removal of all effects from Bus B does not remove the second bus. The Mixer utility pedal remains in the Pedal area, even when a single stompbox (effect) is in the Pedal area. This allows parallel routing of wet and dry signals.
In Logic Pro and MainStage, you use a controller assignment or create a Workspace knob for “Macro A–H Value.” MIDI hardware switches, sliders, or knobs can then be used to control the mapped Pedalboard Macro A–H target parameters in real time. See the Logic Pro User Manual or the MainStage User Manual for details. Click the triangle at the bottom left to hide or show the Macro Controls area. Note: The Macro Controls area is only available in Logic Pro and MainStage.
Stompbox Description 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. The Squash knob sets the threshold for the internal compression circuit. Contour sets the amount of nonlinear distortion applied to the signal. Mix sets the ratio between the source and distorted signals.
Stompbox Description Vintage Drive Overdrive effect that emulates the distortion produced by a field effect transistor (FET), which is commonly used in solid-state amplifiers. When saturated, FETs generate a warmer sounding distortion than bipolar transistors (such as those emulated by Grinder). Drive sets the saturation amount for the input signal. Tone sets the frequency for the high cut filter, resulting in a softer or harsher tone.
Stompbox Description Retro Chorus A subtle, vintage chorus effect. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling 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. Robo Flanger Flexible flanging effect.
Stompbox Description Total Tremolo A flexible tremolo effect (modulation of the signal level). Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling 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 alter the waveform shape of the LFO.
Stompbox Description Tru-Tape Delay A vintage tape delay effect. The Norm/Reverse switch changes the delay playback direction. Reverse mode is indicated by a blue LED and Normal mode is indicated by a red LED. Hi Cut and Lo Cut activate a fixed frequency filter. Dirt sets the amount of input signal gain, which can introduce an overdriven, saturated quality. Flutter emulates speed fluctuations in the tape transport mechanism.
Stompbox Description Squash Compressor A simple compressor. Sustain sets the threshold level. Signals above this are reduced in level. Level determines the output gain. The Attack switch can be set to Fast for signals with fast attack transients, such as drums, or to Slow for signals with slow attack phases, such as strings. Utility Pedals This section describes the parameters of the Mixer and Splitter pedals.
Delay Effects 2 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 provides control over the following aspects of each tap: • Level and pan position • Highpass and lowpass filtering • Pitch transposition (up or down) Further effect-wide parameters include synchronization, quantization, and feedback. As the name implies, Delay Designer offers significant sound design potential. You can use it for everything from a basic echo effect to an audio pattern sequencer. You can create complex, evolving, moving rhythms by synchronizing the placement of taps.
• Tap parameter bar: Offers a numeric overview of the current parameter settings for the selected tap. You can view and edit the parameters of each tap in this area. See Editing Taps in Delay Designer’s Tap Parameter Bar. • Tap pads: You can use these two pads to create taps in Delay Designer. See Creating Taps in Delay Designer. • Sync section: You can set all Delay Designer synchronization and quantization parameters in this section. See Synchronizing Taps in Delay Designer.
• Toggle buttons: Click to enable or disable the parameters of a particular tap. The parameter being toggled is chosen with the view buttons. The label at the left of the toggle bar always indicates the parameter being toggled. For more information, see Using Delay Designer’s Tap Toggle Buttons. • Tap display: Represents each tap as a shaded line. Each tap contains a bright bar (or dot for stereo panning) that indicates the value of the parameter.
Zooming and Navigating Delay Designer’s Tap Display 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. To zoom the Tap display Do one of the following: µ Vertically drag the highlighted section (the bright rectangle) of the Overview display.
µ To move to different sections of the Tap display Horizontally drag the (middle of the) bright rectangle in the Overview display. The zoomed view in the Tap display updates as you drag. Creating Taps in Delay Designer You can create new delay taps in three different ways: by using the Tap pads, by creating them in the Identification bar, or by copying existing taps. To create taps with the Tap pad 1 Click the upper pad (Start).
µ To create taps in the Identification bar Click at the appropriate position. µ To copy taps in the Identification bar Option-drag a selection of one or more taps to the appropriate position. The delay time of copied taps is set to the drag position. Delay Designer Tap Creation Suggestions The fastest way to create multiple taps is to use the Tap pads.
The Identification bar shows the letter of each visible tap. The Tap Delay field of the Tap parameter bar displays the letter of the currently selected tap, or the letter of the tap being edited when multiple taps are selected (for details, see Selecting Taps in Delay Designer). Selecting Taps in Delay Designer There will always be 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.
µ Open the pop-up menu to the right of the Tap name, and choose the appropriate tap letter. To select multiple taps Do one of the following: µ µ Drag across the background of the Tap display to select multiple taps. Shift-click specific taps in the Tap display to select multiple nonadjacent taps. Moving and Deleting Taps in Delay Designer 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.
µ Select a tap letter in the Identification bar and drag it downward, out of the Tap display. This method also works when more than one tap is selected. µ To delete all selected taps Control-click (or right-click) a tap, and choose “Delete tap(s)” from the shortcut menu. Using Delay Designer’s Tap Toggle Buttons Each tap has its own toggle button in the Toggle bar. These buttons offer you a quick way to graphically activate and deactivate parameters.
Note: The first time you edit a filter or pitch transpose parameter, the respective module automatically turns on. This saves you the effort of manually turning on the filter or pitch transposition module before editing. After you manually turn either of these modules off, however, you need to manually switch it back on. Editing Parameters in Delay Designer’s Tap Display You can graphically edit any tap parameter that is represented as a vertical line in Delay Designer’s Tap display.
Parameter values change to match the mouse position as you drag across the taps. Command-dragging across several taps allows you to draw value curves, much like using a pencil to create a curved line on a piece of paper. Aligning Delay Designer Tap Values You can use Delay Designer’s Tap display to graphically align tap parameter values that are represented as vertical lines. To align the values of several taps 1 Command-click in the Tap display, and move the pointer while holding down the Command key.
The values of taps that fall between the start and end points are aligned along the line. Editing Filter Cutoff in Delay Designer’s Tap Display Whereas the steps outlined in Editing Parameters in Delay Designer’s Tap Display apply to most graphically editable parameters, the Cutoff and Pan parameters work in a slightly different fashion. In Cutoff view, each tap actually shows two parameters: highpass and lowpass filter cutoff frequency.
If the highpass filter’s cutoff frequency value is above that of the lowpass filter cutoff frequency, the filter switches from serial operation to parallel operation, meaning that the tap passes through both filters simultaneously. In this case, the space between the two cutoff frequencies represents the frequency band being rejected—in other words, the filters act as a band-rejection filter.
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. In stereo input/stereo output configurations, the Pan parameter adjusts the stereo balance, not the position of the tap in the stereo field. The Pan parameter appears as a dot on the tap, which represents stereo balance. Drag the dot up or down the tap to adjust the stereo balance. By default, stereo spread is set to 100%.
Editing Taps in Delay Designer’s Tap Parameter Bar The Tap parameter bar provides instant access to all parameters of the chosen tap. The Tap parameter bar also provides access to several parameters that are not available in the Tap display, such as Transpose and Flip. Editing in the Tap parameter bar is fast and precise when you want to edit the parameters of a single tap. All parameters of the selected tap are available, with no need to switch display views or estimate values with vertical lines.
• Pan field: Controls the pan position for mono input signals, stereo balance for stereo input signals, and surround angle when used in surround configurations. • Pan displays a percentage between 100% (full left) and −100% (full right), which represents the pan position or balance of the tap. A value of 0% represents the center panorama position. • When used in surround, a surround panner replaces the percentage representation. For more information, see Working with Delay Designer in Surround.
To reset the value of a tap Do one of the following: µ In the Tap display, Option-click a tap to reset the chosen parameter to its default setting. If multiple taps are selected, Option-clicking any tap will reset the chosen parameter to its default value for all selected taps. µ In the Tap parameter bar, Option-click a parameter value to reset it to the default setting.
Note: Delay Designer offers a maximum delay time of 10 seconds. This means that if you load a setting into a project with a slower tempo than the tempo at which it was created, some taps may fall outside the 10-second limit. In such cases, these taps will not be played but will be retained as part of the setting. • Sync button: Enables or disables synchronized mode. • Grid pop-up menu: Provides several grid resolutions, which correspond to musical note durations.
