Final-Cut-Pro-Logic-Effects 2
Final-Cut-Pro-Logic-Effects 2
Logic Effects
for Mac
Contents
Overview 5
Distortion effects 9
Echo effects 20
Equalizers 43
Intro to equalizers 43
AutoFilter 44
Channel EQ 49
Linear Phase EQ 53
Vintage EQ collection 57
Modulation effects 88
Pitch effects 95
Copyright 159
The most common processing options include EQs, levels (dynamic processors),
modulations, distortions, spaces (reverbs), and echo (delays).
Further advanced features include precise signal meters and analyzers, noise reduction,
bass enhancement, and vocal effects.
All effects, processors, and utilities provide an intuitive interface that simplifies operation,
enabling you to work quickly. Outstanding audio quality is assured when needed, or—at the
other end of the spectrum—extreme processing is possible when you need to radically alter
your audio. All effects and processors are highly optimized for efficient CPU usage.
Distortion Bitcrusher
Clip Distortion
Distortion
Distortion II
Overdrive
Phase Distortion
Ringshifter
EQ AutoFilter
Channel EQ
Linear Phase EQ
Vintage EQ collection
Modulation Chorus
Ensemble
Flanger
Phaser
Scanner Vibrato
Tremolo
Reverb ChromaVerb
SilverVerb
Space Designer
Legacy DeEsser
Denoiser
Fat EQ
PlatinumVerb
2. In the Effects browser, select a type of audio effect, then select an effect from the Logic
category on the right.
• Preview what an effect sounds like using the audio from the currently selected
timeline clip: Move the pointer over the audio effect thumbnail.
• Filter the list of effects that appear: Type text in the Effects browser search field.
• Drag the effect to an audio clip (or a video clip with audio) in the timeline.
• Double-click the effect icon to apply the effect to the selected clip.
The effect appears in the Effects section of the Audio inspector and in the Audio Animation
editor.
2. To open the Audio inspector, click the Inspector button on the right side of the toolbar
(or press Command-4), then click the Audio button at the top.
You can preview your adjustments by using the skimmer or playing the clip in the
timeline.
To return the effect’s values to their default settings, click the effect’s Reset button .
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products, go to the general Apple Support website. You’ll also have access to product
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frequently asked questions about Final Cut Pro, go to:
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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. When
overdriven, they produce a type of distortion that many people find musically pleasing, and
that has become a familiar part of the sound of rock and pop music. Analog tube distortion
adds a distinctive warmth and bite to the signal.
There are also distortion effects that intentionally cause clipping and digital distortion
of the signal. These can be used to modify vocal, music, and other clips to produce an
intense, unnatural effect, or to create sound effects.
Distortion effects include parameters for tone, which let you shape the way the distortion
alters the signal (often as a frequency-based filter), and for gain, which let you control how
much the distortion alters the output level of the signal.
WARNING: When set to high output levels, distortion effects can damage your hearing—
and your speakers. When you adjust effect settings, it is recommended that you lower the
output level of the clip, and raise the level gradually when you are finished.
To add the Bitcrusher effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Mode buttons: Set the distortion mode to Fold, Clip, or Wrap. Signal peaks that exceed
the clip level are processed.
Note: The Clip Level parameter has a significant impact on the behavior of all three
modes. This is reflected in the waveform display, so try each mode button and adjust
the Clip Level slider to get a feel for how this works.
• Fold button: Set a softer distortion by halving the level of the center portion of
the signal above the threshold. The start and end levels of the clipped signal are
unchanged.
• Clip button: Enable to cause an abrupt distortion when the clipping threshold is
exceeded. Clipping that occurs in most digital systems is closest to Cut mode.
• Wrap button: Set a less severe distortion by offsetting the start, mid, and end levels of
the signal above the threshold. This parameter smooths signal levels when they cross
the threshold. The center portion of the clipped signal is also softer than in Cut mode.
• Drive knob and field: Set the amount of gain applied to the input signal.
Note: Raising the Drive level also tends to increase the amount of clipping at the effect
output.
• Resolution knob and field: Set the bit rate (between 1 and 24 bits) to alter the
calculation precision of the process. Lower values increase the number of sampling
errors, generating more distortion. At extremely low bit rates, the amount of distortion
can be greater than the level of the usable signal.
• Downsampling knob and field: Reduce the sample rate. A value of 1x has no effect on
the signal, a value of 2x halves the sample rate, and a value of 10x reduces the sample
rate to one-tenth of the original. (For example, if you set Downsampling to 10x, a
44.1 kHz signal is sampled at just 4.41 kHz.)
• Mix knob and field: Set the balance between the dry and crushed signal.
• Clip Level handle and field: Set the point (below the clipping threshold of the channel)
at which the signal starts clipping. Drag the green dot at the top left, or drag vertically
in the field, to set the clip level.
Clip Distortion has an unusual combination of serially connected filters. The incoming
signal is amplified by the Drive value, passes through a highpass filter, then is subjected
to nonlinear distortion. Following the distortion, the signal passes through a lowpass filter.
The effect signal is then recombined with the original signal, and this mixed signal is sent
through a further lowpass filter. All three filters have a slope of 6 dB/octave.
This unique combination of filters allows for gaps in the frequency spectra that can sound
good with this sort of nonlinear distortion.
To add the Clip Distortion effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Input Gain knob and field: Set the amount of gain applied to the plug-in input signal.
This behaves like a preamplifier for the Drive parameter.
• Drive knob and field: Set the amount of additional gain (distortion) applied to the input
signal. After being amplified by Drive, the signal passes through a highpass filter.
• Tone handle and field: Set the cutoff frequency (in hertz) of the highpass filter.
• Symmetry handle and field: Set the amount of nonlinear (asymmetrical) distortion
applied to the signal.
• Clip Filter handle and field: Set the cutoff frequency (in hertz) of the first lowpass
filter.
• Mix knob and field: Set the ratio between the effect (wet) signal and original (dry)
signals, following the clip filter.
• High Shelving knob and field: Set 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. Unlike these types of treble
controls, however, you can boost or cut the signal by up to ±30 dB with the Gain
parameter.
• LP Filter knob and field: Set the cutoff frequency (in hertz) of the lowpass filter. This
processes the mixed signal.
• Gain knob and field: Set the amount of gain applied to the output of signals above the
high shelving filter frequency.
• Output Gain knob and field: Set the amount of gain applied to the plug-in output signal.
To add the Distortion effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Drive knob and field: Set the amount of saturation applied to the signal.
• Tone knob and field: Set the frequency for the high cut filter. Filtering the harmonically
rich distorted signal produces a softer tone.
• Level Compensation button: Turn on to reference the overall processing of the signal to
0 dB. This compensates for increases in loudness caused by adding distortion.
To add the Distortion II effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Pre Gain knob and field: Set the amount of gain applied to the input signal.
• Drive knob and field: Set the amount of saturation applied to the signal.
• Tone knob and field: Boost the integrated high shelf filter gain both pre- and post-
distortion, to achieve a different tone.
• Growl: Emulates a two-stage tube amplifier similar to the type found in a Leslie 122
speaker cabinet, which is often used with the Hammond B3 organ.
• Nasty: Produces hard distortion, suitable for creating very aggressive sounds.
To add the Overdrive effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Drive knob and field: Set the saturation amount for the simulated FET transistor.
• Tone knob and field: Set the frequency of the high cut filter. Filtering the harmonically
rich distorted signal produces a softer tone.
• Level Compensation button: Turn on to reference the overall processing of the signal to
0 dB. This compensates for increases in loudness caused by using overdrive.
The input signal only passes the delay line and is not affected by any other process. The
Mix parameter blends the effect signal with the original signal.
To add the Phase Distortion effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Monitor button: Turn on to hear the input signal in isolation. Turn off to hear the mixed
signal.
• Cutoff knob and field: Set the (center) cutoff frequency of the lowpass filter.
• Resonance knob and field: Emphasize frequencies surrounding the cutoff frequency.
• Intensity knob and field: Set the amount of modulation applied to the signal.
• Phase Reverse button: Turn on 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.
• Mix slider and field: Set the percentage of the effect signal mixed with the original
signal.
Ringshifter
The ring modulator modulates the amplitude of the input signal using either the internal
oscillator or a side-chain signal. The frequency spectrum of the resulting effect signal
equals the sum and difference of the frequency content in the two original signals.
Its sound is often described as metallic or clangorous. The ring modulator was used
extensively on jazz rock and fusion records in the early 1970s.
Note: Frequency shifting should not be confused with pitch shifting. Pitch shifting
transposes the original signal, leaving its harmonic frequency relationship intact.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Oscillator controls: Configure the internal sine wave oscillator, which modulates the
amplitude of the input signal in both of the frequency shifter modes as well as in the
ring modulator OSC mode. See Oscillator controls.
• Envelope Follower controls: Modulate the oscillator frequency and output signal with an
envelope follower. See Envelope Follower controls.
• LFO modulation controls: Modulate the oscillator frequency and output signal with an
LFO. See LFO modulation controls.
• Output controls: Set feedback, stereo width, and the amount of dry and wet signals. See
Output controls.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• 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).
• Dual (Frequency Shifter) button: The frequency shifting process produces one shifted
effect signal for each stereo channel—one is shifted up, and the other is shifted down.
The oscillator Frequency control determines the shift direction in the left versus the
right channel.
• OSC (Ring Modulator) button: The ring modulator uses the internal sine wave oscillator
to modulate the input signal.
• Side Chain (Ring Modulator) button: The ring modulator modulates the amplitude of
the input signal with the audio signal assigned via the side-chain input. The sine wave
oscillator is switched off, and the Frequency controls are not accessible when Side
Chain mode is active.
• In the frequency shifter modes, the Frequency control adjusts the amount of frequency
shifting (up or down) applied to the input signal.
• In the ring modulator OSC mode, the Frequency control adjusts the frequency content
(timbre) of the resulting effect. This timbre can range from subtle tremolo effects to
clangorous metallic sounds.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Lin (Linear) and Exp (Exponential) buttons: Switch the scaling of the Frequency control:
• Exp: Exponential scaling offers extremely small increments around the 0 point, which
is useful for programming slow-moving phasing and tremolo effects.
• Lin: Linear scaling resolution is even across the entire control range.
• Env Follow (Envelope Follower) slider and field: Determine the impact of incoming signal
levels on the oscillator modulation depth.
• LFO slider and field: Determine the amount of oscillator modulation by the LFO.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Time knob and field: Set the delay time. This is in Hz when running freely, or in note
values (including triplet and dotted notes) when the Sync button is active.
• Sync button: This is used to sync the delay to the project tempo in Logic Pro and is
disabled for use in Final Cut Pro.
• Level knob and field: Set 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).
The envelope follower analyzes the amplitude (volume) of the input signal and uses this
to create a continuously changing control signal—a dynamic volume envelope of the input
signal. This control signal can be used for modulation purposes.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Sens (Sensitivity) slider and field: Determine how responsive the envelope follower
is to the input signal. At lower settings, the envelope follower reacts only to the most
dominant signal peaks. At higher settings, the envelope follower tracks the signal more
closely, but may react less dynamically.
• Attack slider and field: Set the response time of the envelope follower.
• Decay slider and field: Control the time it takes the envelope follower to return from a
higher to a lower value.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Symmetry and Smooth sliders and fields: Change the shape of the LFO waveform.
• Rate knob and field: Set the waveform cycle speed of the LFO.
• Sync button: This is used to sync the LFO rate with the project tempo in Logic Pro and
is disabled for use in Final Cut Pro.
To add the Ringshifter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Dry/Wet knob and field: Set the mix ratio of the dry input signal and the wet effect
signal.
• Feedback knob and field: Set 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). Use of longer delay times, in conjunction with
high Feedback settings, creates continuously rising and falling frequency shift effects.
• Stereo Width knob and field: Determine the breadth of the effect signal in the stereo
field. Stereo Width affects only the effect signal of Ringshifter, not the dry input signal.
• Env Follow (Envelope Follower) slider and field: Determine the amount of Dry/Wet
parameter modulation by the input signal level.
• LFO slider and field: Set the LFO modulation depth of the Dry/Wet parameter.
The held, and delayed, signal is repeated after a given time period, creating a repeating
echo effect, or delay. 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.
You can use delays to double individual sounds to resemble a group of instruments playing
the same melody, to create echo effects, to place the sound in a large “space,” to generate
rhythmic effects, or to enhance the stereo position of an audio clip.
Echo effects are generally used as individual audio clip effects. They are rarely used on an
overall mix, unless you’re trying to achieve an unusual effect.
Delay Designer
You can use Delay Designer with mono, stereo, or surround clips. See Use Delay Designer
in surround for details on using it in surround.
To add the Delay Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Main display: Provides a graphic representation of all taps. You can see, and edit, the
parameters of each tap in this area. See Main display controls.
• 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 Tap
parameter bar .
• Tap pads: You can use these two pads to create taps in Delay Designer. See Create
taps.
• Sync section: This is used to sync tempo in Logic Pro and is disabled for use with
Final Cut Pro.
• Master section: This area contains the global Mix and Feedback controls. See Master
section controls.
• View buttons: Determine the parameter or parameters represented in the Tap display.
See View buttons.
• Autozoom button: Zoom the Tap display out, making all taps visible. Turn Autozoom off
if you want to zoom the display in (by dragging vertically on the Overview display) to
view specific taps.
• Overview display: Shows all taps in the time range. See Navigate the Tap display.
• 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. See Use the 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. You can directly edit
tap parameters in the Tap display area. See Edit taps in the Tap display.
• Identification bar: Shows an identification letter for each tap. It also serves as a time
position indicator for each tap. You may freely move taps backward or forward in time
along this bar/timeline. See Move and delete taps.
• Cutoff button: Show the highpass and lowpass filter cutoff frequencies of taps.
• Reso (Resonance) button: Show the filter resonance value of each tap.
• For mono to stereo channels, each tap contains a line showing its pan position.
• For stereo to stereo channels, each tap contains a dot showing its stereo balance. A
line extending outwards from the dot indicates the tap’s stereo spread.
• For surround channels, each tap contains a line representing its surround angle (see
Use Delay Designer in surround).
Tip: You can temporarily switch the Tap display to Level view from one of the other
view modes by pressing Command-Option.
Tip: If the Overview display is hidden behind a tap, you can move it to the foreground by
pressing and holding Shift.
To add the effect and show its controls, see Add Logic effects to clips.
• Vertically drag the highlighted section (the bright rectangle) of the Overview display.
• Horizontally drag the highlighted bars—to the left or right of the bright rectangle—in
the Overview display.
Note: The Autozoom button needs to be disabled when manually zooming with the
Overview display. When you zoom in on a small group of taps, the Overview display
continues to show all taps. The area shown in the Tap display is indicated by the bright
rectangle in the Overview display.
To add the effect and show its controls, see Add Logic effects to clips.
2. Horizontally drag the middle of the bright rectangle in the Overview display.
The fastest way to create multiple taps is to use the Tap pads. If you have a specific rhythm
in mind, you might find it easier to tap out your rhythm on dedicated hardware controller
buttons, instead of using mouse clicks. If you have a MIDI controller, you can assign the
Tap pads to buttons on your device. For information about assigning controllers, see the
Logic Pro Control Surfaces Support manual.
After a tap has been created, you can freely adjust its position, or you can remove it if it
was created accidentally. See Move and delete taps.
To add the effect and show its controls, see Add Logic effects to clips.
Note: Whenever you click the Start pad, it automatically erases all existing taps. To
avoid erasing your initial taps, create subsequent taps by clicking in the Identification
bar.
The upper pad label changes to Tap, and a red tap recording bar appears in the strip
below the view buttons.
These are created at the exact moments in time of each click, adopting the rhythm of
your click pattern.
This adds the final tap, ending tap recording and assigning the last tap as the feedback
tap (for an explanation of the feedback tap, see Master section controls).
Note: If you don’t click the Last Tap button, tap recording automatically stops after
10 seconds or when the 26th tap is created, whichever comes first.
To add the effect and show its controls, see Add Logic effects to clips.
To add the effect and show its controls, see Add Logic effects to clips.
Taps are assigned letters, based on their order of creation. The first tap to be created
is assigned as Tap A, the second tap is assigned as Tap B, and so on. Once assigned,
each tap is always identified by the same letter, even when moved in time and therefore
reordered. For example, if you initially create three taps, they are named Tap A, Tap B, and
Tap C. If you then change the delay time of Tap B so that it precedes Tap A, it’s still called
Tap B.
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 (see Select taps).
Select a tap
1. In the Final Cut Pro timeline, select a clip with the Delay Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• Click one of the arrows to the left of the tap name to select the next or previous tap.
• Click the pop-up menu to the right of the tap name, and choose the appropriate tap
letter.
To add the effect and show its controls, see Add Logic effects to clips.
• 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.
Note: When you move a tap, you are actually editing its delay time.
To add the effect and show its controls, see Add Logic effects to clips.
2. Select the tap in the Identification bar, and drag it to the left to go forward in time, or to
the right to go backward in time.
This method also works when more than one tap is selected.
Note: Editing the Delay Time parameter in the Tap Delay field of the Tap parameter bar also
moves a tap in time. See Tap parameter bar .
Delete taps
1. In the Final Cut Pro timeline, select a clip with the Delay Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
You can also 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.
• Delete all selected taps: Control-click (or right-click) a tap, and choose “Delete
tap(s)” from the shortcut menu.
• Reso view: Toggle buttons switch the filter slope between 6 dB and 12 dB.
To add the effect and show its controls, see Add Logic effects to clips.
When you release the Option and Command keys, the toggle buttons return to their
standard functionality in the active view mode.
Note: The first time you edit a filter or pitch transpose control, 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.
To add the effect and show its controls, see Add Logic effects to clips.
3. Vertically drag the bright line of the tap you want to edit (or one of the selected taps, if
multiple taps are selected).
If you have chosen multiple taps, the values of all selected taps are changed relative to
each other.
Note: The method outlined above is slightly different for the Filter Cutoff and Pan
parameters. See Edit filter cutoff and Edit pan.
To add the effect and show its controls, see Add Logic effects to clips.
2. Command-drag horizontally and vertically across several taps in the 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.
To add the effect and show its controls, see Add Logic effects to clips.
2. Command-click in the Tap display, and move the pointer while holding down the
Command key.
3. Click the appropriate position to mark the end point of the line.
The values of taps that fall between the start and end points are aligned along the line.
In Cutoff view, each tap actually shows two parameters: highpass and lowpass filter cutoff
frequency. You can adjust the filter cutoff values independently by dragging the specific
cutoff frequency line—the upper line is lowpass and the lower line is highpass—or you can
adjust both cutoff frequencies by dragging between them.
When the highpass filter cutoff frequency value is lower than that of the lowpass cutoff
frequency, only one line is shown. This line represents the frequency band that passes
through the filters—in other words, the filters act as a bandpass filter. In this configuration,
the two filters operate serially, meaning that the tap passes through one filter first, then the
other.
