Summary of Contents for Universal Audio Ampex ATR-102
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CHAPTER 1 Documentation & Support UAD Documentation Overview This section describes the various instructional and technical resources that are available for installing, using, and troubleshooting UAD Powered Plug- Ins. Documentation for the product line is available in written, video, and on- line formats.
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Direct Developers UAD Powered Plug-Ins includes plug-ins from our Direct Developer partners. Documentation for these 3rd-party plug-ins are separate files that are written and provided by the plug-in developers themselves. The filenames for these plug-ins are the same as the plug-in names. Installed Locations The UAD and Direct Developer manual files are copied to the hard drive by the software installer to the following locations:...
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When the content of and function of the software represented in a screenshot is identical on both platforms, no differentiation is made in the screenshot title. If there is a significant difference between platforms, screenshots from both platforms are included. The technical support pages on our website offer a wealth of helpful informa- Online Documentation...
Customer Support Customer support is provided by Universal Audio staff to all registered UAD Powered Plug-Ins users. Support Hours Our support specialists are available to assist you via email and telephone during our normal business hours, which are from 9am to 5pm, Monday through Friday, Pacific Standard Time.
Tape Recorder has turned music recordings into records. With its cohesive sound, punch, and ability to provide subtle-to-deep tape saturation and color, the Ampex ATR-102 is a fixture in major recording and mastering studios — and is considered by many engineers to be the best-sounding tape machine for final mixdown.
Like magnetic tape, users can dial in a clean sound, or just the right amount of harmonic saturation. Mixdown Tape The primary purpose of the UAD Ampex ATR-102 is to obtain tape mixdown Deck sonics within the DAW environment. To obtain the classic tape mixdown...
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“noiseless” digital recorders), noise is still an ever-present character- istic of the sound of using tape and tape machines. The UAD Ampex ATR-102 models the hum, hiss, wow, flutter, and crosstalk characteristics of the original hardware. These noise components can be in-...
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(tape) EQ, and record bias can easily be altered for manual calibration and/or creative purposes. Manual UAD Ampex ATR-102 includes the full suite of tools required to manually cal- Calibration Tools ibrate the recorder. Manual calibration tools are provided so expert users can calibrate the system to their preferred methods for obtaining desired results.
VU modes. Figure 3. One side of the Ampex ATR-102 “penthouse” showing meter and I/O controls The plug-in operates at an internal level of –12 dBFS. Therefore a digital sig- nal with a level of –12 dB below full scale digital (0 dBFS) at the plug-in input...
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0 – 10, are arbitrary and do not reflect a particular dB value. Tip: Click the “RECORD” label text to return the Record value to 0 dB. UAD Powered Plug-Ins Manual - 25 - Chapter 2: Ampex ATR-102...
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Note that the Reproduce control is to the left of the Record control, which is atypical of most signal flow designs, where inputs usually precede outputs (flowing from left to right). This quirky arrangement of the Ampex ATR-102 I/O controls, where the input control “follows” the output control, is true to the original hardware design.
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“British sound” during tape’s heyday. NAB (also known as IEC2) was the American standard with its own sound. AES is truly standardized at 30 IPS and is the sole EQ found on the Ampex ATR-102 at 30 IPS. Power is the plug-in bypass control. When set to OFF, em-...
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Tape Type Tape Type selects the active tape stock formulation. Seven Tape Types are modeled in the UAD Ampex ATR-102. To se- lect the Tape Type, click the TAPE button to cycle through the available types, or click directly on the Tape Type value label.
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Type are shown in Table Table 1. Tape Manufacturer’s Recommended Calibration Levels Tape Type Calibration Flux Level +0 dB 177 nWb/m 35-90 +3 dB 251 nWb/m +3 dB 251 nWb/m UAD Powered Plug-Ins Manual - 29 - Chapter 2: Ampex ATR-102...
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+9 dB 502 nWb/m Tip: The UAD Ampex ATR-102 default presets bank offers a variety of preset Tape Type, Tape Speed, CAL level, and EQ configurations that are commonly used for the recording of specific genres. This control specifies the active tape head model. Head Widths of Head Width 1/4,”...
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The “spin state” is saved until it is changed again. Note: Spinning reels automation is not supported in all hosts. In Sonar, the plug-in must be configured as a “tempo-based effect” for reels animation. UAD Powered Plug-Ins Manual - 31 - Chapter 2: Ampex ATR-102...
The secondary controls panel is accessed by clicking the OPEN button beneath the AMPEX label. Figure 4. UAD Ampex ATR-102 secondary controls The Ampex ATR-102 has individual calibration controls for ad- Auto Cal justing sync (record) EQ, reproduction (playback) EQ, and record bias, which are used to compensate for the inherent non-linearities of tape systems.
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The Repro EQs are used as filters to shape the frequency response of the sys- tem in maintaining a flat response and enable compensation for any tape fre- quency loss or head wear. UAD Powered Plug-Ins Manual - 33 - Chapter 2: Ampex ATR-102...
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The Hum frequency is dependent on the Emphasis EQ control (page 27). The frequency is 60 Hz when set to NAB (US) and 50 Hz when set to CCIR (Eu- ropean). UAD Powered Plug-Ins Manual - 34 - Chapter 2: Ampex ATR-102...
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Wow is a by-product of capstan irregularities, while flutter is a by-product of tape stretching and sticking. Both can be effectively used for creative purposes. UAD Powered Plug-Ins Manual - 35 - Chapter 2: Ampex ATR-102...
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Transformer These ON/OFF buttons enable and disable the transformer cir- Enable cuit of the Ampex ATR-102. For an overview of this feature, see “Modeled Transformer” on page Tape Delay These parameters control the built-in Tape Delay, which creates tape echo effects. For an overview of this feature, “Tape Delay”...
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(see tip below). Tip: To retain custom delay times when changing Tape Speed, hold Shift when changing Tape Speed. UAD Powered Plug-Ins Manual - 37 - Chapter 2: Ampex ATR-102...
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Magnetic Reference Laboratory (“MRL”) is a company that produces align- ment tapes. The MRL tapes used in the UAD Ampex ATR-102 are fringing compensated. In-depth discussions about fringing compensation and system alignment are beyond the scope of this manual; thorough resources are avail- able from the MRL website at: http://www.mrltapes.com...
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Cal Levels. Note: When the Manual Cal knob is set to MRL, the Distortion Meter is inac- tive (there is no distortion display in this mode). UAD Powered Plug-Ins Manual - 39 - Chapter 2: Ampex ATR-102...
(Tape Type, Tape Speed, Emphasis EQ, and Head Width) are inadvertently modified. Preparation • Reduce monitoring system volume to avoid loud sine wave Tones. • Insert UAD Ampex ATR-102 on the DAW output bus (see note below). • Set Path Select Repro mode (Sync mode is not supported for manual cal).
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To manually calibrate UAD Ampex ATR-102: 1. Set the Manual Cal Knob to the “MRL” position. The built-in alignment tape Repro Level Calibration tone will sound and its level can be viewed on the Meters. 2. Set the Tones frequency to 1 kHz.
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If HF EQ is adjusted by a large amount, it may be necessary to reca- librate the record level (steps 12, 13). The manual calibration procedure is complete. For related information, see the Manual Calibration Notes in the next section. UAD Powered Plug-Ins Manual - 42 - Chapter 2: Ampex ATR-102...
Cal Levels. • We recommend leaving the record SHELF EQ control in its default position. • The Ampex ATR-102 hardware has an additional gain control via a set- screw (like Repro HF/LF, Bias, etc) which is usually used for manual gain calibrations.
1/4” 35-90 CCIR Ampex ATR-102 The Ampex ATR-102 uses an internal upsampling technique. This upsampling Latency results in a larger latency than most other UAD plug-ins. See “UAD Delay Compensation” in the UAD System Manual for more information. UAD Powered Plug-Ins Manual...
CHAPTER 3 Cambridge EQ Overview The UAD Cambridge EQ plug-in is a mastering-quality, no-compromise equal- izer that enables powerful tonal shaping of any audio source. Its algorithm was modeled from various high-end analog filters, providing a sonically rich foundation for timbral manipulation. Special attention was given to the han- dling of higher frequencies, resulting in a much smoother and more satisfying high-end response than is found in most digital filters.
Cambridge EQ Controls Each feature of the Cambridge EQ interface is detailed below. Response Curve The Response Curve Display plots the frequency response of the current Cam- Display bridge EQ settings. It provides instant visual feedback of how audio is being processed by the equalizer.
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Zoom Buttons The vertical scale of the Curve Display can be increased or reduced with the Zoom buttons. This function allows the resolution of the Curve Display to be changed for enhanced visual feedback when very small or very large amounts of boost or cut are applied.
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Master Level This control adjusts the signal output level of Cambridge EQ. This Knob may be necessary if the signal is dramatically boosted or re- duced by the EQ settings. The available range is ±20 dB. The A/B Selector switches between two separate sets of Cam- A/B Selector Button bridge EQ plug-in values.
Low Cut / High Cut Filters The Low Cut and High Cut filters are offered in ad- dition to the five parametric/shelf bands. A wide range of filter types is provided to facilitate tonal creativity. Many filters that are available are repre- sented.
Cut Frequency This knob determines the cutoff frequency for the Cut filters. The available Knob range is from 20 Hz – 5 kHz for the low cut filter, and 20 Hz – 20 kHz for the high cut filter. EQ Bands All five of the EQ bands can be used in parametric or shelf mode.
Table 7. Available ranges for the Band Frequency parameter Low Frequencies (LF) 20-400 Hz Low-Mid Frequencies (LMF) 30-600 Hz Mid Frequencies (MF) 100-6 kHz High-Mid Frequencies (HMF) 900-18 kHz High Frequencies (HF) 2k-20 kHz When operating at sample rates less than 44.1 kHz, the maximum fre- Note: quency will be limited.
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Note that the Q numeric value in relation to its knob position is warped (i.e. not linear) and varies according to the parametric type. When set to Type I, the bandwidth remains at a fixed Q regardless of the gain Type I setting for the band;...
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Note that the Q value increases as gain is boosted but the knob position does not change The Q value is approached as gain increases, and reaches the knob position at maximum gain. See Figure Figure 11. Parametric Type II response When set to Type III, there is a Q/Gain dependency on boost and attenua- Type III tion.
Shelf EQ Shelf Enable Each band can be switched from parametric mode to shelf Button mode by clicking the shelf enable button. The button is off by default. To enable shelving on any band, click the shelf button. The button is green when shelving is enabled. Additionally, the control bat associated with the band has a horizontal shelf indicator line in the response curve display (see Figure 14 on page...
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Figure 13. Shelf Type A Shelf Mode Indicator Line Figure 14. Shelf Type B Figure 15. Shelf Type C UAD Powered Plug-Ins Manual - 56 - Chapter 3: Cambridge EQ...
CHAPTER 4 Cooper Time Cube Dual Mechanical Delay Line The original Cooper Time Cube was a Duane H. Cooper and Bill Putnam col- laborative design that brought a garden hose-based mechanical delay to the world in 1971 and has achieved cult status as the most unique delay ever made.
Design Overview The original UREI/Universal Audio Model 920-16 Cooper Time Cube hard- ware (see “Cooper Time Cube Hardware” on page 62) has two audio chan- nels, A and B. Each channel is transduced to/from a coiled length of plastic tubing which provides the acoustic “sound columns” that define its distinctive sonic character.
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HP Filter The 12 dB per octave high pass filter is used to reduce low frequencies at the input to the delays when desired. The high pass filter affects the delayed (wet) signals only. The available frequency range is from 20 Hz to 12 kHz. Turn the knob clockwise to reduce low frequencies into the delay processors.
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Color The Color switch toggles between the original filter emphasis of the hardware in position A and the “leveled” filter in position B which allows for greater De- cay ranges. Unlike the other parameters, the A and B labels for Color are for reference only.
Channel Controls The channel controls affect each channel of the processor independently. The control functionality is identical for each channel. “A” indicates the left chan- nel and “B” is the right channel. Figure 18. The channel controls Delay controls the delay time for each channel of the processor. The selected Delay A/B value is shown in the Echo display (“Echo A/B”...
Echo Volume This control determines the volume of the delayed signal. Rotate the control clockwise for louder echo. Up to +10 dB of gain is available at the maximum setting. Reducing the control to its minimum value will mute the delay. Tip: Click the “ECHO VOL”...
CHAPTER 5 CS-1 Channel Strip Overview The CS-1 Channel Strip provides the EX-1 Equalizer and Compressor, DM-1 Delay Modulator, and RS-1 Reflection Engine combined into one plug-in. In- dividual effects in the CS-1 Channel Strip can be bypassed when not in use to preserve UAD DSP use.
EX-1 Equalizer and Compressor Figure 22. The EX-1 EQ/Compressor plug-in window The EX-1 plug-in consists of a five-band parametric EQ and compressor. EX-1 Equalizer Controls The Equalizer portion of the EX-1 is a five-band fully parametric EQ. Each band has its own set of controls. The first two bands can also be enabled to function as low-shelf or high-pass filter.
Gain (G) Knob The Gain control determines the amount by which the frequency setting is boosted or attenuated. The available range is ±18 dB. Determines the center frequency to be boosted or attenuated by the Gain set- Frequency (fc) Knob ting.
Ratio Knob Determines the amount of gain reduction used by the compression. For exam- ple, a value of 2 (expressed as a 2:1 ratio) reduces the signal by half, with an input signal of 20 dB being reduced to 10 dB. A value of 1 yields no com- pression.
DM-1 Delay Modulator Figure 23. The DM-1 Delay Modulator plug-in window The DM-1 Delay Modulator provides stereo effects for delay, chorus, and flange. DM-1 Controls Sync Button This button puts the plug-in into Tempo Sync mode. See Chapter 8 in the UAD System Manual for more information.
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Mode Pop-up Determines the DM-1 effect mode. The available modes are: Chorus, Menu Chorus180, QuadChorus, Flanger1, Flanger2, Dual Delay, and Ping Pong Delay. In addition to reconfiguring the DM-1’s settings, the Mode also deter- mines the available parameter ranges for L/R Delay and Depth. In Chorus mode, both oscillators (or modulating signals) are in phase.
The RECIR units are expressed as a percentage in all Modes except Dual De- lay and Ping Pong. In these modes, RECIR values are expressed as T60 time, or the time before the signal drops 60 decibels. This low pass filter reduces the amount of high frequencies in the signal. Turn Damping Knob down this control to reduce the brightness.