Use subtle variations of the grid position of every second increment (values between 45% and 55%) to create a less rigid rhythmic feel. This can deliver very human timing variations. Use of extremely high Swing values are unsubtle as they place every second increment directly beside the subsequent increment. Make use of higher values to create interesting and intricate double rhythms with some taps, while retaining the grid to lock other taps into more rigid synchronization with the project tempo.
Note: If Feedback is enabled and you begin creating taps with the Tap pads, Feedback is automatically turned off. When you stop creating taps with the Tap pads, Feedback is automatically re-enabled. • Mix sliders: Independently set the levels of the dry input signal and the post-processing wet signal. Working with Delay Designer in Surround Delay Designer’s design is optimized for use in surround configurations.
Echo This simple echo effect always synchronizes the delay time to the project tempo, allowing you to quickly create echo effects that run in time with your composition. • Time pop-up menu: Sets 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: Determines how often the delay effect is repeated.
Every sample at a frequency of 44.1 kHz is equivalent to the time taken for a sound wave to travel 7.76 millimeters. If you delay one channel of a stereo microphone by 13 samples, this will emulate an acoustic (microphone) separation of 10 centimeters. • Delay slider and field (L and R in stereo version): Determines the number of samples that the incoming signal will be delayed by. • Link L & R button (only in stereo version): Ensures that the number of samples is identical for both channels.
Note: If you use the effect on mono channel strips, the track or bus will have two channels from the point of insertion—all Insert slots after the chosen slot will be stereo. As the parameters for the left and right delays are identical, the descriptions below only cover the left channel—the right channel information is provided in brackets, if named differently. Parameters that are common to both channels are shown separately.
Common Parameters • Beat Sync button: Synchronizes delay repeats to the project tempo, including tempo changes. • Output Mix (Left and Right) sliders and fields: Independently control the left and right channel signals. • Low Cut and High Cut sliders and fields: Frequencies below the Low Cut value and above the High Cut value are filtered out of the source signal.
• Groove slider and field: Determines the proximity of every second delay repeat to the absolute grid position—in other words, how close every second delay repeat is. A Groove setting of 50% means that every delay has the same delay time. Settings below 50% result in every second delay being played earlier in time. Settings above 50% result in every second delay being played later in time. When you want to create dotted note values, move the Groove slider all the way to the right (to 75%).
Distortion Effects 3 You can use Distortion effects to recreate the sound of analog or digital distortion and to radically transform your audio. 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, and they are still used in musical instrument amplifiers today.
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. • Drive slider and field: Sets the amount of gain in decibels applied to the input signal. Note: Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well.
• Displaced: The start, center and end levels of the signal (above the threshold) are offset, resulting in a distortion which is less severe as signal levels cross the threshold. The center portion of the clipped signal is also softer than in Cut mode. • Clip Level slider and field: Sets the point (below the clipping threshold of the channel strip) at which the signal starts clipping. • Mix slider and field (Extended Parameters area): Sets the balance between dry (original) and wet (effect) signals.
• Mix slider and field: Sets the ratio between the effect (wet) signal and original (dry) signals, following the Clip Filter. • Sum LPF knob and field: Sets the cutoff frequency (in Hertz) of the lowpass filter. This processes the mixed signal. • (High Shelving) Frequency knob and field: Sets the frequency (in Hertz) of the high shelving filter. If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a mixer channel strip or a stereo hi-fi amplifier.
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 use it creatively when designing new sounds. • PreGain knob: Sets the amount of gain applied to the input signal. • Drive knob: Sets the amount of saturation applied to the signal. • Tone knob: Sets the frequency of the highpass filter. Filtering the harmonically rich distorted signal produces a softer tone.
• Display: Shows the impact of parameters on the signal. • Tone knob and field: Sets the frequency for the high cut filter. Filtering the harmonically rich distorted signal produces a softer tone. • Output slider and field: Sets the output level. This allows you to compensate for increases in loudness caused by using Overdrive. Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (see Modulation Effects).
• Phase Reverse checkbox (Extended Parameters area): Enable to reduce the delay time on the right channel when input signals that exceed the cutoff frequency are received. Available only for stereo instances of the Phase Distortion effect.
Dynamics Processors 4 The 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.
Types of Dynamics Processors There are four types of dynamics processors included in Logic Studio. These are each used for different audio processing tasks. • Compressors: Logic Studio features a number of downward compressors. These behave like an automatic volume control, lowering the volume whenever it rises above a certain level, called the threshold.
Adaptive Limiter The Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds. It works by rounding and smoothing peaks in the signal, producing an effect similar to an analog amplifier being driven hard. Like an amplifier, it can slightly color the sound of the signal. You can use the Adaptive Limiter to achieve maximum gain, without introducing generally unwanted distortion and clipping, which can occur when the signal exceeds 0 dBFS.
• Out Ceiling knob and field: Sets the maximum output level, or ceiling. The signal will not rise above this. • Output meters (to the right): Show output levels, allowing 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): Choose the type of peak smoothing: • OptFit: Limiting follows a linear curve, which allows signal peaks above 0 dB.
Compressor Parameters The Compressor offers the following parameters: • Circuit Type pop-up menu: Choose the type of circuit emulated by the Compressor. The choices are Platinum, Class(ic) A_R, Class(ic) A_U, VCA, FET, and Opto (optical). • Side Chain Detection pop-up menu: Determines if the Compressor uses the maximum level of each side-chained signal (Max) or the summed level of all side-chained signals (Sum) to exceed or fall below the threshold.
• Compressor Threshold slider and field: Sets the threshold level—signals above this threshold value are reduced in level. • Peak/RMS buttons: Determines whether signal analysis is with the Peak or RMS method, when using the Platinum circuit type. • Gain slider and field: Sets the amount of gain applied to the output signal. • Auto Gain pop-up menu: Choose a value to compensate for volume reductions caused by compression. The choices are Off, 0 dB, and −12 dB.
Setting Suitable Compressor Envelope Times The Attack and Release parameters shape the dynamic response of the Compressor. The Attack parameter determines the time it takes after the signal exceeds the threshold level before the Compressor starts reducing the signal. Many sounds, including voices and musical instruments, rely on the initial attack phase to define the core timbre and characteristic of the sound.
Note: If you activate Auto Gain and RMS simultaneously, the signal may become over-saturated. If you hear any distortion, switch Auto Gain off and adjust the Gain slider until the distortion is inaudible. Using a Side Chain with the Compressor Use of a side chain with a compressor is common. This allows you to use the dynamics (level changes) of another channel strip as a control source for compression.
The Detector parameters are on the left side of the DeEsser window, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider. DeEsser Detector Section • Detector Frequency knob and field: Sets the frequency range for analysis. • Detector Sensitivity knob and field: Sets the degree of responsiveness to the input signal.
Ducker Ducking is a common technique used in radio and television broadcasting: When the DJ or announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level. Ducker provides a simple means of achieving this result with existing recordings. It does not work in real time. Note: For technical reasons, Ducker can only be inserted in output and aux channel strips.
• Attack slider and field: Controls how quickly the volume is reduced. If you want the music mix signal to be gently faded out, set this slider to a high value. This value also controls whether or not the signal level is reduced before the threshold is reached. The earlier this occurs, the more latency is introduced. Note: This only works if the ducking signal is not live—the ducking signal must be an existing recording.
Enveloper The Enveloper is an unusual processor that lets you shape the attack and release phases of a signal—the signal’s transients, in other words. This makes it a unique tool that can be used to achieve results that differ from other dynamic processors. • Threshold slider and field: Sets the threshold level. Signals that exceed the threshold have their attack and release phase levels altered. • (Attack) Gain slider and field: Boosts or attenuates the attack phase of the signal.
Boosting the attack phase can add snap to a drum sound, or it can amplify the initial pluck or pick sound of a stringed instrument. Attenuating the attack causes percussive signals to fade in more softly. You can also mute the attack, making it virtually inaudible. A creative use for this effect is alteration of the attack transients to mask poor timing of recorded instrument parts. Boosting the release phase also accentuates any reverb applied to the affected channel strip.
