Final Cut Pro X Logic Effects Reference
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Contents Preface 5 An Introduction to Logic Effects for Final Cut Pro X 5 About the Logic Effects included with Final Cut Pro X 7 Additional Resources Chapter 1 9 10 11 12 13 13 14 15 Distortion Effects Bitcrusher Clip Distortion Distortion Effect Distortion II Overdrive Phase Distortion Ringshifter Chapter 2 23 23 42 44 45 Echo Effects Delay Designer Modulation Delay Stereo Delay Tape Delay Chapter 3 47 47 53 56 57 Equalizers AutoFilter Channel EQ Fat EQ Linear Phase EQ Chapter 4 63 63 65 66 7
75 78 81 83 Multipressor Noise Gate Spectral Gate Surround Compressor Chapter 5 89 90 90 92 93 94 96 Modulation Effects Chorus Effect Ensemble Effect Flanger Effect Phaser Effect Scanner Vibrato Effect Tremolo Effect Chapter 6 97 Spaces Effects 98 Plates, Digital Reverb Effects, and Convolution Reverb 99 PlatinumVerb Chapter 7 103 104 105 109 115 117 123 Space Designer Convolution Reverb Getting to Know the Space Designer Interface Working with Space Designer’s Impulse Response Parameters Workin
Preface An Introduction to Logic Effects for Final Cut Pro X Final Cut Pro X comes bundled with an extensive range of Logic Effects, digital signal processing (DSP) effects and processors that are used to color or tonally shape existing audio recordings and audio sources—in real time. These will cover almost every audio processing and manipulation need you will encounter in your day-to-day work.
Effect category Included effects Overdrive Phase Distortion Ringshifter Echo Delay Designer Modulation Delay Stereo Delay Tape Delay EQ AutoFilter Channel EQ Fat EQ Linear Phase EQ Levels Adaptive Limiter Compressor Enveloper Expander Gain Plug-in Limiter Multichannel Gain Multipressor Noise Gate Spectral Gate Surround Compressor Modulation Chorus Effect Ensemble Effect Flanger Effect Phaser Effect Scanner Vibrato Effect Tremolo Effect Spaces PlatinumVerb Space Designer Convolution Reverb Specia
Effect category Included effects MultiMeter Stereo Spread SubBass Test Oscillator Vocal DeEsser Pitch Correction Effect Pitch Shifter II Vocal Transformer Additional Resources In addition to the documentation that comes with Final Cut Pro, there are a variety of other resources you can use to find out more. Final Cut Pro Website For general information and updates, as well as the latest news on Final Cut Pro, go to: • http://www.apple.
Distortion Effects 1 You can use Distortion effects to recreate the sound of analog or digital distortion and to radically transform your audio. Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology, and they are still used in musical instrument amplifiers today.
Bitcrusher Bitcrusher is a low-resolution digital distortion effect. You can use it to emulate the sound of early digital audio devices, to create artificial aliasing by dividing the sample rate, or to distort signals until they are unrecognizable. • Drive slider and field: Sets the amount of gain in decibels applied to the input signal. Note: Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well.
• Displaced: The start, center, and end levels of the signal (above the threshold) are offset, resulting in a distortion which is less severe as signal levels cross the threshold. The center portion of the clipped signal is also softer than in Cut mode. • Clip Level slider and field: Sets the point (below the clipping threshold of the clip) at which the signal starts clipping. • Mix slider and field (Extended Parameters area): Sets the balance between dry (original) and wet (effect) signals.
• Mix slider and field: Sets the ratio between the effect (wet) signal and original (dry) signals, following the Clip Filter. • Sum LPF knob and field: Sets the cutoff frequency (in Hertz) of the lowpass filter. This processes the mixed signal. • (High Shelving) Frequency knob and field: Sets the frequency (in Hertz) of the high shelving filter. If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a stereo hi-fi amplifier.
Distortion II Distortion II emulates the distortion circuit of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect, or use it creatively when designing new sounds. • PreGain knob: Sets the amount of gain applied to the input signal. • Drive knob: Sets the amount of saturation applied to the signal. • Tone knob: Sets the frequency of the highpass filter. Filtering the harmonically rich distorted signal produces a softer tone.
• Display: Shows the impact of parameters on the signal. • Tone knob and field: Sets the frequency for the high cut filter. Filtering the harmonically rich distorted signal produces a softer tone. • Output slider and field: Sets the output level. This allows you to compensate for increases in loudness caused by using Overdrive. Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (see Modulation Effects).
• Phase Reverse checkbox (Extended Parameters area): Enable to reduce the delay time on the right channel when input signals that exceed the cutoff frequency are received. Available only for stereo instances of the Phase Distortion effect. Ringshifter The Ringshifter effect combines a ring modulator with a frequency shifter effect. Both effects were popular during the 1970s, and are currently experiencing something of a renaissance.
Getting to Know the Ringshifter Interface The Ringshifter interface consists of six main sections. Mode buttons Oscillator parameters Delay parameters Envelope follower parameters Output parameters LFO parameters • Mode buttons: Determine whether the Ringshifter operates as frequency shifter or ring modulator. See Setting the Ringshifter Mode.
Setting the Ringshifter Mode The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator. • Single (Frequency Shifter) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted up (positive value) or down (negative value).
• In the ring modulator OSC mode, the Frequency parameter controls the frequency content (timbre) of the resulting effect. This timbre can range from subtle tremolo effects to clangorous metallic sounds. • Frequency control: Sets the frequency of the sine oscillator.
• Sync button: Synchronizes the delay to the project tempo. You can choose musical note values with the Time knob. • Level knob and field: Sets the level of the delay added to the ring-modulated or frequency-shifted signal. A Level value of 0 passes the effect signal directly to the output (bypass).
Modulating the Ringshifter with the LFO The oscillator Frequency and Dry/Wet parameters can be modulated with the LFO—and the envelope follower (see Modulating the Ringshifter with the Envelope Follower). The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. The LFO produces continuous, cycled control signals. • Power button: Turns the LFO on or off and enables the following parameters.