Link Button This button links the left and right delay knobs so that when you move one de- lay knob, the other follows. The ratio between the two knobs is maintained. Figure 24. The DM-1L includes a Link button RS-1 Reflection Engine Figure 25.
RS-1 Controls Sync Button This button puts the plug-in into Tempo Sync mode. See Chapter 8 in the UAD System Manual for more information. Determines the shape of the reverberant space, and the resulting reflective Shape Pop-up Menu patterns. Table 8. Available RS-1 Shapes Cube Square Plate Rectangular Plate...
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Recirculation allows both positive and negative values. The polarity refers to the phase of the delays as compared to the original signal. If Recirculation dis- plays a positive value, all the delays will be in phase with the source. If it dis- plays a negative value, then the phase of the delays flips back and forth be- tween in phase and out of phase.
CHAPTER 6 dbx 160 Compressor/Limiter Overview The dbx® 160 Compressor/Limiter is an officially licensed and faithful emu- lation of the legendary dbx 160 hardware compressor/limiter — still widely considered the best VCA compressor ever made. Originally designed and sold by David Blackmer in 1971, this solid-state design set the standard for performance and affordability.
dbx 160 Controls The minimal controls on the UAD dbx 160 make it very simple to operate. Threshold Knob The Threshold knob defines the level at which the onset of compression occurs. Incoming signals that exceed the Threshold level are compressed. Signals below the Thresh- old are unaffected.
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Output Gain Output Gain controls the signal level that is output from the plug-in. The available range is ±20 dB. Generally speaking, adjust the Output control after the de- sired amount of compression is achieved with the Threshold and Compression controls. Output does not affect the amount of compression.
CHAPTER 7 DreamVerb Overview DreamVerb™, Universal Audio’s flagship stereo reverb plug-in, draws on the unparalleled flexibility of RealVerb Pro. Its intuitive and powerful interface lets you create a room from a huge list of different materials and room shapes. These acoustic spaces can be customized further by blending the different room shapes and surfaces with one another, while the density of the air can be changed to simulate different ambient situations.
Screenshot Figure 27. The DreamVerb plug-in window Signal Flow Figure 28 illustrates the signal flow for DreamVerb. The input signal is equal- ized then delay lines are applied to the early reflection and late field gener- ators. The resulting direct path, early reflection, and late-field reverberation are then independently positioned in the soundfield.
(Figure 27 on page 77) is similarly organized. The DreamVerb user interface Reflected energy equalization is controlled with the Resonance panel. The pattern of early reflections (their relative timing and amplitudes) is determined by the room shapes in the Shape panel (Figure 31 on page 80).
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Bypass switch Band Amplitude control bats Band 1 (low Band 5 (high Band 2, 3, and 4 shelving) control Edge control bats shelving) control Figure 29. DreamVerb Resonance panel The equalizer can be disabled with this switch. When the switch is off (black Bypass instead of grey), the other resonance controls have no effect.
Shelving The simplest (and often most practical) use of the equalizer is for low and/or high frequency shelving. This is achieved by dragging the left-most or right- most horizontal line (the ones without control bats) up or down, which boosts or cuts the energy at these frequencies.
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Shape Menus DreamVerb lets you specify two room shapes that can be blended to create a hybrid of early reflection pat- terns. The first and second shape each have their own menu. The available shapes are the same for each of the two shape menus.
Materials Panel The parameters in the Materials panel, in conjunction with the Shape panel (Figure 31 on page 80) and Reverberation panel (Figure 34 on page 86) ef- fect the spatial characteristics of the reverb. The material composition of an acoustical space effects how different fre- quency components decay over time.
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Materials Menus DreamVerb lets you specify two room materials, which can be blended to create a hybrid of absorption and re- flection properties. The first and second room material each has its own menu. The available materials are the same for each of the two materials menus. The first material is displayed in the lower left area of the Materials panel, and the second material is displayed in the lower right area.
Figure 32 on page 82) are used to blend the Materials The Materials Blending Bars (see Blending Bars three materials together at any ratio. The materials are not just mixed together with the bars; the reverberation algorithm itself is modified by blending. Materials Blending Blend the two materials by dragging the vertical Blending Bar horizontally.
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ER End control bat (time & amplitude) Bypass switch Materials Filtering control bat ER Start control bat (predelay & amplitude) Late-field relative timing display Figure 33. DreamVerb Reflections panel Bypass The early reflections can be disabled with this switch. When the switch is off (black instead of grey), the other Reflections controls have no effect.
Late-Field To highlight the relative timing relationship between the early reflections and Relative Timing late-field reverberation components, the shape and timing of the late-field is represented as an outline in the Reflections panel. The shape of this outline is modified by parameters in the Reverberations panel, not the Reflections panel.
Late-Field Start This parameter defines when the late-field reverb tail begins (the delay be- tween the dry signal and the onset of the LF) in relation to the dry signal. This bat controls two late-field parameters. Dragging the bat vertically con- Amplitude &...
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Figure 35. DreamVerb Positioning panel Direct These two sliders control the panning of the dry signal. The upper Direct slider controls the left audio channel, and the lower Direct slider controls the right audio channel. A value of <100 pans the signal hard left; a value of 100> pans the signal hard right.
Distance DreamVerb allows you to control the distance of the perceived source with this slider. In reverberant environments, sounds originating close to the lis- tener have a different mix of direct and reflected energy than those originat- ing further from the listener. Larger percentages yield a source that is farther away from the listener.
Mute This switch mutes the signal at the input to DreamVerb. This allows the reverb tail to play out after mute is applied, which is helpful for auditioning the sound of the reverb. Mute is on when the button is gray and off when the button is black.
DreamVerb Presets DreamVerb includes 100+ presets in addition to the internal factory bank. Presets in the internal factory bank are accessed via the host application’s pre- set menu. The additional presets are copied to disk by the UAD installer and can be loaded using the Settings menu in the UAD Toolbar (see “Using UAD Powered Plug-Ins”...
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• The EQ is often most useful for a simple Lf or Hf roll-off/boost, or to notch out bothersome frequencies for particular sources. For full mix ambience/mas- tering presets, use the EQ to cut most of all LF input, which yields added am- bience without mucking up the mix.
FATSO Sr. offers a “Tranny” saturation control, sidechain filtering, and deeper com- pression parameters including Threshold, Attack and Release. These special FATSO Sr. mods are only available with Universal Audio’s plug-in version of this unique studio tool. Note: The additional controls in the FATSO Sr.
FATSO Screenshots Figure 37. The FATSO Jr. plug-in window Figure 38. The FATSO Sr. plug-in window FATSO Functional Overview The FATSO was essentially designed to integrate frequencies in a musical Four Processing Types manner and provide some foolproof vintage sounding compression. Gener- ally, it is difficult to make the unit sound unnatural due to its vintage topology.
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2nd and 3rd get increasingly harsh and unmusical, and therefore should be lower in amplitude (<-60 dB) to keep within our line of thinking. Second har- monic is considered to be the warmest and most “consonant” harmonic dis- tortion. Warmth High Frequency Saturation Processor This circuit is meant to simulate the softening of the high frequencies that oc-...
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Transformer design and use is an art, and there are always trade-offs. How- ever, it has been widely known that a good audio transformer circuit can do wonderful things to an audio signal. This was the goal of the Tranny circuit. The hardware designers tried to emulate the desirable characteristics of the good old input/output transformers in a consistent musical way, and in a se- lectable fashion.
Tracking Tracking mode (green and yellow LED) is an 1176 type compressor that is great for tracking instruments and vocals during the recording process or dur- ing mixdown. Spank Spank mode (red LED) is a radical limiter type compressor that was specifi- cally designed to emulate the nice squeeze of the older SSL talkback compres- sors from the 70's &...
Channel Controls Input The Input knob defines the signal level going into the plug-in. Higher levels result in a more saturated signal. Levels above 0 VU provide dramatically higher distortion characteristics, especially when clipped (as indicated by the Pinned LED). See THD Indicators below.
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Table 9. Compressor Mode LED States Compressor Mode LED State Active Compressor Mode(s) All Unlit Compressor inactive Green Buss Yellow General Purpose (G.P.) Green + Yellow Tracking (most versatile ratio) Spank Red + Green Spank + Buss Red + Yellow Spank + General Purpose Red + Green + Yellow Spank + Tracking...
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Bypass/Tranny This black button is a multifunction control. Clicking the button re- peatedly cycles through Tranny, Bypass, and Tranny Off modes. The currently active mode is indicated by the adjacent LED's. Tranny (green LED) The Tranny processor is active in this mode (see “The Tranny Processor”...
Global Controls The global controls are not channel-specific; they apply to both channels. Link Compress The control signal sidechains of the gain reduction processors for channels 1 and 2 can be linked using the Link Compress function. To activate Link Compress, click the LINK COMPRESS text or LED on Ch1, on the left.
Controls Unlinked Unlink the controls when dual-mono operation is desired. Channel 1 and 2 controls are completely independent in this mode, and automation data is written and read by each channel separately. Link Controls is disabled when the FATSO is used in a mono-in/mono-out configuration. Important: Unlike the other controls for channels 1 and 2, which are iden- tical on the left and right sides of the interface, the Link CONTROLS function...
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Filter Filter regulates the cutoff frequency of the filter on the compres- (HP SIDE FILT) sor's control signal sidechain. When active, frequencies below the filter value are not passed to the sidechain. Values of 60 Hz, 120 Hz, 240 Hz, 480 Hz, and Off are available. The filter slope is 6 dB per octave.
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Attack LED’s Unlit – Compressor Inactive When the compressor is disabled and all Attack LED’s are unlit, the button is disabled. Note: This control has no effect when the compressor is inactive, or when it is in “pure” Spank mode (see “Compressor Mode”...
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Tranny Level This control determines the amount of Tranny process- ing (see “The Tranny Processor” on page 95 for a de- tailed description). Higher values make the Tranny ef- fect more prominent. Increasing the Tranny level also increases the signal THD (see “THD Indicators”...
Plant Studios in Sausalito, CA. That’s nearly two thou- sand pounds of sound in one plug-in! Impractical or impossible for most be- fore, Universal Audio brings this classic mix tool within the reach of everyone. Licensed and endorsed by EMT Studiotechnik GmbH as the world’s only au- thentic plate reverb emulation.
EMT 140 Controls The EMT 140 interface is an amalgam of controls found at the plate amplifier itself and the remote damper controls, plus a few DAW-friendly controls that we added for your convenience. The GUI incorporates the original look and feel of those controls, and utilizes that look for the DAW-only controls.
Reverb Controls Plate reverb systems are extremely simple: A re- mote damper setting, and a high pass or shelf fil- ter found at the plate itself. Additional manipula- tion is often used, including reverb return equalization, which is typically achieved at the console.
Stereo Controls Width Width allows you to narrow the stereo image of EMT 140. The range is from 0 – 100%. At a value of zero, EMT 140 returns a monophonic reverb. At 100%, the stereo reverb field is as wide as possible.
Because this is a shelving EQ, all frequencies below this setting will be af- fected by the low band Gain value. This parameter determines the amount by which the transition frequency set- Low Gain ting for the low band is boosted or attenuated. The available range is ±12 dB, in increments of 0.5 dB (fine control) or 1.0 dB (coarse control).
Output Meter The vintage-style VU Meter represents the plug-in output level. It is active when the Power switch is on, and slowly returns to zero when Power is switched off. Blend Controls The amount of time between the dry signal and the onset of the Predelay reverb is controlled with this knob.
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Power Switch This toggle switch enables or disables EMT 140. You can use it to compare the processed settings to the original signal, or to by- pass the plug-in which reduces (but not eliminates) the UAD DSP load (unless UAD-2 DSP LoadLock is enabled). The red EMT power indicator glows brighter when the plug-in is enabled.
CHAPTER 10 EMT 250 Electronic Reverberator Introduction Unveiled by EMT at the AES convention in 1976 and inducted into the TEC Hall of Fame in 2007, the EMT 250 was the first digital reverberation device to create ambience through a purely electronic system. With its single reverb program and iconic lever-driven control surface, the EMT 250 is still an indis- pensable tool within the record-making elite and is widely considered one of the best-sounding reverbs ever made.
EMT 250 Screenshot Figure 40. The EMT 250 plug-in window Functional Overview Program Modes The EMT 250 offers six effect types: Reverb, Delay, Phase, Chorus, Echo, and Space. These effects are called “program modes” in the EMT 250. Only one mode can be active at a time.
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Each unique parameter in the plug-in retains a distinct value, but only the pa- rameters that are active in the current program mode are visible in the graph- ical user interface. All parameters are always visible in Controls View (see Chapter 7 of the UAD System Manual), even when they are not active in the current program mode.
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In some program modes, the yellow “LED ring” around the control is illumi- nated to indicate that changing the switch position will change the sound. For program modes that do not offer quadraphonic processing (e.g., Delay), the switch is re-purposed to sum the processed outputs to mono. In Echo mode, it functions as an input mute.
Program Mode Controls The details of each unique program mode are below, followed by descrip- tions of the global controls, which affect all program modes. Control Table 10 displays the parameter that each control is mapped to for each of Functions the EMT 250 program modes.
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Reverb Reverb program mode offers the same all-time classic reverb algorithm that made the EMT 250 famous. Decay Time (Lever 1) Lever 1 controls the main reverb tail decay time. The red LEDs on the left side of lever 1 indicate the current decay time; the green LEDs on the right side of lever 1 are inactive.
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Delay Delay program mode offers two independent delay processors, one each for the left and right output channels. Up to 375ms delay time is available for each channel. Delay repeats (feedback) are not available in Delay mode; use Echo mode if delay feedback is desired. Note: The maximum per-channel delay time of 375ms in Delay mode is ob- tained by setting the coarse, fine, and predelay times to their respective max-...
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Important: In Delay mode, lever 3 does not control a “real” parameter; it is only used to select the active channel for other parameters in the graphical user interface. For this reason, the parameter is not exposed for external con- trol surfaces or automation, nor is it saved in sessions or presets.
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In Phase mode the green LEDs to the of right lever 1 are active, but the panel markings (0–300ms) do not represent the actual phase delay time values. In- stead, the LEDs indicate the relative value between 0–15ms. Predelay (Lever 4) Lever 4 can be used as a common predelay to both phase delays.