Expander The Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic range above the threshold level. You can use the Expander to add liveliness and freshness to your audio signals. • Threshold slider and field: Sets the threshold level. Signals above this level are expanded. • Peak/RMS buttons: Determine whether the Peak or RMS method is used to analyze the signal.
Limiter The 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. The Limiter is used primarily when mastering.
Multipressor The Multipressor (an abbreviation for multiband compressor) is an extremely 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 you to apply more compression to the bands that need it, without affecting other bands.
Multipressor Parameters The parameters in the Multipressor window are grouped into three main areas: the graphic display in the upper section, the set of controls for each frequency band in the lower section, and the output parameters on the right. Graphic display section Frequency band section Output section Multipressor Graphic Display Section • Graphic display: Each frequency band is represented graphically. The amount of gain change from 0 dB is indicated by blue bars.
• Expnd Thrsh(old) fields: Set the expansion threshold for the selected band. Setting the parameter to its minimum value (−60 dB), means that only signals that fall below this level are expanded. • Expnd Ratio fields: Set the expansion ratio for the selected band. • Expnd Reduction fields: Set the amount of downward expansion for the selected band. • Peak/RMS fields: Enter a smaller value for shorter peak detection, or a larger value for RMS detection, in milliseconds.
Setting Multipressor Compression Parameters The Compression Threshold and Compression Ratio parameters are the key parameters for controlling compression. Usually the most useful combinations of these two settings are a low Compression Threshold with a low Compression Ratio, or a high Compression Threshold with a high Compression Ratio.
Noise Gate Parameters The Noise Gate has the following parameters. • Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level. • Reduction slider and field: Sets the amount of signal reduction. • Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold. • Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.
Using the 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 the Noise Gate.
The filters allow only very high (loud) signal peaks to pass. In the drum kit example, you could remove the hi-hat signal, which is higher in frequency, with the High Cut filter and allow the snare signal to pass. Turn monitoring off to set a suitable Threshold level more easily. Silver Compressor The Silver Compressor is a simplified version of the Compressor (for usage tips, see Using the Compressor). • Gain Reduction meter: Shows the amount of compression in real time.
Silver Gate The Silver Gate is a simplified version of the Noise Gate (for usage tips, see Using the Noise Gate). • Lookahead slider and field: Sets how far ahead the noise gate analyzes the incoming signal, allowing the Silver Gate to respond more quickly to peak levels. • Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level.
You can link channels by assigning them to one of three groups. When you adjust the threshold or output parameter of any grouped channel, the parameter adjustment is mirrored by all channels assigned to the group. Link section Main section LFE section The Surround Compressor is divided into three sections: • The Link section at the top contains a series of menus where you assign each channel to a group. See Surround Compressor Link Parameters.
• Grp. (Group) pop-up menus: Set group membership for each channel (A, B, C, or no group (indicated by -). Moving the Threshold or Output Level slider for any grouped channel will move the sliders for all channels assigned to that group. Tip: Press Command and Option while moving the Threshold or Output Level slider of a grouped channel to temporarily unlink the channel from the group.
• Release knob and field: Sets the amount of time it takes to return to 0 compression, after the signal falls below the threshold. • Auto button: When the Auto button is enabled, the release time dynamically adjusts to the audio material. • Limiter button: Turns limiting for the main channels on or off. • Threshold knob and field: Sets the threshold for the limiter on the main channels.
Equalizers 5 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 commonly used audio processes, both for music projects and in post-production work for video. You can use EQ to subtly or significantly shape the sound of an audio file, instrument, or project by adjusting specific frequencies or frequency ranges.
Channel EQ The Channel EQ is a highly versatile multiband EQ. It provides eight frequency bands, including lowpass and highpass filters, low and high shelving filters, and four flexible parametric bands. It also features an integrated Fast Fourier Transform (FFT) Analyzer that you can use to view the frequency curve of the audio you want to modify, allowing you to see which parts of the frequency spectrum may need adjustment.
• Resolution pop-up menu: Sets the sample resolution for the Analyzer, with the following menu items: low (1024 points), medium (2048 points), and high (4096 points). Channel EQ Graphic Display Section • Band On/Off buttons: Click to turn the corresponding band on or off. Each button icon indicates the filter type: Band 1 is a highpass filter. Band 2 is a low shelving filter. Bands 3 through 6 are parametric bell filters. Band 7 is a high shelving filter. Band 8 is a lowpass filter.
• Gain-Q Couple Strength pop-up menu (Extended Parameters area): Choose the amount of Gain-Q coupling. • Choose “strong” to preserve most of the perceived bandwidth. • Choose “light” or “medium” to allow some change as you raise or lower the gain. • The asymmetric settings feature a stronger coupling for negative gain values than for positive values, so the perceived bandwidth is more closely preserved when you cut, rather than boost, gain.
Using the Channel EQ Analyzer The Analyzer, when active, makes uses of a mathematical process called a Fast Fourier Transform (FFT) to provide a real-time 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/frequency ranges.
• Low Shelf slider and field: Sets the amount of gain for the low shelving filter. Fat EQ The Fat EQ is a versatile multiband EQ which can be used on individual sources or overall mixes. The Fat EQ provides up to five individual frequency bands, graphically displays EQ curves, and includes a set of parameters for each band. The Fat EQ offers the following parameters. • Band Type buttons: Located above the graphic display.
• Q fields: Sets the Q or bandwidth of each band—the range of frequencies around the center frequency that are altered. At low Q factor values, the EQ covers a wider frequency range. At high Q values, the effect of the EQ band is limited to a narrow frequency range. The Q value can significantly influence how audible your changes are—if you’re working with a narrow frequency band, you’ll generally need to cut or boost more drastically to notice the difference.
Linear Phase EQ Parameters The left side of the Channel EQ window includes the Gain and Analyzer controls. The central area of the window includes the graphical display and parameters for shaping each EQ band. Linear Phase EQ Gain and Analyzer Controls • Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands. • Analyzer button: Turns the Analyzer on or off.
• Graphic display: Shows the current curve of each EQ band. • Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band. • Drag vertically in the section of the display that encompasses each band to adjust the gain of each band (except bands 1 and 8). The display reflects your changes immediately. • Drag the pivot point in each band to adjust the Q factor. Q is shown beside the cursor when the mouse is moved over a pivot point.
Using the Linear Phase EQ The Linear Phase EQ is typically used as a mastering tool and is, therefore, generally inserted into master or output channel strips. The way you use the Linear Phase EQ is obviously dependent on the audio material and what you intend to do with it, but a useful workflow for many situations is as follows: Set the Linear Phase EQ to a flat response (no frequencies boosted or cut), turn on the Analyzer, and play the audio signal.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top parameter, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and −40 dB. The Analyzer display is always dB-linear. Note: When choosing a resolution, be aware that higher resolutions require significantly more processing power.
Match EQ Parameters The Match EQ offers the following parameters. Match EQ Analyzer Parameters • Analyzer button: Turns the Analyzer function on or off. • Pre/Post button: Determines whether the Analyzer looks at the signal before (Pre) or after (Post) the filter curve is applied. • View pop-up menu: Sets the information shown on the graphic display. Choices are: • Automatic: Displays information for the current function, as determined by the active button below the graphic display.
• Select menu (Surround instances only): The Select buttons are replaced by the Select menu, enabling you to choose an individual channel or all channels. Changes to the filter curve will affect the chosen channel when a single channel is selected. • Channel Link slider and field: Refines the settings made with the Select buttons or Select menu. • When set to 100%, all channels (L and R for stereo, or all surround channels) are represented by a common EQ curve.
• A value of 100% has no impact on the filter curve. • Smoothing slider and field: Sets the amount of smoothing for the filter curve, using a constant bandwidth set in semitone steps. A value of 0.0 has no impact on the filter curve. A value of 1.0 means a smoothing bandwidth of one semitone. A value of 4.0 means a smoothing bandwidth of four semitones (a major third). A value of 12.0 means a smoothing bandwidth of one octave, and so on.
5 When you are done, click Current Material Match (this automatically disengages the Current Material Learn button). 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.
• Copy Current Spectrum: Copies the current spectrum to the Clipboard (this can be used by any Match EQ instance in the current project). • Paste Current Spectrum: Pastes the Clipboard contents to the current Match EQ instance. • Load Current Material Spectrum from setting file: Loads the spectrum from a stored setting file. • Generate Current Material Spectrum from audio file: Generates a frequency spectrum for an audio file that you have chosen.