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%. To adjust this, drag either side of the dot. As you
do so, the width of the line extending outward from the dot changes. Keep an eye on the
Spread parameter in the Tap parameter bar while you’re adjusting.
In surround configurations, the bright line represents the surround angle. See Use Delay
Designer in surround.
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. If you choose multiple taps in the Tap
display, the values of all selected taps are changed relative to each other.
Option-click a parameter value to reset it to the default setting. If multiple taps are
selected, Option-clicking a parameter of any tap resets all selected taps to the default
value for that parameter.
• Filter On/Off button: Enable or disable the highpass and lowpass filters for the selected
tap.
• HP-Cutoff-LP fields: Set the cutoff frequencies (in Hz) for the highpass and lowpass
filters.
• Slope buttons: Determine the steepness of the highpass and lowpass filter slope. Click
the 6 dB button for a gentler filter slope, or click the 12 dB button for a steeper, more
pronounced filtering effect.
Note: You cannot set the slope of the highpass and lowpass filters independently.
• Reso (Resonance) field: Set the amount of filter resonance for both filters.
• Tap Delay fields: Show the number and name of the selected tap in the upper section
and the delay time in the lower section.
• Pitch On/Off button: Enable or disable pitch transposition for the selected tap.
• Transp (Transpose) fields: The left field sets the amount of pitch transposition in
semitones. The right field fine-tunes each semitone step in cents (1/100 of a semitone).
• Flip buttons: Swap the left and right side of the stereo or surround image. Clicking
these buttons reverses the tap position from left to right, or vice versa. For example, if a
tap is set to 55% left, clicking the flip button changes it to 55% right.
• Pan field: Control 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.
• Level field: Determine the output level for the selected tap.
• Copy sound parameters: Copies all parameters (except the delay time) of the selected
tap or taps into the Clipboard.
• Paste sound parameters: Pastes the tap parameters from the Clipboard into the
selected tap or taps. If there are more taps in the Clipboard than are selected in the Tap
display, the extra taps in the Clipboard are ignored.
• Reset sound parameters to default values: Resets all parameters of all selected taps
(except the delay time) to the default values.
• 2 x delay time: Doubles the delay time of all selected taps. For example, the delay times
of three taps are set as follows: Tap A = 250 ms, Tap B = 500 ms, Tap C = 750 ms. If you
select these three taps and choose the “2 x delay time” shortcut menu command, the
taps will be changed as follows: Tap A = 500 ms, Tap B = 1000 ms, Tap C = 1500 ms. In
other words, a rhythmic delay pattern will unfold half as fast. (In musical terms, it will be
played in half time.)
• 1/2 x delay time: Halves the delay time of all selected taps. Using the example above,
choosing the “1/2 x delay time” shortcut menu command changes the taps as follows:
Tap A = 125 ms, Tap B = 250 ms, Tap C = 375 ms. In other words, a rhythmic delay
pattern will unfold twice as fast. (In musical terms, it will be played in double time.)
1. In the Final Cut Pro timeline, select a clip with the Delay Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• 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 resets the chosen parameter to
its default value for all selected taps.
• In the Tap parameter bar, Option-click a parameter value to reset it to the default
setting.
If multiple taps are selected, Option-clicking a parameter of any tap resets all
selected taps to the default value for that parameter.
In simple delays, the only way for the delay to repeat is to use feedback. Because
Delay Designer offers 26 taps, you can use these taps to create repeats, rather than
requiring discrete feedback controls for each tap.
Delay Designer’s global Feedback parameter does, however, allow you to send the output
of one user-defined tap back through the effect input, to create a self-sustaining rhythm or
pattern. This tap is known as the feedback tap.
• Feedback Level knob and field: Set the feedback tap output level before it is routed
back into Delay Designer’s input.
• A value of 100% sends the feedback tap back into Delay Designer’s input at full
volume.
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 reenabled.
• Mix sliders: Independently set the levels of the dry input signal and the post-processing
wet signal.
When you use Delay Designer in any surround configuration, the Pan parameter on the Tap
parameter bar is replaced with a surround panner, which lets you determine the surround
position of each tap.
Note: In the Tap display’s Pan view, you can only adjust the angle of taps. You must use the
surround panner on the Tap parameter bar to adjust diversity.
1. In the Final Cut Pro timeline, select a clip with the Delay Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
Note: Delay Designer generates separate automation data for stereo pan and surround
pan operations. This means that when you use it in surround channels, it will not react
to existing stereo pan automation data, and vice versa.
Although rich, combined flanging and chorus effects are possible, 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.
To add the Modulation Delay effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Time knob and field: Set the basic delay time. Set to the far left position to create
flanger effects, to the center for chorus effects, and to the far right to hear clearly
discernible delays.
• Feedback knob and field: Set the amount of effect signal routed back to the input. Use a
high Feedback value for strong modulations. If you want to double the signal, don’t use
Feedback. Negative values invert the phase of the feedback signal, resulting in more
chaotic effects.
• De-Warble button: Turn on to make sure the pitch of the modulated signal remains
constant.
• Constant Mod (Modulation) button: Turn on to make sure the modulation width remains
constant, regardless of the modulation rate.
Note: When Constant Mod is turned on, higher modulation frequencies reduce the
modulation width.
• D-Mode button: Turn on to introduce a spatial filtering effect that resembles a well-
known vintage processor.
• LFO 1 and LFO 2 Rate knobs and fields: Set the modulation rate for the left and right
stereo channels. In surround instances, the center channel is assigned the middle value
of the left and right LFO Rate knobs. The other channels are assigned values between
the left and right LFO rates.
Note: The right LFO Rate knob is available only in stereo and surround instances, and it
can be set separately only if Link L & R is not turned on.
• Mix slider and fields: Determine the balance between the two LFOs.
• Phase knob and field: Control the phase relationship between individual channel
modulations. Available only in stereo and surround instances.
• At 0°, the extreme values of the modulation are achieved simultaneously for all
channels.
• At 180° or −180°, you achieve the greatest possible distance between the modulation
phases of the channels.
Note: The Phase controls are available only if Link L & R is turned on.
• Distribution pop-up menu: Choose how phase offsets between individual channels are
distributed in the surround field. Choose Circular, Left↔Right, Front↔Rear, Random, or
New Random. Available only in surround instances.
Note: When you load a setting that uses the Random option, the saved phase offset
value is recalled. If you want to randomize the phase setting again, choose New Random
from the Distribution pop-up menu.
• Filter button: Turn on to introduce an additional allpass filter into the signal path. This
filter shifts the phase angle of a signal, influencing its stereo image.
• Low Cut and High Cut sliders and fields: Set the frequency at which the phase shift
crosses 90°—the halfway point of the total 180°—for each of the stereo channels. In
surround instances, the other channels are automatically assigned values that fall
between the two settings.
• Volume Mod (Modulation) slider and field: Determine the impact of LFO modulation on
the amplitude of the effect signal.
• Output Mix slider and field: Set the balance between dry and wet signals.
Note: If you use Stereo Delay on mono channel strips, the track or bus switches to two-
channel operation from the point of insertion—all Insert slots after the chosen slot are
stereo.
To add the Stereo Delay effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
Channel controls
• Input pop-up menu: Choose the input signal for the two stereo sides. Options include
Off, Left, Right, L + R, and L − R.
• Delay Time knob and field: Set the delay time in milliseconds or in note values when
synchronized with the clip. Notes (and dots) are displayed around the Delay Time knob
when synchronized with the clip. Click the notes or dots (or rotate the knob) to choose
an exact synchronization value.
Note: Clicking note or dot values resets the Deviation parameter value. Choose a value
from the Note pop-up menu to retain the current Deviation value.
• Note pop-up menu: Choose the grid resolution for the delay time when Tempo Sync is
turned on.
• Low Cut and High Cut slider and field: Cut frequencies below the Low Cut value and
above the High Cut value from the source signal.
• Feedback knob and field: Set the amount of feedback for the left and right delay
signals.
• Feedback Phase button: Invert the phase of the corresponding channel feedback signal.
• Crossfeed Left to Right (or Right to Left) knob and field: Transfer the feedback signal of
the left channel to the right channel, and vice versa.
• Crossfeed Phase button: Invert the phase of the crossfed feedback signals.
• Tempo Sync button: This is used to sync delay repeats with the clip in Logic Pro and is
disabled for use in Final Cut Pro.
• Stereo Link button: Turn Stereo Link on to make the same adjustments for both
channels. Adjusting one channel value adjusts the other. Relative values are maintained.
• Press Command to temporarily flip stereo linking, allowing you to adjust a control in
a linked fashion even when Stereo Link is not turned on.
• Command-drag a slider to adjust a single channel when Stereo Link is turned on.
Note: Stereo Link can be automatically turned off when you choose a new routing or
setting. To turn it back on, click the Stereo Link button.
• Output Mix sliders and fields: Independently control the level of the left and right
channel signals.
To add the Tape Delay effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Delay Time knob and field: Set the delay time in milliseconds. Notes (and dots) are
displayed around the Delay Time knob when the delay time is synchronized with the
clip. Click the notes or dots (or rotate the knob) to choose an exact synchronization
value.
Note: Clicking note or dot values resets the Deviation parameter value. Choose a value
from the Note pop-up menu to retain the current Deviation value.
• Note pop-up menu: Set the grid resolution for the delay time.
• Smoothing slider and field: Even out the LFO and flutter effect. See the LFO and Flutter
parameter descriptions, below.
• Clip Threshold knob: Set the level of the distorted tape saturation signal. Higher values
produce no additional audible distortion. Lower values result in an aggressive distortion.
This behavior is influenced by high Feedback values, which result in eventual distortion,
irrespective of the Clip Threshold value. However, aggressive distortion and signal
breakup are achieved far more rapidly when a low Clip Threshold level is used.
• Spread knob and field: Set the width of the effect signal in stereo instances. This
parameter is not available in mono instances.
• Tape Head Mode buttons: Click either Clean or Diffuse to emulate a different tape head
position. This affects the behavior of other parameters, such as Flutter and Feedback.
• Low Cut and High Cut sliders and fields: Cut frequencies below the Low Cut value and
above the High Cut value to shape the sound of taps (delay repeats) with the highpass
and lowpass filters. The filters are located in the feedback circuit, which means that the
filtering effect increases in intensity with each delay repeat. If you want an increasingly
muddy and confused tone, move the High Cut slider toward the left. For ever thinner
echoes, move the Low Cut slider toward the right. If you can’t hear the effect, check the
Dry and Wet controls and the filter settings.
• LFO Rate knob and field: Set the speed of the LFO.
• LFO Intensity knob and field: Set the amount of LFO modulation. A value of 0 turns off
delay modulation.
• Flutter Rate and Flutter Intensity knobs and fields: Simulate the speed irregularities of
tape transports used in analog tape delay units.
• Feedback knob: Set the amount of delayed and filtered signal that is routed back to
the input. Set Feedback to the lowest possible value to generate a single echo. Set
Feedback to 100% to endlessly repeat the signal. The levels of the original signal and
taps (echo repeats) tend to accumulate and may cause distortion. Use the Character
controls to change the color of these overdriven signals.
• Freeze button: Capture current delay repeats and sustain them until Freeze is turned off.
• Dry and Wet sliders and fields: Independently control the amount of original and effect
signal.
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.
All EQs are specialized filters that allow certain frequencies to pass through unchanged
while raising (boosting) or lowering (cutting) the level of other frequencies. Some EQs can
be used in a “broad-brush” fashion, to boost or cut a large range of frequencies. Other
EQs, particularly parametric and multiband EQs, can be used for more precise control.
The simplest types of EQs are single-band EQs, which include low cut and high cut,
lowpass and highpass, shelving, and parametric EQs.
Multiband EQs (such as the Channel EQ or Linear Phase EQ) combine several filters in
one unit, enabling you to control a large part of the frequency spectrum. Multiband EQs
allow you to independently set the frequency, bandwidth, and Q factor of each frequency
spectrum band. This provides extensive, and precise, tone-shaping on any audio source,
be it an individual audio signal or an overall mix.
The effect works by analyzing incoming signal levels through use of a threshold parameter.
Any signal level that exceeds the threshold is used as a trigger for a synthesizer-style
ADSR envelope or a low-frequency oscillator (LFO). These control sources are used to
dynamically modulate the filter cutoff.
AutoFilter lets you choose between different filter types and slopes, control the amount of
resonance, add distortion for more aggressive sounds, and mix the original, dry signal with
the processed signal.
To add the AutoFilter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
The AutoFilter window is divided into sections for Filter, Envelope, Distortion, LFO, and Out
(output) controls. An extended controls section is also available.
• Filter controls: Control the tonal color of the filtered sound. See Filter controls.
• Envelope controls: Define how the filter cutoff frequency is modulated over time. See
Envelope controls.
• Distortion controls: Distort the signal both before and after the filter. See Distortion
controls.
• LFO controls: Define how the filter cutoff frequency is modulated by the LFO. See LFO
controls.
• Out (output) controls: Set the level of both the dry and effect signal. See Output
controls.
• Cutoff knob and field: Set the cutoff frequency for the filter. Higher frequencies are
attenuated, whereas lower frequencies are allowed to pass through in a lowpass filter.
The reverse is true in a highpass filter. When the State Variable filter is set to bandpass
(BP) mode, the filter cutoff determines the center frequency of the frequency band that
is allowed to pass.
• Resonance knob and field: Boost or cut signals in the frequency band that surrounds
the cutoff frequency. Very high Resonance values cause the filter to begin oscillating
at the cutoff frequency. This self-oscillation occurs before you reach the maximum
Resonance value.
• State Variable buttons: Switch the filter between highpass (HP), bandpass (BP), lowpass
(LP), and peak (PK) modes.
• 4-Pole Lowpass buttons: Set the slope of the lowpass filter to 6, 12, 18, or 24 dB per
octave.
Note: Clicking one of these buttons automatically chooses the lowpass (LP) filter and
slope, overriding any active State Variable filter button.
• Fatness slider and field: Boost the level of low-frequency content. When you set
Fatness to its maximum value, adjusting Resonance has no effect on frequencies below
the cutoff frequency. This parameter is used to compensate for a weak or “brittle”
sound caused by high resonance values, when in the lowpass filter mode.
• Envelope slider and field: Determine the impact of the envelope on cutoff frequency.
• LFO slider and field: Determine the impact of the LFO on cutoff frequency.
• Threshold knob and field: Set an input level that—if exceeded—triggers the envelope
or LFO that dynamically modulates filter cutoff frequency. See LFO controls and Filter
controls.
Note: Retriggering of the envelope or LFO occurs only if Retrigger is set to On. See
Extended controls.
• Dynamic knob and field: Determine the input signal modulation amount. You can
modulate the peak value of the envelope section by adjusting this control.
• Attack handle and field: Drag the handle horizontally (or drag in the field vertically) to
set the envelope attack time.
• Decay handle and field: Drag the handle horizontally (or drag in the field vertically) to
set the envelope decay time.This handle sets both the decay time and the sustain level.
• Sustain handle and field: Drag the handle vertically (or drag in the field vertically) to set
the envelope sustain level. If the input signal falls below the threshold level before the
envelope sustain phase, the release phase is triggered.
• Release handle and field: Drag the handle horizontally (or drag in the field vertically) to
set the envelope release time. This is triggered as soon as the input signal falls below
the threshold.
• Pre Filter knob and field: Set the amount of distortion applied before the filter section
processes the signal.
• Post Filter knob and field: Set the amount of distortion applied after the filter section
processes the signal.
• Mode pop-up menus: Choose the distortion type. The options are Classic, Tube, and
Scream.
• Sync button: This is used to sync the LFO with the clip tempo in Logic Pro and is
disabled for use in Final Cut Pro.
• Rate knob and field: Set the speed of LFO modulation. See Extended controls.
• Sync Phase knob and field: This is used to set the phase relationship between the LFO
rate and the project tempo in Logic Pro (when Sync is active) and is disabled for use in
Final Cut Pro.
• Stereo Phase knob and field: Set the phase relationship of the LFO modulations
between the two channels (stereo only).
• Waveform buttons: Select the shape of the LFO waveform. Choose descending
sawtooth, ascending sawtooth, triangle, pulse wave, or random.
• Pulse Width knob and field: Alter the curve shape of the selected waveform.
• Dry Signal slider and field: Set the amount of original, dry signal added to the filtered
signal.
• Main Out slider and field: Set the overall output level. This compensates for higher
levels caused by the use of distortion or by the filtering process itself.
• Rate Modulation slider and field: Set the LFO frequency, independent of the input signal
level. Typically, when the input signal exceeds the threshold, the modulation width of
the LFO increases from 0 to the Rate Mod. value. This parameter allows you to override
this behavior.
• Retrigger button: Start the LFO waveform at 0 each time the threshold is exceeded.
• Region Gate On and Off buttons: Turn on to keep the filter open for the entire region
length. Turn off to use other AutoFilter parameters to control filter time.
• LFO Decay/Delay slider and field: Set the time required for the LFO to move from zero to
the maximum value.
You can use Channel EQ to shape the sound of an individual clip. The Analyzer and graphic
controls make it easy to view and change the audio signal in real time.
Using Channel EQ
The way you use Channel 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 Channel EQ
to a flat response (no frequencies boosted or cut), then turn on the Analyzer and play
the audio signal. Keep an eye on the graphic display to see which parts of the frequency
spectrum have frequent peaks and which parts of the spectrum stay at a low level. Pay
particular attention to sections where the signal distorts or clips. Use the graphic display
or parameter controls to adjust the frequency bands.
You can reduce or eliminate unwanted frequencies, and you can raise quieter frequencies
to make them more pronounced. You can adjust the center frequencies of bands 2 through
7 to affect a specific frequency—either one you want to emphasize, such as the root note
of the music, or one you want to eliminate, such as hum or other noise. While doing so,
change the Q parameter so that only a narrow range of frequencies are affected, or widen
the range to alter a broad area.
Each EQ band has a different color in the graphic display. You can graphically adjust the
frequency of a band by dragging horizontally. Drag vertically to adjust the amount of gain
for the band. For bands 1 and 8, the slope values can be changed only in the parameter
area below the graphic display. Each band has a pivot point (a small circle on the curve) at
the location of the band’s frequency; you can adjust the Q or width of the band by dragging
the pivot point vertically.
You can also adjust the decibel scale of the graphic display by vertically dragging either
the left or right edge of the display, where the dB scale is shown, when the Analyzer is not
active. When the Analyzer is active, dragging the left edge adjusts the linear dB scale, and
dragging the right edge adjusts the Analyzer dB scale.
To increase the resolution of the EQ curve display in the most interesting area around the
zero line, drag the dB scale, on the left side of the graphic display, upward. Drag downward
to decrease the resolution.