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Positions “I” and “II” are of a simpler nature, while “III” and “IV” are more complex. Position “I” duplicates the Left Front and Right Front outputs of the hardware. “II” duplicates Left Rear and Right Rear outputs of the hardware. “III”...
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Note: Levers 1 and 2 both control the echo time, but these parameters are not individually exposed for external control surfaces and automation. In- stead, a single echo time parameter is exposed, and levers 1 and 2 in the plug-in interface are both updated to match the value. HF Decay (Lever 3) Lever 3 controls the high frequency damping in Echo mode.
Predelay (Lever 4) Lever 4 is used as a typical reverb predelay parameter. See “Lever 4 Prede- lay” on page 115 for more information. Front/Rear In Space mode, the Front/Rear Outputs switch is illuminated. Changing the switch setting will yield a slightly different effect. See “Front/Rear Outputs”...
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Dry/Wet The Dry/Wet slider control determines the balance between the original and the pro- cessed signal. The range is from 0% (dry, unprocessed) to 100% (wet, processed sig- nal only). This control uses a logarithmic scale to provide increased resolution when se- lecting lower values.
*EP-34 Tape Echo is not affiliated with, sponsored nor endorsed by any companies currently using the Echoplex name. The EP-34 Tape Echo name, as well as the EP-3 and EP-4 model names, are used solely to identify the classic effects emulated by Universal Audio’s product. UAD Powered Plug-Ins Manual...
EP-34 Tape Echo Screenshot Figure 41. The UAD EP-34 Tape Echo plug-in window EP-34 Controls Echo Delay Echo Delay controls the delay time of the unit. The selected value is shown in the Echo display (page 128). The parameter can be adjusted by using the metallic “slider handle” or the “slider nose”...
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The available delay range is 80 to 700 milliseconds. When Sync is active, beat values from 1/64 to 1/2 can be selected. When the beat value is out of range, the value is displayed in parenthesis. This occurs in Sync mode when the time of the note value exceeds 700ms (as defined by the current tempo of the host application).
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Echo Volume This knob determines the wet/dry mix of the delayed signal. In the minimum position, the “dry” signal is colored by the cir- cuitry of the modeled emulation. Rotate the control clockwise for louder echo. Reducing the control to its minimum value will mute the delay.
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Echo Pan Pan sets the position of the delayed (wet) signal in the stereo field; it does not affect the unprocessed (dry) signal. Tip: Click the “Echo” control text to return the knob to center. Note: When the plug-in is used in a mono-in/mono-out (“MIMO”) configuration, the Pan knob does not function and cannot be adjusted.
Sync This switch engages Sync mode for the plug-in. In Sync mode, delay times are synchronized to (and therefore dependent upon) the mas- ter tempo of the host application. When Sync is toggled, parameter units are converted between milliseconds and beats to the closest matching value.
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*EP-34 Tape Echo is not affiliated with, sponsored nor endorsed by any companies currently using the Echoplex name. The EP-34 Tape Echo name, as well as the EP-3 and EP-4 model names, are used solely to identify the classic effects emulated by Universal Audio’s product. UAD Powered Plug-Ins Manual...
Control” back in the 1950’s when it was originally sold. It is still revered for its extremely smooth, artifact-free sound, and now Universal Audio has made it easily accessible to you. And you don’t have to let it warm up for 30 minutes...
Fairchild Screenshot Figure 43. The Fairchild plug-in window 2 Compressors, 4 Modes There are two compressors within the Fairchild 670. They can be used as dual L/R, dual mono/stereo, or they can be linked together and used on ei- ther the L/R or mono/stereo signals. The mode in which the compressors operate is determined by the combina- “Fairchild Modes”...
Controls Overview Most of the controls are associated with one or the other of the compressors, as opposed to being strictly associated with one channel of input/output (de- pends on active mode). These controls include Threshold, Time Constant, Bias Current Balance, and DC Bias. There are two sets of controls that always work on the left and right signals: input level and output level.
Fairchild Modes Dual Left/Right In Dual L/R mode, the Fairchild operates as two monophonic compressors with completely independent controls for the left and right channels. There is no interaction between the left and right channels. In Lat/Vert mode, the 670 acts on the lateral and vertical (the sum and differ- Lateral –...
Stereo, coupled This mode, like stereo couple left/right, causes the two compressors to be mono/stereo linked together so that they always compress the same amount. But here, the inputs to the two compressors are fed with the mono and stereo components of the signal.
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Meter Select Switch This switch determines what is displayed on the VU meters. If GR is selected, the meter will show gain reduction in dB for the corresponding compressor channel (which is not necessarily left or right; depends on the active mode). If the AGC switch has been set to left/right, the GR shown will be for the left or right channel.
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Time Constant This 6-position switch provides fixed and variable time constants (attack and release times) to accommodate various types of program material. Positions 1-4 provide successively slower behavior, and 5 and 6 provide program de- pendent response. The values published by Fairchild for each position are in Table 12 below.
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DC Bias DC Bias controls the ratio of compression as well as the knee width. As the knob is turned clockwise, the ratio gets lower and the knee gets broader. The threshold also gets lower as the knob is turned clockwise. The ‘factory cal’ tick mark should be aligned with the screw slot “dot”...
An original 4032 still resides in Florida with Thriller engineer and Bill Putnam protégé Bruce Swedien, where he continues his love affair with the desk he calls “marvelous sounding.” Universal Audio’s plug-in version of the all-important 32C EQ module is measured from Mr.
Harrison 32C EQ and Harrison 32C SE Controls Note: Knob settings, when compared to the graphical user interface silk- screen numbers, may not match the actual parameter values. This behavior is identical to the original hardware, which we modeled exactly. When the plug-in is viewed in parameter list mode (Controls View), the actual parameter values are displayed.
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Low Pass (high cut) This control determines the cutoff frequency for the low pass filter. The avail- able range is 1.6 kHz to 20 kHz. Each of the four EQ bands have Four EQ Bands similar controls. The band center frequency is controlled the top row of knobs, and the band gain is controlled by the bottom row.
Hi Gain This control determines the amount by which the frequency setting for the high band is boosted or attenuated. The available range is ±10 dB. The Gain knob controls the signal level that is Gain output from the plug-in. The default value is 0 dB.
Harrison 32C Latency The Harrison 32C (but not the Harrison 32C SE) uses an internal upsampling technique to facilitate its amazing sonic accuracy. This upsampling results in a slightly larger latency than other UAD plug-ins. See Chapter 9 “UAD Delay Compensation”...
Helios consoles for their personal use. Of all the Helios consoles produced between 1969 and 1979, the original “Type 69” is considered the most musical. Universal Audio modeled the EQ section of the very first Type 69. This console was originally found at Island’s Basing Street Studio in London;...
Helios Type 69 Controls Overview The simple yet powerful Helios Type 69 Passive EQ adds a unique sonic tex- ture to the music that passes through it. It can be pushed to its most extreme boost settings while retaining openness and clarity. The Type 69 Passive EQ replicates all the controls of the original hardware.
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Bass The Bass knob has a dual purpose. It specifies the amount of attenuation when the low band is in shelving mode, and specifies the frequency of the low frequency peak filter when the Bass Gain knob is not zero. When Bass is set to one of the frequency values (60 Hz, 100 Hz, 200 Hz, or 300 Hz) the low band is in peak mode.
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Whether gain or attenuation is applied is determined by the Mid Type Note: control. Mid Type specifies whether the midrange band is in Peak or Trough Mid Type mode. When switched to Peak, the Mid Gain control will boost the midrange.
duced if UAD-2 DSP LoadLock is enabled). Click the switch to toggle the state; the switch is illuminated in green when the plug-in is active. Helios 69 Latency The Helios 69 uses an internal upsampling technique to facilitate its amazing sonic quality.
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Basing Street—Home of the original Type 69 Helios desk The same desk, now in Berkeley’s Morningwood, nearly 40 years later UAD Powered Plug-Ins Manual - 151 - Chapter 14: Helios Type 69 Equalizer...
CHAPTER 15 LA-2A Overview The LA-2A Classic Leveling Amplifier long ago achieved classic status. It’s a given in almost any studio in the world – relied upon daily by engineers whose styles range from rock to rap, classical to country and everything in be- tween.
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Figure 48 depicts the input and output characteristics of a compressor and perfect amplifier. When operated within its specified range, an amplifier pro- vides a constant amount of gain regardless of the input signal level. In Figure 48, the signal level of a perfect amplifier is represented with a constant output gain of 10 dB.
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Compression region 10 dB of compression 2:1 compression ratio Output –10 knee Level (dB) –20 –30 –30 –20 –10 Input Level (dB) Figure 49. Input and output curve of compressor with 2:1 ratio and –20 dB threshold As mentioned previously, the compression ratio is defined as the ratio of the increase of the level of the input signal to the increase in the level of the output Figure 49, the input level is increased by 10 dB while the output...
Teletronix LA-2A Leveling Amplifier Background Audio professionals passionate about their compressors revere the LA-2A. The original was immediately acknowledged for its natural compression char- acteristics. A unique electro-optical attenuator system allows instantaneous gain reduction with no increase in harmonic distortion – an accomplishment at the time, still appreciated today.
LA-2A Controls Figure 51. The LA-2A plug-in window. Changes the characteristics of the compressor I/O curve. When set to Com- Limit/Compress press, the curve is more gentle, and presents a low compression ratio. When set to Limit, a higher compression ratio is used. Gain Adjusts the output level (by up to 40 dB).
CHAPTER 16- LA-3A Compressor Overview The original Teletronix LA-3A Audio Leveler made its debut at the 1969 New York AES show. Marking a departure from the tube design of the LA-2A Lev- eling Amplifier, the solid-state LA-3A offered a new sound in optical gain re- duction, with faster attack and release characteristics that were noticeably dif- ferent from its predecessor.
LA-3A Controls Background For detailed information about compressors, see “Compressor Basics” on page 152. This switch changes the characteristics of the compressor I/O curve. When Comp/Lim set to Compress, the curve is more gentle, and presents a low compression ra- tio.
CHAPTER 17 Lexicon 224 Classic Digital Reverb From the moment it was unleashed on the audio industry in 1978, the original Lexicon 224 Digital Reverb — with its tactile, slider-based controller and fa- mously lush reverb tail — almost single-handedly defined the sound of an en- tire era.
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Parameters Every tunable parameter from the original is present in the Lexicon 224 plug-in, and exposed as dedicated controls — inviting easy experimentation and sonic exploration. All seven algorithms/nine programs are available un- der the Program selection. Lexicon’s distinctive Bass/Mid “split decay” adjust- ments and Crossover control set the highly tunable reverb image, along with the Treble Decay for rolling off high frequencies.
Operational Overview Graphical User The original Lexicon 224 consists of two hardware elements. The “main- Interface frame” rack-mountable 4U chassis contains the power supply, circuitry, and audio input/output connectors. The remote control unit has a display, buttons, and sliders which control the 224 parameters and functionality. Some of these buttons and sliders have dual and even triple functionality, which makes using certain “buried”...
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Like the original hardware, UAD Lexicon 224 buttons are momentary-style Lexicon 224 Buttons and don’t latch in a down position. When a function is unavailable within a particular program, the button’s LED will not illuminate when clicked (the LEDs also don’t illuminate for the increment/decrement buttons). The first click of an increment/decrement button displays the current value of the parameter;...
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When configured as mono-in/mono-out (“MIMO”), output A is used exclu- sively except in programs 2, 4, and 9, where outputs A and C are summed into one monophonic signal. This implementation is recommended in the orig- inal hardware manual. If Rear Outs is enabled in MIMO mode, outputs B and D are used instead of A and C.
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Display Hold is set to 1.5 (the default value), after parameters are edited, the value displayed here reverts after 1.5 seconds to a reverb time which is related to the combined Bass and Mid slider values. This relationship is based on approximations designed by the original Lexicon engineers;...
Primary Controls Program The Program buttons (Figure 55) are used to specify which of the nine default Lexicon programs, and its associated algorithm, is active. See “Lexicon 224 Programs” on page 162 for an overview. Eight reverb programs and one chorus program are available. Click a reverb program button 1 –...
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Bass The Bass slider defines the reverb decay time for the frequencies below the Crossover value. Higher Bass values result in longer bass frequency decay times (when Crossover is not set too low). The Bass reverb decay time value, in seconds, is shown in the Numerical Display. The available range is 0.6 sec- onds to 70 seconds.
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Treble Decay Treble Decay sets a frequency above which decay is very rapid. Lower values will produce a “darker” reverb with less high frequency content. If Treble De- cay is set very low, then adjusting Bass, Mid, and Crossover may have little to no audible effect.
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Higher Diffusion values are frequently desirable when the material has a lot of percussion. Higher Diffusion can also contribute to a smoother-sounding re- verb. With low Diffusion values the early reverb will be “grainy” and sparse, but will produce a clear, bright sound that is very useful with strings, horns, and vocals.
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Important: When Immediate is off and a program is changed, previously modified parameter values are lost, unless the settings were saved as a preset or if the session file was previously saved so it can be recalled. This UAD-only control enables or disables the modeled inherent dy- System Noise namic system noise of the original Lexicon 224 hardware.
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Mode Mode Enhancement makes the sound of the Lexicon 224 programs more nat- Enhancement ural by preventing room modes from ringing in the reverb tail. Mode En- hancement works by continuously modulating certain delay lines (taps) within the program algorithms, which increases the effective density without thicken- ing the reverb itself.
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The Pitch Shift controls are accessed in the Hidden Controls panel. See page 173 for access details. Decay Optimization improves the Lexicon 224 reverb clarity and naturalness Decay Optimization by dynamically reducing reverb diffusion and coloration in response to input signal levels.
Solo When Solo is activated, the Dry/Wet mix is set to 100% wet and the Dry/Wet controls are deactivated. Solo mode is optimal when using Lexicon 224 in the “classic” reverb configuration (placed on an effect group/bus that is configured for use with channel sends). When Lexicon 224 is used on a channel insert, Solo should be deactivated.
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Access The hidden controls are exposed by click- ing the “OPEN” text to the right of the Dis- play Panel. Conversely, the exposed panel is closed by clicking the “CLOSE” text while the panel is open. Note: The last-used state of the Hidden Controls panel (open or closed) is re- tained when a new Lexicon 224 plug-in is instantiated.