The left scale—and the right, if the Analyzer is inactive—shows the dB values for the filter curve in an appropriate color. Single-Band EQs The sections below provide descriptions for the following Logic Studio single-band EQ effects: • Low Cut and High Cut Filter • High Pass and Low Pass Filter • High Shelving and Low Shelving EQ • Parametric EQ You can find these effects by opening the plug-in menu and choosing EQ > Single Band.
In contrast, the Low Pass Filter affects the frequency range above the selected frequency. • Frequency slider and field: Sets the cutoff frequency. • Order slider and field: Sets the filter order. The more orders used, the stronger the filtering effect. • Smoothing slider and field: Adjusts the amount of smoothing, in milliseconds. High Shelving and Low Shelving EQ The Low Shelving EQ affects only the frequency range that falls below the selected frequency.
Parametric EQ The Parametric EQ is a simple filter with a variable center frequency. It can be used to boost or cut any frequency band in the audio spectrum, either with a wide frequency range, or as a notch filter with a very narrow range. A symmetrical frequency range on either side of the center frequency is boosted or cut. • Gain slider and field: Sets the amount of cut or boost. • Frequency slider and field: Sets the cutoff frequency. • Q-Factor slider and field: Adjusts the Q (bandwidth).
• Gain slider and field: Sets the amount of cut or boost for the parametric EQ. • Low Shelf slider and field: Sets the level of the low shelving EQ. • Low Frequency slider and field: Sets the cutoff frequency for the low shelving EQ.
Filter Effects 6 Filters are used to emphasize or suppress frequencies in an audio signal, resulting in a change to the tonal color of the audio. Logic Studio 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.
Getting to Know the AutoFilter Interface The main areas of the AutoFilter window are the Threshold, Envelope, LFO, Filter, Distortion, and Output parameter sections. Threshold parameter Envelope parameters Filter parameters Output parameters LFO parameters Distortion parameters • Threshold parameter: Sets an input level that—if exceeded—triggers the envelope or LFO, which are used to dynamically modulate the filter cutoff frequency. See AutoFilter Threshold Parameter.
AutoFilter Threshold Parameter The Threshold parameter analyzes the level of the input signal. If the input signal level exceeds the set threshold level, the envelope and LFO are retriggered—this applies only if the Retrigger button is active. The envelope and LFO can be used to modulate the filter cutoff frequency. AutoFilter Envelope Parameters The envelope is used to shape the filter cutoff over time. When the input signal exceeds the set threshold level, the envelope is triggered.
AutoFilter LFO Parameters The LFO is used as a modulation source for filter cutoff. • Coarse Rate knob, Fine Rate slider and field: Used to set the speed of LFO modulation. Drag the Coarse Rate knob to set the LFO frequency in Hertz. Drag the Fine Rate slider (the semicircular slider above the Coarse Rate knob) to fine-tune the frequency. Note: The labels shown for the Rate knob, slider, and field change when you activate Beat Sync. Only the Rate knob (and field) is available.
AutoFilter Filter Parameters The Filter parameters allow you to precisely tailor the tonal color. • Cutoff knob and field: Sets 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 Parameters The Distortion parameters can be used to overdrive the filter input or filter output. The distortion input and output modules are identical, but their respective positions in the signal chain—before and after the filter, respectively—result in remarkably different sounds. • Input knob and field: Sets the amount of distortion applied before the filter section. • Output knob and field: Sets the amount of distortion applied after the filter section.
EVOC 20 Filterbank The 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 Parameters 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 • High and Low Frequency parameters: Determine the lowest and highest frequencies allowed to pass by the filter banks.
• Highest button: Click to determine whether the highest filter band acts as bandpass or lowpass filter. In the Bandpass setting, the frequencies below the lowest bands and above the highest bands are ignored. In the Lowpass setting, all frequencies above the highest bands are filtered. • Resonance knob: Determines the basic sonic character of both filter banks. Increasing Resonance emphasizes the center frequency of each band.
• Waveform buttons: Set the waveform type used by the Shift LFO on the left side or Fade LFO on the right side. You can choose between triangle, falling and rising sawtooth, square up and down around zero (bipolar, good for trills), square up from zero (unipolar, good for changing between two definable pitches), a random stepped waveform (S&H), and a smoothed random waveform for each LFO. • LFO Fade Intensity slider: Controls the amount of Fade AB modulation by the Fade LFO.
• Stereo Width knob: Distributes the output signals of the filter bands in the stereo field. • At the left position, the outputs of all bands are centered. • At the centered position, the outputs of all bands ascend from left to right. • At the right position, the bands are output—alternately—to the left and right channels. EVOC 20 TrackOscillator The EVOC 20 TrackOscillator is a vocoder with a monophonic pitch tracking oscillator.
Vocoding, as a process, is not strictly limited to vocal performances. You could use a drum loop as the analysis signal to shape a string ensemble sound arriving at the synthesis input. How Does a Vocoder Work? The speech analyzer and synthesizer features of a vocoder are actually two bandpass filter banks. Bandpass filters allow a frequency band—a slice—in the overall frequency spectrum to pass through unchanged, and cut the frequencies that fall outside the band’s range.
Getting to Know the EVOC 20 TrackOscillator Interface The EVOC 20 TrackOscillator window is divided into several parameter sections. Synthesis In parameters Analysis In parameters Formant Filter parameters Output parameters Tracking Oscillator parameters LFO parameters U/V Detection parameters • Analysis In parameters: Determine how the input signal is analyzed and used by the analysis filter bank. See EVOC 20 TrackOscillator Analysis In Parameters.
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. You should be as precise as possible with these parameters, to ensure the best possible speech intelligibility and accurate tracking. • Attack knob: Determines how quickly each envelope follower—coupled to each analysis filter band—reacts to rising signals.
Setting the Release Time Longer release times cause the analysis input signal transients to sustain for a longer period, at the vocoder’s output. A long release time on percussive input signals, such as a spoken word or hi-hat part, will translate into a less articulated vocoder effect. Use of extremely short release times results in rough, grainy vocoder sounds. Release values of around 8 to 10 ms are useful starting points.
If speech containing voiced and unvoiced sounds is used as a vocoder’s analysis signal, but the synthesis engine doesn’t differentiate between voiced and unvoiced sounds, the result will sound rather weak. To avoid this problem, the synthesis section of the vocoder must produce different sounds for the voiced and unvoiced parts of the signal. The EVOC 20 TrackOscillator includes an Unvoiced/Voiced detector for this specific purpose.
Important: Take care with the Level knob, particularly when a high Sensitivity value is used, to avoid internally overloading the EVOC 20 TrackOscillator. EVOC 20 TrackOscillator Synthesis In Parameters The Synthesis In section controls various aspects of the tracking signal for the synthesizer. The tracking signal is used to trigger the internal synthesizer. • Synthesis In pop-up menu: Sets the tracking signal source. The choices are: • Oscillator (Osc.
Important: The parameters discussed in this section are available only if the Synthesis In menu is set to Osc. • FM Ratio field: Sets the ratio between Oscillators 1 and 2, which defines the basic character of the sound. Even-numbered values or their multiples produce harmonic sounds, whereas odd-numbered values or their multiples produce inharmonic, metallic sounds. • An FM Ratio of 1.000 produces results resembling a sawtooth waveform. • An FM Ratio of 2.
• Pitch Quantize Glide slider: Sets the amount of time the pitch correction takes, allowing sliding transitions to quantized pitches. • Root/Scale keyboard and pop-up menu: Define the pitch or pitches that the tracking oscillator is quantized to. • Max Track value field: Sets the highest frequency. All frequencies above this threshold are cut, making pitch detection more robust.
The Formant Filter display is divided in two by a horizontal line. The upper half applies to the Analysis section and the lower half to the Synthesis section. Parameter changes are instantly reflected in the Formant Filter display, providing invaluable feedback on what is happening to the signal as it is routed through the two formant filter banks. • High and Low Frequency parameters: Determine the lowest and highest frequencies allowed to pass by the filter section.