• Feedback knob and field: Sets the amount of the signal that is routed back to the effect input. Feedback adds an edge to the Ringshifter sound and is useful for a variety of special effects. It produces a rich phasing sound when used in combination with a slow oscillator sweep. Comb filtering effects are created by using high Feedback settings with a short delay time (less than 10 ms).
Echo Effects 2 Echo effects store the input signal—and hold it for a short time—before sending it to the effect input or output. The held, and delayed, signal is repeated after a given time period, creating a repeating echo effect, 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.
• Highpass and lowpass filtering • Pitch transposition (up or down) Further effect-wide parameters include synchronization, quantization, and feedback. As the name implies, Delay Designer offers significant sound design potential. You can use it for everything from a basic echo effect to an audio pattern sequencer. You can create complex, evolving, moving rhythms by synchronizing the placement of taps.
• Sync section: This is used for syncing tempo in Logic Pro and is disabled for use with Final Cut Pro. • Master section: This area contains the global Mix and Feedback parameters. See Using Delay Designer’s Master Section. Getting to Know Delay Designer’s Main Display Delay Designer’s main display is used to view and edit tap parameters. You can freely determine the parameter shown, and quickly zoom or navigate through all taps.
• 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. For more details, see Editing Parameters in Delay Designer’s 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.
Zooming and Navigating Delay Designer’s Tap Display You can use Delay Designer’s Overview display to zoom and to navigate the Tap display area. Overview display Tip: If the Overview display is hidden behind a tap, you can move it to the foreground by holding down Shift. To zoom the Tap display Do one of the following: µ Vertically drag the highlighted section (the bright rectangle) of the Overview display.
µ To move to different sections of the Tap display Horizontally drag the (middle of the) bright rectangle in the Overview display. The zoomed view in the Tap display updates as you drag. Creating Taps in Delay Designer You can create new delay taps in three different ways: by using the Tap pads, by creating them in the Identification bar, or by copying existing taps. To create taps with the Tap pad 1 Click the upper pad (Start).
µ To create taps in the Identification bar Click at the appropriate position. µ To copy taps in the Identification bar Option-drag a selection of one or more taps to the appropriate position. The delay time of copied taps is set to the drag position. Delay Designer Tap Creation Suggestions The fastest way to create multiple taps is to use the Tap pads.
The Identification bar shows the letter of each visible tap. The Tap Delay field of the Tap parameter bar displays the letter of the currently selected tap, or the letter of the tap being edited when multiple taps are selected (for details, see Selecting Taps in Delay Designer). Selecting Taps in Delay Designer There will always be at least one selected tap. You can easily distinguish selected taps by color—the toggle bar icons and the Identification bar letters of selected taps are white.
µ Open the pop-up menu to the right of the Tap name, and choose the appropriate tap letter. To select multiple taps Do one of the following: µ µ Drag across the background of the Tap display to select multiple taps. Shift-click specific taps in the Tap display to select multiple nonadjacent taps. Moving and Deleting Taps in Delay Designer You can move a tap backward or forward in time, or completely remove it. Note: When you move a tap, you are actually editing its delay time.
µ Select a tap letter in the Identification bar and drag it downward, out of the Tap display. This method also works when more than one tap is selected. µ To delete all selected taps Control-click (or right-click) a tap, and choose “Delete tap(s)” from the shortcut menu. Using Delay Designer’s Tap Toggle Buttons Each tap has its own toggle button in the Toggle bar. These buttons offer you a quick way to graphically activate and deactivate parameters.
Note: The first time you edit a filter or pitch transpose parameter, the respective module automatically turns on. This saves you the effort of manually turning on the filter or pitch transposition module before editing. After you manually turn either of these modules off, however, you need to manually switch it back on. Editing Parameters in Delay Designer’s Tap Display You can graphically edit any tap parameter that is represented as a vertical line in Delay Designer’s Tap display.
Parameter values change to match the mouse position as you drag across the taps. Command-dragging across several taps allows you to draw value curves, much like using a pencil to create a curved line on a piece of paper. Aligning Delay Designer Tap Values You can use Delay Designer’s Tap display to graphically align tap parameter values that are represented as vertical lines. To align the values of several taps 1 Command-click in the Tap display, and move the pointer while holding down the Command key.
The values of taps that fall between the start and end points are aligned along the line. Editing Filter Cutoff in Delay Designer’s Tap Display Whereas the steps outlined in Editing Parameters in Delay Designer’s Tap Display apply to most graphically editable parameters, the Cutoff and Pan parameters work in a slightly different fashion. In Cutoff view, each tap actually shows two parameters: highpass and lowpass filter cutoff frequency.
If the highpass filter’s cutoff frequency value is above that of the lowpass filter cutoff frequency, the filter switches from serial operation to parallel operation, meaning that the tap passes through both filters simultaneously. In this case, the space between the two cutoff frequencies represents the frequency band being rejected—in other words, the filters act as a band-rejection filter.
Lines above the center position indicate pans to the left, and lines below the center position denote pans to the right. Left (blue) and right (green) channels are easily identified. In stereo input/stereo output configurations, the Pan parameter adjusts the stereo balance, not the position of the tap in the stereo field. The Pan parameter appears as a dot on the tap, which represents stereo balance. Drag the dot up or down the tap to adjust the stereo balance. By default, stereo spread is set to 100%.
Editing Taps in Delay Designer’s Tap Parameter Bar The Tap parameter bar provides instant access to all parameters of the chosen tap. The Tap parameter bar also provides access to several parameters that are not available in the Tap display, such as Transpose and Flip. Editing in the Tap parameter bar is fast and precise when you want to edit the parameters of a single tap. All parameters of the selected tap are available, with no need to switch display views or estimate values with vertical lines.