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If the left and right controls have different values when link is inactive and Link is engaged, the left channel value is copied to the right channel (thereby over- writing the right channel value). When Link is active, automation data is written and read for the left channel only.
Program Descriptions P1 Small Concert This program emulates the sound of a small concert hall, with moderate initial Hall B density and moderately non-uniform decay. It is optimized for reverb times of 1.5 to 5 seconds (for longer decay times, P3 Large Concert Hall B is recom- mended instead).
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P8 Constant In naturally occurring reverb, new reflections are continuously added to the Density Plate A decaying sound over time. This sonic build-up increases density and colora- tion in the reverb tail. P8 Constant Density Plate A has high initial density and coloration (giving a “plate”...
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Table 14. Lexicon 224 Outputs Used With Monophonic Output Program Output(s) Program Output(s) 1. Small Concert Hall B 6. Small Concert Hall A 2. Vocal Plate A + C 7. Room A 3. Large Concert Hall B 8. Constant Density Plate A 4.
CHAPTER 18 Little Labs IBP Overview The Little Labs IBP Phase Alignment Tool easily eliminates the undesirable hol- low comb-filtered sound when combining out-of-phase and partially out-of-phase audio signals. Designed as a phase problem-solving device, the award-winning Little Labs IBP (“In-Between Phase”) has established itself with audio engineers as not only a “fix it”...
Delay Adjust The Delay Adjust parameter is unique to Universal Audio’s “workstation” ver- sion of the Little Labs IBP. Delay Adjust is a continuously variable control that simply delays the input signal from 0.0 to 4.0 milliseconds.
Phase Adjust This switch determines the range of the Phase Adjust parameter. This is useful 90°/180° when finer Phase Adjust resolution is desired. When the switch is disengaged, the Phase Adjust range is 180°. When the switch is engaged (darker), the Phase Adjust range is 90°. This switch sets the range of frequency emphasis.
CHAPTER 19 Little Labs VOG Bass Resonance Processor For many top engineers, the Little Labs VOG (Voice Of God) is the ultimate bass resonance tool for mixing. Available for the first time as a plug-in, the Lit- tle Labs-authenticated VOG for the UAD-2 platform accurately models the sonic characteristics of this unique 500-series hardware audio processor in every detail.
Little Labs VOG Screenshot Figure 58. The UAD Little Labs VOG plug-in window Operational Overview Two simple knobs allow you to dial in the VOG’s desired frequency and effect amplitude. The center of the sweepable frequency range is selected via two push-buttons of 40 Hz and 100 Hz, or you can set the center to 200 Hz by pressing both buttons simultaneously.
cies above remain intact. The higher the amplitude of the peak resonance frequency, the more you cut off the mud below, effectively performing two functions at once. A dedicated “flat” button allows you to quickly audition A/B comparisons. The VOG is intended for mixing, mastering, post-production sweetening, In Use sound design, and audio restoration.
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Like the original hardware, increasing the Frequency value (rotating clock- wise) actually lowers the target frequency (“increases the low end”). Changes to this setting are heard only if Amplitude is set above zero. Note: The control values for Frequency, which range from 0 – 10, are arbi- trary and do not reflect a particular frequency value.
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The Little Labs VOG hardware unit UAD Powered Plug-Ins Manual - 186 - Chapter 19: Little Labs VOG...
CHAPTER 20 Manley Massive Passive EQ Overview Universal Audio’s UAD Powered Plug-In versions of the Manley Massive Pas- sive EQs represent UA’s most ambitious and detailed EQ model to date. The two-channel, four-band Manley Massive Passive tube EQ utilizes design strengths from choice console, parametric, graphic, and Pultec EQs —...
Massive Passive Screenshots Figure 59. The Massive Passive plug-in window Figure 60. The Massive Passive Mastering plug-in window Unusual EQ Conventions The Massive Passive has design and operation characteristics that make it unique in the EQ world. Some of these factors mean the “Massivo” may not respond in a manner that you would expect from typical EQs.
Unique Shelves Most EQs offer a shelving mode for the edge bands only. Massive Passive of- fers the shelving option on all bands for expanded sonic possibilities, such as “staircase” EQ curves. One result of not using negative feedback loops in the design is that the gain No negative feedback loops control for a band cannot have a “bipolar”...
Standard vs. Mastering Versions The layout and function of the Massive Passive controls are essentially identi- cal for both the Standard and Mastering versions. The exact control differ- ences between the controls are detailed in Table 17 below. Table 17. Control differences between Massive Passive versions Standard Mastering Channel Gain Range...
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The Shelf/Bell toggle switch defines the shape of the filter band. A Shelf/Bell unique aspect of this control is that unlike other EQs where only the edge frequencies offer a shelving mode, with Massive Passive all bands can be used in either mode for expanded sonic possibilities. Note: The Bandwidth control (page...
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Gain has a fair amount of interaction with the Bandwidth control. The maxi- mum band gain is available in Shelf mode when Bandwidth is fully counter-clockwise; less band gain is available in Shelf mode as the Band- width is decreased (rotated clockwise). Conversely, the maximum gain is available in Bell mode when Bandwidth is fully clockwise;...
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As Bandwidth is increased in Shelf mode, a bell curve begins to be intro- duced in the opposite direction (i.e., overshoot). For example, if the Shelf is boosted, a dip is created at higher Bandwidth values. At maximum Band- width, this overshoot curve is pronounced. The effect of the Bandwidth control in Shelf mode is shown in Figure 62 below.
Frequency This control defines the center frequency (Bell mode) or edge frequency (Shelf mode) for the band. Each band provides a wide range of specially tuned overlapping and interleaving frequency choices. The available frequencies for each band are listed in Table 18 below.
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The Channel Gain controls are intended to help match levels between “By- pass” and “EQ enabled” modes so that the EQ effect can be more accurately judged. With drastic EQ there may not be enough range to match levels, but with drastic EQ this kind of comparison is of little use.
Figure 63. High Pass and Low Pass filter response curves (standard version) Mastering The Low Pass/High Pass filter frequencies in the mastering version are tuned Filters specifically for mastering, and the slopes are flatter until the knee. The slopes are 18 dB per octave on the mastering filters except for the highest value (52K) which is 30 dB/octave.
When set to Link (up position), modifying any channel one or channel two control causes its adjacent stereo counterpart control to snap to the same po- sition (channel 1 & 2 controls are ganged together in Link mode). When Link is active, automation data is written and read for channel one only.
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• You may also find yourself getting away with what seems like massive amounts of boost. Where the knobs end up, may seem scary particularly for mastering. Keep in mind that, even at maximum boost, a wide bell might only max out at 6 dB of boost (less for the lowest band) and only reaches 20 dB at the narrowest bandwidth.
Additional Information The original (and rather lengthy) user manual written by Manley Labs for the hardware unit contains a wealth of great information about the philosophy, design decisions, and use of the Massive Passive EQ. It is highly recom- mended reading for those interested in technical details. The manual can be found on their website, along with info about their other great products: •...
CHAPTER 21 Moog Multimode Filter Overview If UA were able to conceive a product with Moog, what would it be? The answer is revealed in the new UAD Moog Multimode Filter, which delivers the first truly analog-sounding VCF (voltage controlled filter) emulation made for mixing, performing, creating, or destroying.
Moog Filter Screenshot Figure 64. The UAD Moog Filter plug-in window Moog Filter Controls The Moog Filter is true stereo, with separate filters for the left and right chan- nels. The dual filters share the same controls. The only time the left and right filters diverge is when Filter Spacing or LFO Offset are not zero.
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Warning: Due to these differences in input structure, cut and pasting of full-to-SE and SE-to-full presets may cause noticeable differences in gain. Keep hold of the master fader! The Drive/Gain multicolor LED indicates the plug-in signal level Drive/Gain LED just after the Drive/Gain control. The Drive/Gain LED operates when the plug-in is in Bypass mode, but not when Power is off.
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Cutoff This parameter defines the cutoff frequency of both filter channels in all modes (lowpass, bandpass, highpass). UA has expanded the available frequency range of 20 Hz to 12 kHz on the MF-101 Moogerfooger to the broader avail- able range of 12 Hz to 12 kHz on the Moog Mulitmode Filter. In lowpass mode, frequencies above the cutoff are attenuated.
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Step/Track This switch is a smoothing control for the filter cutoff frequency parameter. Smoothing is most obvious on continuous filter sweeps when varying the cut- off rapidly with the knob or automation. Step mode can be desirable when sudden cutoff changes are automated and other creative purposes. Smoothing is on in the Track position, and off in the Step position.
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The LFO (low frequency oscillator) modulates the filter cutoff frequency. Sev- eral waveform shapes are available. The LFO can be synchronized to the tempo of the host (see Free/Sync below). Amount controls the depth of the LFO filter cutoff modulation. A higher value Amount will have a broader filter sweep.
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Offset Offset adjusts the polarity between LFO signals for the left and right channels. The available range is ±180 degrees. Offset can create great stereo spacial effects. When the filter is in Mono mode, both filters are still heard. Tip: Click the knob label (“OFFSET”) to return the value to zero.
Moog Filter SE Overview The UAD Moog Filter SE is derived from the UAD Moog Filter. Its algorithm has been revised (primarily the elimination of the Drive circuit) in order to pro- vide sonic characteristics very similar to the Moog Filter but with significantly less DSP usage.
Moog Filter Latency The Moog Filter (but not the Moog Filter SE) uses an internal upsampling tech- nique to facilitate its amazing sonic quality. This upsampling results in a slightly larger latency than other UAD plug-ins. See Chapter 9 “UAD Delay Compensation”...
CHAPTER 22 MXR Flanger/Doubler Classic Electronic Flanging For more than 30 years, musicians and engineers have relied upon the MXR Flanger/Doubler as one of best-sounding bucket-brigade flanging effects ever made. Through its signature flanging, doubling, and delay effects, the MXR Flanger/Doubler imprints a unique stamp on guitars, bass, keys, drums, or just about any source needing movement and depth.
MXR Flanger/Doubler Screenshot Figure 66. The UAD MXR Flanger/Doubler plug-in window Operational Overview Model 126 The MXR “Model 126” Flanger/Doubler is an analog delay processor that uses “bucket-brigade” technology to create short signal delays. The delay time can be modulated manually, or automatically with a low frequency os- cillator (LFO).
Software-Only The UAD MXR Flanger/Doubler plug-in has some features not included in the Features original hardware. The LFO rate can be synchronized to the tempo of the DAW session; the LFO can be reset; Stereo mode can apply processing to both sides of a stereo signal;...
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Flanger When in the “down” (gray) position, Flanger mode is active. This is the de- fault setting. Doubler Doubler mode is active when the button is in the “up” (white) position. This software-only switch modifies the processed signals at the out- Stereo Mode puts when used in a stereo-output configuration.
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This continuous control determines the delay time of the Manual processor. The delay time is modulated by the Sweep LFO when the Width value is higher than 0%. The available range of the control depends on the set- ting of the Effect button. In Flanger mode, the available delay time range is 4.9 milliseconds to 0.33 millisec- onds.
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This continuous control adjusts the blend between the original dry signal and the processed wet signal(s). The available range is 0 – 100%. When set to minimum, only the dry signal is heard. When set to maximum, the signal is almost entirely wet, however a small amount of dry signal is present (like the original hardware).
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Sync The speed of the Sweep LFO can be synchronized to the tempo of the host application by engaging the Sync button. Tempo Sync is engaged when the button is in the “down” (gray) position and the LED above the button is illuminated. See Chapter 8 “Tempo Sync”...
MXR Flanger/Doubler Latency This plug-in uses an internal upsampling technique. The upsampling results in a slightly larger latency than most other UAD plug-ins. See Chapter 9 “UAD Delay Compensation” in the UAD System Manual for more information. Note: Compensating for additional latency is not required if the host appli- cation supports full plug-in delay compensation throughout the signal path, or when it is used only on the outputs.
Modeling the 3-band EQ and high-pass filter in painstaking detail and thor- oughness, Universal Audio's Neve 1073 EQ will deliver the same sonic ex- perience expected from its analog cousin with exacting detail. Bundled to- gether come two versions: The 1073 EQ with absolute sonic accuracy, and the 1073SE EQ for high instance counts.
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Input Gain The Input Gain control sets the level at the input of the plug-in. The range is from –20 dB to +10 dB. When the Input Gain knob “snaps” to the OFF position, plug-in processing is disabled and UAD DSP usage is reduced (unless UAD-2 DSP LoadLock is enabled).
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The available midrange center frequencies are 360 Hz, 700 Hz, 1.6 kHz, 3.2 kHz, 4.8 kHz, 7.2 kHz, and OFF. When OFF is specified, the band is dis- abled. UAD CPU usage is not reduced when the band is OFF. The low frequency band is controlled by dual-concentric Low Band knobs, delivering smooth shelving equalization.
Phase The PHASE button inverts the polarity of the signal. When the switch is in the “In” (darker) position, the phase is inverted. Leave the switch “Out” (lighter) position for normal phase. The equalizer is engaged when the EQL switch is in the “In” (darker) position.
Neve 1073 Latency The Neve 1073 (but not the 1073SE) uses an internal upsampling technique to facilitate its amazing sonic quality. This upsampling results in a slightly larger latency than other UAD plug-ins. The latency, and its compensation, is identical to that of the UAD Precision Equalizer. See Chapter 9 “UAD Delay Compensation”...
The Rolling Stones to The Red Hot Chili Peppers. Universal Audio’s Neve 1081 EQ delivers the same sonic experience as its analog cousin with exacting detail. The 1081 EQ also includes a DSP opti- mized 1081SE EQ for higher instance counts.
Neve 1081 and 1081SE Controls Overview The Neve 1081 channel module is a four-band EQ with high and low cut fil- ters. The 1081 features two parametric midrange bands, with “Hi-Q” selec- tions for tighter boosts or cuts. Both the high and low shelf filters have select- able frequencies and may be switched to bell filters.
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High Band The high band delivers smooth high frequency shelving or peak equalization. The inner knob controls the band gain, and the outer ring selects the frequency or band disable. High Gain The equalization gain for the high band is selected with the inner knob of the dual-concentric control.
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High-Mid Frequency The high-midrange band frequency is selected with the outer ring of the dual- concentric knob controls. The ring control can be dragged with the mouse, or click directly on the “silkscreen” text to specify a frequency or disable the band.