• Resonance knob: Resonance is responsible for the basic sonic character of the vocoder—low settings result in a soft character, whereas high settings lead to a more snarling, sharp character. Technically, increasing the Resonance value emphasizes the middle frequency of each frequency band. Using Formant Stretch and Formant Shift Formant Stretch and Formant Shift are significant Formant Filter parameters that you can use separately or in combination (see EVOC 20 TrackOscillator Formant Filter Parameters).
• Waveform buttons: Set the waveform type used by the LFO. You can choose between triangle, falling and rising sawtooth, square up and down around zero (bipolar, good for trills), square up from zero (unipolar, good for changing between two definable pitches), a random stepped waveform (S&H), and a smoothed random waveform for each LFO. • LFO Rate knob and field: Determines the speed of modulation.
• Stereo Mode pop-up menu: Sets the input/output mode of the EVOC 20 Filterbank. The choices are m/s (mono input to stereo output), and s/s (stereo input to stereo output). Note: Set Stereo Mode to m/s if the input signal is mono, and to s/s if the input signal is stereo. In s/s mode, the left and right stereo channels are processed by separate filter banks. When you use m/s mode on a stereo input signal, the signal is first summed to mono before it is passed to the filter banks.
• Wah parameters: Provide control over the type and tone of the wah wah effect. See Wah Parameters. • Auto Wah parameters: Set the depth and envelope times for the automatic wah wah effect. See Auto Wah Parameters. • Fuzz parameters: Set the compression ratio, and control the tone and level of the integrated distortion circuit. See Fuzz Parameters. Effect Order Buttons These buttons determine the signal flow of the Fuzz-Wah effect. Click Wah-Fuzz or Fuzz-Wah to choose the desired flow.
• Morl1: This setting mimics the sound of a popular wah wah pedal. It features a slight peak characteristic. • Morl2: This setting mimics the sound of a popular distortion wah wah pedal. It has a constant Q(uality) Factor setting. • Auto Gain button: The wah wah effect can cause wide variations in the output level. Turning Auto Gain on compensates for this behavior, and limits the output signal dynamics (see Setting the Wah Wah Level with Auto Gain).
Fuzz Parameters These parameters control the integrated distortion and compression circuits. The compressor always precedes the Fuzz effect. • Comp (Compression) Ratio knob: Sets the compression ratio. • Fuzz Gain knob: Sets the level of the Fuzz, or distortion, effect. • Fuzz Tone knob: Adjusts the tonal color of the fuzz effect. Low settings tend to be warmer, and high settings are brighter and harsher. Using the Fuzz-Wah The following section provides practical tips for the Fuzz-Wah parameters.
You can set the upper and lower limits of the range independently by dragging the left and right handles of the slider bracket. You can move the entire range by dragging the center section of the slider bracket. Spectral Gate The 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.
• Center Freq. (Frequency) knob and field: Sets the center frequency of the band that you want to process. • Bandwidth knob and field: Sets the width of the frequency band that you want to process. • Super Energy knob and field: Controls the level of the frequency range above the threshold. • High Level slider and field: Blends the frequencies of the original signal—above the selected frequency band—with the processed signal.
b Use the High Level slider to blend frequencies above the defined frequency band with the processed signal. 5 You can modulate the defined frequency band using the Speed, CF Modulation, and BW Modulation parameters. a Speed determines the modulation frequency. b CF (Center Frequency) Modulation defines the intensity of the center frequency modulation. c BW (Band Width) Modulation controls the amount of bandwidth modulation.
Imaging Processors 7 The Logic Studio Imaging processors are tools for manipulating the stereo image. This enables you 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. This chapter covers the following: • Binaural Post-Processing (p. 163) • Direction Mixer (p. 164) • Stereo Spread (p.
Note: When using multiple Binaural Panners on several channels, you should turn the integrated conditioning off and route the output of all binaurally panned signals to an aux channel. Insert a Binaural Post-Processing plug-in into this aux channel and apply diffuse-field compensation to all Binaural Panner outputs at once. This approach is simpler to manage, better sonically, and reduces computer processing requirements.
The Direction Mixer works with any type of stereo recording, regardless of the miking technique used. For information about XY, AB, and MS recordings, see Getting to Know Stereo Miking Techniques. • Input buttons: Click the LR button if the input signal is a standard left/right signal, and click the MS button if the signal is middle and side encoded. • Spread slider and field: Determines the spread of the stereo base in LR input signals. Determines the level of the side signal in MS input signals.
Using the Direction Mixer’s Direction Parameter 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. The following applies when working with LR signals: • At 90°, the center of the stereo base is panned hard left. • At −90°, the center of the stereo base is panned hard right.
Understanding 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 extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels. This is done alternately—middle frequencies to the left channel, middle frequencies to the right channel, and so on. This greatly increases the perception of stereo width without making the sound totally unnatural, especially when used on mono recordings.
Metering Tools 8 You can use the Metering tools to analyze audio in a variety of ways. These plug-ins offer different facilities to the meters shown in channel strips. They have no effect on the audio signal and are designed 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.
The LED shows the current analysis status. If the LED is flashing, a tempo measurement is taking place. When the LED is continuously lit, analysis is complete, and the tempo is displayed. The measurement ranges from 80 to 160 beats per minute. The measured value is displayed with an accuracy of one decimal place. Click the LED to reset the BPM Counter. Note: The BPM Counter also detects tempo variations in the signal and tries to analyze them accurately.
Stereo instances of the Level Meter show independent left and right bars, whereas mono instances display a single bar. Surround instances display a bar for each channel—in a vertical rather than horizontal orientation. The current peak values are displayed numerically, superimposed over the graphic display. You can reset these values by clicking in the display. The Level Meter can be set to display levels using Peak, RMS, or Peak & RMS characteristics.
You can view either the Analyzer or Goniometer results in the main display area. You switch the view and set other MultiMeter parameters with the controls on the left side of the interface. Main display in Analyzer view Analyzer parameters Peak parameters Goniometer parameters Level Meter Correlation Meter While you can insert the MultiMeter directly into any channel strip, it is more commonly used in the master channel strip of the host application—when you are working on the overall mix.
Using the MultiMeter Analyzer In Analyzer mode, the 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 • Analyzer button: Switches the main display to Analyzer mode.
Using the MultiMeter Goniometer 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 cancellations along the center line (M—mid/mono). The idea of the goniometer was born with the advent of early two-channel oscilloscopes.
Using the MultiMeter’s 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.
• When the Correlation Meter moves into the red area to the left of the center position, out-of-phase material is present. This will lead to phase cancellations if the stereo signal is combined into a mono signal. Using the MultiMeter Peak Parameters The MultiMeter Peak parameters are used to enable/disable the peak hold function and to reset the peak segments of all meter types. You can also determine a temporary peak hold duration.
Although you can insert the Surround MultiMeter directly into any channel strip, it is more commonly used in the master channel strip of the host application—when you are working on the overall surround mix. Analyzer parameters Goniometer parameters Peak parameters Balance/Correlation button Main display (Goniometer shown) Using the Surround MultiMeter Analyzer In Analyzer mode, the MultiMeter’s main display shows the frequency spectrum of the input signal as 31 independent frequency bands.
• Sum and Max buttons: Determine whether a summed or maximum level is displayed in the Analyzer results in the main display. These buttons are relevant only when multiple channels are selected with the channel buttons. • Channel buttons: Used to select a single channel or a combination of channels for metering. The number and appearance of these buttons varies when different surround modes are chosen.
Because the Surround MultiMeter Goniometer is dealing with multichannel signals, the display is divided into multiple segments, as shown in the image. Each segment indicates a speaker position. When the surround panner is moved in a channel strip, the indicator changes accordingly. This indicates not only left and right channel coherence, but also the front-to-rear coherence. • Goniometer button: Displays the Goniometer results in the main display.
Using the Surround 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.
Depending on the chosen surround format, a number of points that indicate speaker positions are shown (L, R, C, Ls, Rs in a 5.1 configuration is displayed in the figure). Lines connect these points. The center position of each connecting line is indicated by a blue marker. A gray ball indicates the surround field/sound placement. As you move the surround panner of the channel strip, the ball in the Correlation Meter mirrors your movements.