• Pan field: Controls the pan position for mono input signals, stereo balance for stereo input signals, and surround angle when used in surround configurations. • Pan displays a percentage between 100% (full left) and −100% (full right), which represents the pan position or balance of the tap. A value of 0% represents the center panorama position. • When used in surround, a surround panner replaces the percentage representation. For more information, see Working with Delay Designer in Surround.
To reset the value of a tap Do one of the following: µ In the Tap display, Option-click a tap to reset the chosen parameter to its default setting. If multiple taps are selected, Option-clicking any tap will reset the chosen parameter to its default value for all selected taps. µ In the Tap parameter bar, Option-click a parameter value to reset it to the default setting.
• 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 re-enabled. • Mix sliders: Independently set the levels of the dry input signal and the post-processing wet signal. Working with Delay Designer in Surround Delay Designer’s design is optimized for use in surround configurations.
Modulation Delay The Modulation Delay is based on the same principles as the Flanger and Chorus effects, but you can set the delay time, allowing both chorus and flanging effects to be generated. It can also be used without modulation to create resonator or doubling effects. The modulation section consists of two LFOs with variable frequencies. Although rich, combined flanging and chorus effects are possible, the Modulation Delay is capable of producing some extreme modulation effects.
Note: The right LFO Rate knob is available only in stereo and surround instances, and it can be set separately only if the Left Right Link button is not enabled. • LFO Left Right Link button: Available only in stereo and surround instances, it links the modulation rates of the left and right stereo channels. Adjustment of either Rate knob will affect the other channels.
Stereo Delay The Stereo Delay works much like the Tape Delay (see Tape Delay), but allows you to set the Delay, Feedback, and Mix parameters separately for the left and right channels. The Crossfeed knob for each stereo side determines the feedback intensity or the level at which each signal is routed to the opposite stereo side. You can freely use the Stereo Delay on mono channels when you want to create independent delays for the two stereo sides.
• Crossfeed Left to Right (Crossfeed Right to Left) knob and field: Transfer the feedback signal of the left channel to the right channel, and vice versa. • Feedback Phase button: Use to invert the phase of the corresponding channel’s feedback signal. • Crossfeed Phase button: Use to invert the phase of the crossfed feedback signals. Common Parameters • Beat Sync button: Synchronizes delay repeats to the project tempo, including tempo changes.
• Delay field: Sets the current delay time in milliseconds (this parameter is dimmed when you synchronize the delay time to the project tempo). • Sync button: Synchronizes delay repeats to the project tempo (including tempo changes). • Tempo field: Sets the current delay time in beats per minute (this parameter is dimmed when you synchronize the delay time to the project tempo).
Equalizers 3 An equalizer (commonly abbreviated as EQ) shapes the sound of incoming audio by changing the level of specific frequency bands. Equalization is one of the most commonly used audio processes, both for music projects and in post-production work for video. You can use EQ to subtly or significantly shape the sound of an audio file, instrument, or project by adjusting specific frequencies or frequency ranges.
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 an LFO (low frequency oscillator). These control sources are used to dynamically modulate the filter cutoff.
AutoFilter Threshold Parameter The Threshold parameter analyzes the level of the input signal. If the input signal level exceeds the set threshold level, the envelope and LFO are retriggered—this applies only if the Retrigger button is active. The envelope and LFO can be used to modulate the filter cutoff frequency. AutoFilter Envelope Parameters The envelope is used to shape the filter cutoff over time. When the input signal exceeds the set threshold level, the envelope is triggered.
AutoFilter LFO Parameters The LFO is used as a modulation source for filter cutoff. • Coarse Rate knob, Fine Rate slider and field: Used to set the speed of LFO modulation. Drag the Coarse Rate knob to set the LFO frequency in Hertz. Drag the Fine Rate slider (the semicircular slider above the Coarse Rate knob) to fine-tune the frequency. Note: The labels shown for the Rate knob, slider, and field change when you activate Beat Sync. Only the Rate knob (and field) is available.
AutoFilter Filter Parameters The Filter parameters allow you to precisely tailor the tonal color. • Cutoff knob and field: Sets the cutoff frequency for the filter. Higher frequencies are attenuated, whereas lower frequencies are allowed to pass through in a lowpass filter. The reverse is true in a highpass filter. When the State Variable Filter is set to bandpass (BP) mode, the filter cutoff determines the center frequency of the frequency band that is allowed to pass.
AutoFilter Distortion Parameters The Distortion parameters can be used to overdrive the filter input or filter output. The distortion input and output modules are identical, but their respective positions in the signal chain—before and after the filter, respectively—result in remarkably different sounds. • Input knob and field: Sets the amount of distortion applied before the filter section. • Output knob and field: Sets the amount of distortion applied after the filter section.
Channel EQ The Channel EQ is a highly versatile multiband EQ. It provides eight frequency bands, including lowpass and highpass filters, low and high shelving filters, and four flexible parametric bands. It also features an integrated Fast Fourier Transform (FFT) Analyzer that you can use to view the frequency curve of the audio you want to modify, allowing you to see which parts of the frequency spectrum may need adjustment. You can use the Channel EQ to shape the sound of an individual clip.
Band 1 is a highpass filter. Band 2 is a low shelving filter. Bands 3 through 6 are parametric bell filters. Band 7 is a high shelving filter. Band 8 is a lowpass filter. • Graphic display: Shows the current curve of each EQ band. • Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band. • Drag vertically in the section of the display that encompasses each band to adjust the gain of each band (except bands 1 and 8).
Note: If you play back automation of the Q parameter with a different Gain-Q Couple setting, the actual Q values will be different than when the automation was recorded. Using the Channel EQ The way you use the 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 the Channel EQ to a flat response (no frequencies boosted or cut), turn on the Analyzer and play the audio signal.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top parameter, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and −80 dB. The Analyzer display is always dB-linear. Note: When choosing a resolution, be aware that higher resolutions require significantly more processing power.