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The available low-mid band center frequencies are 220 Hz, 270 Hz, 330 Hz, 390 Hz, 470 Hz, 560 Hz, 680 Hz, 820 Hz, 1000 Hz,1200 Hz, and OFF. When OFF is specified, the band is disabled. UAD CPU usage is not re- duced when the band is OFF.
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Cut Filters The independent low and high cut filters are controlled by the dual-concentric knobs to the right of the low band (see Figure 70 on page 223). The controls specify the fixed frequency of the cut filter. The cut filters have an 18 dB per octave slope.
Neve 1081SE Figure 71. The Neve 1081SE plug-in window Overview The UAD Neve 1081SE is derived from the UAD Neve 1081. Its algorithm has been revised in order to provide sonic characteristics very similar to the 1081 but with significantly less DSP usage. It is provided to allow 1081-like sound when DSP resources are limited.
N-Roses. Artists ranging from Primus and Metallica to My Morning Jacket and The Red Hot Chili Peppers have also called on the distinct tone of the Neve 8068 and 31102 EQ in the studio. Universal Audio’s officially licensed and endorsed Neve 31102 EQ plug-in delivers the same sonic experience as its analog cousin with exacting detail.
Neve 31102 and 31102SE Controls Input Gain The Input Gain control sets the level at the input of the plug- in, and doubles as a plug-in bypass control. The range is from –20 dB to +10 dB, and off. When the Input Gain knob “snaps” to the off position, plug-in processing is disabled and UAD DSP usage is re- duced.
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Midrange Band The midrange band is controlled by dual-concentric knobs, delivering smooth semi-parametric midrange equalization with a choice of two bandwidths. The inner knob controls the band gain, and the outer ring selects the frequency or band disable. Midrange Gain The equalization gain for the midrange band is selected with the inner knob of the dual-concentric control.
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Rotate the control clockwise to boost the selected low band frequency, or counter-clockwise to reduce the bass response. Low Frequency The low frequency is selected with the outer ring of the dual-concentric knob controls. The ring knob pointer can be dragged with the mouse, or click the shelving symbol above the knob to cycle through the available frequencies (shift+click to step back one frequency).
The equalizer is engaged when the EQL switch is in the “In” (darker) position. To disable the EQ, put the switch in the “Out” (lighter) posi- tion. Click the button to toggle the state. In the hardware 31102, the audio is still slightly colored even when the EQL switch is in the Out position.
Neve 31102 Latency The Neve 31102 (but not the 31102SE) uses an internal upsampling tech- nique to facilitate its amazing sonic quality. This upsampling results in a slightly larger latency than other UAD plug-ins. The latency and its compen- sation is identical to that of the other UAD Neve EQ’s. See Chapter 9 “UAD Delay Compensation”...
CHAPTER 26 Neve 33609 Compressor Overview Derived from the original Neve 2254 compressor, circa 1969, the 33609 stereo bus compressor/limiter utilizes a bridged-diode gain reduction circuit and many custom transformers. The uniquely musical character of this circuit made the 33609 a studio standard since its release. The UAD Neve 33609 is the only Neve-sanctioned software recreation of the Neve 33609 (revision C).
Neve 33609 Screenshot Figure 74. The Neve 33609 plug-in window Operation The UAD Neve 33609 is a two-channel device capable of running in stereo or dual-mono modes. The active mode is determined by the mono/stereo switch (see “Mono/Stereo” on page 240).
Technical Article The article “Ask the Doctors: Modeling of the Neve 33609 compressor/lim- iter” contains interesting technical details about the 33609. It is available at our online webzine: • http://www.uaudio.com/webzine/2006/august/index2.html Neve 33609 and 33609SE Controls Limiter Controls in this section only function when the limiter is enabled with the “limit In”...
Compressor Controls in this section only function when the compressor is enabled with the “compress In” switch (the Power switch must also be on). The compressor precedes the limiter (see “Signal Flow” on page 236). Note: Threshold determines how much compression will occur. When the input sig- Compressor Threshold nal exceeds the threshold level, the compressor engages.
Other Controls The interface elements that are not directly contained within the compressor or limiter are detailed below. This control is a software-only addition not found on the original hardware. It Output Gain is an overall makeup gain stage at the output of the plug-in to compensate for reduced levels as a result of compression and/or limiting.
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Link When set to link (down position), modifying any channel one or channel two control causes its adjacent stereo counterpart control to snap to the same po- sition (channel 1 & 2 controls are ganged together in link mode). When link is active, automation data is written and read for channel one only.
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• If you feed the same signal into both channels, you can have a lower thresh- old with a lower ratio on one channel, and a higher threshold with a higher ratio on the other channel. In this case, you will get a double knee, with the lower ratio being used between the knees, and the higher ratio above both knees.
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Headroom Switch The Headroom switch is provided to accommodate applications where high amounts of gain reduction are not desired. Headroom simply lowers the in- ternal operating level so that the plug-in is not “pushed” into gain reduction as much. Headroom can be set to 22 dB, 18 dB, or 14 dB. At 22 dB, signals will push the plug-in into gain reduction (and more non-linearity and “good”...
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Keep in mind there are no hard and fast rules. Use the above recom- Note: mendations as guidelines and feel free to experiment with the various posi- tions of the headroom switch regardless of the audio source. If it sounds good, use it! The UAD Neve 33609/33609SE includes a bank of factory presets.
Neve 33609SE Figure 76. The Neve 33609SE plug-in window Overview The UAD Neve 33609SE is derived from the UAD Neve 33609. Its algorithm has been revised in order to provide sonic characteristics very similar to the 33609 but with significantly less DSP usage. It is provided to allow 33609- like sound when DSP resources are limited.
CHAPTER 27 Neve 88RS Channel Strip Overview In 2001, Neve launched the 88 Series: A new, large-format analog console that represented the best of all Neve designs that came before it. Considered the ultimate console for modern features and reliability, it is also heralded as one of the best-sounding consoles ever made by veterans of both the audio and film communities.
Neve 88RS Controls Overview The UAD Neve 88RS controls are divided into four main sections: dynamics, EQ, cut filters, and global. Each section and control is detailed below. In the UAD Neve 88RS plug-in, 0 dBFS is calibrated to +4 dBU plus 18 dB of headroom, so 0 dBFS is equivalent to 22 dBU.
Gate/Expander The gate/expander module operates in either gate or expansion mode. In gate mode, signals below the threshold are attenuated by the range (RGE) amount (see Figure 79 on page 248), and hysteresis is available (see Figure 80 on page 249).
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Gate/Exp The Hysteresis knob sets the difference in threshold for signals Hysteresis that are either rising or falling in level. Signals that are rising (HYST) in level are passed when the level reaches the threshold value plus the hysteresis value. Signals that are falling in level are not passed at the lower threshold level.
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The available range is –25 dB to +15 dB. A range of –25 dB to –65 dB is available when the –40 dB switch is engaged (see “Gate/Exp Threshold –40 dB” on page 250). In typical use it’s best to set the threshold value to just above the noise floor of the desired signal (so the noise doesn’t pass when the desired signal is not present), but below the desired signal level (so the signal passes when present).
Slower release times can smooth the transition that occurs when the signal dips below the threshold, which is especially useful for material with frequent peaks. Fast release times are typically only suitable for certain types of percus- Note: sion and other instruments with very fast decays. Using fast settings on other sources may produce undesirable results.
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An example: When compressing a snare track with a standard compressor, if the snare hits are sparse, the compressor will release between each hit, so that each hit has a squashed sound. With the 88R compressor, distortion will be reduced, because the compressor will not come out of compression as much between the snare hits.
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As the Threshold control is increased and more processing occurs, out- Note: put level is typically reduced. Adjust the Gain control to modify the output of the module to compensate if desired. The –20 dB switch increases the sensitivity of the limiter/compres- L/C Threshold –20 dB sor by lowering the range of the available threshold values.
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Slower release times can smooth the transition that occurs when the signal dips below the threshold, which is especially useful for material with frequent peaks. However, if the release is too long, compression for sections of audio with loud signals may extend to sections of audio with lower signals. Fast release times are typically only suitable for certain types of percus- Note: sion and other instruments with very fast decays.
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88RS EQ Band Layout EQ module enable switch High Frequency (HF) band controls High Midrange Frequency (HMF) band controls Low Midrange Frequency (LMF) band controls Low Frequency (LF) band controls Figure 81. Neve 88RS EQ Controls Layout This button activates the equalizer module. The module is ac- EQ Enable (EQ) tive when the button is gray and the green indicator illumi- nates.
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HF Freq This parameter determines the HF band center frequency to be boosted or attenuated by the band Gain setting. The available range is 1.5 kHz to 18 kHz. This control determines the amount by which the frequency set- HF Gain ting for the HF band is boosted or attenuated.
LMF Freq This control determines the LMF band center frequency to be boosted or at- tenuated by the LMF Gain setting. The available range is 120 Hz to 2 kHz. This control determines the amount by which the frequency setting for the LMF LMF Gain band is boosted or attenuated.
Cut Enable This button activates the cut filter. The cut filter is active when the button is gray and the red indicator illuminates. Cut Frequency This knob determines the cutoff frequency for the cut filter. The available range is 7.5 kHz to 18 kHz for the high cut fil- ter (lighter blue control), and 31.5 Hz to 315 Hz for the low cut filter (darker blue control).
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Power The Power switch determines whether the plug-in is active. This is useful for comparing the processed settings to the original signal or bypassing the plug-in to reduce the UAD DSP load (load is not reduced if UAD-2 DSP LoadLock is enabled). Toggle the switch to change the Power state;...
CHAPTER 28 Precision Buss Compressor Overview The Precision Buss Compressor is a dual-VCA-type dynamic processor that yields modern, transparent gain reduction characteristics. It is specifically designed to “glue” mix elements together for that cohesive and polished sound typical of master section console compressors. A flexible and intuitive tool, the Precision Buss Compressor is intended primarily for controlling the final output of your mix, but can be usefully applied to a variety of sources from drum busses or overheads to vocal groups, or even as a channel...
Precision Buss Compressor Screenshot Figure 82. The Precision Buss Compressor plug-in window Precision Buss Compressor Controls Control knobs for the Precision Buss Compressor behave the same way as with all UAD plug-ins. Parameters with text values can be modified with text entry.
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When Ratio is changed, the Threshold value is updated accordingly: When Ratio is set to 2:1, the Threshold range is –55 dB to 0 dB. When Ratio is set to 4:1, the Threshold range is –45 dB to +10 dB. When Ratio is set to 10:1, the Threshold range is –40 dB to +15 dB.
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Slower release times can smooth the transition that occurs when the signal dips below the threshold, especially useful for material with frequent peaks. However, if you set too large of a Release time, compression for sections of audio with loud signals may extend to lengthy sections of audio with lower signals.
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Toggling the Fade switch causes an already active fade to reverse direction, without a jump in output level. The Fade Set rate is constant even if an active fade is interrupted. For example: If the Fade Set value is 30 seconds and a fade out is initiated, then Fade is clicked again after 20 seconds, it will take 20 seconds to fade back in.
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Gain Reduction The Gain Reduction meter dis- Meter plays the amount of gain re- duction occurring within the compressor. More blue bars moving to the left indicate more gain reduction is occurring. The meter range is from –16 dB to 0 dB. Signal peaks are held for 3 seconds before resetting.
CHAPTER 29 Precision De-Esser Overview The Precision De-Esser seamlessly and accurately removes sibilance from in- dividual audio tracks or even composite mixes via its intuitive interface and sophisticated yet transparent filter processing. The Threshold knob dials in the amount of sibilance reduction, while the two- position “Speed”...
Precision De-Esser Controls Control knobs for the Precision De-Esser behave the same way as all UAD plug-ins. Threshold, Frequency, and Width values can be modified with text entry. Threshold controls the amount of de-essing by defining Threshold the signal level at which the processor is activated. Ro- tate Threshold counter-clockwise for more de-essing.
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When Solo is active, changes to the Threshold and Split controls cannot Note: be heard. Width controls the bandwidth of the de-essing Width sidechain when in bandpass mode. Bandpass mode is active when the control is in any position except fully clockwise.
Gain Reduction The Gain Reduction meter provides a visual indica- tion of how much attenuation (compression) is oc- curing. Signal peaks are held for 3 seconds before resetting. When Split is on, the amount of sidechain attenuation is displayed. When Split is off, it displays the attenuation of the entire signal.
These harmonics stimulate a psychoa- coustic bass-enhancing effect in the listener, giving even the smallest speakers greater translation of low frequency sources. Universal Audio’s unique ap- proach to this common problem combines a simple control set that yields ex- acting results with minimal adjustment and allows the widest range of tonality available in its class, from subtle to decidedly audible.
Precision Enhancer Hz Controls Control knobs for the Precision Enhancer Hz behave the same way as with all UAD plug-ins. Effect, Hz Frequency, and Output values can be modified with text entry. The Effect Knob controls the amount of pro- Effect Knob cessing that occurs in the plug-in.
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Mode B (Bass 2) Mode B is primarily for electric and DI bass with balanced mid range harmonics to help the bass stick out of the mix. Mode C (Synth) Mode C is tuned specifically for synth bass and other full-range material. It produces a wider range of harmonics than the Bass modes A and B.
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Hz Frequency Through filter isolation of the original bass content, the Hz Frequency parameter defines the cutoff fre- quency for the enhancement process. Frequencies be- low this value are enhanced by the processor. The available range is 16 Hz to 320 Hz. Hz Solo isolates the original bass signal and can be combined with Ef- Hz Solo fect Solo.
Precision Enhancer Hz Usage Notes • The Precision Enhancer Hz effect can serve multiple purposes. When the fre- quency control is set low, the effect extends into the audible low end. Lower frequencies work well for adding a low end thump or beefing up percussive bass/kicks, but be careful not to overdo it.
CHAPTER 31 Precision Enhancer kHz Overview The Precision Enhancer kHz is a sophisticated tool with a simple control set, primarily designed to bring dull or poorly recorded tracks to life. However, with five distinct Enhancement “Modes”, the Precision Enhancer kHz will find uses on virtually any source.
Precision Enhancer kHz Screenshot Figure 85. The Precision Enhancer kHz plug-in window Precision Enhancer kHz Controls Control knobs for the Precision Enhancer kHz behave the same way as with all UAD plug-ins. Effect, kHz Frequency, and Output values can be modified with text entry.