The bands derived from FFT analysis are divided in a logarithmic scale—there are more
bands in higher octaves than in lower ones.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top
control, 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
−80 dB. The Analyzer display is always dB-linear.
To add the Channel EQ effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Band 1 vertical and horizontal lines: Drag to change the slope and Q value.
• Band 2 On/Off button: A low shelving filter that adjusts the level of low frequencies and
has a minimal impact on frequencies above the cutoff (set) frequency. When band 2 is
active, you can directly change band parameters in the graphic display.
• Band 2 background or dot: Drag to change the frequency and gain values.
• Band 2 vertical and horizontal lines: Drag to change the gain and Q values.
• Bands 3–6 On/Off buttons: Parametric bell filters with three controls. Frequency sets a
center frequency. Q sets the width of the frequency band around the center frequency.
Gain sets the level of the band. When one of these bands is active, you can directly
change band parameters in the graphic display.
• Band 3–6 background or dot: Drag to change the frequency and gain values.
• Band 3–6 vertical and horizontal lines: Drag to change the gain and Q values.
• Band 7 On/Off button: A high shelving filter that adjusts the level of high frequencies
and has a minimal impact on frequencies below the cutoff (set) frequency. When band 7
is active, you can directly change band parameters in the graphic display.
• Band 7 background or dot: Drag to change the frequency and gain values.
• Band 7 vertical and horizontal lines: Drag to change the gain and Q values.
• Band 8 On/Off button: A lowpass filter that allows low frequencies to pass and reduces
the level of high frequencies near the cutoff (set) frequency. When band 8 is active, you
can directly change band parameters in the graphic display.
• Band 8 vertical and horizontal lines: Drag to change the slope and Q value.
Tip: Press and hold the Command key while performing any of the operations below
to limit dragging to vertical or horizontal movement.
• Drag anywhere in the colored band to adjust gain and the center frequency.
• Drag the vertical lines that encompass the band to adjust the Q (bandwidth) only.
• Drag the horizontal line in the band to adjust the gain only. If Q-Coupling is enabled,
both the gain and the bandwidth are adjusted.
• Drag the intersection of vertical and horizontal lines to adjust the gain and Q
simultaneously.
Note: Horizontally dragging the frequency handle in band 1 and band 8 adjusts both
the frequency and Q.
• Gain/Slope field: Set the amount of gain for the selected band. For bands 1 and 8, this
changes the slope of the filter.
• Q field: Adjust the Q factor or resonance of the affected range around the center
frequency in the selected band.
Note: The Q parameter of band 1 and band 8 has no effect when the slope is set to
6 dB/Oct. When the Q parameter is set to an extremely high value, such as 100, these
filters affect only a very narrow frequency band and can be used as notch filters.
• Master Gain slider and field: Set the overall output level of the signal. Use it after
boosting or cutting individual frequency bands.
• Analyzer Pre/Post button: Set to display the frequency curve before or after EQ is
applied, when the Analyzer is on. See Intro to Channel EQ.
• Processing pop-up menu: Choose which channel or channels to process (Stereo, Left
Only, Right Only, Mid Only, or Side Only).
• Analyzer Resolution pop-up menu: Choose the sample resolution for the Analyzer.
Choose “low (2048 points),” “medium (4096 points),” or “high (8192 points).”
• Analyzer Decay slider and field: Set the decay rate (in dB per second) of the Analyzer
curve. This is shown as a peak decay in Peak mode or an averaged decay in RMS mode.
• Gain-Q Couple Mode pop-up menu: Choose the amount of Gain-Q coupling.
• Light or medium: Allows some change as you raise or lower the gain.
• Asymmetric: These 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.
Note: If you play back automation of the Q parameter with a different Gain-Q Couple
Mode setting, the actual Q values are different than when the automation was recorded.
Linear Phase EQ
A further difference is that Linear Phase EQ uses a fixed amount of CPU resources,
regardless of how many bands are active. Linear Phase EQ also introduces greater amounts
of latency. Therefore, it is strongly recommended that you use it for mastering previously
recorded audio.
Each EQ band has a different color in the graphic display. You can graphically adjust the
frequency of a band by dragging horizontally. Drag vertically to adjust the amount of gain
for the band. For bands 1 and 8, the slope values can be changed only in the control area
below the graphic display. Each band has a pivot point (a small circle on the curve) at the
location of the band’s frequency; you can adjust the Q or width of the band by dragging the
pivot point vertically.
You can also adjust the decibel scale of the graphic display by vertically dragging either
the left or right edge of the display, where the dB scale is shown, when the Analyzer is not
active. When the Analyzer is active, dragging the left edge adjusts the linear dB scale, and
dragging the right edge adjusts the Analyzer dB scale.
To increase the resolution of the EQ curve display in the most interesting area around the
zero line, drag the dB scale, on the left side of the graphic display, upward. Drag downward
to decrease the resolution.
The bands derived from FFT analysis are divided in accordance with the frequency linear
principle—there are more bands in higher octaves than in lower ones.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top
control, 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.
To add the Linear Phase EQ effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Band 1 vertical and horizontal lines: Drag to change the slope and Q value.
• Band 2 On/Off button: A low shelving filter that adjusts the level of low frequencies and
has a minimal impact on frequencies above the cutoff (set) frequency. When band 2 is
active, you can directly change band parameters in the graphic display.
• Band 2 background or dot: Drag to change the frequency and gain values.
• Band 2 vertical and horizontal lines: Drag to change the gain and Q values.
• Bands 3–6 On/Off buttons: Parametric bell filters with three controls. Frequency sets a
center frequency. Q sets the width of the frequency band around the center frequency.
Gain sets the level of the band. When a band is active, you can directly change band
parameters in the graphic display.
• Band 3–6 background or dot: Drag to change the frequency and gain values.
• Band 3–6 vertical and horizontal lines: Drag to change the gain and Q values.
• Band 7 background or dot: Drag to change the frequency and gain values.
• Band 7 vertical and horizontal lines: Drag to change the gain and Q values.
• Band 8 On/Off button: A lowpass filter that allows low frequencies to pass and reduces
the level of high frequencies near the cutoff (set) frequency. When band 8 is active, you
can directly change band parameters in the graphic display.
• Band 8 vertical and horizontal lines: Drag to change the slope and Q value.
• Graphic display: Shows the current curve of each EQ band. The scale is shown in dB.
The color of each band matches the corresponding button above the display. Each
colored band (and corresponding Frequency, Gain, and Q field) is highlighted as you
move the pointer across it. To select a band for editing, click a curve line segment, the
(center frequency) handle, or in the colored space between the zero line and EQ curve.
Tip: Press and hold the Command key while performing any of the operations below
to limit dragging to vertical or horizontal movement.
• Drag anywhere in the colored band to adjust gain and the center frequency.
• Drag the vertical lines that encompass the band to adjust the Q (bandwidth) only.
• Drag the horizontal line in the band to adjust the gain only. If Q-Coupling is enabled,
both the gain and bandwidth are adjusted.
• Drag the intersection of vertical and horizontal lines to adjust the gain and Q
simultaneously.
Note: Horizontally dragging the frequency handle in band 1 and band 8 adjusts both
the frequency and Q.
• Gain/Slope field: Set the amount of gain for the selected band. For bands 1 and 8, this
changes the slope of the filter.
• Q field: Adjust the Q factor or resonance of the affected range around the center
frequency in the selected band.
Note: The Q parameter of band 1 and band 8 has no effect when the slope is set to
6 dB/Oct. When the Q parameter is set to an extremely high value, such as 100, these
filters affect only a very narrow frequency band and can be used as notch filters.
• Analyzer Pre/Post button: Set to display the frequency curve before or after EQ is
applied, when the Analyzer is on. See Intro to Linear Phase EQ.
• Processing pop-up menu: Choose to process both sides of a stereo signal, the Left
Only, Right Only, Mid Only, or Side Only signal.
• Analyzer Resolution pop-up menu: Choose the sample resolution for the Analyzer.
Choose “low (2048 points),” “medium (4096 points),” or “high (8192 points).”
• Analyzer Decay slider and field: Set the decay rate (in dB per second) of the Analyzer
curve. These are shown as a peak decay in Peak mode or an averaged decay in RMS mode.
• Gain-Q Couple Mode pop-up menu: Choose the amount of Gain-Q coupling.
• Light or medium: Allows some change as you raise or lower the gain.
• Asymmetric: These 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.
Note: If you play back automation of the Q parameter with a different Gain-Q Couple
Mode setting, the actual Q values are different than when the automation was recorded.
Vintage EQ collection
The unique output stage of each unit is also modeled, allowing you to pair the output stage
of any unit with the original or other EQ models.
Further enhancements include fully sweepable frequency controls that allow more detailed
signal contouring than the fixed-frequency options found on some of the original devices.
Each vintage EQ unit provides a distinct tonal signature that imparts a sonic color on
signals, unlike precise, clean modern equalizers such as the other Logic EQs.
All vintage EQ models share a set of common Output controls, along with unique controls
that are discussed in each section.
• Drive knob: Set the amount of gain/saturation of the chosen vintage EQ output stage.
This imparts the distortion and coloration of the original hardware output stage, even if
all EQ bands are in a neutral position.
• Output Model pop-up menu: Choose a vintage EQ model output stage. You can use the
matching output stage model for the active EQ or choose the output stage of another
unit. The output stage allows you to add harmonic distortion to your signals.
• Silky (Tube EQ): The output stage of the Vintage Tube EQ.
• Punchy (Graphic EQ): The output stage of the Vintage Graphic EQ.
• Smooth (Console EQ): The output stage of the Vintage Console EQ.
• Phase pop-up menu: Set the processing mode of the EQ and the chosen output stage.
Natural mirrors the cut/boost phase shifts of the original EQ. Linear allows EQ changes
without phase shifts of the source signal.
Each analog EQ introduces phase shifts of the signal which can have an audible (and
often desirable) effect on the sound. In some situations, however, phase shifts can
affect transients. This is especially the case when using steep cut filters, or high boost
of narrow filters. Linear phase filters let you change only the gain of a certain frequency
area of your material by retaining the phase, with slightly higher latency than in natural
mode.
• Volume field: Drag vertically to set the overall effect output level. The range is ±25 dB.
The original console module is regarded as a cult classic by many recording engineers, and
has been used on countless hit records over the past 40 to 50 years.
• Low Cut button: Turn the low cut/highpass filter on or off. This is a third-order filter set
at 18 dB per octave.
• Low Cut knob: Set the low cut/highpass filter frequency at 50, 80, 160, or 300 Hz, or set
values between these increments. Frequencies below this are rolled off at a fixed 18 dB
per octave.
• Low Gain knob: Set the low shelving filter level. The gain range is ±16 dB.
• Low Freq (Frequency) knob: Set the low shelving filter center frequency at 35, 60, 110,
or 220 Hz, or set values between these increments.
• Mid Gain knob: Set the midrange filter level. The gain range is ±18 dB.
• Mid Freq (Frequency) knob: Set the midrange filter center frequency at 0.36, 0.7, 1.6,
3.2, 4.8, or 7.2 kHz, or set values between these increments.
• High button: Turn the high shelving filter on or off. The level is fixed at 12 kHz.
• High Gain knob: Set the high shelving level. The gain range is ±16 dB.
Frequencies aren’t fixed at the default values, and you can proportionally scale all bands to
provide more focus on a portion of the overall frequency spectrum. This flexibility makes
it great for precise signal shaping and also a useful tool for tasks such as tuning difficult
rooms.
You can change the standard frequency by dragging the Tune field.
• Tune field: Drag to set the frequency of all band sliders. Scaling of frequencies is
proportional. This can be used to tune the bands to your project key.
Tip: When the frequency is set to +12, you can boost 32 kHz, which results in a very
smooth high-end boost.
• EQ band sliders: Drag to cut or boost the selected frequency of the incoming signal by
±12 dB.
The main original unit (upper) that Vintage Tube EQ is based on is a valve-equipped analog
design. It is a lossless passive equalizer. This means that the signal level remains constant
even if the EQ is switched out. The original unit is noted for the “musical” quality of its
filters, making it a versatile tool for mixing and mastering.
The second emulated EQ model (lower) is often paired with the original unit. It’s the perfect
partner for the upper unit, adding midrange flexibility that lets you fine-tune signals in this
frequency spectrum, with a beautifully matched tonal signature.
• Low Boost knob: Set the amount of low-frequency boost, up to 13.5 dB.
• Low Atten (Attenuation) knob: Set the amount of low-frequency attenuation (cut), up to
17.5 dB.
• Low Freq (Frequency) knob: Set the low-range center frequency to 20, 30, 60, or 100
Hz, or values between these increments.
• High Bandwidth knob: Set the Q, or bandwidth, of the high-frequency range from
narrow to broad.
• High Atten (Attenuation) knob: Set the amount of high-frequency attenuation (cut), up
to 16 dB.
• High Atten (Attenuation) Sel knob: Set the high-range shelving frequency to 5, 10, or 20
kHz, or values between these increments.
• Low Freq (Frequency) knob: Set the low-range center frequency to 0.2, 0.3, 0.5, 0.7, or
1.0 kHz, or values between these increments.
• Dip Freq (Frequency) knob: Set the Dip (attenuation) center frequency to 0.2, 0.3, 0.5,
0.7, or 1, 1.5, 2, 3, 4, 5, or 7 kHz, or values between these increments.
• Dip knob: Set the amount of attenuation (cut) for the selected Dip frequency, up to 10
dB.
• High Freq (Frequency) knob: Set the high-range center frequency to 1.5, 2, 3, 4, or 5
kHz, or values between these increments.
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 clips. This can be to increase
the perceived loudness or to highlight the most important sounds, while ensuring that
softer sounds are not lost in the mix.
There are four types of dynamics processors included in Final Cut Pro, each used for
different audio processing tasks.
By reducing the highest parts of the signal, called peaks, a compressor raises the
overall level of the signal, increasing the perceived volume. This gives the signal more
focus by making the louder (foreground) parts stand out, while keeping the softer
background parts from becoming inaudible. Compression also tends to make sounds
tighter or punchier because transients are emphasized, depending on attack and
release settings, and because the maximum volume is reached more swiftly.
In addition, compression can make a project sound better when played back in different
audio environments. For example, the speakers of a television set or in a car typically
have a narrower dynamic range than the sound system in a cinema. Compressing
the overall mix can help make the sound fuller and clearer in lower-fidelity playback
situations.
Compressors are typically used on dialogue clips to make the speech more intelligible in
an overall mix. They are also commonly used on music and sound effects clips, but they
are rarely used on ambience clips.
• Expanders: Expanders are similar to compressors, except that they raise, rather than
lower, the signal when it exceeds the threshold. Expanders are used to add life to audio
signals.
• Noise gates: Noise gates alter the signal in a way that is opposite to that used by
compressors or limiters. Whereas a compressor lowers the level when the signal is
louder than the threshold, a noise gate lowers the signal level whenever it falls below
the threshold. Louder sounds pass through unchanged, but softer sounds, such as
ambient noise or the decay of a sustained instrument, are cut off. Noise gates are often
used to eliminate low-level noise or hum from an audio signal.
Adaptive Limiter is typically used on the final mix, where it can be placed after a
compressor, such as Multipressor, and before a final gain control, resulting in a mix of
maximum loudness. Adaptive Limiter can produce a louder-sounding mix than can be
achieved by normalizing the signal.
Note: Using Adaptive Limiter adds latency when the Lookahead parameter is active. The
effect is typically used for mixing and mastering previously recorded tracks, not while
recording. Bypass Adaptive Limiter while recording.
To add the Adaptive Limiter effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Input meters: Show input levels in real time. The Margin field shows the peak input level.
You can reset the Margin field by clicking it.
• Reduction meter: Shows the amount of gain reduction. The Margin field shows the peak
reduction level. You can reset the Margin field by clicking it.
• Output meters: Show output levels of the limited signal. The Margin field shows the
peak output level. You can reset the Margin field by clicking it.
• Out Ceiling knob and field: Set the maximum output level, or ceiling. The signal does not
rise above this.
• Lookahead knob and field: Set the playback buffer size (how far in the future the file is
analyzed for peaks). Also see the description of the Optimal Lookahead control, below.
Values lower than the optimal buffer size are indicated in red.
• Remove DC Offset button: Turn on to activate a highpass filter that removes direct
current (DC) from the signal. DC can be introduced by lower-quality audio hardware.
• True Peak Detection button: Turn on to detect inter-sample peaks in the signal.
• Optimal Lookahead field and button: Use the Apply button to set the optimal playback
buffer size. This changes the value shown for the Lookahead parameter.
Compressor
You can use Compressor with individual clips, including voice, instrumental, and effects
clips.
The Ratio parameter is a percentage of the overall level; the more the signal exceeds the
threshold, the more it is reduced. A ratio of 4:1 means that increasing the input by 4 dB
results in an increase of the output by 1 dB, if above the threshold.
For example, with the Threshold set at −20 dB and the Ratio set to 4:1, a −16 dB peak in
the signal (4 dB louder than the threshold) is reduced by 3 dB, resulting in an output level
of −19 dB.
Many sounds, including voices and musical instruments, rely on the initial attack phase
to define the core timbre and characteristic of the sound. When compressing these types
of sounds, you should set higher Attack values to ensure that the attack transients of the
source signal aren’t lost or altered.
When attempting to maximize the level of an overall mix, it’s best to set the Attack
parameter to a lower value, because higher values often result in no, or minimal,
compression.
The Release parameter determines how quickly the signal is restored to its original level
after it falls below the threshold level. Set a higher Release value to smooth out dynamic
differences in the signal. Set a lower Release value if you want to emphasize dynamic
differences.
Important: The results of your settings for the Attack and Release parameters depend
not only on the type of source material but also on the compression ratio and threshold
settings.
Compressor knee
The Knee parameter determines whether the signal is slightly, or severely, compressed as
it approaches the threshold level.
Setting a Knee value close to 0 (zero) results in no compression of signal levels that fall
just below the threshold, while levels at the threshold are compressed by the full Ratio
amount. This is known as hard knee compression, which can cause abrupt and often
unwanted transitions as the signal reaches the threshold.
Increasing the Knee parameter value increases the amount of compression as the
signal approaches the threshold, creating a smoother transition. This is called soft knee
compression.
You can also use the Auto Gain parameter to compensate for the level reduction caused by
compression (choose either 0 dB or −12 dB).
When you use the Platinum circuit type, Compressor can analyze the signal using one
of two methods: Peak or root mean square (RMS). Although Peak is more technically
accurate, RMS provides a better indication of how people perceive the signal loudness.
Note: If you turn on Auto Gain and RMS simultaneously, the signal may become over-
saturated. If you hear any distortion, turn off Auto Gain and adjust the Make Up knob until
the distortion is inaudible.