• Band 5: Click the high shelving or lowpass button. • Graphic display: Shows the EQ curve of each frequency band. • Frequency fields: Sets the frequency for each band. • Gain knobs: Set the amount of gain for each band. • Q fields: Sets the Q or bandwidth of each band—the range of frequencies around the center frequency that are altered. At low Q factor values, the EQ covers a wider frequency range. At high Q values, the effect of the EQ band is limited to a narrow frequency range.
Linear Phase EQ Parameters The left side of the Channel EQ window includes the Gain and Analyzer controls. The central area of the window includes the graphical display and parameters for shaping each EQ band. Linear Phase EQ Gain and Analyzer Controls • Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands. • Analyzer button: Turns the Analyzer on or off.
• Graphic display: Shows the current curve of each EQ band. • Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band. • Drag vertically in the section of the display that encompasses each band to adjust the gain of each band (except bands 1 and 8). The display reflects your changes immediately. • Drag the pivot point in each band to adjust the Q factor. Q is shown beside the cursor when the mouse is moved over a pivot point.
Using the Linear Phase EQ The Linear Phase EQ is typically used as a mastering tool and is, therefore, generally inserted into master or output audio. The way you use the Linear Phase EQ is obviously dependent on the audio material and what you intend to do with it, but a useful workflow for many situations is as follows: Set the Linear Phase EQ to a flat response (no frequencies boosted or cut), turn on the Analyzer, and play the audio signal.
As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top parameter, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and −40 dB. The Analyzer display is always dB-linear. Note: When choosing a resolution, be aware that higher resolutions require significantly more processing power.
Levels Effects 4 The levels effects control the perceived loudness of your audio, add focus and punch to clips, and optimize the sound for playback in different situations. The dynamic range of an audio signal is the range between the softest and loudest parts of the signal—technically, between the lowest and highest amplitudes. Dynamics processors enable you to adjust the dynamic range of individual audio clips.
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.
Adaptive Limiter The Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds. It works by rounding and smoothing peaks in the signal, producing an effect similar to an analog amplifier being driven hard. Like an amplifier, it can slightly color the sound of the signal. You can use the Adaptive Limiter to achieve maximum gain, without introducing generally unwanted distortion and clipping, which can occur when the signal exceeds 0 dBFS.
• Out Ceiling knob and field: Sets the maximum output level, or ceiling. The signal will not rise above this. • Output meters (to the right): Show output levels, allowing you to see the results of the limiting process. The Margin field shows the highest output level. You can reset the Margin field by clicking it. • Mode buttons (Extended Parameters area): Choose the type of peak smoothing: • OptFit: Limiting follows a linear curve, which allows signal peaks above 0 dB.
Compressor Parameters The Compressor offers the following parameters: • Circuit Type pop-up menu: Choose the type of circuit emulated by the Compressor. The choices are Platinum, Class(ic) A_R, Class(ic) A_U, VCA, FET, and Opto (optical). • Side Chain Detection pop-up menu: Determines if the Compressor uses the maximum level of each side-chained signal (Max) or the summed level of all side-chained signals (Sum) to exceed or fall below the threshold.
• Compressor Threshold slider and field: Sets the threshold level—signals above this threshold value are reduced in level. • Peak/RMS buttons: Determines whether signal analysis is with the Peak or RMS method, when using the Platinum circuit type. • Gain slider and field: Sets the amount of gain applied to the output signal. • Auto Gain pop-up menu: Choose a value to compensate for volume reductions caused by compression. The choices are Off, 0 dB, and −12 dB.
Setting Suitable Compressor Envelope Times The Attack and Release parameters shape the dynamic response of the Compressor. The Attack parameter determines the time it takes after the signal exceeds the threshold level before the Compressor starts reducing the signal. Many sounds, including voices and musical instruments, rely on the initial attack phase to define the core timbre and characteristic of the sound.
Note: If you activate Auto Gain and RMS simultaneously, the signal may become over-saturated. If you hear any distortion, switch Auto Gain off and adjust the Gain slider until the distortion is inaudible. Enveloper The Enveloper is an unusual processor that lets you shape the attack and release phases of a signal—the signal’s transients, in other words. This makes it a unique tool that can be used to achieve results that differ from other dynamic processors.
Using the Enveloper The most important parameters of the Enveloper 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.
Expander The Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic range above the threshold level. You can use the Expander to add liveliness and freshness to your audio signals. • Threshold slider and field: Sets the threshold level. Signals above this level are expanded. • Peak/RMS buttons: Determine whether the Peak or RMS method is used to analyze the signal.
Gain Plug-in Gain amplifies (or reduces) the signal by a specific decibel amount. • Gain slider and field: Sets the amount of gain. • Phase Invert Left and Right buttons: Invert the phase of the left and right channels, respectively. • Balance knob and field: Adjusts the balance of the incoming signal between the left and right channels. • Swap L/R (Left/Right) button: Swaps the left and right output channels. The swapping occurs after the Balance parameter in the signal path.
The Limiter is used primarily when mastering. Typically, you apply the 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. The 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, the Limiter reduces the level before clipping can occur. The Limiter cannot, however, fix audio that is clipped during recording.
Multichannel Gain Multichannel Gain allows you to independently control the gain (and phase) of each channel in a surround mix. • Master slider and field: Sets the master gain for the combined channel output. • Channel gain sliders and fields: Set the amount of gain for the respective channel. • Phase Invert buttons: Invert the phase of the selected channel. • Mute buttons: Mute the selected channel.
Downward expansion works as a counterpart to compression. Whereas the compressor compresses the dynamic range of higher volume levels, the downward expander expands the dynamic range of the lower volume levels. With downward expansion, the signal is reduced in level when it falls below the threshold level. This works in a similar way to a noise gate, but rather than abruptly cutting off the sound, it smoothly fades the volume with an adjustable ratio.
Multipressor Frequency Band Section • Compr(ession) Thrsh(old) fields: Set the compression threshold for the selected band. Setting the parameter to 0 dB results in no compression of the band. • Compr(ession) 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(old) fields: Set the expansion threshold for the selected band.