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Mode C Mode C dynamically enhances the high frequency content. The enhancement amount is increased as the input signal level increases. Mode D Mode D dynamically enhances both high and low frequency content. The en- hancement amount is increased as the input signal level increases. The kHz Frequency parameter is disabled in this mode.
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Generally speaking, adjust the Output control after the desired amount of pro- cessing is achieved with the Effect and kHz Frequency controls. Output does not affect the amount of enhancement processing, nor does it have any effect when the plug-in is disabled. Output Meter The Output Meter displays the signal level at the output of the plug-in.
CHAPTER 32 Precision Equalizer Overview The Universal Audio Precision Equalizer™ is a stereo or dual-mono four band EQ and high-pass filter designed primarily for mastering program material. The Precision Equalizer may also be used in recording and mixing where the utmost in EQ quality is required.
Precision Equalizer Controls The easy to use Precision Equalizer features stepped controls throughout for easy recall. Both the left and right channels feature four bands of EQ, grouped in two overlapping pairs. There are two bands for low frequencies (L1 and L2), and two for highs (H1 and H2). There is also a shelving or peak/notch filter available for each band, along with five peak/notch (Q) re- sponses per band.
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Dual Mode In Dual mode (dual-mono mode), the left and right parameters can be inde- pendently adjusted so that each side of the stereo signal can have different EQ settings. Note that this mode is infrequently used during mastering be- cause phase, imaging, and level inconsistencies may be induced in the result- ing stereo signal.
Power Switch The Power Switch determines whether the plug-in is active. This is useful for comparing the processed settings to the original signal, or to bypass the plug-in to reduce the UAD DSP load (load is not reduced if UAD-2 DSP LoadLock is enabled).
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Frequency Knob The Frequency knob determines the center frequency of the filter band to be boosted or attenuated by the band Gain setting. This knob is stepped with 41 values for easy reproducibility during mastering. To double the resolution of the available knob values (for fine control), press the shift key on the computer keyboard while adjusting the knob.
Precision Equalizer Latency The Precision Equalizer uses an internal sample rate of 192 kHz to facilitate its amazing sonic quality. This upsampling results in a slightly larger latency than other UAD plug-ins. See Chapter 9 “UAD Delay Compensation” in the UAD System Manual for more information.
Universal Audio’s Precision K-Stereo™ Ambience Recovery plug-in is a psy- choacoustic processor conceived and created by famed mastering engineer Bob Katz. Collaborating with Universal Audio to bring this unique, patented process to the UAD platform, Bob Katz’s K-Stereo process uses elements of the Haas effect and other psychoacoustic principles to create a transparent, phase-accurate “ambience recovery”...
Operational Overview Ambience The primary function of Precision K-Stereo is for ambience recovery and en- Recovery hancement. The plug-in doesn’t add new ambience or change the balance of the mix. Instead, it extracts, recovers, polishes, and embellishes the ambience that already exists in a source recording. Precision K-Stereo does not use any comb filters, matrixing, phase processing, or related techniques which are used in typical image processors.
Configurations Precision K-Stereo is optimized for use in stereo-in/stereo-out configurations. However, ambience recovery is possible on monophonic source recordings when the plug-in is used in a mono-in/stereo-out configuration. Mono signals are “stereo-ized” in the MISO context. Note: Due to the inherent stereo nature of the ambience recovery process, Precision K-Stereo is not intended for use in mono-in/mono-out configurations.
Ambience Recovery Refer to Figure 88 below for the controls in this section. Figure 88. The Ambience Recovery controls Recover Enable This switch enables/disables the ambience recovery process. See “Ambience Recovery” on page 286 for an overview. Recover Enable must be engaged for the Ambience Gain, Enhance Deep/Wide, and Ambience Filters controls to have an effect.
Ambience Filters The Ambience Filters (EQ) provide for frequency adjustments to the ambient portion of the signal. These controls do not affect the direct (dry) portion of the signal. The Ambience Filters consist of one Low Cut filter, one High Cut filter, and a single-band parametric bell filter.
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Bell Filter The bell filter is fully parametric, with independent control of frequency, gain, Q (bandwidth), and gain. Bell Frequency This control sets the center frequency of the bell filter. The available range is 150 Hz to 10 kHz. Bell Q Bell Q determines the bandwidth of the bell filter.
Mid/Side Controls The Mid/Side controls allow for level adjustments to the middle (center) and side portions of signals within a stereo field. Mid/Side adjustments can be made when ambience recovery is active or disabled. Note that the ambience recovery process does not use or require mid/side techniques to achieve am- bience recovery or enhancement.
Output Gain Controls Refer to Figure 91 below for the controls in this section. Figure 91. The Output Gain controls This switch enables/disables the Left/Right Gain and Link controls. L/R Gain Enable Link This switch links (gangs) the Left/Right Gain controls for ease of operation when both channels require the same value.
Output Level Meters The stereo peak/hold meters display the signal level at the output of the plug- in. The meter range is from –30 dB to 0 dBFS. Signal peaks are held for 3 sec- onds before resetting. Power When the Power switch is in the OFF position, the interface elements do not illuminate, plug-in processing is disabled, and UAD DSP usage is reduced (unless UAD-2 LoadLock is enabled).
Factory Preset Notes by Bob Katz Precision K-Stereo includes factory presets designed by Bob Katz for use with his signature plug-in. Descriptions about these presets are listed below. The Default preset and presets with the “BK” prefix are the settings created by Mr.
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BK-MSTR-Full Same as Full Orchestra A, but the high cut filter is set to 10 kHz which softens Orchestra B the high frequency ambience to tighten the percussion. If the hall or chamber in the original recording is missing some presence, BK-MSTR-Clear Presence here’s a suggestion on what to do.
CHAPTER 34 Precision Limiter Overview The Universal Audio Precision Limiter™ is a single-band, look-ahead, brick- wall limiter designed primarily for mastering with program material. The easy- to-use Limiter achieves 100% attack within a 1.5ms look-ahead window, which prevents clipping and guarantees zero overshoot performance. Both the attack and release curves are optimized for mastering, which minimizes aliasing.
Precision Limiter Screenshot Figure 92. The Precision Limiter plug-in window Controls Overview Control knobs for the Precision Limiter behave the same way as all UAD plug- ins. Input, Output, and Release values can be modified with text entry. The Precision Limiter introduced a new control style for UAD plug-ins. For the Mode, Meter, Scale, and Clear parameters, click the parameter label, the value text, or the LED to toggle between available values.
Release The Release knob sets the value of the limiter release time. The default value is Auto. The available range is from 1 second to 0.01 milliseconds. Auto Mode When the Release knob is fully clockwise, Automatic mode is active. In Auto mode, release time is program-dependent.
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with 20-20 kHz pink noise on an SPL meter set to C-weighted slow (i.e. aver- age) response. It is this calibrated meter/monitor relationship that establishes a consistent average “perceived loudness” with reference to 0 dB on the meter. Sliding Meter Scale With the K-System, programs with different amounts of dynamic range and headroom can be produced by using a loudness meter with a sliding scale, because the moveable 0 dB point is always tied to the same calibrated mon-...
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Each of these modes displays the The RMS and instantaneous peak levels, “Meter Response” on which follow the signal, and the peak-hold level (see page 301). PK-RMS K-20 K-14 K-12 Figure 93. Precision Limiter Meter Types K-20 K-20 mode displays 0 dB at –20 dB below full scale. K-20 is intended for ma- terial with very wide dynamic range, such as symphonic music and mixing for film for theatre.
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Note: When the meters are in the K-modes, the displayed RMS level is 3.01 dB higher when compared to the same signal level in the Peak-RMS mode. This is done to conform to the AES-17 specification, so that peak and average measurements are referenced to the same decibel value with sine waves.
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Figure 94. Precision Limiter meter scale in PK-RMS Zoom mode The main level meters in Normal mode, and the gain reduction meter in both Normal and Zoom modes, are linear (level differences between LED segments is the same). In PK-RMS and K-20 Zoom modes however, the main level meters use two different linear ranges for increased accuracy.
Hold The meter Hold Time switch determines how much time will pass before the peak values for the main meter and the gain reduction meter are reset. It af- fects both the peak LED’s and the peak text display. Values of 3 seconds, 10 seconds, or Infinite (indicated by the lazy-8 symbol) can be selected.
CHAPTER 35 Precision Maximizer Overview The Precision Maximizer is a dynamic impact processor that uniquely en- hances the apparent loudness, warmth, and presence of individual tracks or program material without appreciably reducing dynamic range or peak level control. Significant audio improvements can be achieved without the fatigu- ing artifacts typically associated with traditional dynamic processors.
Precision Maximizer Screenshot Figure 95. The Precision Maximizer plug-in window Precision Maximizer Controls Control knobs for the Precision Maximizer behave the same way as all UAD plug-ins. Input, Shape, Mix, and Output values can be modified with text en- try. The stereo peak Input Meter displays the signal level at the input Input Meter of the processor, after the Input control.
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Shape The Shape knob is the primary saturation control for the Maximizer effect. It contours the harmonic content and ap- parent dynamic range of the processor by changing the small-signal gain of the saturator. The available range is 0–100%. At lower settings, apparent loudness is not as dramatic but harmonic processing still occurs, producing a richer sound with minimal reduction of dynamic range.
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The crossover frequencies in three-band mode are 200 Hz and 2.45 kHz. Click the Bands button to change the mode. Alternately, you can click+hold the LED area and drag like a slider to change the value. UAD DSP usage is increased when three-band mode is active (unless Note: UAD-2 DSP LoadLock is enabled).
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Output The Output knob controls the signal level that is output from the plug-in. The available range is –12 dB to 0 dB. Note that when Limit is not engaged, it is possible for the output level to exceed 0 dB. In this case, Output can be lowered to eliminate any associated clipping.
Operating Tips • As a starting point for general loudness enhancement, set Precision Maxi- mizer to one-band mode with Limit engaged, with Mix at 100% and Shape at 50%. Then set Input so signals peak at around 0 dB on the Input Meters. These settings offer good results under most conditions, producing more presence with a warmer sound and enhanced detail (especially with lower frequencies), while retaining the apparent dynamic range of the original...
CHAPTER 36 Precision Multiband Overview The Precision Multiband is a specialized mastering tool that provides five spectral bands of dynamic range control. Compression, expansion or gate can be chosen separately for each of the five bands. The unparalleled flexi- bility and easy to follow graphical design of the Precision Multiband make it the ideal tool for the novice as well as the seasoned mastering engineer.
Precision Multiband Interface The Precision Multiband interface is designed to make this complex processor easier to use. Five separate frequency bands are available for processing. Each band is identified by a unique color, and all controls specific to the band have the same color.
Band Controls The Band Controls contain the parameters that are used to specify all the settings for each band (except the frequencies; see “Frequency Controls” on page 319). The Band Controls for each of the five bands are identical. Only one set of Band Controls is displayed at a time.
EQ Display Selection A band can also be selected by clicking within the area of the band in the EQ Display. For example, clicking within the area shown here will se- lect the LMF band. Band Parameters Because the Band Controls for each of the five bands are identical, they are only described once.
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Relative mode is not available for the Type parameter because the Note: available Type values are discrete. Click and shift-click both activate Absolute mode for Type. Absolute Link In Absolute mode, changes to a band control will force the same control in the other bands to snap to the same value as the current band.
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COMPRESS When a band is set to Compress, the dynamic range of the band will be re- duced (dependent upon the band threshold and input level). This is the typical value in multiband compression. EXPAND When a band is set to Expand, the dynamic range of the band will be in- creased (dependent upon the band threshold and input level).
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Attack Attack sets the amount of time that must elapse once the input signal reaches the Threshold level before processing is applied. The faster the Attack, the more rapidly processing is applied to signals above the threshold. The available range is 50 microseconds to 100 milliseconds. Release sets the amount of time it takes for processing to cease once the input Release signal drops below the threshold level.
EQ Display In the EQ Display, the entire audio spectrum from 20 Hz to 20 kHz is dis- played along the horizontal axis. Gain and attenuation of the five band fre- quencies (up to ±12 dB) are displayed along the vertical axis. Figure 97.
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Adjusting Gain The gain of a band can be adjusted by click-drag- ging the top of its colored line. In this case the cursor changes to an up/down arrow when hovered over the hot spot to indicate the direction available for dragging.
Frequency Controls The crossover frequency (xF) between the bands and the center frequency (cF) of the Mid bands is shown at the bottom of the EQ Display (see “EQ Display” on page 317). The frequencies for each band can be modified by entering the values directly and by manipulating the colored band curves.
Dynamics Meters Realtime display of Precision Multiband dy- namics processing is shown in the Dynamics Meters. This area also contains the band en- able and band solo controls. There is one vertical dynamics meter for each band. They are color coded to match the bands, and represent (from left to right) the LF, LMF, MF, HMF, and HF bands respectively.
The band is soloed when its Solo button is red. Click the button to toggle the solo state of the band. Soloing bands does not reduce UAD CPU usage. When a band is in Solo mode, its curve Solo Display in the EQ Display is highlighted.
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Output Level The stereo Meter peak/hold Out- put Meter dis- plays the signal level at the output of the plug-in. Signal peaks are held for 3 seconds before resetting. Output Level The Output Level knob controls the signal level that is output from the plug-in. Knob The default value is 0 dB.
Power Switch The Power Switch determines whether the plug-in is active. Click the toggle button or the UA logo to change the state. When the Power switch is in the Off position, plug-in processing is disabled and UAD DSP usage is reduced (unless UAD-2 DSP LoadLock is enabled). When the plug-in is bypassed with this switch (but not by the host bypass), the I/O meters and the Input Level knob remain active.
CHAPTER 37 Pultec and Pultec-Pro Overview The Pultec EQP-1A Program Equalizer and Pultec MEQ-5 plug-ins are faithful electronic reproductions of the classic hardware equalizers. Our DSP wizards have ensured that every revered sonic nuance of these vintage processors are faithfully maintained. UAD Pultec and UAD Pultec-Pro The UAD Pultec plug-in is the EQP-1A Program Equalizer that was introduced in version 2.2 to much acclaim.
Pultec Latency The Pultec and Pultec-Pro plug-ins introduce an additional 13 samples of de- lay due to upsampling when the session sample rate is below 100 kHz. This additional latency does not occur at sample rates above 100 kHz. You may enter a value of 13 in the “Samples”...