To add the Compressor effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• FET (Field Effect Transistor) compressors are known for their fast transient
response. They can deliver a clean or colored tone (particularly in the midrange),
and can be pushed to a somewhat “crunchy” tone on transients. FET compressors
are ideal for drums, vocals, guitars, and other signals with a fast attack phase. FET
compressors can only attenuate the signal.
• Opto (Optical) compressors are known for their fast transient response and nonlinear
release handling. They are very clean and are ideal for vocals and guitars. They are
also often used as limiting amplifiers across busses or outputs.
• Side Chain and Output buttons: View Side Chain or Output parameters.
• Gain Reduction meter/graph: Click either the Meter or Graph button to change the real-
time compression amount display.
• Threshold knob and field: Set the threshold level—signals above this threshold value are
reduced in level.
• Ratio knob and field: Set the compression ratio—the ratio of signal reduction when the
threshold is exceeded.
• Make Up knob and field: Set the amount of gain applied to the compressed signal.
• Auto Gain buttons: The Off button disables autogain. The 0 dB and –12 dB buttons
compensate for volume reductions caused by compression.
• Knee knob and field: Set the strength of compression at levels close to the threshold.
Lower values result in more severe or immediate compression (hard knee). Higher
values result in gentler compression (soft knee).
• Attack knob and field: Set the time it takes for Compressor to react when the signal
exceeds the threshold.
• Release knob and field: Set the time it takes for Compressor to stop reducing the signal
after the signal level falls below the threshold. This control works in conjunction with
the Auto button when Auto is turned on.
• Auto button: Make the release time dynamically adjust to the audio material. The
automatic release time adjustment and compression results change when different
Release parameter values are used.
• Output Gain knob and field: Set the overall level of the compressor output.
• Output Gain meter: Displays the overall level of the compressor output.
Output controls
• Limiter button: Turn the integrated limiter on or off. Limiting prevents the Compressor
output from exceeding the threshold value.
• Limiter Threshold knob and field: Set the threshold level for the limiter.
• Distortion knob: Choose whether to apply clipping above 0 dB, and the type of clipping.
Soft, Hard, and Clip reduce the signal around the 0 dB line in different ways, resulting in
a smoothed or squared-off distortion of the signal peaks.
• Mix knob and field: Set the balance between dry (source) and wet (effect) signals. This
enables you to either reduce signal peaks (dry) or increase the level of softer signals
(wet).
• Max button: Turn on to compress both channels if either stereo channel exceeds or
falls below the threshold.
• Sum button: When Sum is turned on, the combined level of both channels must
exceed the threshold before compression occurs.
• Peak and RMS buttons: Use in conjunction with the Max and Sum buttons. Click Peak
or RMS to determine whether signal peaks or a signal average is used for detection.
These can help avoid artifacts such as clicks in the processed signal, depending on
the type of audio material and parameter settings (notably Attack).
• Filter buttons: Turn the filter on or off. Turn on Listen to monitor the side-chain signal.
• Filter mode knob: Choose the type of filter used to process the incoming side-chain
signal. Filtering the side-chain input signal can enhance the precision of trigger signals,
resulting in more surgical compression. The choices are LP (lowpass), BP (bandpass),
HP (highpass), ParEQ (parametric), and HS (high shelving).
• Frequency knob and field: Set the center frequency for the side-chain filter.
• Q knob and field: Set the width of the frequency band affected by the side-chain filter.
• Gain knob and field: Set the amount of gain applied to the side-chain signal.
DeEsser 2
You can use DeEsser 2 on a vocal track to reduce sibilance without affecting other
frequencies on the track. DeEsser 2 attenuates the selected frequency only if it exceeds
a set threshold level, preventing the sound from becoming darker when no sibilance
is present. It has extremely fast attack and release response times for the shortest of
transients, helping the recording retain a natural sound.
DeEsser 2 provides two operating modes—Relative and Absolute—for working with high- or
low-level audio signals. Also included are two filter shapes and range parameters that you
can use to define and control the affected frequency range.
To add the DeEsser 2 effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Detection max field: Shows the maximum level of the selected frequency. Click to reset.
• Detection meter slider: Drag to set the Threshold, or amplification level, above which
gain reduction of the selected frequency is applied.
• Reduction max field: Shows the maximum level (peak hold). Click to reset.
• Reduction meter slider: Drag to set the maximum amount of dynamic gain reduction
applied to the selected frequency.
• Threshold knob and field: Set the Threshold, or amplification level, above which gain
reduction of the selected frequency is applied.
• Max Reduction knob and field: Set the maximum amount of dynamic gain reduction
applied to the selected frequency.
• Frequency knob and field: Set the center or maximum frequency of the detection filter,
depending on the chosen filter.
• Mode pop-up menu: Choose Relative or Absolute mode. Relative is highly responsive
and works with both high- and low-level signals. Absolute works with high-level signals,
serving as a classic de-esser.
• Relative: In this mode, the level of the filtered signal (determined by the Range,
Frequency, and Filter settings) is compared with the full-bandwidth level of the
incoming signal. The Threshold parameter value determines the amplification level of
the filtered signal (because the level of the filtered signal will always be lower than
the full-bandwidth signal). When the amplified, filtered signal level is lower than the
full-bandwidth signal, the Detection meter shows a blue meter below the Threshold
value and no processing occurs. When the amplified, filtered signal level is higher
than the full-bandwidth level, the Detection meter shows a yellow meter above the
Threshold value and processing takes place.
• Absolute: The Detection level meter shows the level of the incoming filtered
signal (determined by the Range, Frequency, and Filter settings). When the level
exceeds the Threshold parameter value, the meter display switches from blue
(not processed) to yellow (processed). Low-level signals can only be processed in
Absolute mode if the Threshold parameter is set to a very low value.
• Range buttons: Set the filter frequency range. Split affects only signals within the set
frequency band. Wide affects the entire frequency range.
• Filter Solo button: Turn on to hear the filtered signal—the split frequency band—in
isolation, when Split is turned on.
1. In the Final Cut Pro timeline, select a clip with the DeEsser 2 effect applied, then open
the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
4. Set the frequency you want to reduce using the Frequency knob.
To make the frequency easier to hear and identify, click the Filter Solo button.
5. Drag the Threshold knob to the level at which DeEsser 2 should start to apply reduction.
To set a narrow frequency range, click the Split Range button. To set a broader range,
click the Wide Range button.
6. Drag the Max Reduction knob to set how much sibilance to reduce.
Note: When reducing sibilance, keep in mind that sibilance is a natural part of speech,
and removing too much may make your vocals sound strange.
Enveloper
The most important Enveloper controls are the two Gain sliders, one on each side of the
central display. These govern the Attack and Release levels of each respective phase.
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.
When using Enveloper, set the Threshold to the minimum value and leave it there. Only
when you seriously raise the release phase, which boosts the noise level of the original
recording, should you raise the Threshold slider a little. This limits Enveloper to affecting
only the useful part of the signal.
Drastic boosting or cutting of either the release or attack phase may change the overall
level of the signal. You can compensate for this by adjusting the Out Level slider.
Generally, you’ll find that Attack Time values of around 20 ms and Release Time values
of 1500 ms are good to start with. Then adjust them for the type of signal that you’re
processing.
The Lookahead slider defines how far into the future of the incoming signal Enveloper
looks, in order to anticipate future events. You generally won’t need to use this feature,
except when processing signals with extremely sensitive transients. If you do raise the
Lookahead value, you may need to adjust the Attack Time to compensate.
To add the Enveloper effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• 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. When
the Gain slider is set to the center position—0%—the signal is unaffected.
• Lookahead slider and field: Sets the pre-read analysis time for the incoming signal. This
enables Enveloper to know in advance what signals are coming, enabling accurate and
fast processing.
• (Attack) Time knob and field: Determines the amount of time it takes for the signal to
increase from the threshold level to the maximum Gain level.
• (Release) Time knob and field: Determines the amount of time it takes for the signal to
fall from the maximum gain level to the threshold level.
• (Release) Gain slider and field: Boosts or attenuates the release phase of the signal.
When the Gain slider is set to the center position—0%—the signal is unaffected.
• Out Level slider and field: Sets the level of the output signal.
To add the Expander effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Expansion meter: Shows the amount of gain (expansion) applied to the signal.
• Threshold knob and field: Set the threshold level. Signals above this level are expanded.
• Ratio knob and field: Set the expansion ratio—the ratio of signal expansion when the
threshold is exceeded.
• Auto Gain button: Turn on to compensate for the level increase caused by expansion.
When Auto Gain is active, the signal sounds softer, even when the peak level remains
the same.
Note: If you dramatically change the dynamics of a signal (with extreme Threshold and
Ratio values), you may need to reduce the Gain knob level to avoid distortion. In most
cases, turning on Auto Gain adjusts the signal appropriately.
• Knee knob and field: Determine the strength of expansion at levels close to the
threshold. Lower values result in more severe or immediate expansion—hard knee.
Higher values result in a gentler expansion—soft knee.
• Attack knob and field: Set the time it takes for Expander to respond to signals that
exceed the threshold level.
• Output Clip pop-up menu: Choose whether to apply clipping above 0 dB, and the type
of clipping. Soft and Hard change the signal around 0 dB in different ways, resulting in a
smoothed or squared-off distortion of signal peaks.
• Peak/RMS buttons: Determine whether the Peak or RMS method is used to analyze the
signal.
To add the Gain effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Phase Invert Left and Right buttons: Invert the phase of the left and right channels,
respectively.
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. For example, if you place microphones above and below a snare drum, inverting
the phase of either microphone can improve (or ruin) the sound. As always, rely on your
ears.
• Balance knob and field: Adjust the balance of the incoming signal between the left and
right channels.
• Swap L/R button: Swap the left and right output channels. Swapping occurs after the
Balance parameter in the signal path. The Swap L/R button is disabled when Mono is
turned on.
• Mono button: Output the summed mono signal on both the left and right channels.
Note: The Gain effect is available in mono, mono-to-stereo, and stereo instances. Only one
Phase Invert button is available in mono and mono-to-stereo modes. In mono mode, the
Balance, Swap L/R, and Mono parameters are also disabled. A separate Multichannel Gain
effect is also available in surround channels. This features per-channel Phase Invert and
Mute buttons, and Level sliders for each channel.
Limiter is used primarily when mastering. Typically, you apply Limiter as the very last
process in the mastering signal chain, where it raises the overall volume of the signal so
that it reaches, but does not exceed, 0 dB.
Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level, it has no
effect on a normalized signal. If the signal clips, Limiter reduces the level before clipping
can occur. Limiter cannot, however, fix audio that is clipped during recording.
To add the Limiter effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Input meters: Show input levels in real time. The Margin field shows the highest input
level. Click the Margin field to reset it.
• Output meters: Show output levels of the limited signal. The Margin field shows the
highest output level. Click the Margin field to reset it.
• Gain knob and field: Set the amount of gain applied to the input signal.
• Release knob and field: Set the time it takes for Limiter to stop processing, after the
signal falls below the threshold level.
• Output Level knob and field: Set the output level of the signal.
• Lookahead knob and field: Adjust how far ahead (in milliseconds) Limiter analyzes the
audio signal. This enables it to react earlier to peak volumes by adjusting the amount of
reduction.
Note: Lookahead causes latency, but this has no perceptible effect when you use
Limiter as a mastering effect on prerecorded material. Set it to higher values if you
want the limiting effect to occur before the maximum level is reached, thus creating a
smoother transition.
• Mode pop-up menu: Choose between Legacy and Precision algorithms. Use Precision
for hard limiting, but be aware that this can introduce distortion artifacts.
• Soft Knee button (Legacy mode): Turn on to limit the signal only when it reaches the
threshold. The transition to full limiting is nonlinear, producing a softer, less abrupt
effect, and reducing distortion artifacts that can be produced by hard limiting (in
Precision mode).
• True Peak Detection button (Precision mode): Turn on to detect intersample peaks in
the signal.
To add the Multichannel Gain effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Master slider and field: Set the master gain for the combined channel output.
• Channel gain sliders and fields: Set the amount of gain for the respective channel.
Multipressor
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. This
avoids the pumping effect often associated with high amounts of compression.
Multipressor allows the use of higher compression ratios on specific frequency bands, so it
can achieve a higher average volume without causing audible artifacts.
Raising the overall volume level can result in a corresponding increase in the existing noise
floor. Each frequency band features downward expansion, which allows you to reduce or
suppress this noise.
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.
Output parameters
The Out slider sets the overall output level. Set Lookahead to higher values when the Peak/
RMS fields are set to higher values (farther toward RMS). Set Auto Gain to On to reference
the overall processing to 0 dB, making the output louder.
To add the Multipressor effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Drag the horizontal bar up or down to adjust the gain make-up for that band.
• Drag the vertical edges of a band to the left or right to set the crossover
frequencies, which adjusts the band’s frequency range.
• Gain Make-up fields: Set the amount of the gain make-up for each band.
• Compr (Compression) Ratio fields: Set the compression ratio for the selected band.
Setting the parameter to 1:1 results in no compression of the band.
• Expnd Thrsh (Expansion Threshold) 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 (Expansion) Ratio fields: Set the expansion ratio for the selected band.
• Expnd (Expansion) 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.
• Attack fields: Set the amount of time before compression starts for the selected band,
after the signal exceeds the threshold.
• Release fields: Set the time required before compression stops on the selected band,
after the signal falls below the threshold.
• Solo buttons: Enable to hear compression on only the selected frequency band.
• Level meters: The bar on the left shows the input level, and the bar on the right shows
the output level.
• Threshold arrows: Two arrows appear to the left of each Level meter.
• Lookahead value field: Adjusts how far the effect looks forward in the incoming audio
signal, in order to react earlier to peak volumes, and therefore achieve smoother
transitions.
Noise Gate
Noise Gate works by allowing signals above the threshold level to pass unimpeded, while
reducing signals below the threshold level. This effectively removes lower-level parts of the
signal, while allowing the desired parts of the audio to pass.
The Attack, Hold, and Release knobs modify the dynamic response of Noise Gate. If you
want the gate to open extremely quickly for percussive signals such as drums, set the
Attack knob to a lower value. For sounds with a slow attack phase, such as string pads, set
Attack to a higher value. Similarly, when working with signals that fade out gradually or that
have longer reverb tails, set a higher Release knob value that allows the signal to fade out
naturally.
The Hysteresis slider provides another option for preventing chattering, without needing
to define a minimum Hold time. Use it to set the range between the threshold values
that open and close Noise Gate. This is useful when the signal level hovers around the
Threshold level, causing Noise Gate to switch on and off repeatedly, producing the
undesirable chattering effect. The Hysteresis slider essentially sets Noise Gate to open
at the Threshold level and remain open until the level drops below another, lower, level.
As long as the difference between these two values is large enough to accommodate the
fluctuating level of the incoming signal, Noise Gate can function without creating chatter.
This value is always negative. Generally, −6 dB is a good place to start.
In some situations, you may find that the level of the signal you want to keep and the level
of the noise signal are close, making it difficult to separate them. For example, when you
are recording a drum kit and using Noise Gate to isolate the sound of the snare drum, the
hi-hat may also open the gate in many cases. To remedy this, use the side-chain controls
to isolate the desired trigger signal with the High Cut and Low Cut filters.
To add the effect and show its controls, see Add Logic effects to clips.
2. Click the Monitor button to hear how the High Cut and Low Cut filters affect the
incoming trigger signal.
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.
To add the Noise Gate effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Gate and Ducker buttons: Set the operating mode. See Use Noise Gate.
• Threshold knob and field: Set the threshold level. Signals that fall below the threshold
are reduced in level.
• Hysteresis slider and field: Set the difference (in decibels) between the threshold values
that open and close the gate. This prevents the gate from rapidly opening and closing
when the input signal level is close to the threshold level.
• Attack knob and field: Set the time it takes to fully open the gate after the signal
exceeds the threshold.
• Hold knob and field: Set the time the gate remains open after the signal falls below the
threshold.
• Release knob and field: Set the time it takes to reach maximum attenuation after the
signal falls below the threshold.
• Lookahead slider and field: Control how far ahead Noise Gate analyzes the incoming
signal, allowing the effect to respond more quickly to peak levels.
• Filter button: Turn on to adjust the High Cutoff and Low Cutoff parameters.
• High Cutoff slider and field: Set the upper cutoff frequency for the side-chain signal.
• Low Cutoff slider and field: Set the lower cutoff frequency for the side-chain signal.
Note: When no external side chain is selected, the input signal is used as the side chain.
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 controls. The
signal ranges above and below the defined band can be individually processed with the
Low Level and High Level controls and the Super Energy and Sub Energy controls. See Use
Spectral Gate.
1. In the Final Cut Pro timeline, select a clip with the Spectral Gate effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Set the frequency band you want to process by using the Center Freq. and Bandwidth
controls.
The graphic display shows the band defined by these two parameters.
All incoming signals above and below the threshold level are divided into upper and
lower frequency ranges.
4. Rotate the Super Energy knob to control the level of the frequencies above the
threshold, and rotate the Sub Energy knob to control the level of the frequencies below
the threshold.
5. To mix the frequencies that fall outside the defined frequency band with the processed
signal, do any of the following:
• Drag the Low Level slider to blend the frequencies below the defined frequency band
with the processed signal.
• Drag the High Level slider to blend frequencies above the defined frequency band
with the processed signal.
• Drag the CF Modulation slider to define the intensity of the center frequency
modulation.
7. Drag the Gain slider to adjust the final output level of the processed signal.
To add the Spectral Gate effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Threshold slider and field: Set a threshold level for dividing frequency ranges. When the
threshold is exceeded, the frequency band defined by the Center Freq. and Bandwidth
parameters is divided into upper and lower frequency ranges.
• Speed slider and field: Set the modulation frequency for the defined frequency band.
• CF (Center Frequency) Modulation slider and field: Set the intensity of center frequency
modulation.
• BW (Bandwidth) Modulation slider and field: Set the amount of bandwidth modulation.
• Graphic display: Shows the frequency band defined by the Center Freq. and Bandwidth
parameters.
• Center Freq. (Frequency) knob and field: Set the center frequency of the band that you
want to process.
• Bandwidth knob and field: Set the width of the frequency band that you want to
process.
• Super Energy knob and field: Control the level of the frequency range above the
threshold.
• High Level slider and field: Mix the frequencies of the original signal—above the
selected frequency band—with the processed signal.
• Low Level slider and field: Mix the frequencies of the original signal—below the selected
frequency band—with the processed signal.
• Gain slider and field: Set the output level of Spectral Gate.
Surround Compressor
You can adjust the compression ratio, knee, attack, and release for the main, side,
surround, and LFE channels, depending on the chosen surround format. All channels
include an integrated limiter and provide independent threshold and output level controls.
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.
• The Link section at the top contains a series of menus where you assign each channel
to a group.
• The Main section includes controls common to all the main channels, and the threshold
and output controls for each channel.
• The LFE section on the lower right includes separate controls for the LFE channel.