Note: This meter must be manually opened by clicking on the Balance/Correlation button. • Hold Time pop-up menu: When peak hold is active, sets the hold time for all metering tools to 2, 4, or 6 seconds—or infinite. • Reset button: Click to reset the peak hold segments of all metering tools. Tuner You can tune instruments connected to your system with the Tuner utility.
• Keynote/Octave displays: The upper, Keynote display shows the target pitch of the note being played (the closest tuned pitch). The lower, Octave display indicates the octave that the incoming note falls into. This matches the MIDI octave scale, with the C above middle C displayed as C4, and middle C displayed as C3. • Tuning Adjustment slider and field: Sets the pitch of the note used as the basis for tuning. By default, the Tuner is set to the project’s Tuning parameter value.
Modulation Effects 9 Modulation effects are used to add motion and depth to your sound. Effects such as chorus, flanging, and phasing are well-known examples. 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.
• Spreader (p. 203) • Tremolo Effect (p. 204) Chorus Effect The Chorus effect delays the original signal. The delay time is modulated with an LFO. The delayed, modulated signal is 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 Ensemble effect can add a great deal of richness and movement to sounds, particularly when you use a high number of voices. It is very useful for thickening parts, but it can also be used to emulate more extreme pitch variations between voices, resulting in a detuned quality to processed material. • Intensity sliders and fields: Set the amount of modulation for each LFO. • Rate knobs and fields: Control the frequency of each LFO.
Note: When you are using the Ensemble effect in surround, the input signal is converted to mono before processing. In other words, you insert the Ensemble effect as a multi-mono instance. 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.
Microphaser The Microphaser is a simple plug-in that allows you to quickly create swooshing, phasing effects. • LFO Rate slider and field: Defines the frequency (the speed) of the LFO. • Feedback slider and field: Determines the amount of the effect signal that is routed back into the input. This can change the tonal color and/or make the sweeping effect more pronounced. • Intensity slider and field: Determines the amount of modulation.
Although rich, combined flanging and chorus effects are possible, the Modulation Delay is capable of producing some extreme modulation effects. These include emulations of tape speed fluctuations and metallic, robot-like modulations of incoming signals. • Feedback slider and field: Determines the amount of the effect signal that is routed back to the input. If you’re going for radical flanging effects, enter a high Feedback value. If simple doubling is what you’re after, don’t use any feedback.
• LFO Phase knob and field: Available only in stereo and surround instances, it controls the phase relationship between individual channel modulations. • At 0°, the extreme values of the modulation are achieved simultaneously for all channels. • 180° or −180° is equal to the greatest possible distance between the modulation phases of the channels. Note: The LFO Phase parameter is available only if the LFO Left Right Link button is active.
Phaser Effect The Phaser effect combines the original signal with a copy that is slightly out of phase with the original. This means that the amplitudes of the two signals reach their highest and lowest points at slightly different times. The timing differences between the two signals are modulated by two independent LFOs. In addition, the Phaser includes a filter circuit and a built-in envelope follower that tracks volume changes in the input signal, generating a dynamic control signal.
Phaser LFO Section • LFO 1 and LFO 2 Rate knobs and fields: Set the speed for each LFO. • LFO Mix slider and fields: Determines the ratio between the two LFOs. • Env Follow slider and field: Determines the impact of incoming signal levels on the speed of LFO 1. • Phase knob and field: Available only in stereo and surround instances. Controls the phase relationship between the individual channel modulations. At 0°, the extreme values of the modulation are achieved simultaneously for all channels.
Getting to Know the Ringshifter Interface The Ringshifter interface consists of six main sections. Mode buttons Oscillator parameters Delay parameters Envelope follower parameters Output parameters LFO parameters • Mode buttons: Determine whether the Ringshifter operates as frequency shifter or ring modulator See Setting the Ringshifter Mode.
Setting the Ringshifter Mode The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator. • Single (Frequency Shifter) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted up (positive value) or down (negative value).
• In the ring modulator OSC mode, the Frequency parameter controls the frequency content (timbre) of the resulting effect. This timbre can range from subtle tremolo effects to clangorous metallic sounds. • Frequency control: Sets the frequency of the sine oscillator.
• Sync button: Synchronizes the delay to the project tempo. You can choose musical note values with the Time knob. • Level knob and field: Sets the level of the delay added to the ring-modulated or frequency-shifted signal. A Level value of 0 passes the effect signal directly to the output (bypass).
Modulating the Ringshifter with the LFO The oscillator Frequency and Dry/Wet parameters can be modulated with the LFO—and the envelope follower (see Modulating the Ringshifter with the Envelope Follower). The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. The LFO produces continuous, cycled control signals. • Power button: Turns the LFO on or off and enables the following parameters.
• Feedback knob and field: Sets the amount of the signal that is routed back to the effect input. Feedback adds an edge to the Ringshifter sound and is useful for a variety of special effects. It produces a rich phasing sound when used in combination with a slow oscillator sweep. Comb filtering effects are created by using high Feedback settings with a short delay time (less than 10 ms).
• Cabinet Type pop-up menu: You can choose from the following cabinet models: • Wood: Mimics a Leslie with a wooden enclosure, and sounds like the Leslie 122 or 147 models. • Proline: Mimics a Leslie with a more open enclosure, similar to a Leslie 760 model. • Single: Simulates the sound of a Leslie with a single, full-range rotor. The sound resembles the Leslie 825 model. • Split: The bass rotor’s signal is routed slightly to the left, and the treble rotor’s signal is routed more towards the right.
• 910: The 910, or Memphis mode, stops the bass drum rotation at slow speed, while the speed of the horn compartment can be switched. This may be desirable, if you’re after a solid bass sound but still want treble movement. • Sync: The acceleration and deceleration of the horn and bass drums are roughly the same. This sounds as if the two are locked, but the effect is clearly audible only during acceleration or deceleration. • Rotor Fast Rate slider: Adjust to set the maximum possible rotor speed (Tremolo).
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. These allow you to set the modulation speed independently for the left and right channels. The stereo parameters of the mono version of the Scanner Vibrato are hidden behind a transparent cover. • Vibrato knob: Use to choose from three Vibrato positions (V1, V2, and V3) or three Chorus positions (C1, C2, and C3).
Spreader Spreader widens the stereo spectrum of a signal. The Spreader effect periodically shifts the frequency range of the original signal, thus changing the perceived width of the signal. The delay between channels can also be specified (in samples), adding to the perceived width and channel separation of a stereo input signal. • Intensity slider and field: Determines the modulation amount. • Speed knob and field: Defines the frequency of the built-in LFO, and therefore the speed of the modulation.
Tremolo Effect The Tremolo effect modulates the amplitude of the incoming signal, resulting in periodic volume changes. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The graphic display shows all parameters, except Rate. • Depth slider and field: Determines the modulation amount. • Waveform display: Shows the resulting waveform. • Rate knob and field: Sets the frequency of the LFO.
Pitch Effects 10 You can use the Pitch effects of Logic Studio 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. This chapter covers the following: • Pitch Correction Effect (p. 205) • Pitch Shifter II (p. 209) • Vocal Transformer (p. 210) Pitch Correction Effect You can use the Pitch Correction effect to correct the pitch of incoming audio signals.
Pitch Correction Parameters The Pitch Correction effect offers the following parameters. • Use Global Tuning button: Enable to use the project’s Tuning settings for the pitch correction process. If disabled, you can use the Ref. Pitch field to freely set the desired reference tuning. See Setting the Pitch Correction Reference Tuning. • Normal and Low buttons: These determine the pitch range that is scanned (for notes that need correction). See Defining the Pitch Correction Effect’s Quantization Grid.
• Correction Amount display: Indicates the amount of pitch change. The red marker indicates the average correction amount over a longer time period. You can use the display when discussing (and optimizing) the vocal intonation with a singer during a recording session. • Response slider and field: Determines how quickly the voice reaches the corrected destination pitch. Singers use portamenti and other gliding techniques.
Excluding 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 will be 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.
Note: Tunings that differ from software instrument tuning can be interesting when you want to individually correct the notes of singers in a choir. If all voices were individually and perfectly corrected to the same pitch, the choir effect would be partially lost. You can prevent this by (de)tuning the pitch corrections individually. Automating the Pitch Correction Effect The Pitch Correction effect can be fully automated.