Figure 99. Control grouping within the Pultec EQP-1A In/Out Toggle This is a signal bypass control. It allows you to compare the processed and un- Switch processed signal. It does NOT reduce UAD DSP load. In the hardware EQP-1A, the audio is still slightly colored even when the switch is in the Out position.
In the documentation supplied with hardware version of the EQP-1A, it Note: is recommended that both Boost and Attenuation not be applied simulta- neously because in theory, they would cancel each other out. In actual use however, the Boost control has slightly higher gain than the Attenuation has cut, and the frequencies they affect are slightly different too.
Pultec MEQ-5 Screenshot Figure 100. The Pultec-Pro MEQ-5 Midrange Equalizer plug-in window Pultec MEQ-5 Controls The MEQ-5 can control three frequency ranges simultaneously, using three groups of interacting parameters. The first group controls the low–mid frequencies and has two controls: fre- quency select and boost.
In the hardware MEQ-5, the audio is still slightly colored even when the switch is in the Out position and the peak/dip controls are at zero. This is due to the fact that the signal is still passing through its circuitry. Because the plug- in emulates the hardware in every regard, the signal will be slightly processed when this switch is in the In position and the peak/dip controls are at zero.
CHAPTER 38 RealVerb Pro Overview RealVerb Pro uses complex spatial and spectral reverberation technology to accurately model an acoustic space. What that gets you is a great sounding reverb with the ability to customize a virtual room and pan within the stereo spectrum.
RealVerb Pro Background Direct Path Source Wet/Dry Input Early Delay Reflections Gain Output Gain & Pans & Mute Distance Late- Field Delay Reverb Figure 105. RealVerb Pro signal flow Figure 105 illustrates the signal flow for RealVerb Pro. The input signal is equalized and applied to the early reflection generator and the late-field re- verberation unit.
Figure 106). Re- The RealVerb Pro user interface is similarly organized (see flected energy equalization is controlled with the Resonance panel. The pat- tern of early reflections (their relative timing and amplitudes) is determined by the room shapes and sizes in the Shape panel; early reflection predelay and overall energy is specified at the top of the Timing panel.
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To configure the room shape and size: Select a room shape from the first (left) pop-up menu. The selected shape appears in the left side of the Shape circle. Adjust the room size with the top horizontal slider. Select a room shape from the second (right) pop-up menu. The selected shape appears in the right side of the Shape circle.
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Second material First material Blending bar First material Second material selector pop- selector pop-up up menu menu First material Thickness Second material control Thickness control Figure 108. RealVerb Pro Material panel Note: While materials are used to control decay rates as a function of fre- quency, the overall decay rate of the late-field reverberation is controlled from Figure 110 on page 342).
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Blend the absorption properties of the two materials by dragging the Blend- ing bar. The relative amount of each material, expressed as a percentage, appears above their respective pop-up menu. Drag the Blending bar to the right to emphasize the first material, and drag it to the left to emphasize the second material.
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tion frequency, the frequency at which the decay rate is halfway between the low-frequency and high-frequency values. At 100% thickness, the ratio of low- frequency to high-frequency decay times is 10:1. This means that the high fre- quencies will decay 10 times faster than the low frequencies. At 200% thick- ness, this is multiplied by two (high frequencies decay at 20x the rate of the low frequencies).
Resonance (Equalization) The Resonance panel has a three-band parametric equalizer that can control the overall frequency response of the reverb, affecting its perceived brilliance and warmth. By adjusting its Amplitude and Band-edge controls, the equal- izer can be configured as shelf or parametric EQs, as well as hybrids be- tween the two.
Adjust the Band-edge controls for the second and third bands so they are adjacent to each other. To raise the frequency for the high-shelf, drag to the right with the Band-edge control for the second band. To lower the fre- quency for the high-shelf, drag to the left with the Band-edge control for the third band.
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Early Amplitude Reflections control display Predelay control Late-Field Amplitude Reverbera- Control tions display Predelay control Diffusion control Decay Time control Figure 110. RealVerb Pro Timing panel To adjust the timing of the early reflections: Drag the Amplitude control for the early reflections up or down (from –80 dB to 0 dB) to affect the energy of the reflections.
Drag the Decay Time control for the late-field reverberations left or right (from 0.10–96.00 seconds) to affect the length of the reverb tail. The De- cay Time is indicated in the text field at the bottom of the Timing panel. To affect how quickly the late-field reverberations become more dense, ad- just the Diffusion control at the right of Late Reflection display in the Timing panel.
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To pan the direct (dry) signal: Drag the Direct slider left or right. A value of <100 pans the signal hard left; a value of 100> pans the signal hard right. A value of <0> places the signal in the center of the stereo field. Set the positioning for the early reflection or late-field reverberation with any of the following methods: Drag the left and right slider handles to adjust the stereo width.
Levels The Levels panel adjusts the Input Gain and Output Gain for RealVerb Pro. These levels are adjusted by dragging the sliders to the desired values. You can mute the input signal by clicking the Mute button. Figure 112. RealVerb Pro Levels panel Morphing All RealVerb Pro controls vary continuously using proprietary technology to smoothly transition between selected values.
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(Figure 114 on page 346), non user-adjustable con When in Morphing mode trols will change their appearance and will no longer be accessible. When in serted on a Send effect, the ‘W’ button automatically turns on (to keep the mix at 100% wet).
RealVerb Pro Preset Management Factory Presets In the preset menu there are thirty factory presets that can be changed by the user. Any modification to a preset will be saved even if you change presets. If you want to return all the presets to their default settings, select “Reset all to Defaults”...
Its unmistakable warm analog stereo chorus and vibrato have been heard on countless tracks; particularly on guitars, bass and electric keys. Universal Audio has been commissioned by Roland to accurately model the CE-1, and the results are nothing short of spectacular.
Boss CE-1 Controls The Boss CE-1 has two operating modes, chorus and vibrato. Only one mode can be active at a time. The operating mode is set using the Vibrato/Chorus switch. The red Clip LED illuminates when signal peaks in the plug-in occur. Clip LED This is an effect bypass switch.
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The hardware CE-1 has only a monophonic input. Its output can be mono (wet and dry signal mixed at one output jack) or stereo (dry signal in one output jack, wet signal in other output jack). We’ve adapted the model for the mod- ern era, enabling a true stereo input.
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Vibrato Controls These two knobs control rate and depth of the vibrato effect when CE-1 is in vibrato mode. Depth Knob The depth knob controls the intensity of the vibrato ef- fect. Rate Knob The rate knob controls the rate of the vibrato LFO. The rate is indicated by the the Rate LED indicator.
This classic 1979 Roland device has been heard on countless records, from luminaries such as Peter Gabriel, Talking Heads and INXS. Entrusted by the Roland company to emulate this classic studio tool, Universal Audio went to great lengths to preserve this Bucket Brigade chorus with all its unique design elements and sonic characteristics.
Roland Dimension D Controls The Roland Dimension D is very simple device to operate; it has only three controls: Power, Mono, and Mode. Each control is detailed below. The Dimension Mode determines the effect inten- Dimension Mode sity. Four different modes are available. Mode 1 is the most subtle effect, and Mode 4 is maximum intensity.
Echo can be heard on numerous recordings, from 70’s space rock like Pink Floyd and David Bowie, to countless Reggae and Dub albums, to more recent bands like Portishead and Radiohead. Universal Audio has been entrusted by Roland to re-create the highly beloved RE-201 unit, considered the best of the Space Echo line.
Roland RE-201 Screenshot Figure 117. The Roland RE-201 plug-in window Roland RE-201 Interface The RE-201 interface is true to the original hardware, with a few customiza- tions to bring it into the digital era. The original mic and instrument volume controls have been replaced with echo/reverb pan controls and an input control.
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The VU is essentially an input meter, therefore it doesn’t react when the Echo/Normal switch is switched from Echo to Normal. The Peak lamp and VU meter measure signal just after the input volume Note: control. However, like the original hardware, echo intensity (feedback) is ap- plied just before the level detection circuit.
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The affect of each knob position is detailed in Table 25 on page 357. Table 25. RE-201 Mode Selector Positions Mode REPEAT REVERB + ECHO REVERB Knob (echo only) ONLY Position Reverb Active • • • • • Tape • •...
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Repeat Rate This knob controls the time interval of the echo effect. Rotating the control clockwise will decrease the delay time, and counter- clockwise rotation will increase the delay time. The available delay times are as follows: • Head 1: 69ms – 177ms •...
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Echo Volume has no affect when the Mode Selector is in the “Reverb Note: Only” position. This switch determines whether the plug-in is active. This is use- Power Switch ful for comparing the processed settings to the original signal, or to bypass the plug-in to reduce the UAD DSP load. Toggle the switch to change the Power state.
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Splice Normally, the splice on the tape loop comes around at regular in- tervals. This interval varies, and is determined by the selected Re- peat Rate. Depending on what Tape Quality is selected, the splice can be subtle or obvious, and can work as a catalyst for chaos es- pecially when the RE-201 is in a state of self-oscillation.
CHAPTER 42 SPL Transient Designer Overview Universal Audio has partnered with German company Sound Performance Lab (SPL) to bring you the Transient Designer, with its unique and compelling Differential Envelope Technology for shaping the dynamic response of a sound. Only two simple audio controls are required to allow you to effortlessly reshape the attack and sustain characteristics.
SPL Transient Designer Controls Containing only two primary controls, the UAD SPL Transient Designer is ex- tremely simple to operate. The technology behind the processor isn't as im- portant as how it sounds. However, for those who desire a deeper under- standing of the process, a deeper explanation of the underlying technology is presented at the end of this chapter (see “Technology”...
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Signal This 4-stage “LED” indicates the presence of audio signals at the input of the plug-in. When the input signal is below –25 dB, the indicator is off. At –25 dB to –19 dB, the indicator glows slightly. At –18 dB to –10 dB, it lights with medium intensity. At –9 dB to 0 dB, it shines brightly.
In addition to creating an amazing piece of hardware, Sound Performance Lab also wrote an extensive user manual for the Transient Designer. Because Universal Audio has full license to make use of the Transient Designer technol- ogy, SPL has graciously authorized us to use their documentation as well.
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• Shorten the sustain period of a snare or a reverb tail in a very musical way to obtain more transparency in the mix. • When recording a live drum set, shorten the toms or overheads without physically damping them. Usual efforts to damp and mike are reduced re- markably.
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Guitars Use the Transient Designer on guitars to soften the sound by lowering the AT- TACK. Increase ATTACK for in-the-face sounds, which is very useful and works particularly well for picking guitars. Or blow life and juice into quietly played guitar parts. Distorted guitars usually are very compressed, thus not very dynamic.
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channels of the reverb return through two separate Transient Designer in- stances. Turn the ATTACK fully right on one instance and reduce SUSTAIN slightly (about –1.5 dB). On the other instance turn the ATTACK fully left and the SUSTAIN to the 3-o‘clock position (about +12 dB). These settings preserve the original complexity of the reflections in the reverb but the maximum intensity of the effect will move from the left to the right in the mix while the reverb will maintain it‘s presence in both channels.
Technology Of course you don‘t have to know how the Transient Designer works in order to use it. However, since it offers a completely novel signal processing, noth- ing shall be concealed from the more curious users. SPL’s DET is capable of level-independent envelope processing and thus Differential Envelope makes any threshold settings unnecessary.
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Figure 120 on page 369 shows the difference between Env 1 and Env 2 that defines the control voltage of the VCA. The shaded area marks the difference between Env 1 and Env 2 that controls the control voltage of the VCA. The am- plitude of the attack is increased if positive ATTACK values are set.
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The SUSTAIN The SUSTAIN control circuitry also plays host to two envelope generators. The Control Circuitry envelope tracker Env 3 again follows the original waveform. The envelope generator Env 4 maintains the level of the sustain on the peak-level over a longer period of time.
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Figure 124 on page 371 displays the processed waveforms with maximum and minimal sustain to compare against the original waveform in diagram 4. Figure 124. SPL Transient Designer Processed Sustain SPL Sound Performance Lab® and Transient Designer® are registered trademarks of SPL Elec- tronics, GmbH Germany and are used under license.
CHAPTER 43 SSL E Channel Strip Large Format Mix Module The SSL 4000 is famous as the console employed on more Platinum-selling records than any other. With its wide range of VCA compression character- istics and intuitive EQ — rich with colorful band interdependencies — it’s easy to hear why.
SSL E Channel Strip Screenshot Figure 125. The UAD SSL E and 4K Channel Strip plug-in windows SSL E Channel Strip Controls The SSL E Channel Strip controls are divided into four main sections: filters, dynamics, EQ, and global. Note: Knob settings, when compared to the graphical user interface silk- screen numbers, may not match the actual parameter values.
Filters In addition to the four-band EQ, UAD SSL E Channel Strip offers individual high and low pass filters. When the Filter control is at minimum value (fully counter-clockwise), the filter is disabled. The control ranges and sonics of these filters can be changed between “Black”...
Dynamics Separate “soft-knee” compressor/limiter and expansion/gate modules are available in the dynamics section. Each module has their own set of controls. Important: Dynamics are not processed unless enabled by the Dynamics selector buttons (“Dynamics In (DYN IN)” on page 378). Compressor/Limiter When UAD SSL E Channel Strip is used in a stereo-in/stereo-out Link...
This compressor has an automatic make-up gain function. As Threshold is low- ered and compression increases (as knob is rotated clockwise), output gain from the module is increased automatically to compensate. Release sets the amount of time it takes for gain reduction to Compress Release cease once the input signal drops below the threshold level.
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Gate 1 (G1) In Gate 1 mode, signals below the Expand Threshold are attenuated by the Expand Range amount. Gate 1 is authentic to the gate mode on earlier hard- ware consoles. Gate 2 (G2) Gate 2 mode operates the same way as Gate 1, but has a different “no-chat- ter”...
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Expand Attack Attack defines the duration between the input signal reaching the thresh- old and processing being applied by the expander/gate. Attack time is normally auto-sensing and program dependent. When Fast Attack is en- abled, attack time is 1ms. Fast Attack is active when the “F.ATT” LED is illuminated. To toggle Fast At- tack, click the LED or its label text.
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The UAD SSL E Channel Strip EQ module is divided into four frequency bands: High Frequency (HF, blue knobs), High Midrange Frequency (HMF, green knobs), Low Midrange Frequency (LMF, yellow knobs), and Low Fre- quency (LF, orange knobs). The high and low bands can be switched from shelving mode into bell (peak/dip) mode.