To add the Surround Compressor effect to a clip and show the effect’s controls, see Add
Logic effects to clips.
Link controls
• Circuit Type pop-up menu: Choose the type of circuit emulated by the Surround
Compressor. The choices are Platinum, Classic A_R, Classic A_U, VCA, FET, and Opto
(optical).
• Grp. (Group) pop-up menus: Set group membership for each channel—A, B, C, or no
group (indicated by -). Moving the Threshold or Output Level slider for any grouped
channel moves the sliders for all channels assigned to that group.
Tip: Press Option and Command while moving the Threshold or Output Level slider
of a grouped channel to temporarily unlink the channel from the group. This allows you
to make independent threshold settings while maintaining the side-chain detection link
necessary for a stable surround image.
• Byp (Bypass) buttons: Bypass the channel. If the channel belongs to a group, all
channels in the group are bypassed.
• Detection pop-up menu: Choose whether Surround Compressor uses the maximum
level of each signal (Max) or the summed level of all signals (Sum) to exceed or fall
below the threshold.
• If Max is chosen, and any of the surround channels exceed or fall below the
threshold, those channels (or grouped channels) are compressed.
• If Sum is chosen, the combined level of all channels must exceed the threshold
before compression occurs.
• Ratio knob and field: Set the ratio of signal reduction when the threshold is exceeded.
• Knee knob and field: Determine the ratio of compression at levels close to the
threshold.
• Attack knob and field: Set the amount of time it takes to reach full compression, after
the signal exceeds the threshold.
• Release knob and field: Set the amount of time it takes to return to 0 compression, after
the signal falls below the threshold.
• Auto button: Make the release time dynamically adjust to the audio material.
• Threshold knob and field: Set the threshold for the limiter on the main channels.
• Main Compressor Thresholds sliders and fields: Set the threshold level for each
channel—including the LFE channel, which also has independent controls.
• Main Output Levels sliders and fields: Set the output level for each channel—including
the LFE channel, which also has independent controls.
• Ratio knob and field: Set the compression ratio for the LFE channel.
• Knee knob and field: Set the knee for the LFE channel.
• Attack knob and field: Set the attack time for the LFE channel.
• Release knob and field: Set the release time for the LFE channel.
• Auto button: Make the release time dynamically adjust to the audio material.
• Threshold knob and field: Set the threshold for the limiter on the LFE channel.
Effects such as chorus, flanging, and phasing are well-known examples. Modulation effects
typically delay the incoming signal by a few milliseconds and use a low-frequency oscillator
(LFO) to modulate the delayed signal. The LFO may also be used to modulate the delay
time in some effects.
An LFO is much like the sound-generating oscillators in synthesizers, but the frequencies
generated by an LFO are so low that they can’t be heard. Therefore, they are used only for
modulation purposes. LFO controls include speed (or frequency) and depth—also called
intensity—controls.
You can also control the ratio of the affected (wet) signal and the original (dry) signal.
Some modulation effects include feedback controls, which add part of the effect’s output
back into the effect input.
Other modulation effects involve pitch. The most basic type of pitch modulation effect
is vibrato. It uses an LFO to modulate the frequency of the sound. Unlike other pitch
modulation effects, vibrato alters only the delayed signal.
More complex Final Cut Pro modulation effects, such as Ensemble, mix several delayed
signals with the original signal.
You can use the Chorus effect to enrich the incoming signal and create the impression that
multiple instruments or voices are being played in unison. The slight delay time variations
generated by the LFO simulate the subtle pitch and timing differences heard when several
musicians or vocalists perform together. Using Chorus also adds fullness or richness to the
signal, and it can add movement to low or sustained sounds.
To add the Chorus effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• D-Mode button: Turn on to introduce a spatial filtering effect that resembles a well-
known vintage processor.
• Mix knob and field: Determine the balance between dry and wet signals.
To add the Ensemble effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• LFO1, LFO2, and Random On/Off buttons: Enable or disable LFO 1, LFO 2, or the random
LFO independently.
• Rate fields: Set the frequency of LFO 1, LFO 2, and random modulation.
• Intensity fields: Set the amount of LFO 1, LFO 2, and random modulation.
• Modulation is represented by a horizontal line and waveform. The line represents the
LFO intensity. The waveform represents the LFO rate. Drag the line or circular handle
vertically (or use the corresponding field) to set the modulation intensity. Drag the
waveform or circular handle horizontally (or use the corresponding field) to set the
modulation rate.
• LFO1 parameters: Drag the green handle to set the modulation rate and intensity.
• LFO2 parameters: Drag the blue handle to set the modulation rate and intensity.
• Random LFO parameters: Drag the white handle to set the modulation rate and
intensity.
• Voices knob and field: Set the number of chorus instances (voices) generated in
addition to the original signal.
• Stereo Spread slider and field: Distribute voices across the stereo or surround field. You
can set a value of 200% to artificially expand the stereo or surround base. Note that
monaural compatibility may suffer if you do this.
Note: When Ensemble is used in surround, the input signal is converted to mono before
processing. In essence, you insert the Ensemble effect as a multi-mono instance.
• Phase knob and field: Control the phase relationship between the individual voice
modulations. The value you choose here depends on the number of voices, which is
why it is shown as a percentage value rather than in degrees. The value 100 (or −100)
indicates the greatest possible distance between the modulation phases of all voices.
• Volume Compensation knob and field: Compensate for effects signal volume changes
caused by adjusting the Voices value.
• Output Mix knob and field: Set the balance between dry and wet signals.
To add the Flanger effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Rate knob and field: Set the frequency, or speed, of the low-frequency oscillator (LFO).
• Feedback knob and field: Set the amount of the effect signal that is routed back to the
input. This can change the tonal color and make the sweeping effect more pronounced.
Negative Feedback values invert the phase of the routed signal.
• Mix knob and field: Determine the balance between dry and wet signals.
Sonically, phasing is used to create whooshing, sweeping sounds that wander through the
frequency spectrum. It is a commonly used guitar effect, but it is suitable for a range of
signals.
To add the Phaser effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Stages knob and field: Choose phaser algorithms (even numbers) or comb filtering (odd
numbers).
• The 4, 6, 8, 10, and 12 settings switch between five different phaser algorithms. All
are modeled on analog circuits, with each designed for a specific application.
• The 5, 7, 9, and 11 settings don’t generate actual phasing effects. The more subtle
comb filtering effects produced by odd-numbered settings can, however, be useful.
• Ceiling and Floor sliders and fields: Determine the frequency range affected by LFO
modulations. Drag the green slider area between Ceiling and Floor to move the entire
range.
• Rate 1 and 2 knobs and fields: Set the speed for each LFO.
• Sync buttons: Synchronize the modulation speed of each LFO with the clip. Choose
musical note values with the Rate 1 and Rate 2 knobs.
• Phase knob and field: Control the phase relationship between individual channel
modulations. Available only in stereo and surround instances. At 0°, extreme modulation
values are achieved simultaneously for all channels. At 180° or −180°, there is the
greatest possible distance between channel modulation phases.
• Mix slider and fields (LFO section): Determine the ratio between the two LFOs.
• Distribution pop-up menu: Choose how phase offsets between individual channels are
distributed in the surround field. Choose Circular, Left↔Right, Front↔Rear, Random, or
New Random. Available only in surround instances.
Note: When you load a setting that uses the Random option, the saved phase offset
value is recalled. If you want to randomize the phase setting again, choose New Random
from the Distribution pop-up menu.
• Level knob and field: Determine the amount of effect signal routed back to the input.
• Warmth button: Enable or disable a distortion circuit, suitable for warm overdrive
effects.
• Low Cut and High Cut sliders and fields: Set the cutoff frequency of the lowpass (LP)
and highpass (HP) filters.
• Mix slider and field (Out section): Determine the balance of dry and wet signals.
Negative values result in a phase-inverted mix of the effect and direct (dry) signal.
You can choose between three different vibrato and chorus types. The stereo version of
the effect features two additional controls—Stereo Phase and Rate Right. These set the
modulation speed independently for the left and right channels.
To add the Scanner Vibrato effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• In each of the Vibrato positions, only the delay line signal is heard. Each vibrato type
has a different intensity.
• In the three Chorus positions (C1, C2, and C3), the signal of the delay line is mixed
with the original signal. Mixing a vibrato signal with an original, statically pitched
signal results in a chorus effect. This organ-style chorus sounds different from the
Chorus plug-in.
• Depth knob and field: Set the intensity of the chosen chorus effect type. If a vibrato
effect type is chosen, this control has no effect.
• Stereo Phase knob and field: Determine the phase relationship between left and right
channel modulations. If you set the knob to Free, you can set the modulation speed of
the left and right channels independently.
• Rate Left knob and field: Set the modulation speed of the left channel when Stereo
Phase is set to Free. If Stereo Phase is set to a value between 0° and 360°, Rate Left
sets the modulation speed for both the left and right channels, and the Rate Right knob
has no function.
• Rate Right knob and field: Set the modulation speed of the right channel when Stereo
Phase is set to Free.
To add the Tremolo effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Rate knob and field: Set the frequency of the low-frequency oscillator (LFO).
• Smoothing slider and field: Change the shape of the LFO waveform. Also see the
description of the Symmetry parameter, below.
• Distribution pop-up menu: Choose how phase offsets between individual channels are
distributed in the surround field. Choose Circular, Left↔Right, Front↔Rear, Random, or
New Random. Available only in surround instances.
• Offset field: Set the amount of left or right movement for the modulation (cycle). This
results in small or large tremolo variations.
• Symmetry field: Skew the balance toward the upward or downward phase of waveform
cycles.
Note: If Symmetry is set to 50% and Smoothing to 0%, the LFO waveform becomes
rectangular. The timing of the highest volume signal is then equal to the timing of the
lowest volume signal, with the switch between both states occurring abruptly.
• Phase field: Control the phase relationship between individual channel modulations in
stereo or surround signals. At 0, modulation values are reached simultaneously for all
channels. At values of 180 or −180, there is the greatest possible distance between the
modulation phases of the channels.
• Waveform display: Shows and lets you edit modulation Offset, Symmetry, and Phase.
Modulation is represented by a repeating waveform cycle for each channel. The height
represents the LFO intensity, or depth. Adjust the Depth knob to change the LFO
intensity. Two green handles and two blue handles let you control Offset, Symmetry, and
Phase values.
• Offset: Drag the field or the green handle at the left of any waveform cycle to set the
left or right modulation movement.
• Symmetry: Drag the field or the green handle at the right of any waveform cycle to
set the balance between upward and downward waveform phases.
• Phase: Drag the field or either blue handle of any waveform cycle to control the
phase relationship between channel modulations.
You can also define a scale to automatically correct some, but not all, sung notes in a vocal
performance, for example. This enables you to effectively perfect an imperfect vocal take.
You can also use pitch correction effects creatively, modifying all pitched notes in a
performance to a single pitch or a particular key.
See Intro to Pitch Correction, Pitch Shifter controls, and Intro to Vocal Transformer.
Pitch Correction
Pitch correction works by accelerating and slowing down the audio playback speed,
ensuring that the input signal (sung vocal) always matches the correct note pitch. If you
try to correct larger intervals, you can create special effects. Natural articulations of the
performance, such as breath noises, are preserved. Any scale can be defined as a pitch
reference (technically speaking, this is known as a pitch quantization grid), with improperly
intonated notes corrected in accordance with this scale.
The Scale pop-up menu allows you to choose different pitch quantization grids. The scale
that is set manually (with the keyboard graphic in the effect window) is called the User
Scale. The default setting is the Chromatic scale. If you’re unsure of the intervals used in
any given scale, choose it in the Scale menu and look at the keyboard graphic. You can
alter any note in the chosen scale by clicking the keyboard keys. Any such adjustments
overwrite the existing user scale settings.
There is only one user scale per project. You can, however, create multiple user scales and
save them as Pitch Correction effect settings files.
Tip: The drone scale uses a fifth as a quantization grid, and the single scale defines a
single note. Neither of these scales is meant to result in realistic singing voices, so if you’re
after interesting effects, you should give them both a try.
Use the Root pop-up menu to choose the root note of the scale. (If you chose User Scale
or Chromatic in the Scale pop-up menu, the Root pop-up menu is nonfunctional.) You may
freely transpose the major and minor scales, and scales named after chords.
Note: The settings are valid for all octave ranges. Individual settings for different octaves
aren’t provided.
You can use the small bypass (byp) buttons above the green (black) and below the blue
(white) keys to exclude notes from correction. This is useful for blue notes. Blue notes are
notes that slide between pitches, making the major and minor status of the keys difficult
to identify. For example, one of the major differences between C minor and C major is the
Eb (E flat) and Bb (B flat), instead of the E and B. Blues singers glide between these notes,
creating an uncertainty or tension between the scales. Using the bypass buttons, you can
exclude particular keys from changes, leaving them as they were.
If you enable the Bypass All button, the input signal is passed through unprocessed and
uncorrected. This is useful for spot corrections to pitch through use of automation. Bypass
All is optimized for seamless bypass enabling or disabling in all situations.
For more information on automating Pitch Correction, open Final Cut Pro and choose
Help > Final Cut Pro Help.
To add the Pitch Correction effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Use Global Tuning button: Turn on to use the project’s Tuning settings for the pitch
correction process. Turn off to set the reference tuning with Ref. Pitch.
• Normal and Low buttons: Set the pitch range that is scanned (for notes that need
correction).
• Ref. (Reference) Pitch field: Set the desired reference tuning, in cents (relative to the
root).
• Root pop-up menu and field: Choose the root note of the scale.
• Keyboard: Click a key to exclude the corresponding note from pitch quantization grids.
This effectively removes this key from the scale, resulting in note corrections that are
forced to the nearest available pitch (key).
• Byp (Bypass) buttons: Exclude the corresponding note from pitch correction. In other
words, all notes that match this pitch will not be corrected. This applies to both user
and built-in scale quantization grids.
• Bypass All button: Compare the corrected and original signal, or use for automation
changes.
• 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: Set how quickly the voice reaches the corrected destination
pitch. Singers use portamenti and other gliding techniques. If you choose a Response
value that’s too high, seamless portamenti turn into semitone-stepped glissandi, but
the intonation is perfect. If the Response value is too low, the pitch of the output signal
doesn’t change quickly enough. The optimum setting for this parameter depends on the
singing style, tempo, vibrato, and accuracy of the original performance.
• Detune slider and field: Detune the output signal by the set value.
• Input Detune slider and field (Extended controls area): Detune the input signal by the
set value, thus affecting it before any pitch correction takes place. This parameter is of
particular benefit when automated.
Pitch Shifter
To add the Pitch Shifter effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Semi Tones knob and field: Set the pitch shift value in semitones.
• Cents knob and field: Control detuning of the pitch shift value in cents (1/100th of a
semitone).
• Mix knob and field: Set the balance between the effect and original signals.
• Latency Comp (Compensation) button: Turn on to compensate for delays that may be
introduced by some algorithms with particular types of source material.
• Stereo Link buttons: Click Normal to retain the source stereo signals. Invert swaps
(inverts) stereo channel signals, with right channel processing occurring on the left, and
vice versa.
• Speech: Provides a balance between both the rhythmic and harmonic aspects of the
signal. This is suitable for complex signals such as spoken-word recordings, rap, and
hybrid signals such as rhythm guitar.
• Vocals: Retains the intonation of the source, making it well-suited for signals that are
inherently harmonic or melodious, such as string pads.
• Manual: Uses the settings of the Delay, Crossfade, and Stereo Link parameters.
Note: The following controls are active only when Manual is chosen from the Timing pop-
up menu.
• Delay slider and field: Set the amount of delay applied to the input signal. The lower
the frequencies of the input signal, the higher (longer) a delay time is required—to
effectively pitch shift the signal.
• Crossfade slider and field: Set the range (shown as a percentage of the original signal)
used to analyze the input signal.
1. In the Final Cut Pro timeline, select a clip with the Pitch Shifter effect applied, then open
the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. To set the amount of transposition, or pitch shift, drag the Semi Tones slider.
4. Click the Timing pop-up menu and choose the algorithm that best matches the material
you’re working with.
If you’re working with material that doesn’t fit any of these categories, experiment with
each of the algorithms (starting with Speech), then compare the results and use the one
that best suits your material.
Tip: While auditioning and comparing different settings, temporarily set the Mix
parameter to 100% because this makes Pitch Shifter artifacts easier to hear.
You can shift the formants independently, which means that you can turn a vocal clip into
a Mickey Mouse voice while maintaining the original pitch. Formants are characteristic
emphases of certain frequency ranges. They are static and do not change with pitch.
Formants are responsible for the specific timbre of a given human voice.
Vocal Transformer is well suited to extreme vocal effects. The best results are achieved
with monophonic signals, including monophonic instrument clips. It is not designed for
polyphonic voices—such as a choir on a single clip—or other choral clips.
As you adjust the Pitch parameter, you might notice that the formants don’t change.
The Pitch parameter is expressly used to change the pitch of a voice, not its character. If
you set negative Pitch values for a female soprano voice, you can turn it into an alto voice
without changing the specific character of the singer’s voice.
The Formant parameter shifts the formants, while maintaining—or independently altering—
the pitch. If you set this parameter to positive values, the singer sounds like Mickey Mouse.
By adjusting the parameter downward, you can achieve vocals reminiscent of Darth Vader.
Tip: If you set Pitch to 0 semitones, Mix to 50%, and Formant to +1 (with Robotize
turned off), you can effectively place a singer (with a smaller head) next to the original
singer. Both will sing with the same voice, in a choir of two. This doubling of voices is quite
effective, with levels easily controlled by the Mix parameter.
The Tracking slider and field work in conjunction with four buttons that immediately set the
slider to the most useful values, as follows:
• 0 button: Sets the slider to 0%. Delivers interesting results, with every syllable of
the vocal clip being sung at the same pitch. Low values turn sung lines into spoken
language.
• 1 button: Sets the slider to 100%. The range of the melody is maintained. Higher values
augment, and lower values diminish, the melody.
The Pitch Base parameter is used to transpose the note that the Tracking parameter
is following. For example, with Tracking set to 0%, the pitch of the (spoken) note is
transposed to the Pitch Base value.
To add the Vocal Transformer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Pitch knob and field: Determine the amount of transposition applied to the input signal.
• Robotize button: Turn on Robotize mode, which is used to augment, diminish, or mirror
the melody.
• Pitch Base slider and field (available only in Robotize mode): Transpose the note that
the Tracking parameter is following.
• Tracking slider, field, and buttons (available only in Robotize mode): Control how the
melody is changed in Robotize mode.
• Mix slider and field: Define the level ratio between the original (dry) and effect signals.
• Formant knob and field: Shift the formants of the input signal.
• Grain Size slider and field (Extended controls area): The Vocal Transformer effect
algorithm is based on granular synthesis. Use the Grain Size parameter to set the size
of the grains, thus affecting the precision of the process. Experiment to find the best
setting (try Auto first).