Using the Multipressor In the graphic display, the blue bars show the gain change—not merely the gain reduction—as with a standard compressor. The gain change display is a composite value consisting of the compression reduction, plus the expander reduction, plus the auto gain compensation, plus the gain make-up. Setting Multipressor Compression Parameters The Compression Threshold and Compression Ratio parameters are the key parameters for controlling compression.
Noise Gate Parameters The Noise Gate has the following parameters. • Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level. • Reduction slider and field: Sets the amount of signal reduction. • Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold. • Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.
Using the Noise Gate In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold value are completely suppressed. Setting Reduction to a higher value attenuates low-level sounds but still allows them to pass. You can also use Reduction to boost the signal by up to 20 dB, which is useful for ducking effects. The Attack, Hold, and Release knobs modify the dynamic response of the Noise Gate.
The filters allow only very high (loud) signal peaks to pass. In the drum kit example, you could remove the hi-hat signal, which is higher in frequency, with the High Cut filter and allow the snare signal to pass. Turn monitoring off to set a suitable Threshold level more easily. Spectral Gate The Spectral Gate is an unusual filter effect that can be used as a tool for creative sound design.
• Center Freq. (Frequency) knob and field: Sets the center frequency of the band that you want to process. • Bandwidth knob and field: Sets the width of the frequency band that you want to process. • Super Energy knob and field: Controls the level of the frequency range above the threshold. • High Level slider and field: Blends the frequencies of the original signal—above the selected frequency band—with the processed signal.
b Use the High Level slider to blend frequencies above the defined frequency band with the processed signal. 5 You can modulate the defined frequency band using the Speed, CF Modulation, and BW Modulation parameters. a Speed determines the modulation frequency. b CF (Center Frequency) Modulation defines the intensity of the center frequency modulation. c BW (Band Width) Modulation controls the amount of bandwidth modulation.
You can link channels by assigning them to one of three groups. When you adjust the threshold or output parameter of any grouped channel, the parameter adjustment is mirrored by all channels assigned to the group. Link section Main section LFE section The Surround Compressor is divided into three sections: • The Link section at the top contains a series of menus where you assign each channel to a group. See Surround Compressor Link Parameters.
Surround Compressor Link Parameters The Surround Compressor’s Link section provides the following parameters: • 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 -).
Surround Compressor Main Parameters The Surround Compressor’s Main section provides the following parameters: • Ratio knob and field: Sets the ratio of signal reduction when the threshold is exceeded. • Knee knob and field: Determines the ratio of compression at levels close to the threshold. • Attack knob and field: Sets the amount of time it takes to reach full compression, after the signal exceeds the threshold.
Surround Compressor LFE Parameters The Surround Compressor’s LFE section provides the following parameters: • Ratio knob and field: Sets the compression ratio for the LFE channel. • Knee knob and field: Sets the knee for the LFE channel. • Attack knob and field: Sets the attack time for the LFE channel. • Release knob and field: Sets the release time for the LFE channel. • Auto button: When the Auto button is enabled, the release time automatically adjusts to the audio signal.
Modulation Effects 5 Modulation effects are used to add motion and depth to your sound. Effects such as chorus, flanging, and phasing are well-known examples. Modulation effects typically delay the incoming signal by a few milliseconds and use an LFO to modulate the delayed signal. The LFO may also be used to modulate the delay time in some effects.
Chorus Effect The Chorus effect delays the original signal. The delay time is modulated with an LFO. The delayed, modulated signal is mixed with the original, dry signal. You can use the Chorus effect to enrich the incoming signal and create the impression that multiple instruments or voices are being played in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several musicians or vocalists perform together.
The Ensemble effect can add a great deal of richness and movement to sounds, particularly when you use a high number of voices. It is very useful for thickening parts, but it can also be used to emulate more extreme pitch variations between voices, resulting in a detuned quality to processed material. • Intensity sliders and fields: Set the amount of modulation for each LFO. • Rate knobs and fields: Control the frequency of each LFO.
Note: When you are using the Ensemble effect in surround, the input signal is converted to mono before processing. In other words, you insert the Ensemble effect as a multi-mono instance. Flanger Effect The Flanger effect works in much the same way as the Chorus effect, but it uses a significantly shorter delay time. In addition, the effect signal can be fed back into the input of the delay line.
Phaser Effect The Phaser effect combines the original signal with a copy that is slightly out of phase with the original. This means that the amplitudes of the two signals reach their highest and lowest points at slightly different times. The timing differences between the two signals are modulated by two independent LFOs. In addition, the Phaser includes a filter circuit and a built-in envelope follower that tracks volume changes in the input signal, generating a dynamic control signal.
Phaser LFO Section • LFO 1 and LFO 2 Rate knobs and fields: Set the speed for each LFO. • LFO Mix slider and fields: Determines the ratio between the two LFOs. • Env Follow slider and field: Determines the impact of incoming signal levels on the speed of LFO 1. • Phase knob and field: Available only in stereo and surround instances. Controls the phase relationship between the individual channel modulations. At 0°, the extreme values of the modulation are achieved simultaneously for all channels.
You can choose between three different vibrato and chorus types. The stereo version of the effect features two additional parameters—Stereo Phase and Rate Right. These allow you to set the modulation speed independently for the left and right channels. The stereo parameters of the mono version of the Scanner Vibrato are hidden behind a transparent cover. • Vibrato knob: Use to choose from three Vibrato positions (V1, V2, and V3) or three Chorus positions (C1, C2, and C3).
Tremolo Effect The Tremolo effect modulates the amplitude of the incoming signal, resulting in periodic volume changes. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The graphic display shows all parameters, except Rate. • Depth slider and field: Determines the modulation amount. • Waveform display: Shows the resulting waveform. • Rate knob and field: Sets the frequency of the LFO.