• Timing pop-up menu (Extended Parameters area): Determines how timing is derived: by following the selected algorithm (Preset), by analyzing the incoming signal (Auto), or by using the settings of the Delay, Crossfade, and Stereo Link parameters, described below (Manual). Note: The following three parameters are active only when “Manual” is chosen in the Timing pop-up menu. • Delay slider and field (Extended Parameters area): Sets the amount of delay applied to the input signal.
Vocal Transformer Parameters The Vocal Transformer offers the following parameters. • Pitch knob and field: Determines the amount of transposition applied to the input signal. See Setting Vocal Transformer Pitch and Formant Parameters. • Robotize button: Enables Robotize mode, which is used to augment, diminish, or mirror the melody. See Using Vocal Transformer’s Robotize Mode.
Setting Vocal Transformer Pitch and Formant Parameters Use the Vocal Transformer’s Pitch parameter to transpose the pitch of the signal upward or downward. Adjustments are made in semitone steps. Incoming pitches are indicated by a vertical line below the Pitch Base field. Transpositions of a fifth upward (Pitch = +7), a fourth downward (Pitch = −5), or by an octave (Pitch = ±12) are the most useful, harmonically. As you alter the Pitch parameter, you might notice that the formants don’t change.
The Tracking slider and field feature is enhanced by four buttons which immediately set the slider to the most useful values, as follows: • −1 (sets the slider to −100%): All intervals are mirrored. • 0 (sets the slider to 0%): Delivers interesting results, with every syllable of the vocal track being sung at the same pitch. Low values turn sung lines into spoken language. • 1 (sets the slider to 100%): The range of the melody is maintained. Higher values augment, and lower values diminish, the melody.
11 Reverb Effects 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).
Plates, Digital Reverb Effects, and Convolution Reverb The first form of reverb used in music production was actually a special room with hard surfaces, called an echo chamber. It was used to add echoes to the signal. Mechanical devices, including metal plates and springs, were also used to add reverberation to the output of musical instruments and microphones. Digital recording introduced digital reverb effects, which consist of thousands of delays of varying lengths and intensities.
• Density/Time slider and field: Determines both the density and duration of the reverb. Low values tend to generate clearly discernible early reflection clusters, generating something similar to an echo. High values result in a more reverb-like effect. • Mix slider and field: Sets the balance between the effect (wet) and direct (dry) signals. EnVerb EnVerb is a versatile reverb effect with a unique feature: It allows you to freely adjust the envelope—the shape—of the diffuse reverb tail.
EnVerb Time Parameters EnVerb offers the following Time parameters: • Dry Signal Delay slider and field: Determines 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%. • Predelay knob and field: Sets the time between the original signal and the starting point of the reverb attack phase—the very beginning of the first reflection. • Attack knob and field: Defines the time it takes for the reverb to climb to its peak level.
EnVerb Sound Parameters EnVerb offers the following tone control parameters: • Density slider and field: Sets the reverb density. • Spread slider and field: Controls the stereo image of the reverb. At 0% the effect generates a monaural reverb. At 200% the stereo base is artificially expanded. • High Cut slider and field: Frequencies above the set value are filtered out of the reverb tail.
GoldVerb 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 The interface is broken down 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. See GoldVerb Early Reflections Parameters.
GoldVerb Early Reflections Parameters The GoldVerb offers the following Early Reflections parameters: • Predelay slider and field: Determines the amount of 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 The GoldVerb offers the following Reverb parameters: • Initial Delay slider and field: Sets the time between the original signal and the diffuse reverb tail. If you’re going for a natural-sounding, harmonic reverb, the transition between the early reflections and the reverb tail should be as smooth and seamless as possible. Set the Initial Delay parameter so that it is as long as possible, without a noticeable gap between the early reflections and the reverb tail.
PlatinumVerb The PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easy to precisely emulate real rooms. Its dual-band Reverb section splits the incoming signal into two bands, each of which is processed and can be edited separately.
PlatinumVerb Early Reflections Parameters The PlatinumVerb offers the following Early Reflections parameters: • Predelay slider and field: Determines the amount of 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.
PlatinumVerb Reverb Parameters The PlatinumVerb offers the following Reverb parameters: • Initial Delay slider and field: Sets the time between the original signal and the diffuse reverb tail. • Spread slider and field: Controls the stereo image of the reverb. At 0%, the effect generates a monaural reverb. At 200%, the stereo base is artificially expanded. • Crossover slider and field: Defines the frequency at which the input signal is split into two frequency bands, for separate processing.
• Diffusion slider and field: Sets the diffusion of the reverb tail. High Diffusion values represent a regular density, with few alterations in level, times, and panorama position over the course of the diffuse reverb signal. Low Diffusion values result in the reflection density becoming irregular and grainy. This also affects the stereo spectrum. As with Density, find the best balance for the signal. • Reverb Time slider and field: Determines the reverb time of the high band.
SilverVerb The SilverVerb is similar to the AVerb, but it provides an additional 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 frequencies usually sound somewhat unpleasant, hamper speech intelligibility, or mask the overtones of the original signals. Long reverb tails with a lot of bottom end generally result in an indistinct mix.
Space Designer Convolution Reverb 12 Space Designer is a convolution reverb effect. You can use it 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 (IR) reverb sample. An impulse response is a recording of a room’s reverb characteristics—or, to be more precise, a recording of all reflections in a given room, following an initial signal spike.
• Automating Space Designer (p. 249) Getting to Know the 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: Used to load, save, or manipulate (recorded or synthesized) impulse response files.
Working with Space Designer’s Impulse Response Parameters Space Designer can use either recorded impulse response files or its own 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), load or create impulse responses, and set the sample rate and length.
Important: To convolve audio in real time, Space Designer must first calculate any parameter adjustments to the impulse response. This requires a moment or two, following parameter edits, and is indicated by a blue progress bar. During this parameter edit processing time you can continue to adjust the parameter. When calculation starts, the blue bar is replaced by a red bar, advising you that calculation is taking place.
Any mono, stereo, AIFF, SDII, or WAV file can be used as an IR. In addition, surround formats up to 7.1, discreet audio files, and B-format audio files that comprise a single surround IR can also be used. Working in Space Designer’s Synthesized IR Mode In Synthesized IR mode, Space Designer generates a synthesized impulse response based on the values of the Length, envelope, Filter, EQ, and Spread parameters.
• If the project sample rate is 44.1 kHz, the options will be 22.05 kHz, 11.025 kHz, and 5512.5 Hz. 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 IR sample actually contains higher frequencies. When you are reducing the sample rate, use your ears to decide if the sonic quality meets your needs.
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, saving you time and effort. When you are using an impulse response file, the Length parameter value cannot exceed the length of the actual impulse response sample. Longer impulse responses (sampled or synthesized) place a higher strain on the CPU.
• All button: Resets all envelopes and the EQ to default values. • Volume Env button: Displays the volume envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Volume Envelope. • Filter Env button: Displays the filter envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Filter.
Setting Space Designer’s Envelope Parameters You can edit the volume and filter envelopes of all IRs and the density envelope of synthesized IRs. All envelopes can be adjusted both graphically in the main display and numerically in the parameter bar. Whereas some parameters are envelope-specific, all envelopes consist of the Attack Time and Decay Time parameters.
Working with Space Designer’s Volume Envelope The volume envelope is used to set the reverb’s initial level and adjust how the volume will change over time. You can edit all volume envelope parameters numerically, and many can also be edited graphically (see Setting Space Designer’s Envelope Parameters). Init Level node Decay Time/End Level node Attack/Decay Time node • Init Level field: Sets the initial volume level of the impulse response attack phase.
Using Space Designer’s 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 Setting Space Designer’s Envelope Parameters. Note: The density envelope is available only in Synthesized IR mode.
Working with Space Designer’s EQ Space Designer features a four-band EQ comprised 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: Enables or disables the entire EQ section. • Individual EQ band buttons: Enable or disable individual EQ bands.
2 Drag the cursor horizontally over the main display. When the cursor is in the access area of a band, the corresponding curve and parameter area is automatically highlighted and a pivot point is displayed. 3 Drag horizontally to adjust the frequency of the band. 4 Drag vertically to increase or decrease the Gain of the band. 5 Vertically drag the (illuminated) pivot point of a parametric EQ band to raise or lower the Q value.