• Formants pop-up menu (Extended controls area): Determine whether Vocal Transformer
processes all formants (“Process always” setting), or only the voiced ones (Keep
Unvoiced Formants setting). The Keep Unvoiced Formants option leaves sibilant sounds
in a vocal performance untouched. This setting produces a more natural-sounding
transformation effect with some signals.
• Detune slider and field (Extended controls area): Detune the input signal by the set
value. This parameter is of particular benefit when automated.
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).
Digital recording introduced digital reverb effects, which consist of thousands of delays
of varying lengths and intensities. The time differences between the original signal and
the arrival of the early reflections can be adjusted by a parameter known as predelay.
The average number of reflections in a given period of time is determined by the density
parameter. The regularity or irregularity of the density is controlled with the diffusion
parameter. An example of a digital reverb is ChromaVerb. See Intro to ChromaVerb.
Computers make it possible to sample the reverb characteristics of real spaces, using
convolution reverbs. These room characteristic sample recordings are known as impulse
responses.
Convolution reverbs work by convolving (combining) an audio signal with the impulse
response recording of a room’s reverb characteristics. Final Cut Pro includes a convolution
reverb called Space Designer. See Intro to Space Designer.
ChromaVerb
The fundamental approach behind ChromaVerb diverges from other methods of reverb
creation. It is based on the principle of a circular structure in which the sound is gradually
absorbed, much like in a real room. The absorption characteristics are dependent on the
chosen room type and reverb parameter settings.
Each room type offers a unique tonal color, ranging from dense rooms to wide spaces and
large halls.
• Main window: Shows common reverb parameters such as Attack, Size, Density,
Distance, and Decay. A visualization of the reverb output is shown in a graphic display
that allows you to directly edit damping factors, thus changing decay frequencies and
dependencies.
• Details window: Provides access to advanced parameters such as Width, Quality, and
Modulation. The graphic display shows an editable Output EQ that you can use to shape
the ChromaVerb output signal.
The details window contains advanced parameters and shows the built-in six-band Output
EQ. Click the Main or Details button at the upper right to switch between windows.
To add the ChromaVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Main/Details button: Switch between the main window and the details window.
The Damping EQ adjusts the frequencies of the decay signal. It is divided into four
bands, with independent high and low shelving EQ bands, and two parametric EQ
bands.
• Low shelving EQ dot: Drag horizontally to adjust the frequency, and drag vertically to
adjust the ratio of the low shelving band in the Damping EQ. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• Low parametric EQ dot: Drag horizontally to adjust the frequency, and drag vertically to
adjust the ratio of the low parametric band in the Damping EQ. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• High parametric EQ dot: Drag horizontally to adjust the frequency, and drag vertically to
adjust the ratio of the high parametric band in the Damping EQ. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• High shelving EQ dot: Drag horizontally to adjust the frequency, and drag vertically to
adjust the ratio of the high shelving band in the Damping EQ. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• Ratio field: Set the Decay parameter timing ratio of the Damping EQ band.
• Q field: Set the band width surrounding the center frequency of the Damping EQ
band.
• Visualization On/Off button: Turn on or turn off the real-time visualization in the graphic
display.
Note: Visualization works only on computers that support the Metal framework, which
helps to display graphics faster by reducing processing bottlenecks.
The details window contains advanced controls and shows the built-in six-band Output EQ.
Click the Main or Details button at the upper right to switch between windows.
To add the ChromaVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Attack knob and field: Set the attack phase of the reverb. This affects either the volume
or the density build-up time, depending on the chosen room type.
• For the following room types, the Attack parameter increases volume over time:
Theatre, Dense Room, Smooth Space, Reflective Hall, Strange Room, Airy.
• For the following room types, the Attack parameter sets the amount of time it takes
for the reverb to reach the maximum density value, determined with the Density
knob: Room, Chamber, Concert Hall, Synth Hall, Digital, Dark Room, Vocal Hall,
Bloomy.
• Size knob and field: Determine the dimensions of the room. Higher values correspond to
a larger space.
• Density knob and field: Adjust the density of the early and late reflections
simultaneously, depending on the room type.
• Predelay field: Set the time between the start of the original signal and the arrival of
the early reflections. A short predelay setting tends to push sounds away, and longer
predelay settings tend to bring sounds more to the forefront.
An extremely short predelay setting can color the sound and make it difficult to pinpoint
the position of the signal source. A very long predelay setting can be perceived as an
unnatural echo and can divorce the original signal from its early reflections, leaving an
audible gap between them. An optimal predelay setting depends on the type of input
signal—or more precisely, the envelope of the input signal. Percussive signals generally
require shorter predelays than signals where the attack fades in gradually. A good
working method is to use the longest possible Predelay value before you start to hear
side effects, such as an audible echo. When you reach this point, reduce the Predelay
setting slightly.
• Predelay sync button: Turn on to restrict Predelay values to divisions synchronized with
the host application tempo.
• Decay sync button: Turn on to restrict Decay values to divisions synchronized with the
host application tempo.
• Freeze button: Turn on to recirculate the signal infinitely inside the chosen room type.
• Distance knob and field: Set the perceived distance from the source by altering early
and late energy.
• Dry and Wet sliders and fields: Independently set the levels of the source and effect
signals.
The main window contains the main window controls and shows a Damping EQ overlay in
the graphic display. Click the Main or Details button at the upper right to switch between
windows.
To add the ChromaVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Output EQ On/Off button: Turn on to enable the Output EQ, which adjusts frequencies
of the overall combined reverb and source signal.
• Main/Details button: Switch between the main window and the details window.
• Graphic display: Shows the six-band Output EQ curve. You can change the Output EQ
curve in the display or use the fields below it.
• Band 1 On/Off button: Turn on a highpass filter that allows high frequencies to pass and
reduces the level of low frequencies near the cutoff (set) frequency. When band 1 is
active, you can change band parameters directly in the graphic display.
• Band 1 background or dot: Drag the red shaded area horizontally to change the
frequency value. Drag the red dot horizontally to change the frequency, and drag
vertically to change the Q value. Option-click to reset to default values. Option-
Command-drag vertically to change the Q value.
• Band 2 background or dot: Drag the orange shaded area or dot horizontally to
change the frequency, and drag vertically to change the gain value. Option-click to
reset to default values. Option-Command-drag vertically to change the Q value.
• Band 3 On/Off button: Turn on a low parametric filter with three controls. Frequency
sets a center frequency. Q sets the width of the frequency band around the center
frequency. Gain sets the level of the band. When band 3 is active, you can change band
parameters directly in the graphic display.
• Band 3 background or dot: Drag the green shaded area or dot horizontally to change
the frequency, and drag vertically to change the gain value. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• Band 4 On/Off button: Turn on a high parametric filter with three controls. Frequency
sets a center frequency. Q sets the width of the frequency band around the center
frequency. Gain sets the level of the band. When band 4 is active, you can change band
parameters directly in the graphic display.
• Band 4 background or dot: Drag the blue shaded area or dot horizontally to change
the frequency, and drag vertically to change the gain value. Option-click to reset to
default values. Option-Command-drag vertically to change the Q value.
• Band 5 On/Off button: Turn on a high shelving filter that adjusts the level of high
frequencies and has a minimal impact on frequencies below the cutoff (set) frequency.
When band 5 is active, you can change band parameters directly in the graphic display.
• Band 5 background or dot: Drag the purple shaded area or dot horizontally to
change the frequency, and drag vertically to change the gain value. Option-click to
reset to default values. Option-Command-drag vertically to change the Q value.
• Band 6 On/Off button: Turn on a lowpass filter that allows low frequencies to pass and
reduces the level of high frequencies near the cutoff (set) frequency. When band 6 is
active, you can change band parameters directly in the graphic display.
• Band 6 background or dot: Drag the pink shaded area to change the frequency
value. Drag the pink dot horizontally to change the frequency, and drag vertically to
change the gain value. Option-click to reset to default values. Option-Command-
drag vertically to change the Q value.
• Gain field: Set the level of the selected EQ band (bands 2 to 5).
• Order field: Set the order (filter slope) for band 1 or band 6.
• Q field: Set the band width surrounding the frequency of the selected EQ band.
The main window contains the main window controls and shows a Damping EQ overlay in
the graphic display. Click the Main or Details button at the upper right to switch between
windows.
To add the ChromaVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Quality pop-up menu: Choose a quality level. Low produces a grainy reverb with
noisy modulation. High sounds clean and precise. Ultra delivers a smooth, expensive-
sounding reverb.
• Mod (Modulation) Speed slider and field: Set the speed of the built-in LFO.
• Mod (Modulation) Depth slider and field: Set the depth of LFO modulation. The range is
determined by the chosen room type.
• Mod (Modulation) Source buttons: Choose a sine, random, or noise waveform for the
LFO.
• Smoothing slider and field: Change the shape of the LFO waveform. The random
waveform is smoothed, and the sine and noise waveforms are saturated.
• Early/Late slider and field: Set the level of early and late reflections. These vary
depending on the Distance parameter value. See Main window controls.
• Width slider and field: Set the stereo width of the reverb.
• Mono Maker On/Off button: Turn on to remove stereo information below the frequency
set with the corresponding slider.
• Mono Maker slider and field: Set a frequency below which stereo information is
removed. This compensates for perceived level losses in the overall low-frequency
range.
Chamber A punchy reverb that emulates a small to medium-size room. It has a fast
attack and high echo density with low coloration.
Concert Hall A large space with long delays in the initial sound, a slow build, minimal
high-end response, and moderate diffusion build.
Synth Hall Wider than the Room model with the sparsest reflections of all room
types.
Dark Room A small to medium-size, dark sounding, less dense room reverb.
Dense Room A small room with a dense reflection pattern that builds very quickly.
Vocal Hall A medium-size to large smooth vocal hall with a midrange number of
reflections.
Reflective Hall A medium-size to large highly reflective hall reverb with a low reflection
density.
FX - Strange Room A medium-size space with midrange reflection density and a distinct color.
FX - Bloomy A large space reverb with moderate reflection density that creates
blooming decays.
To add the SilverVerb effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Predelay knob and field: Set the time between the original signal and the reverb signal.
• Reflectivity knob and field: Define how reflective the imaginary walls, ceiling, and floor
are.
• Size knob and field: Define the dimensions of the simulated room.
• Density/Time knob and field: Determine both the density and the duration of the reverb.
• Low Cut slider and field: Filter frequencies below the set value out of the reverb signal.
This affects only the tone of the reverb signal, not the original signal.
• High Cut slider and field: Filter frequencies above the set value out of the reverb signal.
This affects only the tone of the reverb signal, not the original signal.
• Modulation On/Off button: Enable or disable the LFO. This affects the Rate, Phase, and
Intensity parameters.
• Rate knob and field: Set the frequency, or speed, of the LFO.
• Phase knob and field: Define the phase of the modulation between the left and right
channels of the reverb signal.
• At 0°, the extreme values (minimum or maximum) of the modulation are achieved
simultaneously on both the left and right channels.
• At a value of 180°, the extreme values opposite each other (left channel minimum,
right channel maximum, or vice versa) are reached simultaneously.
• Intensity slider and field: Set the modulation amount. A value of 0 turns off the delay
modulation.
• Dry and Wet sliders and fields: Set the balance between the effect (wet) and original
(dry) signals.
Convolution can be used to place your audio signal inside any space, including a speaker
cabinet, a plastic toy, a cardboard box, and so on. All you need is an IR recording of the
space.
Space Designer also offers features such as envelopes, filters, EQ, and stereo/surround
balance controls, which provide precise control over the dynamics, timbre, and length of
the reverberation.
Space Designer can operate as a mono, stereo, true stereo (meaning each channel is
processed discretely), or surround effect.
You can, however, record, edit, and play back any movement of the following
Space Designer parameters:
• Stereo X-Over
• Direct Output
• Reverb Output
• Envelope, filter, and EQ controls: Use the buttons in the display mode bar to switch
the main display and parameter bar between envelope, filter, and EQ views. Use the
main display to edit parameters graphically, or use the parameter bar to edit them
numerically. See Intro to envelope and EQ controls.
• Global controls: After your impulse response is loaded or generated, use these controls
to determine how Space Designer operates on the overall signal and impulse response.
Included are input and output controls, predelay, and more. See Intro to global controls.
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 waveform changes in the main display.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
When you first click the Sampled IR button, you’re prompted to choose an impulse
response file from any folder.
• IR Sample pop-up menu: Shows the name of the loaded impulse response. Choose an IR
sample command. The format of the impulse response is shown to the right of the pop-
up menu. See Intro to global controls.
• Load IR: Loads an impulse response sample without changing the envelopes.
• Load IR & Init (Initialize): Loads an impulse response sample and initializes all
envelopes.
• Show in Finder: Opens a Finder window that shows the location of the current
impulse response.
• Open IR Utility: Opens the Impulse Response Utility, which lets you create your own
impulse response files.
• Quality pop-up menu: Choose the sample rate. Lo-Fi produces a grainy reverb. Low
halves the host application sample rate. Medium matches the host application sample
rate. High is smooth and clean sounding.
• Reverse button: Reverse the impulse response and envelopes. When the impulse
response is reversed, you are effectively using the tail rather than the front end of
the sample. You may need to adjust the Predelay and other parameter values when
reversing. See Intro to global controls.
• Size knob and field: Adjust the sample rate of the loaded impulse response file, thereby
changing the perceived size of the reverb by widening or narrowing the room. Size can
also be used to preserve the original length of the impulse response when changing the
sample rate with the Quality pop-up menu.
The Size knob value has an impact on the decay because it is multiplied with the Length
knob value. To explain, a Length knob value of 100% and a Size knob value of 100%
result in a decay that is the full length of the loaded impulse response.
To add the effect and show its controls, see Add Logic effects to clips.
When you first click the Sampled IR button, you’re prompted to choose an impulse
response file.
To add the effect and show its controls, see Add Logic effects to clips.
2. Click the IR Sample pop-up menu, then choose one of the following commands:
• Load IR: Loads an impulse response sample without changing the envelopes.
• Load IR & Init (Initialize): Loads an impulse response sample and initializes the
envelopes.
• Show in Finder: Opens a Finder window that shows the location of the currently
loaded IR file.
• Open IR Utility: Opens the Impulse Response Utility window. This application lets you
create your own impulse response files.
All impulse responses that ship with Final Cut Pro are installed in the /Library/Audio/Impulse
Responses/Apple folder. Deconvolution files have the filename extension .sdir.
Any mono, stereo, AIFF, SDII, or WAV file can be used as an IR. In addition, surround
formats up to 7.1, discrete audio files, and B-format audio files that comprise a single
surround IR can also be used.
Note: You can switch between a loaded impulse response sample and a synthesized
impulse response without losing the settings of the other.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
Repeated clicks of the Synthesized IR button randomly generate new impulse responses
with slightly different reflection patterns. The current impulse response state is
saved with the setting file, including parameter and other values that represent the IR
reflection patterns and characteristics.
To add the effect and show its controls, see Add Logic effects to clips.
2. To set the sample rate of an impulse response, click the Quality pop-up menu and
choose an option:
• Lo-Fi: This setting divides the sample rate by four. If the project sample rate is
96 kHz, the impulse response sample rate is converted to 24 kHz. If the project
sample rate is 44.1 kHz, the impulse response sample rate is converted to
11.025 kHz, and so on.
• Low: This setting effectively halves the sample rate. If the project sample rate is
96 kHz, the impulse response sample rate is converted to 48 kHz. If the project
sample rate is 44.1 kHz, the impulse response sample rate is converted to
22.05 kHz, and so on.
When you select a half sample rate, the impulse response becomes twice as long.
The highest frequency that can be reverberated is halved. This results in a behavior
that is much like doubling every dimension of a virtual room—multiplying the volume
of a room by eight. The Low (and Lo-Fi) setting can also be used for interesting
tempo, pitch, and retro digital effects. Another benefit of reducing the sample rate
is that processing requirements drop significantly, making the lower quality settings
useful for large, open spaces.
This behavior also applies when you choose Lo-Fi, but the sample rate is divided by
four and the impulse response is multiplied in length four times.
• Medium: Space Designer uses the current project sample rate. The sample rate of
a loaded impulse response is automatically converted to match the current project
sample rate, if necessary. For example, this allows you to load a 44.1 kHz impulse
response into a project with a sample rate of 96 kHz, and vice versa.
3. To retain the original length of the impulse response when the sample rate is changed,
adjust the Size knob value.
Using this parameter with your Quality pop-up menu choice can lead to interesting
results.
If you’re using Space Designer in a project that has a higher sample rate than the
impulse response, you may also want to reduce the impulse response sample rate.
Adjust the Size knob value to reduce CPU processing time without compromising reverb
quality.
Tip: You can make similar adjustments in Synthesized IR mode. Most typical reverb
sounds don’t contain an excessive amount of high-frequency content. If your project
has a sample rate of 96 kHz, for example, you would need to use lowpass filtering to
obtain the mellow frequency response characteristics of many reverb sounds. A better
approach would be to first reduce the high frequencies by choosing a lower rate from
the Quality pop-up menu, and then use the lowpass filter, thus conserving significant
CPU resources. Keep in mind that longer impulse responses (sampled or synthesized)
place a higher strain on the CPU.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Rotate the Length knob to set the length of the impulse response—sampled or
synthesized.
The Length knob setting changes the decay value, depending on the current Size knob
value. To clarify, a Length value of 100% and a Size value of 100% result in a decay that
is the full length of the loaded impulse response.
Note: When you’re using a sampled impulse response file, the combined Length (and
Size) parameter values cannot exceed the length of the underlying impulse response
sample.
• Main display: Shows the impulse response waveform and all active envelopes. You can
graphically edit envelopes or the Output EQ curve.
• Parameter bar: View and numerically edit the parameter values of the selected envelope
or EQ curve.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• In Sampled IR mode: The IR Sample pop-up menu is shown in the display mode bar,
beside the Volume Env and Filter Env buttons. See Intro to impulse responses.
• In Synthesized IR mode: The Density Env buttons are shown in the display mode bar,
beside the Volume Env and Filter Env buttons. See Density envelope controls.
• Volume Env (Envelope) button: Show the volume envelope in the foreground of the main
display. Other active envelope curves are shown as transparencies in the background.
See Volume envelope controls.
• Filter Env (Envelope) On/Off button: Enable or disable the filter envelope. This also
automatically turns the filter on or off.
• Filter Env (Envelope) button: Show the filter envelope in the foreground of the main
display. Other active envelope curves are shown as transparencies in the background.
See Filter and filter envelope controls.
• Density Env (Envelope) On/Off button: Enable or disable the density envelope.
• Density Env (Envelope) button: Show the density envelope in the foreground of the main
display. Other active envelope curves are shown as transparencies in the background.
See Density envelope controls.