6 Spaces Effects You can use spaces effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or an open space. Sound waves repeatedly bounce off the surfaces—walls, ceilings, windows, and so on—of any space, or off objects within a space, gradually dying out until they are inaudible. These bouncing sound waves result in a reflection pattern, more commonly known as a reverberation (or reverb).
Plates, Digital Reverb Effects, and Convolution Reverb The first form of reverb used in music production was actually a special room with hard surfaces, called an echo chamber. It was used to add echoes to the signal. Mechanical devices, including metal plates and springs, were also used to add reverberation to the output of musical instruments and microphones. Digital recording introduced digital reverb effects, which consist of thousands of delays of varying lengths and intensities.
PlatinumVerb The PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easy to precisely emulate real rooms. Its dual-band Reverb section splits the incoming signal into two bands, each of which is processed and can be edited separately.
PlatinumVerb Early Reflections Parameters The PlatinumVerb offers the following Early Reflections parameters: • Predelay slider and field: Determines the amount of time between the start of the original signal and the arrival of the early reflections. Extremely short Predelay settings can color the sound and make it difficult to pinpoint the position of the signal source.
PlatinumVerb Reverb Parameters The PlatinumVerb offers the following Reverb parameters: • Initial Delay slider and field: Sets the time between the original signal and the diffuse reverb tail. • Spread slider and field: Controls the stereo image of the reverb. At 0%, the effect generates a monaural reverb. At 200%, the stereo base is artificially expanded. • Crossover slider and field: Defines the frequency at which the input signal is split into two frequency bands, for separate processing.
• Diffusion slider and field: Sets the diffusion of the reverb tail. High Diffusion values represent a regular density, with few alterations in level, times, and panorama position over the course of the diffuse reverb signal. Low Diffusion values result in the reflection density becoming irregular and grainy. This also affects the stereo spectrum. As with Density, find the best balance for the signal. • Reverb Time slider and field: Determines the reverb time of the high band.
Space Designer Convolution Reverb 7 Space Designer is a convolution reverb effect. You can use it to place your audio signals in exceptionally realistic recreations of real-world acoustic environments. Space Designer generates reverb by convolving, or combining, an audio signal with an impulse response (IR) reverb sample. An impulse response is a recording of a room’s reverb characteristics—or, to be more precise, a recording of all reflections in a given room, following an initial signal spike.
• Automating Space Designer (p. 123) Getting to Know the Space Designer Interface The Space Designer interface consists of the following main sections: Impulse response parameters Envelope and EQ parameters Main display Button bar Global parameters Global parameters Filter parameters Parameter bar • Impulse response parameters: Used to load, save, or manipulate (recorded or synthesized) impulse response files.
Working with Space Designer’s Impulse Response Parameters Space Designer can use either recorded impulse response files or its own synthesized impulse responses. The circular area to the left of the main display contains the impulse response parameters. These are used to determine the Impulse Response mode (IR Sample mode or Synthesized IR mode), load or create impulse responses, and set the sample rate and length.
Important: To convolve audio in real time, Space Designer must first calculate any parameter adjustments to the impulse response. This requires a moment or two, following parameter edits, and is indicated by a blue progress bar. During this parameter edit processing time you can continue to adjust the parameter. When calculation starts, the blue bar is replaced by a red bar, advising you that calculation is taking place.
Any mono, stereo, AIFF, SDII, or WAV file can be used as an IR. In addition, surround formats up to 7.1, discreet audio files, and B-format audio files that comprise a single surround IR can also be used. Working in Space Designer’s Synthesized IR Mode In Synthesized IR mode, Space Designer generates a synthesized impulse response based on the values of the Length, envelope, Filter, EQ, and Spread parameters.
• If the project sample rate is 44.1 kHz, the options will be 22.05 kHz, 11.025 kHz, and 5512.5 Hz. Changing the sample rate upward increases—or changing it downward decreases—the frequency response (and length) of the impulse response, and to a degree the overall sound quality of the reverb. Upward sample rate changes are of benefit only if the original IR sample actually contains higher frequencies. When you are reducing the sample rate, use your ears to decide if the sonic quality meets your needs.
All envelopes are automatically calculated as a percentage of the overall length, which means that if this parameter is altered, your envelope curves will stretch or shrink to fit, saving you time and effort. When you are using an impulse response file, the Length parameter value cannot exceed the length of the actual impulse response sample. Longer impulse responses (sampled or synthesized) place a higher strain on the CPU.
• All button: Resets all envelopes and the EQ to default values. • Volume Env button: Displays the volume envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Volume Envelope. • Filter Env button: Displays the filter envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Filter.
Setting Space Designer’s Envelope Parameters You can edit the volume and filter envelopes of all IRs and the density envelope of synthesized IRs. All envelopes can be adjusted both graphically in the main display and numerically in the parameter bar. Whereas some parameters are envelope-specific, all envelopes consist of the Attack Time and Decay Time parameters.
Working with Space Designer’s Volume Envelope The volume envelope is used to set the reverb’s initial level and adjust how the volume will change over time. You can edit all volume envelope parameters numerically, and many can also be edited graphically (see Setting Space Designer’s Envelope Parameters). Init Level node Decay Time/End Level node Attack/Decay Time node • Init Level field: Sets the initial volume level of the impulse response attack phase.
Using Space Designer’s Density Envelope The density envelope allows you to control the density of the synthesized impulse response over time. You can adjust the density envelope numerically in the parameter bar, and you can edit the Init Level, Ramp Time, and End Level parameters using the techniques described in Setting Space Designer’s Envelope Parameters. Note: The density envelope is available only in Synthesized IR mode.
Working with Space Designer’s EQ Space Designer features a four-band EQ comprised of two parametric mid-bands plus two shelving filters (one low shelving filter and one high shelving filter). You can edit the EQ parameters numerically in the parameter bar, or graphically in the main display. EQ On/Off button Individual EQ band buttons • EQ On/Off button: Enables or disables the entire EQ section. • Individual EQ band buttons: Enable or disable individual EQ bands.