Using Space Designer’s Main Filter Parameters The main filter parameters are found at the lower-left corner of the interface. • Filter On/Off button: Switches the filter section on and off. • Filter Mode knob: Determines the filter mode. • 6 dB (LP): Bright, good general-purpose filter mode. It can be used to retain the top end of most material, while still providing some filtering. • 12 dB (LP): Useful where you want a warmer sound, without drastic filter effects.
Note: Activation of the filter envelope automatically enables the main filter. Controls the Attack Time endpoint (and Decay Time startpoint) and Break Level parameters simultaneously. Controls the Decay endpoint and End Level parameters simultaneously. • Init Level field: Sets the initial cutoff frequency of the filter envelope. • Attack Time field: Determines the time required to reach the Break Level (see below). • Break Level field: Sets the maximum filter cutoff frequency that the envelope reaches.
Space Designer Global Parameters: Upper Section These parameters are found around the main display. Output sliders Input slider Latency Compensation button Definition area Rev Vol Compensation button • Input slider: Determines how Space Designer processes a stereo or surround input signal. For more information, see Using Space Designer’s Input Slider. • Latency Compensation button: Switches Space Designer’s internal latency compensation feature on or off.
• Spread and Xover knobs (synthesized IRs only): Spread adjusts the perceived width of the stereo or surround field. Xover sets the crossover frequency in Hertz. Any synthesized impulse response frequency that falls below this value will be affected by the Spread parameter. See Using Space Designer’s Spread Parameters. Using Space Designer’s Input Slider The Input slider behaves differently in stereo or surround instances. The slider does not appear in mono or mono to stereo instances.
Using Space Designer’s Latency Compensation Feature The complex calculations made by Space Designer take time. This time results in a processing delay, or latency, between the direct input signal and the processed output signal. When activated, the Latency Compensation feature delays the direct signal (in the Output section) to match the processing delay of the effect signal. Note: This is not related to latency compensation in the host application.
Using Space Designer’s Rev Vol Compensation Rev Vol Compensation (Reverb Volume Compensation) attempts to match the perceived (not actual) volume differences between impulse response files. It is enabled by default and should generally be left in this mode, although you may find that it isn’t successful with all types of impulse responses. If this is the case, turn it off and adjust input and output levels accordingly.
Space Designer Surround Output Configuration Parameters • C(enter) slider: Adjusts the output level of the center channel independently of other surround channels. • Bal(ance) slider: Sets the level balance between the front (L-C-R) and rear (Ls-Rs) channels. • In 7.1 ITU surround, the balance pivots around the Lm-Rm speakers, taking the surround angles into account. • With 7.1 SDDS surround, the Lc-Rc speakers are considered front speakers.
This can be useful for eliminating level peaks at the beginning of the impulse response sample. Its use also affords a number of creative options, particularly when combined with the Reverse function. See Using Space Designer’s Button Bar. Note: The IR Start parameter is not available or required in Synthesized IR mode because, by design, the Length parameter provides identical functionality.
You can, however, record, edit, and play back any movement of the following Space Designer parameters in a suitable host application: • Stereo Crossfeed • Direct Output • Reverb Output 250 Chapter 12 Space Designer Convolution Reverb
Specialized Effects and Utilities 13 Logic Studio includes a bundle of specialized effects and utilities designed to address tasks often encountered during audio production. As examples of where these processors can help: Denoiser eliminates or reduces noise below a threshold level. Enhance Timing enhances the timing of audio recordings. Exciter can add life to your recordings by generating artificial high frequency components. Grooveshifter enabes you to create rhythmic variations in your recordings.
2 Play the audio signal and set the Reduce value to the point where noise reduction is optimal but little of the appropriate signal is reduced. 3 If you encounter artifacts, use the smoothing parameters. Denoiser Main Parameters The Denoiser offers the following main parameters: Threshold slider and field Noise Type slider and field Graphic display Reduce slider and field • Threshold slider and field: Sets the threshold level. Signals that fall below this level are reduced by the Denoiser.
Note: If the noise floor of your recording is very high (more than −68 dB), reducing it to a level of −83 to −78 dB should be sufficient, provided this doesn’t introduce any audible side effects. This effectively reduces the noise by more than 10 dB, to less than half of the original (noise) volume. • Noise Type slider and field: Determines the type of noise that you want to reduce. • A value of 0 equals white noise (equal frequency distribution).
While effective on suitable material, this type of real-time quantization has some limitations. It does not work well on recordings of performances that have been played too far off the beat. The same is true for very complex, layered drum tracks. It will, however, provide noticeable timing improvements on reasonably tight percussive and melodic material (played in an eighth or quarter note feel).
You can use the Exciter to add life to recordings. It is especially well suited to audio tracks with a weak treble frequency range. The Exciter is also useful as a general tool for enhancing guitar tracks. • Frequency display: Shows the frequency range used as the source signal for the excite process. • Frequency slider and field: Sets the cutoff frequency (in Hertz) of the highpass filter. The input signal passes through the filter before (harmonic) distortion is introduced.
Note: The Grooveshifter relies on perfect matching of the project tempo with the tempo of the treated recording. Any tempo variations deliver less precise results. Grooveshifter Source Material Parameters • Beat and Tonal buttons: Switch between two algorithms, each optimized for different types of audio material. • Beat algorithm: Optimized for percussive input material. The Grain Size slider has no effect when Beat is chosen. • Tonal algorithm: Optimized for tonal input material.
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. • Denoise slider and field: Determines (your estimation of ) the noise floor in your recording and, therefore, how much noise should be eliminated. Settings towards 100 dB allow more noise to pass.
SubBass The SubBass plug-in generates frequencies below those of the original signal, resulting in artificial bass content. The simplest use for the 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. See Using SubBass. SubBass creates two bass signals, derived from two separate portions of the incoming signal.
• High Bandwidth knob and field: Sets the width of the upper band. • Graphic display: Shows the selected upper and lower frequency bands. • Freq. Mix slider and field: Adjusts the mix ratio between the upper and lower frequency bands. • Low Ratio knob and field: Adjusts the ratio between the generated signal and the original lower band signal. • Low Center knob and field: Sets the center frequency of the lower band. • Low Bandwidth knob and field: Sets the width of the lower band.
Utilities and Tools 14 The tools found in the Utility category can help with routine tasks and situations that you may encounter during production, such as the following: Gain plug-ins are used to adjust the level or phase of input signals. I/O Utility enables you to integrate external audio effects into your host application mixer. Test Oscillator generates a static frequency or sine sweep. This chapter covers the following: • Down Mixer (p. 261) • Gain Plug-in (p. 262) • I/O Utility (p.
Channel mapping, panning, and downmixing are automatically handled behind the scenes. You do, however, have some control over the mix. Use the Level sliders to control the respective channel levels. The number (and names) of sliders is dependent on the chosen plug-in format. Gain Plug-in Gain amplifies (or reduces) the signal by a specific decibel amount.
Using Phase Inversion Inverting phase is useful for dealing with time alignment problems, particularly those caused by simultaneous recording with multiple microphones. When you invert the phase of a signal heard in isolation, it sounds identical to the original. When the signal is heard in conjunction with other signals, however, phase inversion may have an audible effect.
Note: Bypassing any latency-inducing plug-ins on the track will provide you with the most accurate reading. • Latency Offset field and slider: Displays the value for the detected latency between the selected output and input. Also allows you to offset the latency manually. To integrate and use an external effects unit with the I/O utility 1 Connect an output (or output pair) of your audio interface with the input (pair) on your effects unit.
Multichannel Gain Multichannel Gain allows you to independently control the gain (and phase) of each channel in a surround mix. • Master slider and field: Sets the master gain for the combined channel output. • Channel gain sliders and fields: Set the amount of gain for the respective channel. • Phase Invert buttons: Invert the phase of the selected channel. • Mute buttons: Mute the selected channel.
In the first mode (default mode), it starts generating the test signal as soon as it is inserted. You can switch it off by bypassing it. In the second mode (activated by clicking the Sine Sweep button), Test Oscillator generates a user-defined frequency spectrum tone sweep—when triggered with the Trigger button. • Waveform buttons: Select the type of waveform to be used for test tone generation.