Note: The density envelope is available only in Synthesized IR mode. The Density Env
buttons aren’t shown when Sampled IR mode is active.
• Output EQ button: Show the six-band Output EQ in the main display. See Output EQ
controls.
The combined total of the volume and filter envelope Attack and Decay time parameter
values is equal to the length of the synthesized or sampled impulse response, unless the
decay time is reduced. See Set impulse response lengths.
The positions of nodes in the main display indicate the current parameter value shown
in the parameter bar below—for Init Level, Attack, Decay, and other envelope parameter
values. If you edit any numerical value in the parameter bar, the corresponding node moves
in the main display, and vice versa.
To add the effect and show its controls, see Add Logic effects to clips.
2. In Space Designer, vertically drag the parameter field—Attack, for example—in the
parameter bar at the bottom of the main display.
The corresponding node (including associated nodes, if applicable) moves in the main
display.
To add the effect and show its controls, see Add Logic effects to clips.
2. In Space Designer, drag the node (large, solid circle) in any available direction.
The corresponding field value changes in the parameter bar below the main display.
To add the effect and show its controls, see Add Logic effects to clips.
2. In Space Designer, drag the envelope curve (the line itself) in the main display.
3. Drag the small nodes (hollow circles) attached to a line for fine adjustments to envelope
curves.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Init (Initial) Level node and field: Set the initial volume level of the impulse response
attack phase. Drag the node vertically. It is expressed as a percentage of the full-scale
volume of the impulse response file. The attack phase is generally the loudest point
of the impulse response. Set Init Level to 100% to ensure maximum volume for early
reflections.
• Attack node and field: Determine the time before the decay phase of the volume
envelope begins. Drag the node horizontally.
• Decay node and field: Set the length of the decay phase. Drag the node horizontally.
Note: The overall decay is determined by the global Length and Size parameter values.
For example, a Length value of 100% and a Size value of 100% result in a decay that is
the full length of the loaded impulse response.
• Lin (Linear) button: The decay curve of the volume envelope is shaped by a linear
algorithm, and results in a less natural-sounding reverb tail.
• End Level node and field: Set the end volume level, expressed as a percentage of the
overall volume envelope. Drag the node vertically.
• If End Level is set to 0%: You can fade out the reverb tail.
• If End Level is set to 100%: You can’t fade out the tail, and the reverb stops abruptly
if the end point falls within the tail. If the end time falls outside the reverb tail, End
Level has no effect.
• Bezier curve node: Drag to create or change convex, concave, or S-curve shapes for the
associated envelope phase.
You can choose from several filter types and also have envelope control over filter cutoff.
Changes to filter settings result in a recalculation of the impulse response rather than a
direct change to the sound as it plays through Space Designer.
The main filter parameters are shown at the right side of the parameter bar when the filter
envelope is selected in the main display.
Click the Filter Env On/Off button to enable the filter envelope and the filter itself. You can
use the envelope to control the filter cutoff frequency over time. You can adjust all filter
envelope parameters, either numerically in the parameter bar or graphically in the main
display using the techniques discussed in Edit envelopes.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• 6 dB (LP): Bright, general-purpose lowpass filter mode that retains the top end of
most material while still providing some filtering.
• 12 dB (LP): Warm, lowpass filter mode without drastic filter effects that is useful for
smoothing bright reverbs.
• BP: 6 dB per octave bandpass design that reduces the low and high ends of the
signal, leaving the frequencies around the cutoff frequency intact.
• HP: 12 dB per octave/two-pole highpass design that reduces the level of frequencies
that fall below the cutoff frequency.
• Resonance field: Emphasize frequencies above, around, or below the cutoff frequency.
The impact of the resonance value on the sound is highly dependent on the chosen
filter mode, with steeper filter slopes resulting in more pronounced tonal changes.
• Attack node and field: Determine the time required to reach the Break Level. Drag the
node horizontally.
• Break Level node and field: Set the maximum filter cutoff frequency. This value also
defines the separation point between the attack and decay phases of the overall filter
envelope. Drag the node vertically. In other words, when the set level is reached after
the attack phase, the decay phase begins. You can create interesting filter sweeps by
setting the Break Level value lower than the Init Level parameter value.
• Decay node and field: Determine the time required after the Break Level point to reach
the End Freq value. Drag the node horizontally.
• End Freq (Frequency) node and field: Set the cutoff frequency at the end of the filter
envelope decay phase. Drag the node vertically.
• Bezier curve node: Drag to create or change convex, concave, or S-curve shapes for the
associated envelope phase.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Ramp Time node and field: Adjust the time between the Initial and End Density levels.
Drag the node horizontally.
• End Density node and field: Set the density of the reverb tail. An End Density value
that’s too low can result in a grainy sounding reverb tail. Drag the node vertically. The
stereo spectrum may also be affected by lower values.
• Reflection Shape field: Determine the steepness (shape) of early reflection clusters as
they bounce off the walls, ceiling, and furnishings of the virtual space.
Low values result in clusters with a sharp contour. High values result in an exponential
slope and a smoother sound. Reflection Shape is useful when re-creating rooms
constructed of different materials. When Reflection Shape is used with suitable
envelope, density, and early reflection settings, you can create rooms of almost any
shape and material.
To add the Space Designer effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Band 1 On/Off button: Turn on a highpass filter that allows high frequencies to pass and
reduces the level of low frequencies near the cutoff (set) frequency. When band 1 is
active, you can change band parameters directly in the graphic display.
• Band 1 background: Drag the red shaded area to change the frequency and gain
values.
• Band 1 node: Drag the red node to change the frequency and Q values.
• Band 2 On/Off button: Turn on a low shelving filter that adjusts the level of low
frequencies and has a minimal impact on frequencies above the cutoff (set) frequency.
When band 2 is active, you can change band parameters directly in the graphic display.
• Band 2 background or node: Drag the orange shaded area or node to change the
frequency and gain values.
• Band 3 On/Off button: Turn on a parametric bell filter with three controls. Frequency
sets a center frequency. Q sets the width of the frequency band around the center
frequency. Gain sets the level of the band. When band 3 is active, you can change band
parameters directly in the graphic display.
• Band 3 background or node: Drag the green shaded area or node to change the
frequency and gain values.
• Band 4 On/Off button: Turn on a parametric bell filter with three controls. Frequency
sets a center frequency. Q sets the width of the frequency band around the center
frequency. Gain sets the level of the band. When band 4 is active, you can change band
parameters directly in the graphic display.
• Band 4 background or node: Drag the blue shaded area or node to change the
frequency and gain values.
• Band 5 On/Off button: Turn on a high shelving filter that adjusts the level of high
frequencies and has a minimal impact on frequencies below the cutoff (set) frequency.
When band 5 is active, you can change band parameters directly in the graphic display.
• Band 5 background or node: Drag the purple shaded area or node to change the
frequency and gain values.
• Band 6 On/Off button: Turn on a lowpass filter that allows low frequencies to pass and
reduces the level of high frequencies near the cutoff (set) frequency. When band 6 is
active, you can change band parameters directly in the graphic display.
• Band 6 background: Drag the pink shaded area to change the frequency and gain
values.
• Band 6 node: Drag the pink node to change the frequency and Q values.
• Order pop-up menu: Choose the filter rolloff amount for the Low Cut and High Cut filter
bands. Higher order filters have a steeper rolloff.
• Q field: Set the Q factor—the width—for the selected band. Low values result in a
narrow frequency band selection. High values encompass a broad frequency band.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Click the EQ On/Off button to enable the EQ, then click one or more EQ band buttons
above the graphic display.
• Click a curve line segment, the (center frequency) node, or anywhere in the colored
space between the zero line and EQ curve to adjust the band.
• Click the node to select a band for editing. Once a band is selected, no other band
node that falls within the active (colored) area of the selected band can be selected.
• Click the graphic display background (outside a colored band) to deselect the
selected band.
• Press and hold the Command key while performing any of the following operations to
limit dragging to vertical or horizontal movement.
• Drag anywhere in the colored band to adjust gain and the center frequency.
Horizontally dragging the node in band 1 and band 6 adjusts both the frequency
and Q.
Global controls
• Quality pop-up menu: Choose the sample rate. Low produces a grainy reverb. High
matches the host application sample rate. Ultra is smooth and clean sounding.
• IR Offset field: Set the playback start point in the impulse response sample.
• Reverse button: Reverse the impulse response and envelopes. When the impulse
response is reversed, you are effectively using the tail rather than the front end of
the sample. You may need to adjust the Predelay and other parameter values when
reversing.
• Definition field: Set a crossover point (as a percentage of the overall length) to reduce
the synthesized impulse response resolution. This emulates reverb diffusion and saves
CPU resources.
• Reset Selected Envelope: Reset the currently displayed envelope to default values.
• Reset All Envelopes: Reset all envelopes to default values.
The reverb volume compensation feature attempts to match the perceived—not the
actual—volume differences between impulse response files. It should generally be
left on, although it may not work with all types of impulse responses. If you have
an impulse response that is of a different level, turn off volume compensation, then
adjust input and output levels accordingly.
• Show Bezier Handles: Enable or disable envelope curve handles (nodes) in the main
display. These enable you to precisely reshape envelopes. See Edit envelopes.
• Input slider: Determine how Space Designer processes the stereo input signal. See Use
global controls.
• Predelay sync button: This is used in Logic Pro to restrict Predelay knob values to
divisions synced with the project tempo and is disabled for use in Final Cut Pro.
• Predelay knob and field: Set the reverb predelay time, or time between the original
signal and the first reflections from the reverb. See Use global controls.
• Length knob and field: Adjust the length of the impulse response. This control works in
conjunction with the Size knob.
• Size knob and field: Adjust the sample rate of the loaded impulse response file, thereby
changing the perceived size of the reverb by widening or narrowing the room. Size can
also be used to preserve the original length of the impulse response when changing the
sample rate with the Quality pop-up menu.
The Size knob value has an impact on the decay because it is multiplied with the Length
knob value. For example, a Length knob value of 100% and a Size knob value of 100%
result in a decay that is the full length of the loaded impulse response.
• Refl (Reflection) Shape knob and field: Adjust to change the perceived shape of the
room in surround instances. This control alters the spacing of early reflections.
• Lo Spread and Hi Spread controls and fields: The Spread controls set the perceived
width of the stereo field.
Note: No Spread controls are shown in the Sampled IR mode of surround instances.
• LFE to Rev (Reverse) slider: Set the output level of the LFE channel independently of
other surround channels.
• C (Center) slider: Set the output level of the center channel independently of other
surround channels.
• Bal (Balance) slider: Set 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.
• Dry and Wet sliders: Set output levels for the dry (source) and wet (effect) signal.
The tasks below cover the use of Space Designer global controls.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• Stereo setting (top of slider): The signal is processed on both channels, retaining the
stereo balance of the original signal.
• XStereo setting (bottom of slider): The signal is inverted, with processing for the
right channel occurring on the left, and vice versa.
Natural reverbs contain most of their spatial information in the first few milliseconds.
Toward the end of the reverb, the pattern of reflections—signals bouncing off walls,
and so on—becomes more diffuse. In other words, the reflected signals become quieter
and increasingly nondirectional, containing far less spatial information. To emulate this
phenomenon, use the full impulse response resolution only at the onset of the reverb, then
use a reduced impulse response resolution toward the end of the reverb.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Vertically drag the Definition field at the top of the global controls section to set the
crossover point—where the switch to the reduced impulse response resolution occurs.
The Definition field is shown as a percentage, where 100% is equal to the length of the
full resolution impulse response.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
The ideal predelay setting for different sounds depends on the properties of—or more
accurately, the envelope of—the original signal. Percussive signals generally require shorter
predelays than signals where the attack fades in gradually, such as strings. A good rule of
thumb is to use the longest predelay possible before unwanted side effects, such as an
audible echo, begin to materialize.
In practice, an extremely short predelay tends to make it difficult to pinpoint the position
of the signal source. It can also color the sound of the original signal. On the other hand,
an excessively long predelay can be perceived as an unnatural echo. It can also divorce the
original signal from its early reflections, leaving an audible gap between the original and
reverb signals.
These guidelines are intended to help you design realistic-sounding spaces that are
suitable for various signals. If you want to create unnatural sound stages or otherworldly
reverbs and echoes, experiment with the Predelay parameter.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Vertically drag the IR Offset field at the top of the global controls section to shift the
playback start point of the impulse response.
This effectively cuts off the beginning of the impulse response, which can be useful for
eliminating level peaks at the start of the sample.
The tasks below cover the use of Space Designer output controls.
Space Designer provides two output sliders—the Dry slider for the direct signal, and the
Wet slider for the reverb signal—when you insert it as a mono, mono to stereo, or stereo
effect.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• Set the level of the Dry slider: Set the level of the non-effect, or dry, signal. Move
the slider to a value of 0 (mute) if Space Designer is inserted in a bus channel or
when you’re using modeling impulse responses, such as speaker simulations.
• Set the level of the Wet slider: Adjust the level of the effect signal.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• Set the level of the LFE to Rev (Reverse) slider: Adjust the output level of the LFE
channel independently of other surround channels.
• Set the level of the C (Center) slider: Adjust the output level of the center channel
independently of other surround channels.
• Set the level of the Bal (Balance) slider: Set 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.
• Set the level of the Dry slider: Set the overall level of the non-effect signal for all
channels. Drag the slider to a value of 0 (mute) when using Space Designer as a bus
effect in an aux channel strip. Use the Send knob of each bussed channel strip to
control the wet/dry balance.
• Set the level of the Wet slider: Adjust the output level of the effect, or wet, signal for
all channels.
1. In the Final Cut Pro timeline, select a clip with the Space Designer effect applied, then
open the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
• Set the level of the Spread knobs and fields: Rotate to extend the stereo base to
frequencies below or above the frequency determined by the X-Over parameter.
• At a Spread value of 1.00, divergence between the left and right channels is at its
maximum.
• Set the X-Over slider and field value: Set the crossover frequency in hertz. Any
impulse response frequency below or above the value you set is affected by the Lo
Spread and Hi Spread parameters (at values over zero).
To add the Correlation Meter effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• A correlation of +1 (the far-right position) means that the left and right channels
correlate 100%—they are completely in phase.
• Correlation values lower than 0 indicate that out-of-phase material is present, which
can lead to phase cancelations if the stereo signal is combined into a monaural signal.
Note: You can also set a size by dragging the lower corners of the effect window.
Direction Mixer works with any type of stereo recording, regardless of the miking technique
used. For information about the most common stereo miking techniques—AB, XY, and MS—
see Stereo miking techniques.
• Input buttons: Set the input signal type. Use LR if the input signal is a standard left/right
signal. Use MS if the signal is middle and side encoded.
• Direction knob and field: Set the pan position for the middle—the center of the stereo
base—of the recorded stereo signal. When Direction is set to a value of 0, the midpoint
of the stereo base in a stereo recording is perfectly centered within the mix.
• Higher values move the center of the stereo base back toward the center of the
stereo mix, but this also has the effect of swapping the stereo sides of the recording.
For example, at a value of either 180° or −180°, the center of the stereo base is dead
center in the mix, but the left and right sides of the recording are swapped.
• Higher values move the middle signal back toward the center of the stereo mix, but
this also has the effect of swapping the side signals of the recording. For example,
at a value of either 180° or −180°, the middle signal is dead center in the mix, but the
left and right sides of the side signal are swapped.
• At a neutral value of 1, the left side of the signal is positioned precisely to the left
and the right side precisely to the right. As you decrease the Spread value, the two
sides move toward the center of the stereo image.
• A value of 0 produces a summed mono signal—both sides of the input signal are
routed to the two outputs at the same level. At values greater than 1, the stereo base
is extended out to an imaginary point beyond the spatial limits of the speakers.
• Values of 1 or higher increase the level of the side signal, making it louder than the
middle signal.
• Split button: Split the signal into independently controlled high and low ranges.
• Crossover field: Set the frequency where the signal is split between high and low
ranges. Drag vertically, or double-click and type a value.
• Direction High and Direction Low knobs and fields: Independently set the central pan
position for the recorded stereo signal in the upper or lower frequency range (set with
Crossover).
• Spread High and Spread Low sliders and fields: Independently set the stereo spread in
LR signals, or set the side signal level in MS signals for the upper/lower frequency range
(set with Crossover).
AB and XY recordings both record left and right channel signals, but the middle signal is
the result of combining both channels.
MS recordings record a middle signal, but the left and right channels are decoded from the
side signal, which is the sum of both left and right channel signals.
The AB technique is commonly used for recording one section of an orchestra, such as the
string section, or perhaps a small group of vocalists. It is also useful for recording piano or
acoustic guitar.
AB is not well suited to recording a full orchestra or group as it tends to smear the stereo
imaging/positioning of off-center instruments. It is also unsuitable for mixing down to mono
because phase cancelations can occur between channels.
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).
XY recordings tend to be balanced in both channels, with good positional information being
encoded. XY recording is commonly used for drum recording and is also suitable for larger
ensembles and many individual instruments.
Typically, XY recordings have a narrower sound field than AB recordings, so they can lack a
sense of perceived width when played back. XY recordings can be mixed down to mono.
MS miking
To make a Middle and Side (MS) recording, two microphones are positioned as closely
together as possible—usually placed on a stand or hung from the studio ceiling. One is a
cardioid (or omnidirectional) microphone that directly faces the sound source you want
to record—in a straight alignment. The other is a bidirectional microphone, with its axes
pointing to the left and right of the sound source at 90° angles. The cardioid microphone
records the middle signal to one side of a stereo recording. The bidirectional microphone
records the side signal to the other side of a stereo recording. MS recordings made in this
way can be decoded by Direction Mixer.
When MS recordings are played back, the side signal is used twice:
• As recorded
MS is ideal for all situations where you need to retain absolute mono compatibility. The
advantage of MS recordings over XY recordings is that the stereo middle is positioned
on the main recording direction (on-axis) of the cardioid microphone. This means that
slight fluctuations in frequency response that occur off the on-axis—as is the case with
every microphone—are less troublesome, because the recording always retains mono
compatibility.
Unlike this process, however, the Exciter distortion process involves passing the input
signal through a highpass filter before feeding it into the harmonics (distortion) generator.
Artificial harmonics are thus added to the original signal, and these added harmonics
contain frequencies at least one octave above the threshold of the highpass filter. The
distorted signal is then mixed with the original, dry signal.
You can use Exciter to add life to recordings, particularly audio tracks with a weak treble
frequency range. You can also use Exciter to enhance guitar tracks.
To add the Exciter effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Frequency field: Drag vertically to set the cutoff frequency of the highpass filter. The
input signal passes through the filter before harmonic distortion is introduced.
• Frequency display: Shows the frequency range used as the source signal for the excite
process. You can drag the green line or handle to set the cutoff frequency.
• Dry Signal button: Turn on to mix the original (pre-effect) signal with the effect signal.
Turn off to hear only the effect signal.