2 Drag the cursor horizontally over the main display. When the cursor is in the access area of a band, the corresponding curve and parameter area is automatically highlighted and a pivot point is displayed. 3 Drag horizontally to adjust the frequency of the band. 4 Drag vertically to increase or decrease the Gain of the band. 5 Vertically drag the (illuminated) pivot point of a parametric EQ band to raise or lower the Q value.
Using Space Designer’s Main Filter Parameters The main filter parameters are found at the lower-left corner of the interface. • Filter On/Off button: Switches the filter section on and off. • Filter Mode knob: Determines the filter mode. • 6 dB (LP): Bright, good general-purpose filter mode. It can be used to retain the top end of most material, while still providing some filtering. • 12 dB (LP): Useful where you want a warmer sound, without drastic filter effects.
Note: Activation of the filter envelope automatically enables the main filter. Controls the Attack Time endpoint (and Decay Time startpoint) and Break Level parameters simultaneously. Controls the Decay endpoint and End Level parameters simultaneously. • Init Level field: Sets the initial cutoff frequency of the filter envelope. • Attack Time field: Determines the time required to reach the Break Level (see below). • Break Level field: Sets the maximum filter cutoff frequency that the envelope reaches.
Space Designer Global Parameters: Upper Section These parameters are found around the main display. Output sliders Input slider Latency Compensation button Definition area Rev Vol Compensation button • Input slider: Determines how Space Designer processes a stereo or surround input signal. For more information, see Using Space Designer’s Input Slider. • Latency Compensation button: Switches Space Designer’s internal latency compensation feature on or off.
• Spread and Xover knobs (synthesized IRs only): Spread adjusts the perceived width of the stereo or surround field. Xover sets the crossover frequency in Hertz. Any synthesized impulse response frequency that falls below this value will be affected by the Spread parameter. See Using Space Designer’s Spread Parameters. Using Space Designer’s Input Slider The Input slider behaves differently in stereo or surround instances. The slider does not appear in mono or mono to stereo instances.
Using Space Designer’s Latency Compensation Feature The complex calculations made by Space Designer take time. This time results in a processing delay, or latency, between the direct input signal and the processed output signal. When activated, the Latency Compensation feature delays the direct signal (in the Output section) to match the processing delay of the effect signal. Note: This is not related to latency compensation in the host application.
Using Space Designer’s Rev Vol Compensation Rev Vol Compensation (Reverb Volume Compensation) attempts to match the perceived (not actual) volume differences between impulse response files. It is enabled by default and should generally be left in this mode, although you may find that it isn’t successful with all types of impulse responses. If this is the case, turn it off and adjust input and output levels accordingly.
• Bal(ance) slider: Sets the level balance between the front (L-C-R) and rear (Ls-Rs) channels. • In 7.1 ITU surround, the balance pivots around the Lm-Rm speakers, taking the surround angles into account. • With 7.1 SDDS surround, the Lc-Rc speakers are considered front speakers. • Rev(erb) slider: Adjusts the output level of the effect (wet) signal. • Dry slider: Sets the overall level of the non-effect signal.
Using Space Designer’s Spread Parameters The Spread and Xover knobs enhance the perceived width of the signal, without losing the directional information of the input signal normally found in the higher frequency range. Low frequencies are spread to the sides, reducing the amount of low frequency content in the center—allowing the reverb to nicely wrap around the mix. The Spread and Xover knobs function only in Synthesized IR mode.
Specialized Effects and Utilities 8 Final Cut Pro includes a bundle of Logic specialized effects and utilities designed to address tasks often encountered during audio production. As examples of where these processors can help: Denoiser eliminates or reduces noise below a threshold level. Exciter can add life to your recordings by generating artificial high frequency components. SubBass generates an artificial bass signal that is derived from the incoming signal.
Denoiser The Denoiser eliminates or reduces any noise below a threshold volume level. The Denoiser uses fast Fourier transform (FFT) analysis to recognize frequency bands of lower volume and less complex harmonic structure. It then reduces these low-level, less complex bands to the appropriate dB level. See Denoiser Main Parameters. If you use the Denoiser too aggressively, however, the algorithm produces artifacts, which are usually less desirable than the existing noise.
Denoiser Main Parameters The Denoiser offers the following main parameters: Threshold slider and field Noise Type slider and field Graphic display Reduce slider and field • Threshold slider and field: Sets the threshold level. Signals that fall below this level are reduced by the Denoiser. • Reduce slider and field: Sets the amount of noise reduction applied to signals that fall below the threshold.
Denoiser Smoothing Parameters The Denoiser offers the following smoothing parameters: Frequency knob and field Transition knob and field Time knob and field • Frequency knob and field: Adjusts how smoothing is applied to neighboring frequencies. If the 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.
The Direction Mixer works with any type of stereo recording, regardless of the miking technique used. For information about XY, AB, and MS recordings, see Getting to Know Stereo Miking Techniques. • Input buttons: Click the LR button if the input signal is a standard left/right signal, and click the MS button if the signal is middle and side encoded. • Spread slider and field: Determines the spread of the stereo base in LR input signals. Determines the level of the side signal in MS input signals.
Using the Direction Mixer’s Direction Parameter When Direction is set to a value of 0, the midpoint of the stereo base in a stereo recording is perfectly centered within the mix. The following applies when working with LR signals: • At 90°, the center of the stereo base is panned hard left. • At −90°, the center of the stereo base is panned hard right.
Understanding XY Miking In an XY recording, two directional microphones are symmetrically angled, from the center of the stereo field. The right-hand microphone is aimed at a point between the left side and the center of the sound source. The left-hand microphone is aimed at a point between the right side and the center of the sound source. This results in a 45° to 60° off-axis recording on each channel (or 90° to 120° between channels).
Unlike these effects, however, the Exciter passes the input signal through a highpass filter before feeding it into the harmonics (distortion) generator. This results in artificial harmonics being added to the original signal. 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 the Exciter to add life to recordings.