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High Frequency HF Gain (HF) Band This control determines the amount by which the frequency value for the band is boosted or at- tenuated. The available range is ±15 dB in both Black and Brown modes. Tip: Click the “0” to return the Gain knob to its center position.
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HMF Q The Q (bandwidth) control defines the proportion of frequencies surrounding the band center frequency to be affected by the band gain control. The filter slopes get steeper (narrower bandwidth) as the control is rotated counter-clockwise. The available range is 0.5 to 2.5 in both Black and Brown modes.
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LF Frequency This control determines the band center frequency to be boosted or attenuated by the band Gain setting. The available range is 30 Hz to 450 Hz in both Black and Brown modes. LF Bell The Bell button switches the LF band from shelf mode to peak/dip mode. In normal (shelf) mode, only frequencies below the frequency value are boosted or attenuated.
Pre-Dynamics (PRE DYN) During “normal” operation (PRE DYN disengaged) the audio signal is output from the dynamics module into the EQ module. Activating PRE DYN reverses this routing, so the EQ is ahead of the dynamics module instead. Pre-dynam- ics is active when the red LED below the button is illuminated. Global I/O Meters The vertical LED-style metering provides a visual indication of the...
Power The Power button determines whether the plug-in is active. Click the Power button to disable the processor. Power is useful for comparing the processed sound to that of the original sig- nal. Usage Notes The SSL E Series channel has been used to mix more hit records than any other in history.
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Compressor or Gate/expander. UAD, UAD-1, UAD-2 and UAD 4K are trademarks of Universal Audio, Inc. All other names are trademarks of Solid State Logic, which are in no way associated or affiliated with Universal Audio, Inc.
SSL 4000 G console’s bus compressor. The undeniable drive and punch of this G Series master compressor — modeled to exacting detail by Universal Audio and fully authenticated by Solid State Logic® — helped make the original 4000 G Series the world's most success- ful studio production console.
SSL G Bus Compressor Controls Threshold Threshold defines the signal level at which the onset of compression oc- curs. Incoming signals that exceed this level are compressed. Signals below the level are unaffected. The control range is ±15 dB. As the Threshold control is de- creased and more compression oc- curs, output level is typically re- duced.
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Available Release times are discrete values of 100ms, 300ms, 600ms, 1.2s, and Auto. The Auto release characteristic for SSL G Bus Compressor has a unique quality that is optimized for program material. Ratio defines the amount of gain re- Ratio duction to be processed by the compressor.
Fade Rate Fade Rate determines the amount of time that will pass between the Fade but- ton being activated and the plug-in output level being reduced to minimum (or being raised to 0 dB in the case of a fade in). The available range is from 1.0 second to 60 seconds.
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SSL's unique fade curves. fade in is also available. UAD, UAD-1, UAD-2 and UAD 4K are trademarks of Universal Audio, Inc. All other names are trademarks of Solid State Logic, which are in no way associated or affiliated with Universal Audio, Inc.
CHAPTER 45 Studer A800 Multichannel Tape Recorder For more than 30 years, artists and engineers alike have been drawn to the warm sound, solid “punchy” low-end, and overall presence of the Studer® A800 Multichannel Tape Recorder. The sheer number of albums recorded on this legendary 2”...
Studer A800 Screenshot Figure 127. The Studer A800 plug-in window Operational Overview The Studer A800 for UAD-2 provides all of the original unit’s desirable ana- log sweetness; like magnetic tape, users can dial in a clean sound, or just the right amount of harmonic saturation using the Input and Output controls.
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+6dB, +7.5dB, or +9dB calibration levels, which can be used at their recom- mended settings, or tweaked for additional tonal options. Input, Sync and Re- pro paths, plus Thru (bypass) are available for authenticity, providing all available circuit options of the A800. A huge time saver, the Studer A800 plug-in features an innovative Gang Controls setting, allowing for instant glo- bal adjustment of any parameters for all Studer A800 instances in your ses- sion.
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Primary & The primary controls (those that are typically most Secondary used) are on the main panel at the bottom portion Controls of the interface. Additional (typically less used) controls are available on the secondary panel. The secondary panel (see Figure 128) is ac- cessed by clicking the Studer A800 label or the...
The UAD Studer A800 implements a control ganging feature that allows easy Ganged Operation simultaneous parameter modification for all instances of the plug-in. The fea- ture enables the DAW to emulate the multitrack tape deck scenario more ac- curately, where a single change to some multitrack machine parameters (such as tape speed, formula, and calibration settings) would affect all tape chan- nels.
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Repro Repro mode models the sound of recording through the record head and playback through the reproduction head, plus all corresponding electronics. Tape Type selects the active tape stock formulation. Four of Tape Type the most popular 2” magnetic tape formulas are modeled in the A800 plug-in: 250, 456, 900, and GP9.
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• 250: +3 Calibration (251 nWb/m) • 456: +6 Calibration (355 nWb/m) • 900: +9 Calibration (502 nWb/m) • GP9: +9 Calibration (502 nWb/m) Note: The noise floor is affected by Cal Level when Noise Enable (page 399) is active. Tip: The UAD Studer A800 default bank offers a variety of preset Tape Type, Tape Speed, CAL level, and EQ configurations that are commonly used for...
Just like real magnetic tape, lower Input levels will have a cleaner sound, while higher levels result in more harmonic saturation and coloration. Higher Input levels will also increase the output level from the plug-in. The Out- put control can be lowered to compensate. Tip: Click the “0”...
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Equaliser The Equaliser buttons determine the active Emphasis EQ val- (Emphasis EQ) ues and the frequency of the hum noise. Click the equaliser buttons to alternate between the two different types. NAB or CCIR curves can be selected when the Tape Speed is 7.5 or 15 IPS.
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While noise is historically considered a negative, and was the attribute that pushed the technical envelope for better machines and formulas, noise is still an ever-present component of the sound of using tape and tape machines. The Studer A800 has individual parameters for Bias, HF Auto Cal Record EQ, and Sync/Repro EQ.
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• Gang Controls is a static control without the ability to make relative offsets. Disable Gang Controls if offsets between the same control within different instantiations is desired. • If Gang Controls is enabled when Auto Cal is enabled, any adjustments made to Tape Type, Tape Speed or Emphasis EQ causes the Calibration Controls...
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Figure 129. The calibration controls for Studer A800 HF Record EQ HF (High Frequency) Record EQ is provided to make up for common residual HF loss due to Bias optimization and system filtering. It is used to tune HF con- tent into the incoming signal prior to the tape non-linearity.
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With the hardware machine, these controls enable compensation for any tape frequency loss or head wear. Under hardware use, the Sync and Repro playback heads are calibrated to normal operating standards and are nearly identical when set correctly. However, they may be tuned incorrectly to achieve a desired sound.
Hum Noise The Hum Noise frequency is dependent on the setting of the Equaliser (Em- phasis EQ) control (page 399). The frequency is 60 Hz when set to NAB (US) and 50 Hz when set to CCIR (European). Note: When IPS (Tape Speed) is set to 30 IPS, the yellow Equaliser (Empha-...
CHAPTER 46 Trident A-Range EQ Overview The original Trident A-Range desk holds near-mythic status in the professional recording industry, and is arguably the best loved of the classic Trident con- sole designs. Particularly noted for its fantastic preamps and the unique band interactions of its colorful EQ section, the Malcolm Toft / Trident-designed A-Range console has made an indelible impact on the sound of record mak- ing.
Operational Overview Unique Band Interactions & Distinct Cut-Filter Combinations The unique inductor-based EQ section of the board is what the Trident A-Range sound is all about. A series of three high pass filters and three low pass filters are arranged at the ends of the EQ section (see Figure 131).
Trident A-Range EQ Controls Phase The Phase (Ø) button inverts the polarity of the signal. The signal is in- verted when the button is engaged (darker). Leave the button inactive (lighter) for normal phase. Phase is independent of the EQ IN setting. Three low pass filters are available, and they can be used simul- Low Pass Filters taneously in any combination.
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Low-Mid Band The low-mid EQ offers peak/dip “bell” equalization for the middle-to- low fre- quencies. Low-Mid Frequency The center frequency of the low-mid filter is specified by this knob. Four center frequencies are available: 2 kHz, 1 kHz, 500 Hz, and 250 Hz. Low-Mid Gain The gain for the low-mid filter is specified by the horizontal slider control.
High Pass Filters Three high pass filters are available, and they can be used si- multaneously in any combination. The available cutoff frequen- cies are 100 Hz, 50 Hz, and 25 Hz with a slope of 18 dB per octave. Each filter is active when its button is engaged (darker). Each high pass filter “adds”...
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The Trident A-Range Console, featuring the Trident A-Range EQ The Trident A-Range Console, featuring the Trident A-Range EQ All visual and aural references to the TRIDENT A-RANGE EQ are trademarks being made with written permission from PMI AUDIO. UAD Powered Plug-Ins Manual - 410 - Chapter 46: Trident A-Range EQ...
— creating a solid framework for any mix, in any genre. Evolved from the popular Universal Audio 175 and 176 tube limiters, the 1176 retained proven qualities of these industry leaders, and set the standard for all limiters to follow.
UA 1176 Screenshots Figure 133. The UA 1176 Rev A plug-in window Figure 134. The UA 1176LN Rev E plug-in window Figure 135. The UA 1176AE plug-in window Figure 136. The 1176LN plug-in window Figure 137. The 1176SE plug-in window UAD Powered Plug-Ins Manual - 412 - Chapter 47: UA 1176 Classic Limiter Collection...
1176 Plug-In Family The complete 1176 family is comprised of five individual plug-ins, as seen on the previous page. Each variation has its own unique sonic characteristics. UA 1176 Limiter The UA 1176 Limiter Collection bundle (introduced in UAD v6.2) provides Collection three distinct 1176 revisions, representing over 40 years of design iterations to the original 1176 —...
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1176LN/SE The original 1176 bundle includes the 1176LN and 1176SE plug-ins. These first-generation versions run on UAD-1 and UAD-2 devices. To accommodate the limited DSP resources of the UAD-1, the input trans- former and I/O distortion characteristics were not modeled in these plug-ins. This makes the legacy LN/SE versions especially useful in situations where less distortion is desirable.
Operational Overview Applications Generally speaking, the primary use for the 1176 plug-ins are as individual inserts for sources that require compression, such as an individual snare, vo- cal, or guitar track, or for multi-instrument sources such as a stereo drum buss. Because the UA 1176 Limiter Collection also models the input and output am- plifiers, these models can also be used as “tone boxes”...
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Grit A simple 1176 trick is turning the attack and release up all the way to their fastest setting. This has the audible effect of adding compression distortion to the audio source, and is especially pronounced in all-buttons mode. What happens here is the attack and release are happening so fast that minute level fluctuations sound like distortion.
1176 Controls Each 1176 plug-in variation has the same control set, so they are only de- tailed once. The parameter descriptions below apply to all models unless oth- erwise noted. Input adjusts the amount of gain reduction as well as the relative threshold. Input Rotate the knob clockwise to increase the compression amount.
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The behavior of the Attack knob varies slightly between the models, as de- tailed below. UA 1176AE Attack The 1176AE offers a unique, fixed 10 ms “SLO” Attack mode when this con- trol is moved to the fully counter-clockwise position. UA 1176 Rev A and UA 1176LN Rev E Attack When Attack is in the OFF position the I/O amplifiers remain active while the compression circuit is bypassed.
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About Program-Dependent Release Program-dependent release is a feature of many compressors. The motivation for having program-dependent release is as follows: After a transient, it is de- sirable to have a fast release to avoid prolonged dropouts. However, while in a continued state of heavy compression, it is better to have a longer release time to reduce the pumping and harmonic distortion caused by repetitive at- tack- release cycles.
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In All Button mode the ratio goes to somewhere between 12:1 and 20:1, and the bias points change all over the circuit, thus changing the attack and re- lease times as well. The unique and constantly shifting compression curve that results yields a trademark overdriven tone that can only be found in this family of limiter/compressors.
Meter Function These four pushbutton switches (to the right of the VU Meter) determine the mode of the VU Meter, and whether the plug-in is enabled. When set to GR, the VU Meter indicates the Gain Reduction level in dB. When set to +8 or +4, the VU Meter indicates the output level in dB;...
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INDEX Customer Support EMT 140 Controls Cut Enable Button EMT 140 Plate Reverb Cut Filter EMT 250 Electronic Reverberator Cut Frequency Knob EMT 250 Functional Overview Cut Type Menu EMT 250 Program Mode Controls EMT 250 Screenshot Enable Button Damper Controls (Reverb Time) Enable/Bypass Switch 65, 66, 69, 72 Damping Knob...
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INDEX Frequency Knob 51, 283 LA-2A LA-3A Compressor Gain (G) Knob LA-3A Controls Gain Knob 52, 283 LA-3A Screenshot Gain Reduction Late Gain Reduction Meter Late-Field Relative Timing Grit Late-Field Start L-Delay Knob Levels Harrison 32C EQ Lexicon 224 Harrison 32C EQ and Harrison 32C SE Controls Lexicon 224 Controls Harrison 32C EQ Screenshot Lexicon 224 Screenshot...
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INDEX Massive Passive Notes Neve 1081 Latency Master Neve 1081 Screenshot 146, 157, 222, 246 Master Level Knob Neve 1081SE Materials Blending Neve 1081SE Controls 144, 228 Materials Blending Bars Neve 31102 Console EQ Materials Menus Neve 31102 Controls Materials Panel Neve 31102 Latency Materials panel Neve 31102 Screenshot...
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INDEX Positioning panel 78, 87 Pultec and Pultec-Pro Pultec EQP-1A Program Equalizer Power 351, 353 Pultec Latency Power Lamp Pultec MEQ-5 Power Switch 112, 282 Pultec-Pro Precision Buss Compressor 73, 260 Precision Buss Compressor Controls 74, 261 Precision Buss Compressor Screenshot Q (Bandwidth) Knob Precision De-Esser 260, 266...
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INDEX Roland RE-201 Screenshot SSL G Channel Strip Controls Room Shape and Material SSL G Channel Strip Screenshot R-Pan Knob 69, 72 Stereo RS-1 Controls Stereo Mode RS-1 Reflection Engine Stereo Operation Studer A800 Studer A800 Screenshot Screen Shots Studer Operational Overview Secondary Controls Support Website Shape...
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