• Harmonics knob and field: Set the ratio between the effect and original signals. If the
Dry Signal button is turned off, this parameter has no effect.
Note: In most cases, it is preferable to select higher Frequency and Harmonics values,
because human ears cannot easily distinguish between the artificial and original high
frequencies.
• Color 1 and Color 2 buttons: Turn on Color 1 to generate a less dense harmonic
distortion spectrum. Color 2 generates more intense harmonic distortion.
• An Analyzer to view the level of each 1/3-octave or major second frequency band
• An integrated Level meter to view the signal level for each channel
You can view either the Analyzer or Goniometer results in the main display area. Use the
Analyzer or Goniometer buttons to switch between modes and to set parameters. The
Loudness/Level and Correlation meters are always visible in the Full display, as are several
common parameters.
You can set a size by dragging the lower corners of the effect window. The View pop-
up menu provides further Display, Meters, or Full items that show a partial view or the
complete effect interface. Control-click in the main display when a partial view is enabled
to switch between modes.
Although you can insert MultiMeter directly into any clip, it is more commonly used when
you are working on the overall mix.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Top and Range fields: Change Analyzer display values by setting the maximum level
(Top) and the overall dynamic range (Range).
• Scale: Indicates the scale of levels. Drag the scale vertically to adjust the Range value.
Changing the scale is useful when analyzing highly compressed material because it
makes it easier to identify small level differences.
• Detection buttons: Determine the channels shown in the Analyzer results in the main
display.
• Mono: Display the spectrum of the mono sum of both (stereo) inputs.
• Mode buttons: Determine how levels are displayed. You can select RMS Slow, RMS Fast,
or Peak.
• The two RMS modes show the effective signal average and provide a representative
overview of perceived volume levels.
• Analyzer Bands pop-up menu: Choose the number of bands shown in the Analyzer
display. You can choose 31 Bands (Third-Octave) or 63 Bands (Major Second).
The idea of the goniometer was born with the advent of early two-channel oscilloscopes.
To use such devices as goniometers, users would connect the left and the right stereo
channels to the X and Y inputs, while rotating the display by 45° to produce a useful
visualization of the signal stereo phase.
The signal trace slowly fades to black, imitating the retro glow of the tubes found in older
goniometers, while also enhancing the readability of the display.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Auto Gain knob and field: Set the amount of display compensation for low input levels.
You can set Auto Gain levels in 10% increments or turn Auto Gain off.
Note: To avoid confusion with the Auto Gain parameter found in other included effects
and processors (such as the compressors), Auto Gain is used only as a display
parameter in the meters. It increases display levels to enhance readability. It does not
change the actual audio levels.
• Decay knob and field: Determine the time it takes for the Goniometer trace to fade to
black.
In the Level meter, the signal level for each channel is represented by a blue bar. Signals
(above the target level) approaching the 0 dB level are represented by a yellow bar. When
the level exceeds 0 dB, the portion of the bar above the 0 dB point turns red.
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.
• Peak and RMS fields: Peak and RMS values are displayed numerically (in dB increments)
above the Level meter. Click the display to reset values.
The Loudness meter shows the momentary loudness level. Loudness indicates the
perceived level of a signal that is indicative of human hearing, making it a useful reference
tool when mixing or mastering. The Loudness meter conforms to the AES 128 specification.
• LU-I field: Loudness Unit-Integrated, which indicates the perceived level from the start
to the end of the program material.
• LU-S field: Loudness Unit-Short term, which indicates the perceived level of the most
recent 3 seconds of program material.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Level pop-up menu: Choose how levels are displayed. The options are Peak, RMS, Peak
& RMS, True Peak, and True Peak & RMS.
• The two RMS options show the effective signal average and provide a representative
overview of perceived volume levels.
• Return Rate pop-up menu: Choose how quickly analyzed signals return from peak
(maximum) levels to zero or incoming signal levels. This is expressed in dB per second.
• Peak pop-up menu: Choose the hold time for the Level meter. Choose 2, 4, or 6
seconds—or an infinite hold time.
Note: The Hold button must be turned on for the selected time value to have an effect.
• Hold button: Turn on to show a small indicator for the most recent peak level. This is
displayed as follows:
• Level meter: A small yellow segment above each stereo level bar indicates the most
recent peak level.
• Correlation meter: The horizontal area around the correlation indicator denotes
phase correlation deviations in real time, in both directions. A vertical line to the
left of the correlation indicator shows the maximum negative phase deviation value.
You can reset this line by clicking it during playback. For more information, see
Correlation meter.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• A +1 correlation value indicates that the left and right channels correlate 100%. In other
words, the left and right signals are in phase and are the same shape.
• Correlation values to the right of the center position indicate that the stereo signal is
mono compatible.
• The middle position indicates the highest allowable amount of left/right divergence,
which is often audible as an extremely wide stereo effect.
• When the Correlation meter moves to the left of the center position, out-of-phase
material is present. This leads to phase cancelations if the stereo signal is combined
into a mono signal.
To add the MultiMeter effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Peak pop-up menu: Choose the hold time for all metering tools. Choose 2, 4, or 6
seconds—or an infinite hold time.
Note: The Hold button must be turned on for the selected time value to have an effect.
• Hold button: Turn on peak hold for all metering tools in MultiMeter. This is displayed in
the following ways:
• Analyzer: A small segment above each 1/3-octave level bar indicates the most recent
peak level.
• Reset button: Reset the peak hold segments of all metering tools.
• Return Rate pop-up menu: Shown only in mono instances. Choose how quickly analyzed
signals return from peak/maximum levels to zero or incoming signal levels. This is
expressed in dB per second.
Note: This parameter is shown below the MultiMeter Level and Loudness meters when a
stereo instance is active.
To add the Stereo Spread effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Lower Int. (Intensity) slider and field: Set the amount of stereo base extension for the
lower frequency bands.
• Upper Int. (Intensity) slider and field: Set the amount of stereo base extension for the
upper frequency bands.
Note: When setting the Lower Int. and Upper Int. sliders, be aware that the stereo effect
is most apparent in the middle and higher frequencies, so distributing low frequencies
between the left and right speakers can significantly alter the energy of the overall mix.
For this reason, use low values for the Lower Int. parameter, and avoid setting the Lower
Freq. parameter below 300 Hz.
• Graphic display: Shows the number of bands the signal is divided into, and the intensity
of the Stereo Spread effect in the upper and lower frequency bands. The upper section
represents the left channel, and the lower section represents the right channel. The
frequency scale displays frequencies in ascending order, from left to right.
• Upper Freq.(Frequency) and Lower Freq. slider and fields: Determine the highest and
lowest frequencies that will be redistributed in the stereo image.
• Order knob and field: Determine the number of frequency bands that the signal is
divided into. A value of 8 is usually sufficient for most tasks, but you can use up to
12 bands.
The simplest use for SubBass is as an octave divider, similar to octaver effect pedals for
electric bass guitars. Whereas such pedals can only process a monophonic input sound
source of clearly defined pitch, SubBass can be used with complex summed signals
as well.
SubBass creates two bass signals, derived from two separate portions of the incoming
signal. These are defined with the High and Low parameters. See SubBass controls.
WARNING: Using SubBass can produce extremely loud output signals. Choose moderate
monitoring levels, and only use loudspeakers that are actually capable of reproducing the
very low frequencies produced. Never try to force a loudspeaker to output these frequency
bands with an EQ.
SubBass is unlike a pitch shifter in that the waveform of the signal generated by SubBass
is not based on the waveform of the input signal, but is sinusoidal—that is, it uses a sine
wave. Given that pure sine waves rarely sit well in complex arrangements, you can control
the amount of—and balance between—the generated and original signals with the Wet and
Dry sliders.
Use the High and Low controls to define the two frequency bands, which SubBass uses to
generate tones. High Center and Low Center define the center frequency of each band,
and High Bandwidth and Low Bandwidth define the width of each frequency band.
The High Ratio and Low Ratio knobs define the transposition amount for the generated
signal in each band. This is expressed as a ratio of the original signal. For example, a Ratio
setting of 2 transposes the signal down one octave.
Important: Within each frequency band, the filtered signal should have a reasonably stable
pitch in order to be analyzed correctly.
In general, narrow bandwidths produce the best results, because they avoid unwanted
intermodulations. Set High Center a fifth higher than Low Center, which means a factor of
1.5 for the center frequency. Derive the sub-bass to be synthesized from the existing bass
portion of the signal, and transpose by one octave in both bands (by setting Ratio to 2).
Do not overdrive the process or you will introduce distortion. If you hear frequency gaps,
move one or both Center frequency knobs, or widen the Bandwidth setting of one or both
frequency ranges a little.
Tip: Be prudent when using SubBass, and compare the extreme low-frequency content
of your mixes with that of other productions. It is very easy to go overboard with it.
To add the SubBass effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• High Ratio knob and field: Adjust the ratio between the generated signal and the original
upper band signal.
• High Center knob and field: Set the center frequency of the upper band.
• High Bandwidth knob and field: Set the width of the upper band.
• Graphic display: Shows the selected upper and lower frequency bands.
• Freq. (Frequency) Mix slider and field: Adjust the mix ratio between the upper and lower
frequency bands.
• Low Ratio knob and field: Adjust the ratio between the generated signal and the original
lower band signal.
• Low Center knob and field: Set the center frequency of the lower band.
• Low Bandwidth knob and field: Set the width of the lower band.
• Dry slider and field: Set the amount of dry (non-effect, original) signal.
• Wet slider and field: Set the amount of wet (effect) signal.
• In Test Tone mode, a test signal is generated immediately when the plug-in is inserted.
You can switch off the test tone by bypassing the plug-in.
• In Sine Sweep mode, a user-defined frequency spectrum tone sweep is generated when
you click the Trigger button.
To add the Test Oscillator effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Anti Aliased button: Enable to use anti-aliased versions of the Square Wave or Needle
Pulse waveforms.
• Frequency knob and field: Set the frequency of the oscillator (the default is 1 kHz).
You can also double-click this field and enter a value ranging from 1 Hz to 22 kHz,
exceeding the possible values that can be set with the knob. If you enter “1,” a 1 Hz test
tone is the result.
• Level knob and field: Set the overall output level. This parameter is common to both
modes.
• Dim button: Reduce the output level by 50%. This parameter is common to both modes.
• Start Frequency and End Frequency knobs and fields: Set the oscillator frequency for
the beginning and end of the sine sweep.
Note: The Frequency field shown below the Start Frequency and End Frequency
controls is a real-time display of the frequency sweep.
• Trigger button: Start the sine sweep of the spectrum set with the Start Frequency and
End Frequency controls.
You can use these effects or replace them with other effects available in Final Cut Pro.
The advantage of using DeEsser rather than an EQ to cut high frequencies is that it
compresses the signal dynamically, rather than statically. This prevents the sound from
becoming darker when no sibilance is present in the signal. DeEsser has extremely fast
attack and release times.
When using DeEsser, you can set the frequency range being compressed (the Suppressor
frequency) independently of the frequency range being analyzed (the Detector frequency).
The two ranges can be easily compared in DeEsser’s Detector and Suppressor frequency
range displays.
The Suppressor frequency range is reduced in level for as long as the Detector frequency
threshold is exceeded.
DeEsser does not use a frequency-dividing network—a crossover utilizing lowpass and
highpass filters. Rather, it isolates and subtracts the frequency band, resulting in no
alteration of the phase curve.
The Detector controls are on the left side of the DeEsser window, and the Suppressor
controls are on the right. The center section includes the Detector and Suppressor displays
and the Smoothing slider.
To add the DeEsser effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Detector Sensitivity knob and field: Set the degree of responsiveness to the input
signal.
• Monitor pop-up menu: Choose Detector to monitor the isolated Detector signal,
Suppressor to monitor the filtered Suppressor signal, Sensitivity to remove the sound
from the input signal in response to the Sensitivity parameter, or Off to hear the
DeEsser output.
Suppressor controls
• Suppressor Frequency knob and field: Set the frequency band that is reduced when the
Detector sensitivity threshold is exceeded.
• Strength knob and field: Set the amount of gain reduction for signals that surround the
Suppressor frequency.
• Smoothing slider: Set the reaction speed of the gain reduction start and end
phases. Smoothing controls both the attack and release times, as they are used by
compressors.
If you use Denoiser too aggressively, however, the algorithm produces artifacts, which are
usually less desirable than the existing noise. If using Denoiser produces these artifacts,
you can use the three Smoothing knobs to reduce or eliminate them. See Denoiser
controls.
1. In the Final Cut Pro timeline, select a clip with the Denoiser effect applied, then open
the effect’s settings in the Audio inspector.
To add the effect and show its controls, see Add Logic effects to clips.
2. Locate a section of the audio where only noise is audible, and set the Threshold value
so that only signals at or below this level are filtered out.
3. 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.
To add the Denoiser effect to a clip and show the effect’s controls, see Add Logic effects
to clips.
• Threshold slider and field: Set the threshold level. Signals that fall below this level are
reduced by Denoiser.
• Reduce slider and field: Set the amount of noise reduction applied to signals that fall
below the threshold. When reducing noise, remember that each 6 dB reduction is
equivalent to halving the volume level (and each 6 dB increase is equivalent to doubling
the volume level).
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: Determine the type of noise that you want to reduce.
• Positive values change the noise type to pink noise (harmonic noise; greater bass
response).
• Negative values change the noise type to blue noise (hissy tape noise).
• Graphic display: Shows how the lowest volume levels of your audio material—which
should be mostly, or entirely, noise—are reduced. Changes to parameters are instantly
reflected here.
Smoothing controls
• Frequency knob and field: Adjust how smoothing is applied to neighboring frequencies.
If Denoiser recognizes that only noise is present on a certain frequency band, the higher
you set the Frequency parameter, the more it changes the neighboring frequency bands
to avoid glass noise.
• Time knob and field: Set the time required by Denoiser to reach (or release) maximum
reduction. This is the simplest form of smoothing.
• Transition knob and field: Adjust how smoothing is applied to neighboring volume levels.
If Denoiser recognizes that only noise is present in a certain volume range, the higher
you set the Transition parameter, the more similar-level values are changed, in order to
avoid glass noise.
To add the Fat EQ effect to a clip and show the effect’s controls, see Add Logic effects to
clips.
• Band Type buttons: Located above the graphic display. For bands 1–2 and 4–5, click
one of the paired buttons to select the EQ type for the corresponding band.
• Master Gain slider and field: Set the overall output level of the signal. Use it after
boosting or cutting individual frequency bands.
To add the PlatinumVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Early Reflections controls: Emulate the original signal’s first reflections as they bounce
off the walls, ceiling, and floor of a natural room.
• Output controls: Set the balance between the effect (wet) and direct (dry) signals.
• Balance ER/Reverb slider: Control the balance between the early reflections and reverb
signal. When you set the slider to either extreme position, the other signal is not heard.
To add the PlatinumVerb effect to a clip and show the effect’s controls, see Add Logic
effects to clips.
• Predelay slider and field: Determine 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. Overly
long Predelay settings can be perceived as an unnatural echo and can divorce the
original signal from its early reflections, leaving an audible gap between them.
The optimum Predelay setting depends on the type of input signal—or more precisely,
the envelope of the input signal. Percussive signals generally require shorter predelays
than signals where the attack fades in gradually. A good working method is to use the
longest possible Predelay value before you start to hear unwanted side effects, such as
an audible echo. When you reach this point, reduce the Predelay setting slightly.
• Room Shape slider and field: Define the geometric form of the room. The numeric value
(3 to 7) represents the number of corners in the room. The graphic display visually
represents this setting.
• Room Size slider and field: Determine the dimensions of the room. The numeric value
indicates the length of the room’s walls—the distance between two corners.
• Stereo Base slider and field: Define the distance between the two virtual microphones
that are used to capture the signal in the simulated room.
Note: Spacing the microphones slightly farther apart than the distance between two
human ears generally delivers the best, and most realistic, results. This parameter is
available only in stereo instances of the effect.
• ER Scale slider and field (Extended controls area): Scale the early reflections along
the time axis, influencing the Room Shape, Room Size, and Stereo Base parameters
simultaneously.
Reverb controls
• Initial Delay slider and field: Set the time between the original signal and the diffuse
reverb tail.
• Spread slider and field: Control the stereo image of the reverb. At 0%, the effect
generates a monaural reverb. At 200%, the stereo base is artificially expanded.
• Low Ratio slider and field: Determine the relative reverb times of the bass and high
bands. The value is expressed as a percentage. At 100%, the reverb time of the two
bands is identical. At values below 100%, the reverb time of frequencies below the
crossover frequency is shorter. At values greater than 100%, the reverb time for low
frequencies is longer.
• Low Freq (Frequency) Level slider and field: Set the level of the low-frequency reverb
signal. At 0 dB, the volume of the two bands is equal. In most mixes, you should set a
lower level for the low-frequency reverb signal. This enables you to boost the bass level
of the incoming signal, making it sound punchier. This also helps to counteract bottom-
end masking effects.
• High Cut slider and field: Frequencies above the set value are filtered from the reverb
signal. Uneven or absorbent surfaces—wallpaper, wood paneling, carpets, and so on—
tend to reflect lower frequencies better than higher frequencies. The High Cut filter
replicates this effect. If you set the High Cut filter so that it is wide open (maximum
value), the reverb will sound as if it is reflecting off stone or glass.
• Density slider and field: Control the density of the diffuse reverb tail. Ordinarily you
want the signal to be as dense as possible. In rare instances, however, a high Density
value can color the sound, which you can fix by reducing the Density value. Conversely,
if you select a Density value that is too low, the reverb tail will sound grainy.
• Diffusion slider and field: Set 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: Determine the reverb time of the high band. Most
natural rooms have a reverb time somewhere in the range of 1 to 3 seconds. This time
is reduced by absorbent surfaces, such as carpet and curtains, and soft or dense
furnishings, such as sofas, armchairs, cupboards, and tables. Large empty halls or
churches have reverb times of up to 8 seconds, with some cavernous or cathedral-like
venues extending beyond that.
Output controls
• Dry slider and field: Control the amount of the original signal.
• Wet slider and field: Control the amount of the effect signal.
Use of the “keyboard” Apple logo (Option-Shift-K) for commercial purposes without the prior written consent
of Apple may constitute trademark infringement and unfair competition in violation of federal and state laws.
Apple, the Apple logo, Apple Books, ChromaVerb, Final Cut, Final Cut Pro, Finder, Logic, Logic Pro, Mac,
Magic Mouse, Metal, and SilverVerb are trademarks of Apple Inc., registered in the U.S. and other countries
and regions.
Apple
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Cupertino, CA 95014
apple.com
Other company and product names mentioned herein are trademarks of their respective companies.
Every effort has been made to ensure that the information in this manual is accurate. Apple is not responsible for
printing or clerical errors.
Some apps are not available in all areas. App availability is subject to change.
028-00588