• A Goniometer for judging phase coherency in a stereo sound field • A Correlation Meter to spot mono phase compatibility • An integrated Level Meter to view the signal level for each channel You can view either the Analyzer or Goniometer results in the main display area. You switch the view and set other MultiMeter parameters with the controls on the left side of the interface.
Using the MultiMeter Analyzer In Analyzer mode, the MultiMeter’s main display shows the frequency spectrum of the input signal as 31 independent frequency bands. Each frequency band represents one-third of an octave. The Analyzer parameters are used to activate Analyzer mode, and to customize the way that the incoming signal is shown in the main display. Analyzer parameters Scale • Analyzer button: Switches the main display to Analyzer mode.
Using the MultiMeter Goniometer A goniometer helps you to judge the coherence of the stereo image and determine phase differences between the left and right channels. Phase problems are easily spotted as trace cancellations along the center line (M—mid/mono). The idea of the goniometer was born with the advent of early two-channel oscilloscopes.
Using the MultiMeter’s Level Meter The Level Meter displays the current signal level on a logarithmic decibel scale. The signal level for each channel is represented by a blue bar. RMS and Peak levels are shown simultaneously, with RMS levels appearing as dark blue bars and Peak levels appearing as light blue bars. When the level exceeds 0 dB, the portion of the bar above the 0 dB mark turns red. Current peak values are displayed numerically (in dB increments) above the Level Meter.
• When the Correlation Meter moves into the red area to the left of the center position, out-of-phase material is present. This will lead to phase cancellations if the stereo signal is combined into a mono signal. Using the MultiMeter Peak Parameters The MultiMeter Peak parameters are used to enable/disable the peak hold function and to reset the peak segments of all meter types. You can also determine a temporary peak hold duration.
Stereo Spread extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels. This is done alternately—middle frequencies to the left channel, middle frequencies to the right channel, and so on. This greatly increases the perception of stereo width without making the sound totally unnatural, especially when used on mono recordings.
The simplest use for the SubBass is as an octave divider, similar to octaver effect pedals for electric bass guitars. Whereas such pedals can only process a monophonic input sound source of clearly defined pitch, SubBass can be used with complex summed signals as well. See Using SubBass. SubBass creates two bass signals, derived from two separate portions of the incoming signal. These are defined with the High and Low parameters. See SubBass Parameters.
• Freq. Mix slider and field: Adjusts the mix ratio between the upper and lower frequency bands. • Low Ratio knob and field: Adjusts the ratio between the generated signal and the original lower band signal. • Low Center knob and field: Sets the center frequency of the lower band. • Low Bandwidth knob and field: Sets the width of the lower band. • Dry slider and field: Sets the amount of dry (non-effect, original) signal. • Wet slider and field: Sets the amount of wet (effect) signal.
In the first mode (default mode), it starts generating the test signal as soon as it is inserted. You can switch it off by bypassing it. In the second mode (activated by clicking the Sine Sweep button), Test Oscillator generates a user-defined frequency spectrum tone sweep—when triggered with the Trigger button. • Waveform buttons: Select the type of waveform to be used for test tone generation.
Vocal Effects 9 You can use the vocal effects of Final Cut Pro to correct the pitch of vocals or enhance audio signals. These effects can also be used for creating unison or slightly thickened parts, or even for creating harmony voices. This chapter covers the following: • DeEsser (p. 143) • Pitch Correction Effect (p. 145) • Pitch Shifter II (p. 148) • Vocal Transformer (p.
The Detector parameters are on the left side of the DeEsser window, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider. DeEsser Detector Section • Detector Frequency knob and field: Sets the frequency range for analysis. • Detector Sensitivity knob and field: Sets the degree of responsiveness to the input signal.
Pitch Correction Effect You can use the Pitch Correction effect to correct the pitch of incoming audio signals. Improper intonation is a common problem with vocal clips, for example. The sonic artifacts that can be introduced by the process are minimal and can barely be heard, as long as your corrections are moderate. 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.
• Scale pop-up menu and field: Click to choose different pitch quantization grids from the Scale pop-up menu. See Defining the Pitch Correction Effect’s Quantization Grid. • 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). See Excluding Notes from Pitch Correction.
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 plug-in 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.
Tip: You’ll often find that it’s best to correct only the notes with the most harmonic gravity. For example, choose “sus4” from the Scale pop-up menu, and set the Root note to match the project key. This will limit correction to the root note, the fourth, and the fifth of the key scale. Activate the bypass buttons for all other notes and only the most important and sensitive notes will be corrected, while all other singing remains untouched.
• Mix slider and field: Sets the balance between the effect and original signals. • Timing pop-up menu (Extended Parameters area): Determines how timing is derived: by following the selected algorithm (Preset), by analyzing the incoming signal (Auto), or by using the settings of the Delay, Crossfade, and Stereo Link parameters, described below (Manual). Note: The following three parameters are active only when “Manual” is chosen in the Timing pop-up menu.
The 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. Vocal Transformer Parameters The Vocal Transformer offers the following parameters. • Pitch knob and field: Determines the amount of transposition applied to the input signal. See Setting Vocal Transformer Pitch and Formant Parameters.
• Detune slider and field (Extended Parameters area): Detunes the input signal by the set value. This parameter is of particular benefit when automated. Setting Vocal Transformer Pitch and Formant Parameters Use the Vocal Transformer’s Pitch parameter to transpose the pitch of the signal upward or downward. Adjustments are made in semitone steps. Incoming pitches are indicated by a vertical line below the Pitch Base field.
Using Vocal Transformer’s Robotize Mode When Robotize is enabled, Vocal Transformer can augment or diminish the melody. You can control the intensity of this distortion with the Tracking parameter. The Tracking slider and field feature is enhanced by four buttons which immediately set the slider to the most useful values, as follows: • −1 (sets the slider to −100%): All intervals are mirrored.
